From braunsquared at gmail.com Wed Dec 1 08:12:51 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Wed, 1 Dec 2010 02:12:51 -0500 Subject: [gst-devel] Compositing and GStreamer In-Reply-To: References: <1290934705.2457.5.camel@deumeu> Message-ID: Hey Everyone, I just wanted to send out a very heartfelt thanks to those that helped me over the last few days. Both via the mail list and irc. Your assistance has been most appreciated. Turns out the issue was that I should have RTFM. During my tests I had overlooked one of the documents on the GStreamer website which turned out to be most useful. Not neglecting the assistance from others, without it I wouldn't have known what to look for and in the end pads were my nemesis. I have a test working, 6 videos, in a single frame all composed with a gnlcomposition with a videomixer in a gnloperation. Very exciting. I also have a second sequence of 6 videos (in a 3x2 grid) in the same composition. Here's my next, and hopefully final, question. The first sequence of videos gnlsources are configured like: - start: 0 - duration: 30 * GST_SECOND - media-start: 0 - media-duration: 30 * GST_SECOND - priority: incrementing by 1 depending on grid location This all works wonderfully. When I add another sequence with the gnlsources configured like: - start: 30 * GST_SECOND - duration: 30 * GST_SECOND - media-start: 0 - media-duration: 30 * GST_SECOND - priority: incrementing by 1 depending on grid location All the videos vanish and I end up with a blank frame with a duration of 60 seconds. It I set the media-start property to 30 * GST_SECOND, it all works but the second sequence of videos are off by 30 seconds. The source files are the same although the elements are unique to it's presence in the stream. This leads me to believe that somethings happening during a seek of some nature. Would adding the second sequence of videos when receiving the EOS signal fix this or is there something else going on? If it should happen during the EOS signal, is there anything I should be aware of when adding elements to the pipeline while handling the signal? Again, most appreciated and when this is all said and done I'd be more than happy to write up a wiki document on what I've learned about gnonlin, pads and troubleshooting links between them. All the Best, Tim On Tue, Nov 30, 2010 at 3:24 AM, Timothy Braun wrote: > Ok, making some headway. I made a 720x480 input, in mpg form, for the > composition which makes the video mixer output the proper size. I've also > dropped the mp4's in favor of mpeg 1 videos as it seemed to be causing an > issue. Using the priority change callback, I've been able to move the > videos around in the frame. Now onto my new issue :). > > This is my current bit of code: > > > GstPipeline *pipeline = GST_PIPELINE(gst_pipeline_new("pipeline")); > > // Create our composition > GstElement *comp = gst_element_factory_make("gnlcomposition", > "composition"); > > GstElement *src1 = gst_element_factory_make("gnlfilesource", NULL); > g_object_set(G_OBJECT(src1), "location", "/Users/tbraun/Documents/Mac > Applications/ccvideorenderer/loop1.mpg", > "start", 0, > "duration", 5 * GST_SECOND, > "media-start", 0, > "media-duration", 5 * GST_SECOND, > "priority", 1, > NULL); > GstElement *src2 = gst_element_factory_make("gnlfilesource", NULL); > g_object_set(G_OBJECT(src2), "location", "/Users/tbraun/Documents/Mac > Applications/ccvideorenderer/loop2.mpg", > "start", 0, > "duration", 5 * GST_SECOND, > "media-start", 0, > "media-duration", 5 * GST_SECOND, > "priority", 2, > NULL); > GstElement *src3 = gst_element_factory_make("gnlfilesource", NULL); > g_object_set(G_OBJECT(src3), "location", "/Users/tbraun/Documents/Mac > Applications/ccvideorenderer/loop3.mpg", > "start", 0, > "duration", 5 * GST_SECOND, > "media-start", 0, > "media-duration", 5 * GST_SECOND, > "priority", 3, > NULL); > GstElement *src6 = gst_element_factory_make("gnlfilesource", NULL); > g_object_set(G_OBJECT(src6), "location", "/Users/tbraun/Documents/Mac > Applications/ccvideorenderer/loop6.mpg", > "start", 0, > "duration", 5 * GST_SECOND, > "media-start", 0, > "media-duration", 5 * GST_SECOND, > "priority", 6, > NULL); > GstElement *bg = gst_element_factory_make("gnlfilesource", NULL); > g_object_set(G_OBJECT(bg), "location", "/Users/tbraun/Documents/Mac > Applications/ccvideorenderer/background.mpg", > "start", 0, > "duration", 5 * GST_SECOND, > "media-start", 0, > "media-duration", 5 * GST_SECOND, > "priority", 7, > > NULL); > GstElement *oper = gst_element_factory_make("gnloperation", NULL); > g_object_set(G_OBJECT(oper), > "start", 0, > "duration", 5 * GST_SECOND, > "media-start", 0, > "media-duration", 5 * GST_SECOND, > "priority", 0, > > NULL); > GstElement *mixer = gst_element_factory_make("videomixer", NULL); > gst_bin_add(GST_BIN(oper), mixer); > > g_object_connect(oper, "signal::input-priority-changed", > onPriorityChange, mixer, NULL ); > > // add the sources to the composition > gst_bin_add(GST_BIN(comp), src1); > gst_bin_add(GST_BIN(comp), src2); > gst_bin_add(GST_BIN(comp), src3); > gst_bin_add(GST_BIN(comp), src6); > gst_bin_add(GST_BIN(comp), bg); > > gst_bin_add(GST_BIN(comp), oper); > > // build the output stream > GstElement *color = gst_element_factory_make("ffmpegcolorspace", > "colorspace"); > > GstElement *enc = gst_element_factory_make("ffenc_mpeg1video", > "encoder"); > GstElement *mux = gst_element_factory_make("ffmux_mpeg", "mux"); > GstElement *queue = gst_element_factory_make("queue", "queue"); > > GstElement *sink = gst_element_factory_make("filesink", "sink"); > g_object_set(sink, "location", "output.mpg", NULL); > > gst_bin_add_many(GST_BIN(pipeline), comp, color, enc, mux, queue, sink, > NULL); > > > g_object_connect (comp, "signal::pad-added", > onPad, color, NULL); > > gst_element_link_many(color, enc, mux, queue, sink, NULL); > > return pipeline; > > This works and outputs a 720x480 video containing the 4 input videos in the > proper places. If I try to add the two additional videos, 6 videos at > 240x240 plus the bg video at 720x480, the process fails while spitting out a > plethora of these: > > *0:00:06.174861000 [331m47454 [00m 0x1006dc460 [33;01mWARN [00m > [00m mp3parse > gstmpegaudioparse.c:1601:head_check: [00m invalid sync* > > It occurs if I add either 1 of the 2 missing videos. > > The other issue seems to be with sync. The source videos for testing > have an embedded timestamp in the frame image. The initial frames are off > for each of the source videos and they all end at a different time. Any > suggestions? > > Again, any input is greatly appreciated. Even a nice "it's not possible" > would be of great assistance. > > All the Best, > Tim > > > > > On Mon, Nov 29, 2010 at 5:00 PM, Timothy Braun wrote: > >> You'll have to excuse my ignorance in my last email. I had the mixer in >> the gnloperation, but I had the src pad of the mixer attached to the final >> output segment rather than the composition src pad. I fixed that, but am >> still running into issues. >> >> So, currently, I have a pipeline which resembles this: >> >> >> +--------------------------------+ >> | gnlcomposition | >> | +----------------------------+ | >> | | Box 1 (gnlfilesource) | | +-------+ >> +-------------------------------+ >> | +----------------------------+ |->| queue |->| filesink (plus encoder >> stuff) | >> | | Box 2 (gnlfilesource) | | +-------+ >> +-------------------------------+ >> | +----------------------------+ | >> | | Video Mixer (gnloperation) | | >> | +----------------------------+ | >> +--------------------------------+ >> >> I hope that comes through ok.. Box 1, box 2 and video mixer all have >> the same duration and the input-priority-changed call is triggering the >> update to xpos and ypos on the video mixer pads. Ultimately there will be 6 >> boxes per frame, but that I'm trying to keep it simple at the moment. >> >> The issue I'm currently running into is with the video mixer. With the >> Box 1 and Box 2 being 240x240 px, the final frame size is 240x240 px. Is >> there a way to feed in a blank 720x480 frame into the mixer so it outputs >> the proper resolution or am I looking at this the wrong way? >> I've tried to input a 720x480 frame from a videotestsrc, but am having >> issues (structurally I believe) getting my test source to exist in a >> gnlsource at a 720x480 resolution. >> >> Again, any help is greatly, greatly appreciated as I'm running short on >> time to get this complete. >> >> Best, >> Tim >> >> >> On Sun, Nov 28, 2010 at 9:06 PM, Timothy Braun wrote: >> >>> Hi Edward, >>> Thanks for the quick reply. It has been most informative. A couple >>> questions if I may. For the video rescale/capsfilter part, what is this >>> doing and how should it look in the pipeline? The input videos are all >>> 240x240 px with a goal output frame size of 720x480px. >>> >>> For a very raw test, I am using this to build my pipeline: >>> >>> GstPipeline *pipeline = GST_PIPELINE(gst_pipeline_new("pipeline")); >>> >>> GstCaps *caps = >>> gst_caps_from_string("video/x-raw-yuv;video/x-raw-rgb"); >>> >>> // Create our composition >>> GstElement *comp = gst_element_factory_make("gnlcomposition", >>> "composition"); >>> g_object_set(G_OBJECT(comp), "caps", caps, NULL); >>> >>> GstElement *src1 = gst_element_factory_make("gnlfilesource", >>> "source1"); >>> g_object_set(G_OBJECT(src1), "location", "loop1.mp4", >>> "start", 0, >>> "duration", 30 * GST_SECOND, >>> "media-start", 0, >>> "media-duration", 30 * GST_SECOND, >>> "priority", 1, >>> "caps", caps, >>> NULL); >>> >>> GstElement *src2 = gst_element_factory_make("gnlfilesource", >>> "source2"); >>> g_object_set(G_OBJECT(src2), "location", "loop2.mp4", >>> "start", 0, >>> "duration", 30 * GST_SECOND, >>> "media-start", 0, >>> "media-duration", 30 * GST_SECOND, >>> "priority", 2, >>> "caps", caps, >>> NULL); >>> >>> GstElement *oper = gst_element_factory_make("gnloperation", NULL); >>> g_object_set(G_OBJECT(oper), "caps", caps, "expandable", TRUE, NULL); >>> >>> GstElement *mixer = gst_element_factory_make("videomixer", NULL); >>> gst_bin_add(GST_BIN(oper), mixer); >>> >>> // listen for the input priority change signal >>> // to update the video mixer pad >>> g_object_connect(oper, "signal::input-priority-changed", >>> onPriorityChange, mixer, NULL ); >>> >>> // add the sources to the composition >>> gst_bin_add(GST_BIN(comp), src1); >>> gst_bin_add(GST_BIN(comp), src2); >>> gst_bin_add(GST_BIN(comp), oper); >>> >>> // build the output stream >>> GstElement *color = gst_element_factory_make("ffmpegcolorspace", >>> "colorspace"); >>> >>> GstElement *identity = gst_element_factory_make("identity", "ident"); >>> g_object_set(identity, "single-segment", TRUE, NULL); >>> >>> GstElement *enc = gst_element_factory_make("ffenc_mpeg1video", >>> "encoder"); >>> GstElement *mux = gst_element_factory_make("ffmux_mpeg", "mux"); >>> GstElement *queue = gst_element_factory_make("queue", "queue"); >>> >>> GstElement *sink = gst_element_factory_make("filesink", "sink"); >>> g_object_set(sink, "location", "output.mpg", NULL); >>> >>> gst_bin_add_many(GST_BIN(pipeline), comp, color, identity, enc, mux, >>> queue, sink, NULL); >>> >>> /* >>> g_object_connect (comp, "signal::pad-added", >>> onPad, mixer, NULL); >>> */ >>> >>> gst_element_link_many(mixer, queue, color, identity, enc, mux, sink, >>> NULL); >>> >>> >>> It seems to link up fine, but fails to stream as it gets stuck in the >>> paused state. I've tried adding queues all around, but no luck. Here is >>> the debug output: >>> >>> ** Message: Creating run loop... >>> ** Message: Building pipeline... >>> ** Message: Attaching to bus... >>> ** Message: Setting state to PLAYING... >>> 0:00:00.141504000 [334m91821 [00m 0x100609d30 [33;01mWARN [00m >>> [00;01;34m gnlsource >>> gnlsource.c:545:gnl_source_change_state: [00m Couldn't find a valid >>> source pad >>> 0:00:00.162296000 [334m91821 [00m 0x100609d30 [33;01mWARN [00m >>> [00;01;35m GST_SCHEDULING gstpad.c:4692:gst_pad_get_range: >>> [00m getrange failed unexpected >>> 0:00:00.191513000 [334m91821 [00m 0x100609d30 [33;01mWARN [00m >>> [00;01;34m gnlsource >>> gnlsource.c:545:gnl_source_change_state: [00m Couldn't find a valid >>> source pad >>> 0:00:00.199956000 [334m91821 [00m 0x100686b10 [33;01mWARN [00m >>> [00m qtdemux qtdemux.c:5801:qtdemux_parse_trak: [00m >>> unknown version 00000000 >>> 0:00:00.200693000 [334m91821 [00m 0x100609d30 [33;01mWARN [00m >>> [00;01;35m GST_SCHEDULING gstpad.c:4692:gst_pad_get_range: >>> [00m getrange failed unexpected >>> 0:00:00.210835000 [334m91821 [00m 0x10064f030 [33;01mWARN [00m >>> [00m qtdemux qtdemux.c:5801:qtdemux_parse_trak: [00m >>> unknown version 00000000 >>> 0:00:00.244706000 [334m91821 [00m 0x101879f60 [33;01mWARN [00m >>> [00;01;34m gnlsource >>> gnlsource.c:221:element_pad_added_cb: [00m We already have >>> (pending) ghost-ed a valid source pad (ghostpad:'':'', pendingblock:1 >>> 0:00:00.244852000 [334m91821 [00m 0x1006501f0 [33;01mWARN [00m >>> [00;01;34m gnlsource >>> gnlsource.c:221:element_pad_added_cb: [00m We already have >>> (pending) ghost-ed a valid source pad (ghostpad:'':'', pendingblock:1 >>> ** Message: Priority Changed: >>> composition::gnloperation0->gnloperation0::sink_1(videomixer0::sink_1) - 2 >>> ** Message: Priority Changed: >>> composition::gnloperation0->gnloperation0::sink_0(videomixer0::sink_0) - 1 >>> >>> I've been battling with this for a bit now, can't seem to make any >>> headway. Any input is, again, much appreciated. >>> >>> Best, >>> Tim >>> >>> >>> On Sun, Nov 28, 2010 at 3:58 AM, Edward Hervey wrote: >>> >>>> Hi, >>>> >>>> On Sat, 2010-11-27 at 13:50 -0500, Timothy Braun wrote: >>>> > Kapil, >>>> > Thanks for the suggestion, but with multifilesrc I would have to >>>> > have the files static with incrementing named files. A single box in >>>> > the 3x2 grid may contain the same file multiple times so I'm afraid it >>>> > won't be the best solution. >>>> > >>>> > I guess, ultimately, there's multiple ways to attack this one from >>>> > what I've been able to find. Here are the two that I've been looking >>>> > at, I'm just not sure which is the better solution at this point: >>>> > * A single gnonlin composition with a 4 gnlsources similar in >>>> > setup as the gst-launch text I have below. >>>> >>>> Using one composition would actually be the 'proper' way. >>>> >>>> > * 6 gnonlin compositions, each feeding to a single videomixer >>>> > which combines them into the final frame. >>>> > * This path I'm currently investigating. I have a test >>>> > written in C, but I'm having some difficulties with >>>> > pad linkage as I still don't have a complete >>>> > understanding of when certain things will exist and >>>> > how to get them. >>>> > * Here's currently whats happening: >>>> > * Create a new pipeline >>>> > * Create a videomixer >>>> > * Create 6 gnonlin compositions each with a >>>> > pad-added signal callback to connect >>>> > gnlcomposition pad to videomixer. >>>> > * ... (this is were it's going wrong) >>>> > * In the pad-added callback I have: >>>> > * static void onPad(GstElement *comp, GstPad >>>> > *pad, GstElement *sink) { >>>> > GstPad *v = gst_element_get_pad(sink, >>>> > "sink"); >>>> > gst_pad_link(pad, v); >>>> > gst_object_unref(v); >>>> > } >>>> > * gst_element_get_pad is not returning a pad >>>> > from the video mixer (sink) which leads me to >>>> > believe that I'm either not asking in the >>>> > right manner or the pad doesn't exist. (I'm >>>> > aware that gst_element_get_pad is deprecated, >>>> > I'm just looking to test at the moment) >>>> > * I noticed in one of the repositories under a >>>> > unit test, the videomixer was attached as a >>>> > gnloperation? Is this the better path to >>>> > take? >>>> > >>>> > This all leads me to a couple more questions as well: >>>> > * A video mixer pad has xpos and ypos properties. This would >>>> > let me shift the video around without needing a video box >>>> > which I believe may be more efficient? >>>> >>>> Yes, it will be more efficient. >>>> >>>> > * If I use the xpos and ypos properties, is the video mixer >>>> > smart enough to change the frame size appropriately or will it >>>> > simply crop the frame to the size of the largest input frame? >>>> > * If so, would it be better to add a videobox to do the >>>> > adjustments for me, or feed in a solid color >>>> > background of the required output size? >>>> >>>> No, it won't change the size, but what you could do is mix the >>>> original sizes with original offsets and then downconvert the video >>>> later. >>>> >>>> Example for one 3x2 segment: >>>> >>>> Create a gnloperation with a videomixer in it with a gnl priority of >>>> 0. >>>> Create a gnlfilesource for each clip with increasing priorities (1->6) >>>> going from left-right and then top to bottom: >>>> 1 2 3 >>>> 4 5 6 >>>> >>>> Connect to the gnloperation 'input-priority-changed' signal. When your >>>> callback is called, you will know which priority is being connected to >>>> which gnloperation ghostpad. You can get the videomixer sink pad by >>>> using the gst_ghost_pad_get_target() method and then setting the proper >>>> xpos/ypos property on that pad based on the priority of the feed being >>>> provided. >>>> >>>> Set 'video/x-raw-yuv;video/x-raw-rgb' as the caps property on all your >>>> sources. >>>> >>>> Set duration and media-duration of *all* gnlobjects to the same >>>> duration. >>>> If you want to add another segment of 3x2 clips, you'll need to re-add >>>> all those 7 objects with a modified 'start' property. >>>> >>>> First connect your composition to an imagesink to make sure the result >>>> is what you want. When it is, insert a videoscale element followed with >>>> a capsfilter with your target resolution. >>>> >>>> Hope this helps. >>>> >>>> > >>>> > Thanks again for the time. I know there's a lot of questions above, >>>> > but any help of any kind is greatly appreciated. >>>> > >>>> > All the best, >>>> > Tim >>>> > >>>> > >>>> > On Fri, Nov 26, 2010 at 1:04 AM, Kapil Agrawal >>>> > wrote: >>>> > Just a quick clue that might help, try using multifilesrc ? >>>> > >>>> > >>>> > On Thu, Nov 25, 2010 at 9:47 PM, Timothy Braun >>>> > wrote: >>>> > >>>> > >>>> > Hello Everyone, >>>> > I'm fairly new to G-Streamer so any input you can >>>> > provide is much appreciated. I'm working on a project >>>> > where we need to generate a 2 minute video which is a >>>> > composite of a total of 24 input videos. The output >>>> > video will have 4 different 30 second sections, each >>>> > containing a 3x2 grid of the smaller input videos. >>>> > The input videos are all naturally at 240x240 with the >>>> > goal of having a final output frame size of 720x480. >>>> > >>>> > Using gst-launch, I've been able to construct a >>>> > sample 30 second clip using a combination of inputs, >>>> > videoboxes and a videomixer. Here is what I've come >>>> > up with so far: >>>> > >>>> > videomixer name=mix ! ffmpegcolorspace ! >>>> > ffenc_mpeg1video ! ffmux_mpeg name=mux ! queue ! >>>> > filesink location=output.mpg >>>> > adder name=adder ! audioconvert ! ffenc_mp2 ! mux. >>>> > filesrc location=loop1.mp4 ! decodebin name=decode1 >>>> > decode1. ! videobox border-alpha=0 top=-240 left=0 ! >>>> > queue ! mix. >>>> > decode1. ! adder. >>>> > filesrc location=loop2.mp4 ! decodebin name=decode2 >>>> > decode2. ! videobox border-alpha=0 top=-240 >>>> > left=-240 ! queue ! mix. >>>> > decode2. ! adder. >>>> > filesrc location=loop3.mp4 ! decodebin name=decode3 >>>> > decode3. ! videobox border-alpha=0 top=-240 >>>> > left=-480 ! queue ! mix. >>>> > decode3. ! adder. >>>> > filesrc location=loop4.mp4 ! decodebin name=decode4 >>>> > decode4. ! videobox border-alpha=0 top=0 left=0 ! >>>> > queue ! mix. >>>> > decode4. ! adder. >>>> > filesrc location=loop5.mp4 ! decodebin name=decode5 >>>> > decode5. ! videobox border-alpha=0 top=0 left=-240 ! >>>> > queue ! mix. >>>> > decode5. ! adder. >>>> > filesrc location=loop6.mp4 ! decodebin name=decode6 >>>> > decode6. ! videobox border-alpha=0 top=0 left=-480 ! >>>> > queue ! mix. >>>> > decode6. ! adder. >>>> > >>>> > Now I need to do this 4 times, each time with a >>>> > potentially different video in each box. I've started >>>> > looking into C interfaces as there's other pieces of >>>> > the puzzle which need to be tied into this, and I am >>>> > trying to determine the best way to tackle this. I >>>> > originally was looking at Gnonlin, but the >>>> > documentation is lacking in regards to how >>>> > gnloperations work. I also recently stumbled upon the >>>> > GES library by Edward Hervey, this looks promising as >>>> > well, but I haven't been able to spend much time on >>>> > it. >>>> > >>>> > If I go the Gnonlin route, I believe I would need 6 >>>> > compositions, one for each box. At the 30 second >>>> > marker, I would swap the filesource to a new one using >>>> > dynamic pads and listening for messages on the >>>> > pipeline bus. Am I far off on this? Any suggestions? >>>> > >>>> > As for the GES library, it looks very promising and >>>> > powerful from the little I read on it. Would this be >>>> > the smarter route to take? If so, does anyone have >>>> > any suggestions for how the pipeline would be >>>> > structured? >>>> > >>>> > Thank you in advance for your time on this and I >>>> > truly appreciate any information you are willing to >>>> > share with me. >>>> > >>>> > Happy Thanksgiving, >>>> > Tim >>>> > >>>> > >>>> > >>>> ------------------------------------------------------------------------------ >>>> > Increase Visibility of Your 3D Game App & Earn a >>>> > Chance To Win $500! >>>> > Tap into the largest installed PC base & get more eyes >>>> > on your game by >>>> > optimizing for Intel(R) Graphics Technology. Get >>>> > started today with the >>>> > Intel(R) Software Partner Program. Five $500 cash >>>> > prizes are up for grabs. >>>> > http://p.sf.net/sfu/intelisp-dev2dev >>>> > _______________________________________________ >>>> > gstreamer-devel mailing list >>>> > gstreamer-devel at lists.sourceforge.net >>>> > >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> > >>>> > >>>> > >>>> > >>>> > -- >>>> > www.mediamagictechnologies.com (Gstreamer, ffmpeg, Red5, >>>> > Streaming) >>>> > twitter handle: @gst_kaps >>>> > http://www.linkedin.com/in/kapilagrawal >>>> > >>>> > >>>> ------------------------------------------------------------------------------ >>>> > Increase Visibility of Your 3D Game App & Earn a Chance To Win >>>> > $500! >>>> > Tap into the largest installed PC base & get more eyes on your >>>> > game by >>>> > optimizing for Intel(R) Graphics Technology. Get started today >>>> > with the >>>> > Intel(R) Software Partner Program. Five $500 cash prizes are >>>> > up for grabs. >>>> > http://p.sf.net/sfu/intelisp-dev2dev >>>> > _______________________________________________ >>>> > gstreamer-devel mailing list >>>> > gstreamer-devel at lists.sourceforge.net >>>> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> > >>>> > >>>> > >>>> ------------------------------------------------------------------------------ >>>> > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >>>> > Tap into the largest installed PC base & get more eyes on your game by >>>> > optimizing for Intel(R) Graphics Technology. Get started today with >>>> the >>>> > Intel(R) Software Partner Program. Five $500 cash prizes are up for >>>> grabs. >>>> > http://p.sf.net/sfu/intelisp-dev2dev >>>> > _______________________________________________ gstreamer-devel >>>> mailing list gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >>>> Tap into the largest installed PC base & get more eyes on your game by >>>> optimizing for Intel(R) Graphics Technology. Get started today with the >>>> Intel(R) Software Partner Program. Five $500 cash prizes are up for >>>> grabs. >>>> http://p.sf.net/sfu/intelisp-dev2dev >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gibrovacco at gmail.com Wed Dec 1 08:50:14 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 1 Dec 2010 09:50:14 +0200 Subject: [gst-devel] Is there plugin for HTTP stream? In-Reply-To: <1291110630581-3065090.post@n4.nabble.com> References: <1291110630581-3065090.post@n4.nabble.com> Message-ID: Hi, On Tue, Nov 30, 2010 at 11:50 AM, wl2776 wrote: > > souphttpsrc I'd say this is the most advanced one in terms of features, even though I'd say it's somewhat gnome-specific (e.g. you need libsoup, gnuTLS and libgcrypt to get it working). Regards > neonhttpsrc > ffdemux_applehttp > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Is-there-plugin-for-HTTP-stream-tp3065081p3065090.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From bard_kuo at compalcomm.com Wed Dec 1 09:10:30 2010 From: bard_kuo at compalcomm.com (bard) Date: Wed, 1 Dec 2010 00:10:30 -0800 (PST) Subject: [gst-devel] How to post gst-openmax Message-ID: <1291191030810-3066790.post@n4.nabble.com> Dear All, I am work on porting gst-openmax , but I didn't find some document described more specifically. Is there a document like porting guide ? or some flowchart available can make me figure out easily. Thanks, Bard -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-post-gst-openmax-tp3066790p3066790.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bard_kuo at compalcomm.com Wed Dec 1 10:22:36 2010 From: bard_kuo at compalcomm.com (bard) Date: Wed, 1 Dec 2010 01:22:36 -0800 (PST) Subject: [gst-devel] out flush in gst-openmax Message-ID: <1291195356075-3066876.post@n4.nabble.com> Hello All, I use gst-openmax0.10.1 to access qcom openmaxIL. command "gst-launch filesrc location=/home/user/src.264 ! omx_h264dec ! fake sink" It turns out that Error from OpenMAX component fill_this_buffer_proxy failure ERROR: from element /GstPipeline:pipeline0/GstOmxH264Dec:omxh264dec0: GStreamer encountered a general stream error. Additional debug info: gstomx_base_filter.c(684): pad_chain (): /GstPipeline:pipeline0/GstOmxH264Dec:omxh264dec0: Error from OpenMAX component ERROR: pipeline doesn't want to preroll. ERROR: Sending OMX_EventError to ClientSetting pipeline to NULL ... What happens? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/out-flush-in-gst-openmax-tp3066876p3066876.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ceyusa at gmail.com Wed Dec 1 10:44:31 2010 From: ceyusa at gmail.com (=?ISO-8859-1?Q?Victor_Manuel_J=E1quez_Leal?=) Date: Wed, 1 Dec 2010 10:44:31 +0100 Subject: [gst-devel] How to post gst-openmax In-Reply-To: <1291191030810-3066790.post@n4.nabble.com> References: <1291191030810-3066790.post@n4.nabble.com> Message-ID: On Wed, Dec 1, 2010 at 9:10 AM, bard wrote: > > Dear All, > > I am work on porting gst-openmax , but I didn't find some document described > more specifically. > Is there a document like porting guide ? or some flowchart available can > make me figure out easily. You could try asking to the gst-openmax mailing list: https://lists.sourceforge.net/lists/listinfo/gstreamer-openmax And you can also scheme the info available at http://freedesktop.org/wiki/GstOpenMAX Porting gst-openmax should be pretty straightforward. Getting OpenMAX working in your setup is another story. vmjl > > Thanks, > Bard > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-post-gst-openmax-tp3066790p3066790.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ceyusa at gmail.com Wed Dec 1 10:46:58 2010 From: ceyusa at gmail.com (=?ISO-8859-1?Q?Victor_Manuel_J=E1quez_Leal?=) Date: Wed, 1 Dec 2010 10:46:58 +0100 Subject: [gst-devel] out flush in gst-openmax In-Reply-To: <1291195356075-3066876.post@n4.nabble.com> References: <1291195356075-3066876.post@n4.nabble.com> Message-ID: On Wed, Dec 1, 2010 at 10:22 AM, bard wrote: > > Hello All, > > I use gst-openmax0.10.1 to access qcom openmaxIL. > command "gst-launch filesrc location=/home/user/src.264 ! omx_h264dec ! fake > sink" > > It turns out that Error from OpenMAX component > > fill_this_buffer_proxy failure > ERROR: from element /GstPipeline:pipeline0/GstOmxH264Dec:omxh264dec0: > GStreamer encountered a general stream error. > Additional debug info: > gstomx_base_filter.c(684): pad_chain (): > /GstPipeline:pipeline0/GstOmxH264Dec:omxh264dec0: > Error from OpenMAX component > ERROR: pipeline doesn't want to preroll. > ERROR: Sending OMX_EventError to ClientSetting pipeline to NULL ... Your OMX H264 component raised an error. This doesn't look like a gstreamer error, neither a gst-openmax one, this looks more like something wrong in you OpenMAX implementation. vmjl > > What happens? > > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/out-flush-in-gst-openmax-tp3066876p3066876.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From bard_kuo at compalcomm.com Wed Dec 1 11:17:52 2010 From: bard_kuo at compalcomm.com (bard) Date: Wed, 1 Dec 2010 02:17:52 -0800 (PST) Subject: [gst-devel] How to post gst-openmax In-Reply-To: References: <1291191030810-3066790.post@n4.nabble.com> Message-ID: <1291198672143-3066977.post@n4.nabble.com> Thanks for reply. I will try asking to the gst-openmax mailing list. >Porting gst-openmax should be pretty straightforward. Getting OpenMAX Is it meant for no modifies needed ? >working in your setup is another story. Thanks, Bard -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-post-gst-openmax-tp3066790p3066977.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bard_kuo at compalcomm.com Wed Dec 1 11:34:47 2010 From: bard_kuo at compalcomm.com (bard) Date: Wed, 1 Dec 2010 02:34:47 -0800 (PST) Subject: [gst-devel] out flush in gst-openmax In-Reply-To: References: <1291195356075-3066876.post@n4.nabble.com> Message-ID: <1291199687805-3067008.post@n4.nabble.com> Thank you vmjl , I will figure out what's wrong in OpenMax. So for gst-openmax , it doesn't need to modify in gstomx_base_filter.c pad_chain (). Right? Bard -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/out-flush-in-gst-openmax-tp3066876p3067008.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ceyusa at gmail.com Wed Dec 1 12:27:36 2010 From: ceyusa at gmail.com (=?ISO-8859-1?Q?Victor_Manuel_J=E1quez_Leal?=) Date: Wed, 1 Dec 2010 12:27:36 +0100 Subject: [gst-devel] out flush in gst-openmax In-Reply-To: <1291199687805-3067008.post@n4.nabble.com> References: <1291195356075-3066876.post@n4.nabble.com> <1291199687805-3067008.post@n4.nabble.com> Message-ID: On Wed, Dec 1, 2010 at 11:34 AM, bard wrote: > > Thank you vmjl , > > I will figure out what's wrong in OpenMax. > > So for gst-openmax , it doesn't need to modify in gstomx_base_filter.c > pad_chain (). Right? In theory, if your omx component follows "correctly" the spec, no changes are needed in gst-openmax vmjl > > > Bard > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/out-flush-in-gst-openmax-tp3066876p3067008.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From mq_han at hotmail.com Wed Dec 1 13:19:45 2010 From: mq_han at hotmail.com (mingqian Han) Date: Wed, 1 Dec 2010 20:19:45 +0800 Subject: [gst-devel] dynamically change video resolution Message-ID: Hi all, Currently I am working with TI Davinci processor, and I build a gstreamer pipeline for video streaming as follows: v4l2src->capsfilter->tee->queue->resizer->h264encoder->rtph264pay->udpsink ->queue->resizer->mjpegencoder->filesink It is just a brief outline, and it works ok. Now I want to change the resolution from D1->CIF on the fly, currently, I do the following: 1. get the src pad of queue 2. block the pad 3. change the properties of resizer and the encoder. 4. unblock the pad but after that, the pipeline stops working. I am new to gstreamer, and wonder if anyone could give me some suggestion. regards, Rafael -------------- next part -------------- An HTML attachment was scrubbed... URL: From dlafferty at gmail.com Wed Dec 1 13:32:56 2010 From: dlafferty at gmail.com (HaroldJRoth) Date: Wed, 1 Dec 2010 04:32:56 -0800 (PST) Subject: [gst-devel] dynamically change video resolution In-Reply-To: References: Message-ID: <1291206776961-3067172.post@n4.nabble.com> When you say 'pipeline stops working', have you tried querying the pipeline state? Just a guess, but you may need to put it back into the PLAYING state. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/dynamically-change-video-resolution-tp3067151p3067172.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From gibrovacco at gmail.com Wed Dec 1 15:05:31 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 1 Dec 2010 16:05:31 +0200 Subject: [gst-devel] dynamically change video resolution In-Reply-To: References: Message-ID: Hi, 2010/12/1 mingqian Han : > Hi all, > ??? Currently I am working with TI Davinci processor, and I build a > gstreamer pipeline for video streaming as follows: > > v4l2src->capsfilter->tee->queue->resizer->h264encoder->rtph264pay->udpsink > > ->queue->resizer->mjpegencoder->filesink > ??? It is just a brief outline, and it works ok. > > ??? Now I want to change the resolution from D1->CIF on the fly, currently, > I do the following: > ??? 1. get the src pad of queue > ??? 2. block the pad > ??? 3. change the properties of resizer and the encoder. > ??? 4. unblock the pad > > ??? but after that, the pipeline stops working. I am new to gstreamer, and > wonder if anyone could give me some suggestion. running the pipeline under GDB and getting a stack trace after the deadlock occurs will give us some more hints. Alternatively (in case you don't have gdb running on the device) you can: 1) execute "ulimit -c unlimited" on your device. 2) send a SIGABRT after the deadlock occurs and 3) get the backtrace from the core with the GDB provided in the BSP (e.g. "gdb /path/to/executable core"). Hope this helps. Regards > > regards, > > Rafael > > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From wl2776 at gmail.com Wed Dec 1 19:26:43 2010 From: wl2776 at gmail.com (wl2776) Date: Wed, 1 Dec 2010 10:26:43 -0800 (PST) Subject: [gst-devel] Framesteping in audio/video pipeline in PAUSED. How to skip buffers? Message-ID: <1291228003293-3067813.post@n4.nabble.com> http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/part-framestep.txt#n71 says that on reception of the step-done message the application must skip the same amount of buffers in the audio-sink. I use playbin2 and can obtain audiosink though its property. Which function should I call to make it skip that buffers? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Framesteping-in-audio-video-pipeline-in-PAUSED-How-to-skip-buffers-tp3067813p3067813.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wl2776 at gmail.com Wed Dec 1 20:23:48 2010 From: wl2776 at gmail.com (wl2776) Date: Wed, 1 Dec 2010 11:23:48 -0800 (PST) Subject: [gst-devel] Framesteping in audio/video pipeline in PAUSED. How to skip buffers? In-Reply-To: <1291228003293-3067813.post@n4.nabble.com> References: <1291228003293-3067813.post@n4.nabble.com> Message-ID: <1291231428564-3067896.post@n4.nabble.com> I've noticed that I always receive two step-done messages, one from video sink and another from audio sink. But the queried position anyway doesn't change without flushing seek if stepping backwards. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Framesteping-in-audio-video-pipeline-in-PAUSED-How-to-skip-buffers-tp3067813p3067896.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wmiller at sdr.com Wed Dec 1 20:25:15 2010 From: wmiller at sdr.com (Wes Miller) Date: Wed, 1 Dec 2010 11:25:15 -0800 (PST) Subject: [gst-devel] get element type name Message-ID: <1291231515380-3067899.post@n4.nabble.com> Hi All, Let's say I have a udpsrc element named "my_src" created thus: recvSrc = gst_element_factory_make( "udpsrc" , "my_src" )); If I use this construct, I get "GstUdpSrc": g_print( "%s\n", g_type_name( G_OBJECT_TYPE( recvSrc )) ); What U'd like to get, instead, is "udpsrc". Is there a function to do that? And what are the proper things to call these two items. Thanks, Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/get-element-type-name-tp3067899p3067899.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From msmith at xiph.org Wed Dec 1 20:49:19 2010 From: msmith at xiph.org (Michael Smith) Date: Wed, 1 Dec 2010 11:49:19 -0800 Subject: [gst-devel] get element type name In-Reply-To: <1291231515380-3067899.post@n4.nabble.com> References: <1291231515380-3067899.post@n4.nabble.com> Message-ID: On Wed, Dec 1, 2010 at 11:25 AM, Wes Miller wrote: > > Hi All, > > Let's say I have a udpsrc element named "my_src" created thus: > > ? ? ? ? recvSrc = gst_element_factory_make( "udpsrc" , ? ? ?"my_src" ?)); > > If I use this construct, I get "GstUdpSrc": > > ? ? ? ? g_print( "%s\n", g_type_name( G_OBJECT_TYPE( recvSrc )) ); > > What U'd like to get, instead, is "udpsrc". ?Is there a function to do that? > > And what are the proper things to call these two items. You have three items, actually: "my_src" - this is your element name "udpsrc" - this is the element factory name "GstUdpSrc" - this the element class name gst_plugin_feature_get_name() to get the name from the element factory. You can get the element factory from the element via its class object. Something like this might work: gst_plugin_feature_get_name(GST_ELEMENT_GET_CLASS(recvSrc)->elementFactory) - though it probably needs some extra casts in there too. Mike From bilboed at gmail.com Wed Dec 1 20:50:26 2010 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 01 Dec 2010 20:50:26 +0100 Subject: [gst-devel] get element type name In-Reply-To: <1291231515380-3067899.post@n4.nabble.com> References: <1291231515380-3067899.post@n4.nabble.com> Message-ID: <1291233026.5576.1.camel@deumeu> On Wed, 2010-12-01 at 11:25 -0800, Wes Miller wrote: > Hi All, > > Let's say I have a udpsrc element named "my_src" created thus: > > recvSrc = gst_element_factory_make( "udpsrc" , "my_src" )); > > If I use this construct, I get "GstUdpSrc": > > g_print( "%s\n", g_type_name( G_OBJECT_TYPE( recvSrc )) ); > > What U'd like to get, instead, is "udpsrc". Is there a function to do that? > > And what are the proper things to call these two items. So you want the factory name ? maybe there's a function to get the factory from an element and maybe there's a function to get the name of a factory hint : gst__get_ Edward > > Thanks, > > Wes > > > From wmiller at sdr.com Wed Dec 1 22:19:53 2010 From: wmiller at sdr.com (Wes Miller) Date: Wed, 1 Dec 2010 13:19:53 -0800 (PST) Subject: [gst-devel] get element type name In-Reply-To: <1291233026.5576.1.camel@deumeu> References: <1291231515380-3067899.post@n4.nabble.com> <1291233026.5576.1.camel@deumeu> Message-ID: <1291238393925-3068067.post@n4.nabble.com> Edward and Michael, Between you I got the answer I needed. Knowing what to call "something" sure help trying to find the gst_something_get_somethingelse(). g_print( " | %-*s", c3w, gst_plugin_feature_get_name( gst_element_get_factory( element )) ); However, when I check the warnings from my compiler I see that the pointer type from gst_element_get_factory() isn't what gst_plugin_feature_get_name() is looking for. Well, of course not, a factory is not a pugin feature. Well, then what about gst_element_factory_get_longname() instead. Nope, that returns "UDP packet receiver". OK, what about gst_element_factory_get_name(). Guess what doesn't exist. And, gst_element_factory_get_icon_name() seems not to exist in my gstreamer. Says it's declared in gst/gst.h. Is that right? So, I guess the question is, if I just cast the factory pointer to a plugin feature pointer and forget about it will that always work as I expect it will since I suspect that gst_plugin_feature_get_name() just wraps some g-object_get_name() with proper pointer typing. Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/get-element-type-name-tp3067899p3068067.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ensonic at hora-obscura.de Wed Dec 1 22:42:06 2010 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 01 Dec 2010 23:42:06 +0200 Subject: [gst-devel] get element type name In-Reply-To: <1291238393925-3068067.post@n4.nabble.com> References: <1291231515380-3067899.post@n4.nabble.com> <1291233026.5576.1.camel@deumeu> <1291238393925-3068067.post@n4.nabble.com> Message-ID: <4CF6C12E.6060504@hora-obscura.de> Am 01.12.2010 23:19, schrieb Wes Miller: > > Edward and Michael, > > Between you I got the answer I needed. Knowing what to call "something" > sure help trying to find the gst_something_get_somethingelse(). > > g_print( " | %-*s", c3w, gst_plugin_feature_get_name( > gst_element_get_factory( element )) ); > > However, when I check the warnings from my compiler I see that the pointer > type from gst_element_get_factory() isn't what gst_plugin_feature_get_name() > is looking for. Well, of course not, a factory is not a pugin feature. Its is as you can see nicely in the "Object Hierarchy" section of GstElementFactory API docs (consult devhelp or various online locations of the API docs). To solve your problem use a cast: gst_plugin_feature_get_name(GST_PLUGIN_FEATURE(gst_element_get_factory(element))); > > Well, then what about gst_element_factory_get_longname() instead. Nope, > that returns "UDP packet receiver". > OK, what about gst_element_factory_get_name(). Guess what doesn't exist. > And, gst_element_factory_get_icon_name() seems not to exist in my gstreamer. > Says it's declared in gst/gst.h. Is that right? If you are looking at the header that matches your binaries everything is going to be fine. Note sure how you get the idea that those are declared in that partiual header and not in some header that gets included by gst/gst.h. Stefan > > So, I guess the question is, if I just cast the factory pointer to a plugin > feature pointer and forget about it will that always work as I expect it > will since I suspect that gst_plugin_feature_get_name() just wraps some > g-object_get_name() with proper pointer typing. > > Wes > > > > From msmith at xiph.org Wed Dec 1 22:46:06 2010 From: msmith at xiph.org (Michael Smith) Date: Wed, 1 Dec 2010 13:46:06 -0800 Subject: [gst-devel] get element type name In-Reply-To: <1291238393925-3068067.post@n4.nabble.com> References: <1291231515380-3067899.post@n4.nabble.com> <1291233026.5576.1.camel@deumeu> <1291238393925-3068067.post@n4.nabble.com> Message-ID: On Wed, Dec 1, 2010 at 1:19 PM, Wes Miller wrote: > > Edward and Michael, > > Between you I got the answer I needed. ?Knowing what to call "something" > sure help trying to find the gst_something_get_somethingelse(). > > ? ? g_print( " | %-*s", ? ? ?c3w, gst_plugin_feature_get_name( > gst_element_get_factory( element )) ); > > However, when I check the warnings from my compiler ?I see that the pointer > type from gst_element_get_factory() isn't what gst_plugin_feature_get_name() > is looking for. ?Well, of course not, a factory is not a pugin feature. A factory _is_ a plugin feature. It's a subclass, as the documentation says. You can cast it; there's a macro for that. Mike From wmiller at sdr.com Wed Dec 1 23:06:34 2010 From: wmiller at sdr.com (Wes Miller) Date: Wed, 1 Dec 2010 14:06:34 -0800 (PST) Subject: [gst-devel] get element type name In-Reply-To: <4CF6C12E.6060504@hora-obscura.de> References: <1291231515380-3067899.post@n4.nabble.com> <1291233026.5576.1.camel@deumeu> <1291238393925-3068067.post@n4.nabble.com> <4CF6C12E.6060504@hora-obscura.de> Message-ID: <1291241194781-3068150.post@n4.nabble.com> Stefan, > If you are looking at the header that matches your binaries everything is > going > to be fine. Note sure how you get the idea that those are declared in that > partiual header and not in some header that gets included by gst/gst.h. By being silly enough to quote the doc that said "include This mail announces the release of GStreamer 0.10.31 "Safety Pins". GStreamer is a streaming media framework that allows the construction of graphs of elements which operate on media data. Applications using this library can do anything from real-time sound processing over playing video to capturing audio, video, and even other types of media data. Its architecture allows for adding new data types or processing capabilities simply by installing new plug-ins. GStreamer is the core module, containing libraries, headers, the basic object hierarchy, and a set of media-agnostic core elements. Highlights of this release: * new GstDateTime API * new GstElementFactory utility functions * new buffering query API to query the already-downloaded areas/ranges * bit reader utility API optimisations * other optimisations and bug fixes For more information, see http://gstreamer.freedesktop.org/modules/gstreamer.html To report bugs, request features or submit patches, please go to https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gstreamer+%28core%29 Direct links: http://gstreamer.freedesktop.org/src/gstreamer/gstreamer-0.10.31.tar.bz2 http://gstreamer.freedesktop.org/src/gstreamer/gstreamer-0.10.31.tar.gz MD5 sums (for tarballs downloaded from gstreamer.freedesktop.org): a21fb08bdb578d972c7c14e77da8fbb6 gstreamer-0.10.31.tar.bz2 939a9b62c720b12fc37c6cdc0e0bdc30 gstreamer-0.10.31.tar.gz Enjoy! -------------- next part -------------- Release notes for GStreamer?0.10.31 "Safety Pins" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the core of the GStreamer streaming media framework. The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series. The 0.10.x series has been reworked for threadsafety. It also features various feature additions and enhancements. This module, gstreamer, only contains core functionality. For actual media playback, you will need other modules. gst-plugins-base contains a basic set of well-supported plug-ins gst-plugins-good contains a set of well-supported plug-ins under our preferred license gst-plugins-ugly contains a set of well-supported plug-ins, but might pose problems for distributors gst-plugins-bad contains a set of less supported plug-ins that haven't passed the rigorous quality testing we expect Features of this release * bin: add "message-forward" property to force forwarding of messages that would usually be filtered such as ASYNC_DONE or EOS * bin: improve tracking of source elements for more efficient event dispatch * bufferlist: add function to add a list of buffers * clock: fix racy shutdown clock id leak * element: add support for arbitrary element class / factory details * element: link_many should activate pads if needed * gst: add math-compat.h header * datetime: add GstDateTime API * elementfactory: add utility functions to filter features by type * plugin: load the gst-python plugin loader with G_MODULE_BIND_LAZY * query: add buffering ranges API to retrieve informations about the areas of the stream currently buffered * value: add int64 range type * info: write debugging output to file if GST_DEBUG_FILE environment variable is set * pad: use more efficient g_object_notify_by_pspec() for caps notifies if compiling against new-enough GLib * pipeline: If the currently used clock gets lost update it the next time when going from PAUSED to playing * plugin: add release datetime field to GstPluginDesc and set it if GST_PACKAGE_RELEASE_DATETIME is defined * utils: speed up pad linking utility functions by not trying pads that will never work * adapter: add function to get a list of buffers; support 0-sized buffers * adapter: optimize gst_adapter_take() and gst_adapter_peek() a little * basesink: only answer the SEGMENT query in pull mode * basesrc: return values in stream time for the POSITION query * basetransform: allow the subclass to add new fields to caps when getting new caps from downstream * basetransform: avoid useless memcpy * basetransform: upstream caps-renegotiation fixes * bitreader: add inlined and unchecked versions of the most important functions * bytewriter: add inline and unchecked variants of all important functions * bytewriter: fix possible infinite loop caused by an overflow * queue: add "silent" property to suppress signal emission (for better performance) * queue: avoid unnecessary g_cond_signal() (for better performance) * queue: push newsegment event when linking in PLAYING * queue2: extend ring buffer to support RAM mode * queue2: in download mode, prevent range corruption due to race * queue2: don't send seeks beyond the end of the file upstream in pull mode (fixes apple trailers and youtube/html5 playback in webkit) * multiqueue: flush the data queue if downstream return WRONG_STATE too * gst-inspect: print GST_PARAM_MUTABLE_* property flags Bugs fixed in this release * 396774 : Make GstElementDetails extensible * 482147 : [queue] Issue with current time level if source task is not started * 579127 : gst-launch: disable CLOCK_LOST message handling * 594504 : Need a GType of " Date AND Time AND Timezone " * 600004 : underrun signal emits when i tested queue overrun test case from file /gstreamerXXXX/tests/check/element/queue.c * 610366 : [gstcollectpads][doc] Add a reminder for 'data' doc * 611918 : leaky queue might not push newsegment event * 618919 : Registry/Plugin Loading Memory Leak * 619522 : basetransform fix for upstream caps-renegotiation * 621299 : make simple queues faster * 621332 : BaseTransform should disable proxy alloc if downstream changes caps * 622740 : GstPad: Do not call gst_pad_accept_caps() when caps change * 623040 : Add release_datetime field to GstPluginDesc * 623121 : [queue2] downloaded areas of the media are not exposed * 623491 : make *_get_type() thread safe * 623541 : [basetransform] Implement POSITION query * 623622 : [basesink/basesrc] Should return values in stream time for POSITION query * 623806 : [pipeline] Doesn't update the clock if the currently used one gets lost and the start time did not change * 623875 : gstregistrybinary.c compatibility with glib > = 2.25.0 * 624203 : gstutils: Make gst_pad_proxy_getcaps() return empty caps if it's what the other side has * 625239 : FTBFS: ./gstreamer-decl.txt:9461: warning: GstTagList has multiple definitions. * 625295 : [info] regression: doesn't flush output stream after every debug print any longer * 625368 : gstdatetime.c doesn't compile in VS 2008 * 625862 : [docs] unused symbol GST_CAT_LEVEL_LOG_valist breaks the build * 626027 : [tag] Add GST_TAG_APPLICATION_NAME * 626181 : GstElementFactory: add listing/filtering API * 626651 : [tag] Photography/capture settings tags * 626784 : element: link_many might assert elements are in paused or playing * 627438 : gst: Add a gst_is_initialized() API * 627826 : GstInt64Range type * 627910 : Warnings emitted when -Wcast-qual used * 627959 : [queue2] on-disk buffering failing for AVI container * 628014 : Deprecate GST_FLOW_IS_FATAL/GST_FLOW_IS_SUCCESS * 628174 : New gstvalue checks cause trouble in thoggen * 628176 : [basetransform] Problems with buffer handling in inplace mode * 628408 : Use GDateTime that has been released * 629241 : Build broken with introspection using gobject-introspection from master * 629410 : GstBaseTransform: position query refers to sink pad, not source pad * 629494 : Latest gst-launch.c doesn't build in Visual Studio 2008 * 629553 : GstAdapter: timestamp not updated when empty buffer is pushed * 629831 : [API] add gst_structure_take_value() and gst_structure_id_take_value() * 629946 : Enumerations have incorrect names of enum values (GEnumValue.value_name) * 630257 : GST_DEBUG_DUMP_DOT_DIR not working anymore * 630436 : basesink: renderdelay needs to be subtracted in adjust_time() * 630437 : basetransform: Make a WARNING into a DEBUG statement * 630439 : clock: fix racy shutdown clock id leak * 631755 : Fix build with glib 2.21.3 * 631853 : [queue2] deadlock when using temp-location and dispatch-properties * 632236 : [gst-inspect] unhelpful uri handler output * 632433 : [basesink] hangs/drops going to PLAYING following flushing step in PAUSED * 632977 : [queue2] qtdemux causes soup to request seeks past the end of the range * 633147 : Simple reverse negotiation pipeline is broken. * 633886 : Visual Studio emits warnings about double defined _USE_MATH_DEFINES * 635031 : [datetime] Fix unix epoch handling * 635389 : Include information on exported packages in GIRs * 635869 : GST_BOILERPLATE_FULL causes warnings in user C++ code * 633176 : recent multiqueue changes break DVD playback API changed in this release - API additions: * gst_is_initialized * gst_buffer_list_iterator_add_list * GstBin:message-forward * GST_TYPE_DATE_TIME * gst_date_time_get_day * gst_date_time_get_hour * gst_date_time_get_microsecond * gst_date_time_get_minute * gst_date_time_get_month * gst_date_time_get_second * gst_date_time_get_time_zone_offset * gst_date_time_get_type * gst_date_time_get_year * gst_date_time_new * gst_date_time_new_from_unix_epoch_local_time * gst_date_time_new_from_unix_epoch_utc * gst_date_time_new_local_time * gst_date_time_new_now_local_time * gst_date_time_new_now_utc * gst_date_time_ref * gst_date_time_unref * gst_tag_list_get_date_time * gst_tag_list_get_date_time_index * GST_TAG_GEO_LOCATION_HORIZONTAL_ERROR * GST_TAG_APPLICATION_DATA * GST_TAG_APPLICATION_NAME * GST_TAG_DATE_TIME * GST_ELEMENT_IS_SOURCE * gst_element_class_set_documentation_uri * gst_element_class_set_icon_name * gst_element_factory_get_documentation_uri * gst_element_factory_get_icon_name * gst_element_factory_list_filter * gst_element_factory_list_get_elements * gst_element_factory_list_is_type * GstElementFactoryListType * GST_ELEMENT_FACTORY_TYPE_ANY * GST_ELEMENT_FACTORY_TYPE_AUDIOVIDEO_SINKS * GST_ELEMENT_FACTORY_TYPE_AUDIO_ENCODER * GST_ELEMENT_FACTORY_TYPE_DECODABLE * GST_ELEMENT_FACTORY_TYPE_DECODER * GST_ELEMENT_FACTORY_TYPE_DEMUXER * GST_ELEMENT_FACTORY_TYPE_DEPAYLOADER * GST_ELEMENT_FACTORY_TYPE_ENCODER * GST_ELEMENT_FACTORY_TYPE_FORMATTER * GST_ELEMENT_FACTORY_TYPE_MAX_ELEMENTS * GST_ELEMENT_FACTORY_TYPE_MEDIA_AUDIO * GST_ELEMENT_FACTORY_TYPE_MEDIA_IMAGE * GST_ELEMENT_FACTORY_TYPE_MEDIA_METADATA * GST_ELEMENT_FACTORY_TYPE_MEDIA_SUBTITLE * GST_ELEMENT_FACTORY_TYPE_MEDIA_VIDEO * GST_ELEMENT_FACTORY_TYPE_MEDIA_ANY * GST_ELEMENT_FACTORY_TYPE_MUXER * GST_ELEMENT_FACTORY_TYPE_PARSER * GST_ELEMENT_FACTORY_TYPE_PAYLOADER * GST_ELEMENT_FACTORY_TYPE_SINK * GST_ELEMENT_FACTORY_TYPE_SRC * GST_ELEMENT_FACTORY_TYPE_VIDEO_ENCODER * GST_ELEMENT_FACTORY_KLASS_DECODER * GST_ELEMENT_FACTORY_KLASS_ENCODER * GST_ELEMENT_FACTORY_KLASS_SINK * GST_ELEMENT_FACTORY_KLASS_SRC * GST_ELEMENT_FACTORY_KLASS_MUXER * GST_ELEMENT_FACTORY_KLASS_DEMUXER * GST_ELEMENT_FACTORY_KLASS_PARSER * GST_ELEMENT_FACTORY_KLASS_PAYLOADER * GST_ELEMENT_FACTORY_KLASS_DEPAYLOADER * GST_ELEMENT_FACTORY_KLASS_FORMATTER * GST_ELEMENT_FACTORY_KLASS_MEDIA_VIDEO * GST_ELEMENT_FACTORY_KLASS_MEDIA_AUDIO * GST_ELEMENT_FACTORY_KLASS_MEDIA_IMAGE * GST_ELEMENT_FACTORY_KLASS_MEDIA_SUBTITLE * GST_ELEMENT_FACTORY_KLASS_MEDIA_METADATA * gst_plugin_feature_list_debug * gst_plugin_feature_rank_compare_func * gst_query_add_buffering_range * gst_query_get_n_buffering_ranges * gst_query_parse_nth_buffering_range * gst_structure_get_date_time * gst_structure_id_take_value * gst_structure_take_value * GST_TYPE_INT64_RANGE * gst_int64_range_get_type * gst_util_fraction_compare * gst_value_get_int64_range_max * gst_value_get_int64_range_min * gst_value_set_int64_range * GST_VALUE_HOLDS_DATE_TIME * gst_adapter_take_list * gst_bit_reader_skip_unchecked * gst_bit_reader_skip_to_byte_unchecked * gst_bit_reader_get_bits_uint16_unchecked * gst_bit_reader_get_bits_uint32_unchecked * gst_bit_reader_get_bits_uint64_unchecked * gst_bit_reader_get_bits_uint8_unchecked * gst_bit_reader_peek_bits_uint16_unchecked * gst_bit_reader_peek_bits_uint32_unchecked * gst_bit_reader_peek_bits_uint64_unchecked * gst_bit_reader_peek_bits_uint8_unchecked * gst_byte_writer_put_int8_unchecked * gst_byte_writer_put_int16_be_unchecked * gst_byte_writer_put_int16_le_unchecked * gst_byte_writer_put_int24_be_unchecked * gst_byte_writer_put_int24_le_unchecked * gst_byte_writer_put_int32_be_unchecked * gst_byte_writer_put_int32_le_unchecked * gst_byte_writer_put_int64_be_unchecked * gst_byte_writer_put_int64_le_unchecked * gst_byte_writer_put_uint8_unchecked * gst_byte_writer_put_uint16_be_unchecked * gst_byte_writer_put_uint16_le_unchecked * gst_byte_writer_put_uint24_be_unchecked * gst_byte_writer_put_uint24_le_unchecked * gst_byte_writer_put_uint32_be_unchecked * gst_byte_writer_put_uint32_le_unchecked * gst_byte_writer_put_uint64_be_unchecked * gst_byte_writer_put_uint64_le_unchecked * gst_byte_writer_put_float32_be_unchecked * gst_byte_writer_put_float32_le_unchecked * gst_byte_writer_put_float64_be_unchecked * gst_byte_writer_put_float64_le_unchecked * gst_byte_writer_put_data_unchecked * gst_byte_writer_fill_unchecked - API deprecations: * GST_FLOW_IS_FATAL * GST_FLOW_IS_SUCCESS Download You can find source releases of gstreamer in the download directory: http://gstreamer.freedesktop.org/src/gstreamer/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications Applications ported to GStreamer 0.10 include Totem, RhythmBox, Sound-Juicer, Gnome Media, Flumotion, Amarok, Jamboree, Pitivi, Istanbul, AnnoAmp, Elisa, and others. Let us know if you want to be added to this list. Contributors to this release * Alessandro Decina * Arun Raghavan * Colin Walters * David Hoyt * David Schleef * Edward Hervey * Evan Nemerson * Havard Graff * Jan Schmidt * Jeffrey S. Smith * Jonas Holmberg * Jonathan Matthew * Mark Nauwelaerts * Michael Bunk * Ognyan Tonchev * Olivier Cr?te * Philippe Normand * Rob Clark * Robert Swain * Sebastian Dr?ge * Shixin Zeng * Stefan Kost * Thiago Santos * Thijs Vermeir * Tim-Philipp M?ller * Trond Andersen * Vincent Penquerc'h * Vladimir Eremeev * Wim Taymans * reynaldo ? From t.i.m at zen.co.uk Thu Dec 2 00:29:33 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 01 Dec 2010 23:29:33 +0000 Subject: [gst-devel] RELEASE: GStreamer Base Plug-ins 0.10.31 "Dance Like It's 1982" Message-ID: <1291246173.12352.137.camel@zingle> This mail announces the release of GStreamer Base Plug-ins 0.10.31 "Dance Like It's 1982". GStreamer Base Plug-ins is a well-groomed and well-maintained collection of GStreamer plug-ins and elements, spanning the range of possible types of elements one would want to write for GStreamer. It also contains helper libraries and base classes useful for writing elements. A wide range of video and audio decoders, encoders, and filters are included. Highlights of this release: * more Orc SIMD optimisations * GstDiscoverer API to collect tags and metadata from files * playbin2 and decodebin2 now officially declared as stable (with playbin/decodebin officially deprecated) * new GstXOverlay API so that window handles can be passed correctly on all platforms * gst_video_convert_frame*() utility functions * many new photography/camera/capture tags * utility functions to extract AAC/MPEG-4/H.264 profile/level information and put it into caps, used in typefinder * countless other bug fixes and enhancements For more information, see http://gstreamer.freedesktop.org/modules/gst-plugins-base.html To report bugs, request features or submit patches, please go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-plugins-base Direct links: http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-0.10.31.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-0.10.31.tar.gz MD5 sums (for tarballs downloaded from gstreamer.freedesktop.org): 9baa0d87e81c88b2477a3554ab629c46 gst-plugins-base-0.10.31.tar.bz2 79a2455ce5f3ff4c239d44a70f606612 gst-plugins-base-0.10.31.tar.gz Enjoy! -------------- next part -------------- Release notes for GStreamer Base Plug-ins?0.10.31 "Dance Like It's 1982" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer Base Plug-ins. The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series. This module contains a set of reference plugins, base classes for other plugins, and helper libraries. This module is kept up-to-date together with the core developments. Element writers should look at the elements in this module as a reference for their development. This module contains elements for, among others: device plugins: x(v)imagesink, alsa, v4lsrc, cdparanoia containers: ogg codecs: vorbis, theora text: textoverlay, subparse sources: audiotestsrc, videotestsrc, gnomevfssrc, giosrc, appsrc network: tcp typefind functions audio processing: audioconvert, adder, audiorate, audioresample, volume visualisation: libvisual video processing: ffmpegcolorspace aggregate elements: uridecodebin, playbin2, decodebin2, decodebin, playbin Other modules containing plug-ins are: gst-plugins-good contains a set of well-supported plug-ins under our preferred license gst-plugins-ugly contains a set of well-supported plug-ins, but might pose problems for distributors gst-plugins-bad contains a set of less supported plug-ins that haven't passed the rigorous quality testing we expect Features of this release * adder: Make sure FLUSH_STOP is always sent after a flushing seek * alsasrc, alsasink: add "card-name" property to get the card name in addition to the device name * appsrc: don't override buffer caps if appsrc caps are NULL; fix element classification * audioclock: add a function to invalidate the clock * audioconvert: optimise remaining conversion code paths with Orc as well * baseaudiosink,baseaudiosrc: post clock-provide and clock-lost messages when going from/to READY to/from PAUSED * baseaudiosink: subtract the render_delay from our latency * decodebin2: don't add non prerolled stream to topology * ffmpegcolorspace: add support for A420 and fix support for 8 bit paletted RGB and IYU1 * gnomevfsrc: set GST_PARAM_MUTABLE_READY flag on the "handle" property * libvisual: add latency query; only drop frames that are really too old * multifdsink: gdp protocol is deprecated. People should use gdppay instead * oggdemux: fix seeking with negative rate with skeleton; fix wrong flowreturn handling * pbutils: AAC profile and level detection utility functions * pbutils: H.264 and MPEG-4 profile and level extraction utility functions * pbutils: new GstDiscoverer utility API for extracting metadata and tags * playbin2, decodebin2: declare stable, deprecate the old playbin/decodebin * playbin2, uridecodebin: add property to configure ring buffer size * rtcpbuffer: add function to manipulation the data in RTCP feedback packets * rtpbuffer: add functions to add RFC 5285 header extensions to GstBufferLists * rtpbuffer: add function to add RTP header extensions with a two bytes header * rtpbuffer: add function to append RFC 5285 one byte header extensions * rtpbuffer: add function to parse RFC 5285 header extensions * rtpbuffer: add function to read RFC 5285 header extensions from GstBufferLists * rtpbuffer: add function to transform a GstBuffer into a GstBufferList * rtsp: improve rtsp timeout calculation and handling * sdp: add methods to convert between uri and message * tags: try ISO-8859-1 as second fallback in case WINDOWS-1252 is not supported * tags: add many more photography/capture tags * tags: EXIF and XMP tag handling improvements * textoverlay: add support for NV12, NV21 and AYUV; configurable text color and position * theoradec: expose telemetry properties only if libtheora was compiled with --enable-telemetry * theoraenc: add support for two-pass encoding; allow change of bitrate and quality on-the-fly * tools: standalone gst-discoverer-0.10 tool for discovering media file properties * typefinding: detect avc1 ftyp as video/quicktime * typefinding: export 3gp profile in caps * typefinding: detect enhanced AC-3 * typefinding: extend AAC typefinder to detect LOAS streams * typefinding: fix ADTS caps stream-format detail * typefinding: more reliable mpeg-ts typefinding * uridecodebin: Only enable progressive downloading if the upstream duration in bytes is known * video: add gst_video_convert_frame*() utility functions * videorate: fixate the pixel-aspect-ratio if necessary * videorate: mark duplicated frames with the GAP flag * videoscale: add support for adding black borders to keep the DAR if necessary ("add-borders" property) * videoscale: Fix caps fixating if the height is fixed but the width isn't * videoscale: only set the PAR if the caps already had a PAR * videoscale: refactor using more Orc code * videotestsrc: new patterns: solid-color, ball, bar and smpte100 * videotestsrc: add "foreground-color" and "background-color" properties, deprecate "colorspec" property * videotestsrc: add support for UYVP format, fix NV21 rendering * volume: use Orc to optimise many code paths * vorbisdec: decode pending buffers upon EOS when doing reverse playback * xoverlay: add set_window_handle() with guintptr argument, deprecate set_xwindow_id() which doesn't work on some platforms * xoverlay: allow render rectangle coordinates to be negative Bugs fixed in this release * 628028 : [uridecodebin] Don't enable progressive downloading for live streams * 623846 : typefinding: add support for " enhanced ac3 " (eac3) * 602437 : [playbin2] [gapless] Completely broken when switching between files with audio/video only * 612264 : Notification needed when the first buffer is pushed by the basertppayloader * 615471 : [videoscale] Interlaced handling makes output worse than no interlaced handling at all * 616392 : videotestsrc colorspec=0/1 does not affect color-matrix in caps * 617314 : pbutils: Add codec-specific utility functions for AAC, H.264, MPEG-4 video * 617506 : [videoscale] Add support for adding black borders if necessary to keep the DAR * 620291 : typefindfunctions: Export 3gp profile in caps * 623663 : [typefinding] mpeg-ts file detected as audio/mpeg * 623807 : [audioclock] Add gst_audio_clock_new_full() with GDestroyNotify for the user_data * 623837 : typefind: only associate .webm with WebM * 623918 : [typefind] Extend AAC typefinder to detect LOAS stream * 624598 : [adder] crash in orc_sse_set_mxcsr() * 624656 : [videoscale] UYVY scaling broken, introduces green lines * 624919 : [videotestsrc] add solid color pattern * 624920 : [textoverlay] configurable text color and position * 624949 : [playbin2] declare playbin2 stable * 625001 : [examples] Don't use GdkDraw/GdkGC * 625118 : [playbin2] Race condition with EOS events in gapless mode * 625944 : [pbutils] GstDiscoverer - API to discover metadata and stream information * 626125 : [alsa] Conditional jump or move depends on uninitialised value(s) * 626570 : [tag] Add resolution tags * 626581 : [playbin2] regression: occasional deadlocks in streamsynchronizer * 626621 : [playbin2] streamsynchronizer regressions * 626629 : [ffmpegcolorspace] doesn't handle palettes any longer * 626718 : playback: Delay usage of GstFactoryList * 627203 : [alsa] alsasrc and alsasink should expose card name via property * 627297 : [regression] build-failure * 627565 : [xoverlay][win64] gulong can't hold a HANDLE * 627768 : add NV12 support to textoverlay * 627780 : GstClockOverlay re-renders string even if it hasn't changed, resulting in very high CPU usage. * 627924 : riff: add support for 2vuy * 628009 : [volume] Float processing with orc broken * 628400 : [videorate] does not generate buffers to fill the duration of the last frame * 628500 : videotestsrc: add moving color bars pattern * 628747 : gst-plugins-base: unable to build because of compiler warning in libggsttag * 629157 : Move video frame conversion from playback plugin to libgstvideo * 629672 : gnomevfsrsrc: " handle " property should also have the GST_PARAM_MUTABLE_READY flag * 629848 : build problem with current gtk+: implicit declaration of function 'gdk_draw_rectangle', GtkStyle' has no member named 'black_gc' * 630303 : theoraenc: Make the bitrate/quality dynamically modifiable * 630353 : [appsrc] Avoid losing buffers' caps * 630440 : ringbuffer: use g_once for type-init * 630443 : baseaudiosink: Add getter and setter for drift tolerance * 630471 : [tag] ligatures " ? " and " ? " are not supported in freeform strings * 630496 : seek example: add new #define to set seek bar graininess * 630802 : videotestsrc.c doesn't compile in Visual Studio 2008 * 631128 : Add methods to manipulate RFC 5285 header extensions * 631312 : [streamsynchronizer] Advancing segments too much * 631633 : [oggdemux] fix seeking with negative rate with skeleton * 631703 : [oggdemux] sintel ogv delay when playing * 631756 : Fix build with glib 2.21.3 * 631773 : [tags] Add new exif tags: sharpness, metering mode, file/capturing source * 631774 : [xvimagesink] sets non-simple caps on pad_alloced buffer * 632167 : [oggdemux] doesn't parse/push all headers in pull mode * 632653 : [seek] Don't use deprecated combo box API * 632656 : [uridecodebin] internal decodebin2 might fail to reach PLAYING in streaming case * 632789 : [PATCH] tests/icles/: adapted test-colorkey.c and test-xoverlay.c to deprecation of gtk_widget_hide_all * 632809 : [regression] build failure in 0.10.30.2 in tools/ * 632988 : [discoverer] gst_caps_ref() critical for substreams of unknown streams * 633023 : [discoverer] Add versionized gst-discoverer tool * 633203 : Regression: streamsynchroniser + disabled deinterlacing = no DVD menu highlights/subtitles * 633311 : discoverer: use specific types in getters, rename some boolean getters * 633336 : [discoverer] Move documentation into the correct section * 633455 : [rtsp] don't let the rtsp connection timeout * 634014 : GTK+3 is a moving target, lets not compile against it by default. * 634584 : decodebin2 docs should mention that " new-decoded-pad " signal may be emitted after " no-more-pads " * 635067 : [*decodebin*] pad template leaked * 635392 : Missing information on exported packages from GIRs * 621349 : [theoraenc] Implement two-pass encoding * 628488 : [theoradec] add properties to enable telemetry overlay * 629746 : Enumerations have incorrect names of enum values (GEnumValue.value_name) * 626869 : The RTP depayloader is sometimes sending partial frames down the pipeline without the DISCONT bit set API changed in this release - API additions: * gst_audio_clock_invalidate() * gst_audio_clock_new_full() * gst_base_audio_sink_get_drift_tolerance() * gst_base_audio_sink_set_drift_tolerance() * gst_x_overlay_got_window_handle() * gst_x_overlay_set_window_handle() * GstXOverlay::set_window_handle() * gst_codec_utils_aac_caps_set_level_and_profile() * gst_codec_utils_aac_get_level() * gst_codec_utils_aac_get_profile() * gst_codec_utils_aac_get_sample_rate_from_index() * gst_codec_utils_h264_caps_set_level_and_profile() * gst_codec_utils_h264_get_level() * gst_codec_utils_h264_get_profile() * gst_codec_utils_mpeg4video_caps_set_level_and_profile() * gst_codec_utils_mpeg4video_get_level() * gst_codec_utils_mpeg4video_get_profile() * gst_discoverer_audio_info_get_bitrate() * gst_discoverer_audio_info_get_channels() * gst_discoverer_audio_info_get_depth() * gst_discoverer_audio_info_get_max_bitrate() * gst_discoverer_audio_info_get_sample_rate() * gst_discoverer_audio_info_get_type() * gst_discoverer_container_info_get_streams() * gst_discoverer_container_info_get_type() * gst_discoverer_discover_uri() * gst_discoverer_discover_uri_async() * gst_discoverer_get_type() * gst_discoverer_info_copy() * gst_discoverer_info_get_audio_streams() * gst_discoverer_info_get_container_streams() * gst_discoverer_info_get_duration() * gst_discoverer_info_get_misc() * gst_discoverer_info_get_result() * gst_discoverer_info_get_stream_info() * gst_discoverer_info_get_stream_list() * gst_discoverer_info_get_streams() * gst_discoverer_info_get_tags() * gst_discoverer_info_get_type() * gst_discoverer_info_get_uri() * gst_discoverer_info_get_video_streams() * gst_discoverer_new() * gst_discoverer_result_get_type() * gst_discoverer_start() * gst_discoverer_stop() * gst_discoverer_stream_info_get_caps() * gst_discoverer_stream_info_get_misc() * gst_discoverer_stream_info_get_next() * gst_discoverer_stream_info_get_previous() * gst_discoverer_stream_info_get_stream_type_nick() * gst_discoverer_stream_info_get_tags() * gst_discoverer_stream_info_get_type() * gst_discoverer_stream_info_list_free() * gst_discoverer_video_info_get_bitrate() * gst_discoverer_video_info_get_depth() * gst_discoverer_video_info_get_framerate_denom() * gst_discoverer_video_info_get_framerate_num() * gst_discoverer_video_info_get_height() * gst_discoverer_video_info_get_max_bitrate() * gst_discoverer_video_info_get_par_denom() * gst_discoverer_video_info_get_par_num() * gst_discoverer_video_info_get_type() * gst_discoverer_video_info_get_width() * gst_discoverer_video_info_is_image() * gst_discoverer_video_info_is_interlaced() * GST_PLUGINS_BASE_VERSION_MAJOR * GST_PLUGINS_BASE_VERSION_MINOR * GST_PLUGINS_BASE_VERSION_MICRO * GST_PLUGINS_BASE_VERSION_NANO * GST_CHECK_PLUGINS_BASE_VERSION * gst_plugins_base_version() * gst_plugins_base_version_string() * gst_rtcp_packet_fb_get_fci() * gst_rtcp_packet_fb_get_fci_length() * gst_rtcp_packet_fb_set_fci_length() * gst_rtp_buffer_add_extension_onebyte_header() * gst_rtp_buffer_add_extension_twobytes_header() * gst_rtp_buffer_get_extension_onebyte_header() * gst_rtp_buffer_get_extension_twobytes_header() * gst_rtp_buffer_list_add_extension_onebyte_header() * gst_rtp_buffer_list_add_extension_twobytes_header() * gst_rtp_buffer_list_from_buffer() * gst_rtp_buffer_list_get_extension_onebyte_header() * gst_rtp_buffer_list_get_extension_twobytes_header() * gst_sdp_message_as_uri() * gst_sdp_message_parse_uri() * GST_TAG_CAPTURING_SOURCE * GST_TAG_CAPTURING_METERING_MODE * GST_TAG_CAPTURING_SHARPNESS * GST_TAG_IMAGE_HORIZONTAL_PPI * GST_TAG_IMAGE_VERTICAL_PPI * GST_TAG_CAPTURING_FLASH_FIRED * GST_TAG_CAPTURING_FLASH_MODE * GST_TAG_CAPTURING_EXPOSURE_PROGRAM * GST_TAG_CAPTURING_EXPOSURE_MODE * GST_TAG_CAPTURING_SCENE_CAPTURE_TYPE * GST_TAG_CAPTURING_GAIN_ADJUSTMENT * GST_TAG_CAPTURING_WHITE_BALANCE * GST_TAG_CAPTURING_CONTRAST * GST_TAG_CAPTURING_SATURATION * GST_TAG_CAPTURING_SHUTTER_SPEED * GST_TAG_CAPTURING_FOCAL_RATIO * GST_TAG_CAPTURING_FOCAL_LENGTH * GST_TAG_CAPTURING_DIGITAL_ZOOM_RATIO * GST_TAG_CAPTURING_ISO_SPEED * GST_VIDEO_FORMAT_UYVP * GST_VIDEO_FORMAT_A420 * gst_video_convert_frame() * gst_video_convert_frame_async() * GstTextOverlay:xpos * GstTextOverlay:ypos * GstTextOverlay:color * GstVideoTestSrc:solid-color * GstVideoTestSrc::foreground-color * GstVideoTestSrc::background-color - API deprecations: * gst_x_overlay_set_xwindow_id() * gst_x_overlay_got_xwindow_id() * GstXOverlay::set_xwindow_id() Download You can find source releases of gst-plugins-base in the download directory: http://gstreamer.freedesktop.org/src/gst-plugins-base/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications Contributors to this release * Alessandro Decina * Alexey Fisher * American Dynamics * Andrzej K. Haczewski * Arun Raghavan * Chris Shoemaker * David Schleef * Edward Hervey * Evan Nemerson * Guillaume Emont * Harri M?h?nen * Havard Graff * Jan Schmidt * Leo Singer * Mark Nauwelaerts * Matthias Clasen * Olivier Cr?te * Parthasarathi Susarla * Philip J?genstedt * Philippe Normand * Ren? Stadler * Rob Clark * Robert Swain * Sebastian Dr?ge * Sreerenj Balachandran * Stefan Kost * Thiago Santos * Thijs Vermeir * Tim-Philipp M?ller * Tristan Matthews * Vincent Penquerc'h * Vladimir * Wim Taymans * Zaheer Abbas Merali ? From t.i.m at zen.co.uk Thu Dec 2 00:35:41 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 01 Dec 2010 23:35:41 +0000 Subject: [gst-devel] RELEASE: GStreamer Good Plug-ins 0.10.26 "Escapades" Message-ID: <1291246541.12352.143.camel@zingle> This mail announces the release of GStreamer Good Plug-ins 0.10.26 "Escapades". GStreamer Good Plug-ins is a set of plug-ins that we consider to have good quality code and correct functionality, under our preferred license (LGPL for the plug-in code, LGPL or LGPL-compatible for the supporting library). Highlights of this release: * more Orc optimisations * new videomixer2 element with better behaviour * new G722 RTP payloader and depayloader elements * add support for RTSP initiation through SDP files * many other bug fixes and enhancements For more information, see http://gstreamer.freedesktop.org/modules/gst-plugins-good.html To file bugs, request features or submit patches, please go to https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-plugins-good Direct links: http://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-0.10.26.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-0.10.26.tar.gz MD5 sums (for tarballs downloaded from gstreamer.freedesktop.org): e1ed191adbf81edff04f348f8ce8e198 gst-plugins-good-0.10.26.tar.bz2 4d9e3a17050da013b0b2e4c23aee10a1 gst-plugins-good-0.10.26.tar.gz Enjoy! -------------- next part -------------- Release notes for GStreamer Good Plug-ins?0.10.26 "Escapades" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer Good Plug-ins. The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series. "Such ingratitude. After all the times I've saved your life." A collection of plug-ins you'd want to have right next to you on the battlefield. Shooting sharp and making no mistakes, these plug-ins have it all: good looks, good code, and good licensing. Documented and dressed up in tests. If you're looking for a role model to base your own plug-in on, here it is. If you find a plot hole or a badly lip-synced line of code in them, let us know - it is a matter of honour for us to ensure Blondie doesn't look like he's been walking 100 miles through the desert without water. This module contains a set of plug-ins that we consider to have good quality code, correct functionality, our preferred license (LGPL for the plug-in code, LGPL or LGPL-compatible for the supporting library). We believe distributors can safely ship these plug-ins. People writing elements should base their code on these elements. Other modules containing plug-ins are: gst-plugins-base contains a basic set of well-supported plug-ins gst-plugins-ugly contains a set of well-supported plug-ins, but might pose problems for distributors gst-plugins-bad contains a set of less supported plug-ins that haven't passed the rigorous quality testing we expect Features of this release * alphacolor: make passthrough work * avidemux: reverse playback fixes; prevent overlap of subsequent fragments * deinterlace: remove assembly code in favor of orc * dvdemux: parse SMPTE time codes * flvdemux: parse and use cts (fixes jittery H.264 playback in some cases) * flvmux: resend onMetada tag when tags changes in streamable mode * g729pay: extend from right parent * gconf: Don't install schemas when GConf is disabled * goom, goom2k1: add latency compensation code, report latency correctly * gstrtpjpegpay: Added Define Restart Interval (DRI) Marker * h264depay: always mark the codec_data as keyframe * icydemux: forward tag events * id3v2mux: Add mapping for album artist * imagefreeze: generate a perfectly timestamped stream * level: avoid division by zero on silence * matroskademux: more robustness for parse errors and corner-cases * matroskademux: extract H.264 profile and level and set on caps * matroskamux: reduce newsegment event spam and set discont flag where needed * pulse: allow setting of pulse stream properties * pulse: fix device_description in READY * pulsesink: Add "client" property to set the PA client name * pulsesink: share the PA context between all clients with the same name * qtdemux: export AAC/MPEG-4/H.264 profile and level in caps * rtp: add G722 payloader and depayloader elements * rtpamr(de)pay: support AMR-WB SID frame * rtpamrpay: proper duration for multiple frame payload; properly support perfect-rtptime * rtpbin: add "ntp-sync" property and "use-pipeline-clock" properties * rtpg729pay: properly support perfect-rtptime * rtph264depay: only set delta unit on all-non-key units * rtpmanager: provide additional statistics * rtpmp4adepay: grab the sampling rate and put into caps * rtpmparobustdepay: properly insert dummy buffers; use valid bitrate for dummy frame * rtpmpvpay: fix timestamping of rtp buffers * rtpsession: Add the option to auto-discover the RTP bandwidth * rtpsession: Calculate RTCP bandwidth as a fraction of the RTP bandwidth * rtpsession: Count sent RTCP packets after they have been finished * rtpsession: relax third-party collision detection * rtpstats: Rectify description of current_time in RTPArrivalStats * rtspext: stop configuration on first failure * rtspsrc: Add property to configure udpsrc buffer size * rtspsrc: add rtsp-sdp protocol support * rtspsrc: don't add /UDP in the transport, it's the default * rtspsrc: fix duration reporting * rtspsrc: handle stale digest authentication session data * rtspsrc: use sdp uri parse method * shapewipe: add optional border parameter and slowdown animation * shapewipe: Force format to AYUV in the example pipeline for the same reason * shapewipe: Force the input to AYUV to prevent negotiation failures in videomixer * spectrum: only aggregate magnitude/phase if user asks for it, performance fixes * v4l2src: add controllable colorbalance parameters, add decimate property * v4l2src: fix using mpegts via the mmap interface; use GstBaseSrc::block-size as fallback size * videomixer2: new videomixer2 element that behaves better than videomixer * vrawdepay: handle invalid payload better Bugs fixed in this release * 625825 : cannot link rtpmp4adepay ! aacparse * 629047 : segfault in seek matroskademux * 537544 : [pulse] allow setting pa context properties * 628996 : pulsesink broken after shared context patch (bug #624338) * 529672 : Big latency and bad framerate while mixing multiple live streams * 581294 : rtspext: extensions configure_stream methods conflict * 598915 : qtdemux: propagate jpeg2000 header data in image/x-j2c * 612313 : qtdemux: Post AAC profile/level in caps * 616521 : qtdemux: Export MPEG-4 video profile and level in stream caps * 617318 : matroskademux, qtdemux: Use pbutils for H.264 profile/level extraction * 620790 : [matroskademux] general stream error when trying to play certain .mkv file * 622390 : [v4l2] add controllable color balance properties / programmable camera * 624338 : [pulsesink] Handle pulse context separate from the ringbuffers and share them * 625547 : imagefreeze unit test fails occasionally * 626048 : [videomixer] needs mode that syncs streams based on timestamps * 626518 : [imagefreeze] better caps negotiation * 627162 : [pulse] better fallback return value for gst_pulse_client_name() * 627174 : [pulsesink] new property to tune the PA client name * 627289 : souphttpsrc: tweak error messages * 627341 : wavparse: strange handling of files less than 12 bytes * 627796 : rtpbin: add ntp clock sync * 628020 : [pulsesink] assertion failure in change_state NULL- > READY * 628058 : Need a way to set the SO_RCVBUF property on rtsp-based sockets. * 628127 : jpeg rtp payloader crashes when there is corruption in the jpeg byte stream. * 628214 : Add support to RTSP initiation through SDP files * 628349 : [v4l2src] Doesn't support capturing mpegts using mmap * 628454 : Matroska demuxer doesn't handle DATE tag if it contains only a year number * 628608 : [alphacolor] element classification is wrong * 629018 : rtpjpegpay: unable to build because of uninitialized variable warning * 629522 : [rtpjpegpay] add support for Define Restart Interval (DRI) * 629839 : [qtdemux] Update xmp tags parsing * 629896 : Error compiling raw1394 (without iec61883) * 630088 : [flvdemux] jerky h.264 video playback * 630205 : [icydemux] Forward tag events downstrem * 630256 : rtph264-pay/depay: doesn't respect timestamps from incomming buffers * 630317 : Getting pulsesink device names doesn't work like for alsasink * 630378 : speexenc/speexdec crash with MSVC * 630446 : rtpmanager: provide additional statistics * 630447 : rtpsession: relax third-party collision detection * 630449 : rtpbin: Unlock before adding pad in new_payload_found * 630451 : rtpbin: Handle rysnc of iterator when looking for free pad name * 630452 : rtpbin: Make cleaning up sources in rtp_session_on_timeout MT safe * 630457 : rtpmanager: packet lost should not be a warning. * 630458 : level: avoid division by zero on silence * 630500 : [rtspsrc] does rtsp setup message always need " /UDP " string? * 630888 : v4l2sink does not cope with v4l2loopback kernel module * 631082 : rtpjitterbuffer: improve document reference * 631303 : [goom] qos warnings if source is GstAudioSrc * 631330 : [flvmux][PATCH] Resend updated onMetada tag when tags changes in streamable mode * 631996 : [h264depay] regression: rtsp://stream.zoovision.com/KibaEp1n900.3gp * 632548 : [rtspsrc] regression; fails to report duration * 632553 : --disable-gconf still tries to install schemas * 632682 : [matroskademux] Handle missing CodecPrivate for Vorbis/Theora * 632945 : rtph264depay in access-unit=true mode does not aggregate the delta unit flag correctly * 633205 : Fix for navigation events in videoflip * 633212 : [goom] Return not-negotiated when bps is unknown * 633970 : [icydemux] broken taglist handling * 635532 : rtspsrc: unexpected eos when using authentication (regression) * 635843 : [rtph264depay] segfault on empty payload * 636179 : [deinterlace] Fields in wrong order * 626463 : [matroskademux] " reading large block of size 14688496 not supported " * 628894 : [matroskademux] sloppy reverse playback * 633294 : deinterlace breaks some DVD menu scenarios Download You can find source releases of gst-plugins-good in the download directory: http://gstreamer.freedesktop.org/src/gst-plugins-good/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications Contributors to this release * Alessandro Decina * American Dynamics * Andoni Morales Alastruey * Arun Raghavan * Bastien Nocera * David Hoyt * David Schleef * Edward Hervey * Havard Graff * IOhannes m zm?lnig * Jan Schmidt * Jonathan Matthew * Mark Nauwelaerts * Olivier Cr?te * Pascal Buhler * Pavel Kostyuchenko * Philip J?genstedt * Philippe Normand * Ren? Stadler * Robert Swain * Sebastian Dr?ge * Sjoerd Simons * Stefan Kost * Thiago Santos * Thibault Saunier * Thijs Vermeir * Tim-Philipp M?ller * Trond Andersen * Vladimir Eremeev * Wim Taymans * Zaheer Abbas Merali ? From parveen.jain at one97.net Thu Dec 2 06:12:53 2010 From: parveen.jain at one97.net (parveen.jain) Date: Thu, 2 Dec 2010 10:42:53 +0530 Subject: [gst-devel] G729 with gstreamer Message-ID: <4cf72c33.e83cec0a.3465.056b@mx.google.com> Hi All, Does Gstreamer(or any of its plugin ) supports the G729 encoder/decoder? Is yes, how is different from intel IPP library support of G729? Regards, Parveen Jain -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.stowers.lists at gmail.com Thu Dec 2 06:58:22 2010 From: john.stowers.lists at gmail.com (John Stowers) Date: Thu, 2 Dec 2010 18:58:22 +1300 Subject: [gst-devel] Another GstAppSrc Question Message-ID: Hi All, I see quite a few similar questions to these on the gst-devel mailing list, but none of them allowed me to get past my problem. I'm building a streaming video thing using gstreamer, the prototype (works fine) is sender: gst-launch -v videotestsrc ! video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink host=127.0.0.1 port=1234 viewer: gst-launch -v udpsrc port=1234 ! theoradec ! autovideosink I have been translating it to C starting with the appsrc-stream example. I can't get it to negotiate the caps I set properly. The relevant warning messages are; basetransform gstbasetransform.c:1047:gst_base_transform_acceptcaps: transform could not transform video/x-raw-rgb, bpp=(int)24, depth=(int)24, width=(int)640, height=(int)480 in anything we support 0:00:00.178133697 10789 0x829a8d0 WARN basetransform gstbasetransform.c:1144:gst_base_transform_setcaps: FAILED to configure caps to accept video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)1/2147483647, width=(int)640, height=(int)480 The relevant code is; app->pipeline = gst_parse_launch("appsrc name=mysource ! video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink host=127.0.0.1 port=1234", NULL); caps = gst_caps_new_simple ("video/x-raw-rgb", "bpp",G_TYPE_INT,24, "depth",G_TYPE_INT,24, "width", G_TYPE_INT, 640, "height", G_TYPE_INT, 480, NULL); gst_app_src_set_caps(GST_APP_SRC(app->appsrc), caps); Full source code is here http://pastebin.com/PJuGVJLC Any pointers? John From vaish05_s at yahoo.com Thu Dec 2 08:25:52 2010 From: vaish05_s at yahoo.com (vaishnavi) Date: Wed, 1 Dec 2010 23:25:52 -0800 (PST) Subject: [gst-devel] Driver communication Message-ID: <1291274752540-3068601.post@n4.nabble.com> Hi, I'm using a driver which performs asynchronous write(aio_write)... I'm trying to develop a plugin which can write to device to perform some processing and then get back the processed data from driver to stream to output.. Is there a single element/plugin which does this..(write to hardware and reading from it)... Thanx -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Driver-communication-tp3068601p3068601.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From gibrovacco at gmail.com Thu Dec 2 08:28:11 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Thu, 2 Dec 2010 09:28:11 +0200 Subject: [gst-devel] G729 with gstreamer In-Reply-To: <4cf72c33.e83cec0a.3465.056b@mx.google.com> References: <4cf72c33.e83cec0a.3465.056b@mx.google.com> Message-ID: Hi, On Thu, Dec 2, 2010 at 7:12 AM, parveen.jain wrote: > Hi All, > > ? Does Gstreamer(or any of its plugin ) supports the G729 encoder/decoder? There's an implementation base on the reference code. It isn't anything distributed with GStreamer, but the elements have been tested against a good variety of cases and works pretty well: http://gitorious.org/gladstone > Is yes, how is different from intel IPP library support of G729? - gladstone is portable across architectures, IPP libraries work only on Intel processors. - gladstone has full sources available (even though you should really ask to ITU-T in order to redistribute the reference code). IPP libs are a "black box": if something doesn't work you'll have to beg Intel for fixing it. The info reported here: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ is misleading, as the codecs themselves are anyway closed source (try to believe ;) ). - IPP libraries are more optimised than the reference code, but there are already a few patches in Gladstone to begin filling the gap. - G729 BA (basically, with DTX support) for gladstone is arriving, I just need a few sanity checks before submitting the patches (they should get to gitorious this weekend, but no promises). Afair IPP libraries don't have DTX support: http://software.intel.com/en-us/forums/showthread.php?t=70652 Regards > > > > Regards, > > Parveen Jain > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From parveen.jain at one97.net Thu Dec 2 08:25:16 2010 From: parveen.jain at one97.net (parveen.jain) Date: Thu, 2 Dec 2010 12:55:16 +0530 Subject: [gst-devel] G729 with gstreamer In-Reply-To: Message-ID: <4cf74b41.10a2960a.5671.195b@mx.google.com> Thanks Marco, This is really what I was looking for. Best Regards, Parveen Jain -----Original Message----- From: Marco Ballesio [mailto:gibrovacco at gmail.com] Sent: Thursday, December 02, 2010 12:58 PM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] G729 with gstreamer Hi, On Thu, Dec 2, 2010 at 7:12 AM, parveen.jain wrote: > Hi All, > > ? Does Gstreamer(or any of its plugin ) supports the G729 encoder/decoder? There's an implementation base on the reference code. It isn't anything distributed with GStreamer, but the elements have been tested against a good variety of cases and works pretty well: http://gitorious.org/gladstone > Is yes, how is different from intel IPP library support of G729? - gladstone is portable across architectures, IPP libraries work only on Intel processors. - gladstone has full sources available (even though you should really ask to ITU-T in order to redistribute the reference code). IPP libs are a "black box": if something doesn't work you'll have to beg Intel for fixing it. The info reported here: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ is misleading, as the codecs themselves are anyway closed source (try to believe ;) ). - IPP libraries are more optimised than the reference code, but there are already a few patches in Gladstone to begin filling the gap. - G729 BA (basically, with DTX support) for gladstone is arriving, I just need a few sanity checks before submitting the patches (they should get to gitorious this weekend, but no promises). Afair IPP libraries don't have DTX support: http://software.intel.com/en-us/forums/showthread.php?t=70652 Regards > > > > Regards, > > Parveen Jain > > ---------------------------------------------------------------------------- -- > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ---------------------------------------------------------------------------- -- Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! Tap into the largest installed PC base & get more eyes on your game by optimizing for Intel(R) Graphics Technology. Get started today with the Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. http://p.sf.net/sfu/intelisp-dev2dev _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From gibrovacco at gmail.com Thu Dec 2 09:11:01 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Thu, 2 Dec 2010 10:11:01 +0200 Subject: [gst-devel] Driver communication In-Reply-To: <1291274752540-3068601.post@n4.nabble.com> References: <1291274752540-3068601.post@n4.nabble.com> Message-ID: Hi, On Thu, Dec 2, 2010 at 9:25 AM, vaishnavi wrote: > > Hi, > > I'm using a driver which performs asynchronous write(aio_write)... > I'm trying to develop a plugin which can write to device to perform some > processing and then get back the processed data from driver to stream to > output.. > > Is there a single element/plugin which does this..(write to hardware and > reading from it)... some examples coming to my mind, unfortunately you need specific hardware to test them: - gstreamer vaapi plugins. - gst-dsp: it's quite simple to understand (or at least it was the last time I hacked with it). - gst-openmax: well, it doesn't grant processing will be made through a driver, but very often it's likely to happen. Regards > > Thanx > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Driver-communication-tp3068601p3068601.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ved.kpl at gmail.com Thu Dec 2 10:39:33 2010 From: ved.kpl at gmail.com (ved kpl) Date: Thu, 2 Dec 2010 15:09:33 +0530 Subject: [gst-devel] Another GstAppSrc Question In-Reply-To: References: Message-ID: it expects the entire rgb caps... Easier way would be to use videoparse after the appsrc. eg. appsrc ! videoparse width=640 height=480 format=rgb ! .... On Thu, Dec 2, 2010 at 11:28 AM, John Stowers wrote: > Hi All, > > I see quite a few similar questions to these on the gst-devel mailing > list, but none of them allowed me to get past my problem. > > I'm building a streaming video thing using gstreamer, the prototype > (works fine) is > > sender: > gst-launch -v videotestsrc ! > video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! > ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink > host=127.0.0.1 port=1234 > > viewer: > gst-launch -v udpsrc port=1234 ! theoradec ! autovideosink > > I have been translating it to C starting with the appsrc-stream example. > I can't get it to negotiate the caps I set properly. The relevant > warning messages are; > > basetransform > gstbasetransform.c:1047:gst_base_transform_acceptcaps: > transform could not transform video/x-raw-rgb, bpp=(int)24, > depth=(int)24, width=(int)640, height=(int)480 in anything we support > 0:00:00.178133697 10789 ?0x829a8d0 WARN ? ? ? ? ? basetransform > gstbasetransform.c:1144:gst_base_transform_setcaps: FAILED > to configure caps to accept video/x-raw-yuv, > format=(fourcc)I420, framerate=(fraction)1/2147483647, width=(int)640, > height=(int)480 > > The relevant code is; > > app->pipeline = gst_parse_launch("appsrc name=mysource ! > video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! > ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink > host=127.0.0.1 port=1234", NULL); > > > > caps = gst_caps_new_simple ("video/x-raw-rgb", > ? ?"bpp",G_TYPE_INT,24, > ? ?"depth",G_TYPE_INT,24, > ? ? "width", G_TYPE_INT, 640, > ? ? "height", G_TYPE_INT, 480, > ? ? NULL); > gst_app_src_set_caps(GST_APP_SRC(app->appsrc), caps); > > Full source code is here > http://pastebin.com/PJuGVJLC > > Any pointers? > > John > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From prabhulingaswamy.bs at globaledgesoft.com Thu Dec 2 10:52:31 2010 From: prabhulingaswamy.bs at globaledgesoft.com (Prabhulinga Swamy B S) Date: Thu, 02 Dec 2010 15:22:31 +0530 Subject: [gst-devel] Another GstAppSrc Question In-Reply-To: References: Message-ID: <4CF76C5F.6010804@globaledgesoft.com> Hi John, You can just leave out the filter you are doing between 'appsrc' and 'ffmpegcolorspace'. That is not required since you are setting the caps of appsrc, that fine. Have you tried like this? app->pipeline = gst_parse_launch("appsrc name=mysource ! " " ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink " " host=127.0.0.1 port=1234", NULL); Prabhu On 12/02/2010 11:28 AM, John Stowers wrote: > Hi All, > > I see quite a few similar questions to these on the gst-devel mailing > list, but none of them allowed me to get past my problem. > > I'm building a streaming video thing using gstreamer, the prototype > (works fine) is > > sender: > gst-launch -v videotestsrc ! > video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! > ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink > host=127.0.0.1 port=1234 > > viewer: > gst-launch -v udpsrc port=1234 ! theoradec ! autovideosink > > I have been translating it to C starting with the appsrc-stream example. > I can't get it to negotiate the caps I set properly. The relevant > warning messages are; > > basetransform > gstbasetransform.c:1047:gst_base_transform_acceptcaps: > transform could not transform video/x-raw-rgb, bpp=(int)24, > depth=(int)24, width=(int)640, height=(int)480 in anything we support > 0:00:00.178133697 10789 0x829a8d0 WARN basetransform > gstbasetransform.c:1144:gst_base_transform_setcaps: FAILED > to configure caps to accept video/x-raw-yuv, > format=(fourcc)I420, framerate=(fraction)1/2147483647, width=(int)640, > height=(int)480 > > The relevant code is; > > app->pipeline = gst_parse_launch("appsrc name=mysource ! > video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! > ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink > host=127.0.0.1 port=1234", NULL); > > > > caps = gst_caps_new_simple ("video/x-raw-rgb", > "bpp",G_TYPE_INT,24, > "depth",G_TYPE_INT,24, > "width", G_TYPE_INT, 640, > "height", G_TYPE_INT, 480, > NULL); > gst_app_src_set_caps(GST_APP_SRC(app->appsrc), caps); > > Full source code is here > http://pastebin.com/PJuGVJLC > > Any pointers? > > John > > > From bard_kuo at compalcomm.com Thu Dec 2 11:42:47 2010 From: bard_kuo at compalcomm.com (bard) Date: Thu, 2 Dec 2010 02:42:47 -0800 (PST) Subject: [gst-devel] out flush in gst-openmax In-Reply-To: References: <1291195356075-3066876.post@n4.nabble.com> <1291199687805-3067008.post@n4.nabble.com> Message-ID: <1291286567587-3068856.post@n4.nabble.com> Thank you! It helps me a lot. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/out-flush-in-gst-openmax-tp3066876p3068856.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From chain at bsd.ee Thu Dec 2 11:17:51 2010 From: chain at bsd.ee (Tonu Jaansoo) Date: Thu, 2 Dec 2010 12:17:51 +0200 Subject: [gst-devel] glupload Message-ID: Hi! I have a strange problem with glupload and a Late2007 imac with latest OS X. I get kernel panics quite randomly. Screen goes just blank or white. Logs show its bad ATI driver, but since i see this happen only when i mess around with glupload, i thought i might ask here. I tested same crashing software on a linux machine and it works fine there. Has anybody seen this and is there a workaround for that. I don't know how to debug kernel. Thanks. T?nu From john.stowers.lists at gmail.com Thu Dec 2 11:53:55 2010 From: john.stowers.lists at gmail.com (John Stowers) Date: Thu, 2 Dec 2010 23:53:55 +1300 Subject: [gst-devel] Another GstAppSrc Question In-Reply-To: References: Message-ID: On Thu, Dec 2, 2010 at 10:39 PM, ved kpl wrote: > it expects the entire rgb caps... > Easier way would be to use videoparse after the appsrc. > eg. appsrc ! videoparse width=640 height=480 format=rgb ! .... It worked fine without any extra caps The videoparse stuff made it work! Thanks a lot John > > On Thu, Dec 2, 2010 at 11:28 AM, John Stowers > wrote: >> Hi All, >> >> I see quite a few similar questions to these on the gst-devel mailing >> list, but none of them allowed me to get past my problem. >> >> I'm building a streaming video thing using gstreamer, the prototype >> (works fine) is >> >> sender: >> gst-launch -v videotestsrc ! >> video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! >> ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink >> host=127.0.0.1 port=1234 >> >> viewer: >> gst-launch -v udpsrc port=1234 ! theoradec ! autovideosink >> >> I have been translating it to C starting with the appsrc-stream example. >> I can't get it to negotiate the caps I set properly. The relevant >> warning messages are; >> >> basetransform >> gstbasetransform.c:1047:gst_base_transform_acceptcaps: >> transform could not transform video/x-raw-rgb, bpp=(int)24, >> depth=(int)24, width=(int)640, height=(int)480 in anything we support >> 0:00:00.178133697 10789 ?0x829a8d0 WARN ? ? ? ? ? basetransform >> gstbasetransform.c:1144:gst_base_transform_setcaps: FAILED >> to configure caps to accept video/x-raw-yuv, >> format=(fourcc)I420, framerate=(fraction)1/2147483647, width=(int)640, >> height=(int)480 >> >> The relevant code is; >> >> app->pipeline = gst_parse_launch("appsrc name=mysource ! >> video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! >> ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink >> host=127.0.0.1 port=1234", NULL); >> >> >> >> caps = gst_caps_new_simple ("video/x-raw-rgb", >> ? ?"bpp",G_TYPE_INT,24, >> ? ?"depth",G_TYPE_INT,24, >> ? ? "width", G_TYPE_INT, 640, >> ? ? "height", G_TYPE_INT, 480, >> ? ? NULL); >> gst_app_src_set_caps(GST_APP_SRC(app->appsrc), caps); >> >> Full source code is here >> http://pastebin.com/PJuGVJLC >> >> Any pointers? >> >> John >> >> ------------------------------------------------------------------------------ >> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >> Tap into the largest installed PC base & get more eyes on your game by >> optimizing for Intel(R) Graphics Technology. Get started today with the >> Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. >> http://p.sf.net/sfu/intelisp-dev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From john.stowers.lists at gmail.com Thu Dec 2 11:54:41 2010 From: john.stowers.lists at gmail.com (John Stowers) Date: Thu, 2 Dec 2010 23:54:41 +1300 Subject: [gst-devel] Another GstAppSrc Question In-Reply-To: <4CF76C5F.6010804@globaledgesoft.com> References: <4CF76C5F.6010804@globaledgesoft.com> Message-ID: On Thu, Dec 2, 2010 at 10:52 PM, Prabhulinga Swamy B S wrote: > Hi John, > > You can just leave out the filter you are doing between 'appsrc' and > 'ffmpegcolorspace'. > That is not required since you are setting the caps of appsrc, that fine. > Have you tried like this? Yeah. The solution was to add a videoparse filter (noted in another reply). Thanks, John > > app->pipeline = gst_parse_launch("appsrc name=mysource ! " > ? ? ? ?" ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink " > ? ? ? ?" host=127.0.0.1 port=1234", NULL); > > > Prabhu > > > > On 12/02/2010 11:28 AM, John Stowers wrote: >> Hi All, >> >> I see quite a few similar questions to these on the gst-devel mailing >> list, but none of them allowed me to get past my problem. >> >> I'm building a streaming video thing using gstreamer, the prototype >> (works fine) is >> >> sender: >> gst-launch -v videotestsrc ! >> video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! >> ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink >> host=127.0.0.1 port=1234 >> >> viewer: >> gst-launch -v udpsrc port=1234 ! theoradec ! autovideosink >> >> I have been translating it to C starting with the appsrc-stream example. >> I can't get it to negotiate the caps I set properly. The relevant >> warning messages are; >> >> basetransform >> gstbasetransform.c:1047:gst_base_transform_acceptcaps: >> transform could not transform video/x-raw-rgb, bpp=(int)24, >> depth=(int)24, width=(int)640, height=(int)480 in anything we support >> 0:00:00.178133697 10789 ?0x829a8d0 WARN ? ? ? ? ? basetransform >> gstbasetransform.c:1144:gst_base_transform_setcaps: ?FAILED >> to configure caps ?to accept video/x-raw-yuv, >> format=(fourcc)I420, framerate=(fraction)1/2147483647, width=(int)640, >> height=(int)480 >> >> The relevant code is; >> >> app->pipeline = gst_parse_launch("appsrc name=mysource ! >> video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! >> ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink >> host=127.0.0.1 port=1234", NULL); >> >> >> >> caps = gst_caps_new_simple ("video/x-raw-rgb", >> ? ? ?"bpp",G_TYPE_INT,24, >> ? ? ?"depth",G_TYPE_INT,24, >> ? ? ? "width", G_TYPE_INT, 640, >> ? ? ? "height", G_TYPE_INT, 480, >> ? ? ? NULL); >> gst_app_src_set_caps(GST_APP_SRC(app->appsrc), caps); >> >> Full source code is here >> http://pastebin.com/PJuGVJLC >> >> Any pointers? >> >> John >> >> >> > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From member at linkedin.com Thu Dec 2 12:35:38 2010 From: member at linkedin.com (Robert Swain via LinkedIn) Date: Thu, 2 Dec 2010 03:35:38 -0800 (PST) Subject: [gst-devel] Invitation to connect on LinkedIn Message-ID: <2004655484.7941168.1291289738753.JavaMail.app@ech3-cdn13.prod> LinkedIn ------------Robert Swain requested to add you as a connection on LinkedIn: ------------------------------------------ Terry, I'd like to add you to my professional network on LinkedIn. - Robert Accept invitation from Robert Swain http://www.linkedin.com/e/8uvs3p-gh7kkj5p-26/I6odceFIR0nwoItulatqxeuuuYZHsrOdOOozdxXIoOow9a7uxmbiOMp/blk/I2489386211_2/1BpC5vrmRLoRZcjkkZt5YCpnlOt3RApnhMpmdzgmhxrSNBszYOnP4NczoUcPAUd399bTBWiPBMpDtkbPAScPoRcjwTej4LrCBxbOYWrSlI/EML_comm_afe/ View invitation from Robert Swain http://www.linkedin.com/e/8uvs3p-gh7kkj5p-26/I6odceFIR0nwoItulatqxeuuuYZHsrOdOOozdxXIoOow9a7uxmbiOMp/blk/I2489386211_2/39vcj4OdzwPejwQcAALqnpPbOYWrSlI/svi/ ------------------------------------------ DID YOU KNOW you can use your LinkedIn profile as your website? Select a vanity URL and then promote this address on your business cards, email signatures, website, etc http://www.linkedin.com/e/8uvs3p-gh7kkj5p-26/ewp/inv-21/ -- (c) 2010, LinkedIn Corporation -------------- next part -------------- An HTML attachment was scrubbed... URL: From haakon.sporsheim at gmail.com Thu Dec 2 12:53:28 2010 From: haakon.sporsheim at gmail.com (Haakon Sporsheim) Date: Thu, 2 Dec 2010 12:53:28 +0100 Subject: [gst-devel] out flush in gst-openmax In-Reply-To: References: <1291195356075-3066876.post@n4.nabble.com> <1291199687805-3067008.post@n4.nabble.com> Message-ID: 2010/12/1 Victor Manuel J?quez Leal : > On Wed, Dec 1, 2010 at 11:34 AM, bard wrote: >> >> Thank you vmjl , >> >> I will figure out what's wrong in OpenMax. >> >> So for gst-openmax , it doesn't need to modify in gstomx_base_filter.c >> pad_chain (). Right? > > In theory, if your omx component follows "correctly" the spec, no > changes are needed in gst-openmax > > vmjl > I've been trying to use gst-openmax on android utilizing the qualcomm snapdragon chip in HTC Passion/Google N1. I'm not sure if this is the same omx core as referenced by Bard. Anyway, I've found that gst-openmax has some shortcomings. First of all, responding to the event OMX_EventPortSettingsChanged. So I'm not sure whether or not gst-openmax "correctly" follows the OpenMAX IL spec. I'm not an expert in this area so you probably want to wait for an answer from one of the gst-openmax maintainers!? Haakon >> >> >> Bard >> -- >> View this message in context: http://gstreamer-devel.966125.n4.nabble.com/out-flush-in-gst-openmax-tp3066876p3067008.html >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> >> ------------------------------------------------------------------------------ >> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >> Tap into the largest installed PC base & get more eyes on your game by >> optimizing for Intel(R) Graphics Technology. Get started today with the >> Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. >> http://p.sf.net/sfu/intelisp-dev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ved.kpl at gmail.com Thu Dec 2 13:11:33 2010 From: ved.kpl at gmail.com (ved kpl) Date: Thu, 2 Dec 2010 17:41:33 +0530 Subject: [gst-devel] Another GstAppSrc Question In-Reply-To: References: <4CF76C5F.6010804@globaledgesoft.com> Message-ID: On Thu, Dec 2, 2010 at 4:24 PM, John Stowers wrote: > On Thu, Dec 2, 2010 at 10:52 PM, Prabhulinga Swamy B S > wrote: >> Hi John, >> >> You can just leave out the filter you are doing between 'appsrc' and >> 'ffmpegcolorspace'. >> That is not required since you are setting the caps of appsrc, that fine. >> Have you tried like this? > > Yeah. The solution was to add a videoparse filter (noted in another reply). Also, u need not set any caps on appsrc. > > Thanks, > > John >> >> app->pipeline = gst_parse_launch("appsrc name=mysource ! " >> ? ? ? ?" ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink " >> ? ? ? ?" host=127.0.0.1 port=1234", NULL); >> >> >> Prabhu >> >> >> >> On 12/02/2010 11:28 AM, John Stowers wrote: >>> Hi All, >>> >>> I see quite a few similar questions to these on the gst-devel mailing >>> list, but none of them allowed me to get past my problem. >>> >>> I'm building a streaming video thing using gstreamer, the prototype >>> (works fine) is >>> >>> sender: >>> gst-launch -v videotestsrc ! >>> video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! >>> ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink >>> host=127.0.0.1 port=1234 >>> >>> viewer: >>> gst-launch -v udpsrc port=1234 ! theoradec ! autovideosink >>> >>> I have been translating it to C starting with the appsrc-stream example. >>> I can't get it to negotiate the caps I set properly. The relevant >>> warning messages are; >>> >>> basetransform >>> gstbasetransform.c:1047:gst_base_transform_acceptcaps: >>> transform could not transform video/x-raw-rgb, bpp=(int)24, >>> depth=(int)24, width=(int)640, height=(int)480 in anything we support >>> 0:00:00.178133697 10789 ?0x829a8d0 WARN ? ? ? ? ? basetransform >>> gstbasetransform.c:1144:gst_base_transform_setcaps: ?FAILED >>> to configure caps ?to accept video/x-raw-yuv, >>> format=(fourcc)I420, framerate=(fraction)1/2147483647, width=(int)640, >>> height=(int)480 >>> >>> The relevant code is; >>> >>> app->pipeline = gst_parse_launch("appsrc name=mysource ! >>> video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! >>> ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink >>> host=127.0.0.1 port=1234", NULL); >>> >>> >>> >>> caps = gst_caps_new_simple ("video/x-raw-rgb", >>> ? ? ?"bpp",G_TYPE_INT,24, >>> ? ? ?"depth",G_TYPE_INT,24, >>> ? ? ? "width", G_TYPE_INT, 640, >>> ? ? ? "height", G_TYPE_INT, 480, >>> ? ? ? NULL); >>> gst_app_src_set_caps(GST_APP_SRC(app->appsrc), caps); >>> >>> Full source code is here >>> http://pastebin.com/PJuGVJLC >>> >>> Any pointers? >>> >>> John >>> >>> >>> >> >> ------------------------------------------------------------------------------ >> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >> Tap into the largest installed PC base & get more eyes on your game by >> optimizing for Intel(R) Graphics Technology. Get started today with the >> Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. >> http://p.sf.net/sfu/intelisp-dev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From bilboed at gmail.com Thu Dec 2 13:31:16 2010 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 02 Dec 2010 13:31:16 +0100 Subject: [gst-devel] get element type name In-Reply-To: <1291241194781-3068150.post@n4.nabble.com> References: <1291231515380-3067899.post@n4.nabble.com> <1291233026.5576.1.camel@deumeu> <1291238393925-3068067.post@n4.nabble.com> <4CF6C12E.6060504@hora-obscura.de> <1291241194781-3068150.post@n4.nabble.com> Message-ID: <1291293076.4291.18347.camel@localhost> On Wed, 2010-12-01 at 14:06 -0800, Wes Miller wrote: > Stefan, > > > If you are looking at the header that matches your binaries everything is > > going > > to be fine. Note sure how you get the idea that those are declared in that > > partiual header and not in some header that gets included by gst/gst.h. > > By being silly enough to quote the doc that said "include not being bright enough to remember that gst.h is pretty much an includer of > include files. And don't forget to read the docs of those functions you're using. Maybe they return something whose reference count you need to decrement after you've used it :) > > Anyway, thanks and also to Edward and Michael. You're welcome > > Happy Christmas, > > Wes From mjoachimiak at gmail.com Thu Dec 2 14:06:25 2010 From: mjoachimiak at gmail.com (Michael Joachimiak) Date: Thu, 2 Dec 2010 15:06:25 +0200 Subject: [gst-devel] out flush in gst-openmax In-Reply-To: References: <1291195356075-3066876.post@n4.nabble.com> <1291199687805-3067008.post@n4.nabble.com> Message-ID: I had the same problem yesterday. I feel that error reporting from OpenMAX is way to short. My problew was the stream size (width or height not divisible over 16). What parameters/pipeline is used to encode src.264? What is the size of the stream? 2010/12/2 Haakon Sporsheim > 2010/12/1 Victor Manuel J?quez Leal : > > On Wed, Dec 1, 2010 at 11:34 AM, bard wrote: > >> > >> Thank you vmjl , > >> > >> I will figure out what's wrong in OpenMax. > >> > >> So for gst-openmax , it doesn't need to modify in gstomx_base_filter.c > >> pad_chain (). Right? > > > > In theory, if your omx component follows "correctly" the spec, no > > changes are needed in gst-openmax > > > > vmjl > > > I've been trying to use gst-openmax on android utilizing the qualcomm > snapdragon chip in HTC Passion/Google N1. I'm not sure if this is the > same omx core as referenced by Bard. Anyway, I've found that > gst-openmax has some shortcomings. First of all, responding to the > event OMX_EventPortSettingsChanged. So I'm not sure whether or not > gst-openmax "correctly" follows the OpenMAX IL spec. I'm not an expert > in this area so you probably want to wait for an answer from one of > the gst-openmax maintainers!? > > Haakon > > >> > >> > >> Bard > >> -- > >> View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/out-flush-in-gst-openmax-tp3066876p3067008.html > >> Sent from the GStreamer-devel mailing list archive at Nabble.com. > >> > >> > ------------------------------------------------------------------------------ > >> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > >> Tap into the largest installed PC base & get more eyes on your game by > >> optimizing for Intel(R) Graphics Technology. Get started today with the > >> Intel(R) Software Partner Program. Five $500 cash prizes are up for > grabs. > >> http://p.sf.net/sfu/intelisp-dev2dev > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >> > > > > > ------------------------------------------------------------------------------ > > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > > Tap into the largest installed PC base & get more eyes on your game by > > optimizing for Intel(R) Graphics Technology. Get started today with the > > Intel(R) Software Partner Program. Five $500 cash prizes are up for > grabs. > > http://p.sf.net/sfu/intelisp-dev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Your Sincerely Michael Joachimiak -------------- next part -------------- An HTML attachment was scrubbed... URL: From wmiller at sdr.com Thu Dec 2 14:23:19 2010 From: wmiller at sdr.com (Wes Miller) Date: Thu, 2 Dec 2010 05:23:19 -0800 (PST) Subject: [gst-devel] get element type name In-Reply-To: <1291293076.4291.18347.camel@localhost> References: <1291231515380-3067899.post@n4.nabble.com> <1291233026.5576.1.camel@deumeu> <1291238393925-3068067.post@n4.nabble.com> <4CF6C12E.6060504@hora-obscura.de> <1291241194781-3068150.post@n4.nabble.com> <1291293076.4291.18347.camel@localhost> Message-ID: <1291296199032-3069320.post@n4.nabble.com> Edwward, > And don't forget to read the docs of those functions you're using. > Maybe they return something whose reference count you need to decrement > 4after you've used it :) Hint, hint, hint..... Checking.... Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/get-element-type-name-tp3067899p3069320.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wmiller at sdr.com Thu Dec 2 14:39:26 2010 From: wmiller at sdr.com (Wes Miller) Date: Thu, 2 Dec 2010 05:39:26 -0800 (PST) Subject: [gst-devel] get element type name In-Reply-To: <1291296199032-3069320.post@n4.nabble.com> References: <1291231515380-3067899.post@n4.nabble.com> <1291233026.5576.1.camel@deumeu> <1291238393925-3068067.post@n4.nabble.com> <4CF6C12E.6060504@hora-obscura.de> <1291241194781-3068150.post@n4.nabble.com> <1291293076.4291.18347.camel@localhost> <1291296199032-3069320.post@n4.nabble.com> Message-ID: <1291297166547-3069329.post@n4.nabble.com> Edward, No, no, and no. No refs to release. gst_plugin_feature_get_name() gst_element_get_factory() gst_element_factory_get_longname() Just redoing my homework. I had already checked yesterday but thought I'd better recheck. Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/get-element-type-name-tp3067899p3069329.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bilboed at gmail.com Thu Dec 2 14:42:48 2010 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 02 Dec 2010 14:42:48 +0100 Subject: [gst-devel] get element type name In-Reply-To: <1291297166547-3069329.post@n4.nabble.com> References: <1291231515380-3067899.post@n4.nabble.com> <1291233026.5576.1.camel@deumeu> <1291238393925-3068067.post@n4.nabble.com> <4CF6C12E.6060504@hora-obscura.de> <1291241194781-3068150.post@n4.nabble.com> <1291293076.4291.18347.camel@localhost> <1291296199032-3069320.post@n4.nabble.com> <1291297166547-3069329.post@n4.nabble.com> Message-ID: <1291297368.4291.18348.camel@localhost> On Thu, 2010-12-02 at 05:39 -0800, Wes Miller wrote: > Edward, > > No, no, and no. No refs to release. > gst_plugin_feature_get_name() > gst_element_get_factory() > gst_element_factory_get_longname() > > Just redoing my homework. I had already checked yesterday but thought I'd > better recheck. Interesting... I would have thought _get_factory would have incremented it. Oh well, better safe than sorry :) Edward > > Wes From thiago.sousa.santos at collabora.co.uk Thu Dec 2 16:07:32 2010 From: thiago.sousa.santos at collabora.co.uk (Thiago Sousa Santos) Date: Thu, 02 Dec 2010 12:07:32 -0300 Subject: [gst-devel] Camerabin2 Prototype and IRC Meeting Message-ID: <1291302452.7078.13.camel@blacksheep> Hello, As mentioned some time ago, we started planning camerabin2. The good news is that a small prototype is ready at http://gitorious.org/gstreamer-camerabin2/gst-plugins-bad It is as simple as it gets, being able to take pictures, record videos and showing a viewfinder. Feedback, code review, patches, tests are welcome. I'm currently trying to get the code in shape to push it as an experimental plugin into -bad module. I'd like arrange meeting with people interested in camerabin2 to discuss desired features and problems with this new design. I'm already poking some people that mentioned interest to set up a date and time. We'd like to do this soon, so expect it to be early next week (Monday or Tuesday). Details on camerabin2 design are on the wiki: http://www.gstreamer.net/wiki/CameraBin -- Thiago From martin.bisson at gmail.com Thu Dec 2 19:04:49 2010 From: martin.bisson at gmail.com (Martin Bisson) Date: Thu, 02 Dec 2010 13:04:49 -0500 Subject: [gst-devel] glupload In-Reply-To: References: Message-ID: <4CF7DFC1.6050403@gmail.com> A couple of years ago, I had problems with ATI drivers. It was under Linux though, with a FireGL. A simple call to glReadPixels would cause a kernel panic. Can you reproduce that problem with a hello world program calling glReadPixels() after rendering a simple scene? Martin On 12/2/2010 5:17 AM, Tonu Jaansoo wrote: > Hi! > > I have a strange problem with glupload and a Late2007 imac with latest OS X. > I get kernel panics quite randomly. Screen goes just blank or white. Logs show its bad ATI driver, but since > i see this happen only when i mess around with glupload, i thought i might ask here. I tested same crashing software on a linux machine and it works fine there. > Has anybody seen this and is there a workaround for that. I don't know how to debug kernel. > > Thanks. > T?nu > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App& Earn a Chance To Win $500! > Tap into the largest installed PC base& get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From john.stowers.lists at gmail.com Thu Dec 2 20:06:06 2010 From: john.stowers.lists at gmail.com (John Stowers) Date: Fri, 3 Dec 2010 08:06:06 +1300 Subject: [gst-devel] Another GstAppSrc Question In-Reply-To: References: <4CF76C5F.6010804@globaledgesoft.com> Message-ID: On Fri, Dec 3, 2010 at 1:11 AM, ved kpl wrote: > On Thu, Dec 2, 2010 at 4:24 PM, John Stowers > wrote: >> On Thu, Dec 2, 2010 at 10:52 PM, Prabhulinga Swamy B S >> wrote: >>> Hi John, >>> >>> You can just leave out the filter you are doing between 'appsrc' and >>> 'ffmpegcolorspace'. >>> That is not required since you are setting the caps of appsrc, that fine. >>> Have you tried like this? >> >> Yeah. The solution was to add a videoparse filter (noted in another reply). > > ? Also, u need not set any caps on appsrc. Is this the recommended way of doing things? It feels like it breaks abstraction / the point of caps. I mean I wanted to support streaming multiple video types by only setting different caps on the appsrc and letting ffmpegcolorspace convert to what was necessary. So I suspect the root cause of why those caps failed to work remains unknown. It does not matter even if I set all the rgb caps, still no success. Do you know how I might solve this while maintaining the abstraction of video type through only the setting of caps? Full code is here; https://gist.github.com/725122 Thanks, John >> >> Thanks, >> >> John > > > >>> >>> app->pipeline = gst_parse_launch("appsrc name=mysource ! " >>> ? ? ? ?" ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink " >>> ? ? ? ?" host=127.0.0.1 port=1234", NULL); >>> >>> >>> Prabhu >>> >>> >>> >>> On 12/02/2010 11:28 AM, John Stowers wrote: >>>> Hi All, >>>> >>>> I see quite a few similar questions to these on the gst-devel mailing >>>> list, but none of them allowed me to get past my problem. >>>> >>>> I'm building a streaming video thing using gstreamer, the prototype >>>> (works fine) is >>>> >>>> sender: >>>> gst-launch -v videotestsrc ! >>>> video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! >>>> ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink >>>> host=127.0.0.1 port=1234 >>>> >>>> viewer: >>>> gst-launch -v udpsrc port=1234 ! theoradec ! autovideosink >>>> >>>> I have been translating it to C starting with the appsrc-stream example. >>>> I can't get it to negotiate the caps I set properly. The relevant >>>> warning messages are; >>>> >>>> basetransform >>>> gstbasetransform.c:1047:gst_base_transform_acceptcaps: >>>> transform could not transform video/x-raw-rgb, bpp=(int)24, >>>> depth=(int)24, width=(int)640, height=(int)480 in anything we support >>>> 0:00:00.178133697 10789 ?0x829a8d0 WARN ? ? ? ? ? basetransform >>>> gstbasetransform.c:1144:gst_base_transform_setcaps: ?FAILED >>>> to configure caps ?to accept video/x-raw-yuv, >>>> format=(fourcc)I420, framerate=(fraction)1/2147483647, width=(int)640, >>>> height=(int)480 >>>> >>>> The relevant code is; >>>> >>>> app->pipeline = gst_parse_launch("appsrc name=mysource ! >>>> video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! >>>> ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink >>>> host=127.0.0.1 port=1234", NULL); >>>> >>>> >>>> >>>> caps = gst_caps_new_simple ("video/x-raw-rgb", >>>> ? ? ?"bpp",G_TYPE_INT,24, >>>> ? ? ?"depth",G_TYPE_INT,24, >>>> ? ? ? "width", G_TYPE_INT, 640, >>>> ? ? ? "height", G_TYPE_INT, 480, >>>> ? ? ? NULL); >>>> gst_app_src_set_caps(GST_APP_SRC(app->appsrc), caps); >>>> >>>> Full source code is here >>>> http://pastebin.com/PJuGVJLC >>>> >>>> Any pointers? >>>> >>>> John >>>> >>>> >>>> >>> >>> ------------------------------------------------------------------------------ >>> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >>> Tap into the largest installed PC base & get more eyes on your game by >>> optimizing for Intel(R) Graphics Technology. Get started today with the >>> Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. >>> http://p.sf.net/sfu/intelisp-dev2dev >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >> >> ------------------------------------------------------------------------------ >> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >> Tap into the largest installed PC base & get more eyes on your game by >> optimizing for Intel(R) Graphics Technology. Get started today with the >> Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. >> http://p.sf.net/sfu/intelisp-dev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From john.stowers.lists at gmail.com Thu Dec 2 20:42:45 2010 From: john.stowers.lists at gmail.com (John Stowers) Date: Fri, 3 Dec 2010 08:42:45 +1300 Subject: [gst-devel] Another GstAppSrc Question In-Reply-To: References: <4CF76C5F.6010804@globaledgesoft.com> Message-ID: On Fri, Dec 3, 2010 at 8:06 AM, John Stowers wrote: > On Fri, Dec 3, 2010 at 1:11 AM, ved kpl wrote: >> On Thu, Dec 2, 2010 at 4:24 PM, John Stowers >> wrote: >>> On Thu, Dec 2, 2010 at 10:52 PM, Prabhulinga Swamy B S >>> wrote: >>>> Hi John, >>>> >>>> You can just leave out the filter you are doing between 'appsrc' and >>>> 'ffmpegcolorspace'. >>>> That is not required since you are setting the caps of appsrc, that fine. >>>> Have you tried like this? >>> >>> Yeah. The solution was to add a videoparse filter (noted in another reply). >> >> ? Also, u need not set any caps on appsrc. > > Is this the recommended way of doing things? It feels like it breaks > abstraction / the point of caps. I mean I wanted to support streaming > multiple video types by only setting different caps on the appsrc and > letting ffmpegcolorspace convert to what was necessary. > > So I suspect the root cause of why those caps failed to work remains > unknown. It does not matter even if I set all the rgb caps, still no > success. Got it, I had the wrong mask values - I was assuming RGBA and it looks like the mask is applied against ARGB. The following works caps = gst_caps_new_simple ("video/x-raw-rgb", "width", G_TYPE_INT, 640, "height", G_TYPE_INT, 480, "bpp", G_TYPE_INT, 24, "depth", G_TYPE_INT, 24, "red_mask", G_TYPE_INT, 0x00ff0000, "green_mask", G_TYPE_INT, 0x0000ff00, "blue_mask", G_TYPE_INT, 0x000000ff, "framerate", GST_TYPE_FRACTION, 25, 1, "endianness", G_TYPE_INT, G_BIG_ENDIAN, NULL); Seems odd that I need to set the framerate on my appsrc (it is streaming, with largely unknown rate), but oh well. Thanks all for the help. John > > Do you know how I might solve this while maintaining the abstraction > of video type through only the setting of caps? > > Full code is here; > https://gist.github.com/725122 > > Thanks, > > John > >>> >>> Thanks, >>> >>> John >> >> >> >>>> >>>> app->pipeline = gst_parse_launch("appsrc name=mysource ! " >>>> ? ? ? ?" ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink " >>>> ? ? ? ?" host=127.0.0.1 port=1234", NULL); >>>> >>>> >>>> Prabhu >>>> >>>> >>>> >>>> On 12/02/2010 11:28 AM, John Stowers wrote: >>>>> Hi All, >>>>> >>>>> I see quite a few similar questions to these on the gst-devel mailing >>>>> list, but none of them allowed me to get past my problem. >>>>> >>>>> I'm building a streaming video thing using gstreamer, the prototype >>>>> (works fine) is >>>>> >>>>> sender: >>>>> gst-launch -v videotestsrc ! >>>>> video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! >>>>> ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink >>>>> host=127.0.0.1 port=1234 >>>>> >>>>> viewer: >>>>> gst-launch -v udpsrc port=1234 ! theoradec ! autovideosink >>>>> >>>>> I have been translating it to C starting with the appsrc-stream example. >>>>> I can't get it to negotiate the caps I set properly. The relevant >>>>> warning messages are; >>>>> >>>>> basetransform >>>>> gstbasetransform.c:1047:gst_base_transform_acceptcaps: >>>>> transform could not transform video/x-raw-rgb, bpp=(int)24, >>>>> depth=(int)24, width=(int)640, height=(int)480 in anything we support >>>>> 0:00:00.178133697 10789 ?0x829a8d0 WARN ? ? ? ? ? basetransform >>>>> gstbasetransform.c:1144:gst_base_transform_setcaps: ?FAILED >>>>> to configure caps ?to accept video/x-raw-yuv, >>>>> format=(fourcc)I420, framerate=(fraction)1/2147483647, width=(int)640, >>>>> height=(int)480 >>>>> >>>>> The relevant code is; >>>>> >>>>> app->pipeline = gst_parse_launch("appsrc name=mysource ! >>>>> video/x-raw-rgb,width=640,height=480,bpp=24,depth=24 ! >>>>> ffmpegcolorspace ! videoscale method=1 ! theoraenc bitrate=150 ! udpsink >>>>> host=127.0.0.1 port=1234", NULL); >>>>> >>>>> >>>>> >>>>> caps = gst_caps_new_simple ("video/x-raw-rgb", >>>>> ? ? ?"bpp",G_TYPE_INT,24, >>>>> ? ? ?"depth",G_TYPE_INT,24, >>>>> ? ? ? "width", G_TYPE_INT, 640, >>>>> ? ? ? "height", G_TYPE_INT, 480, >>>>> ? ? ? NULL); >>>>> gst_app_src_set_caps(GST_APP_SRC(app->appsrc), caps); >>>>> >>>>> Full source code is here >>>>> http://pastebin.com/PJuGVJLC >>>>> >>>>> Any pointers? >>>>> >>>>> John >>>>> >>>>> >>>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >>>> Tap into the largest installed PC base & get more eyes on your game by >>>> optimizing for Intel(R) Graphics Technology. Get started today with the >>>> Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. >>>> http://p.sf.net/sfu/intelisp-dev2dev >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>> >>> ------------------------------------------------------------------------------ >>> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >>> Tap into the largest installed PC base & get more eyes on your game by >>> optimizing for Intel(R) Graphics Technology. Get started today with the >>> Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. >>> http://p.sf.net/sfu/intelisp-dev2dev >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >> >> ------------------------------------------------------------------------------ >> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >> Tap into the largest installed PC base & get more eyes on your game by >> optimizing for Intel(R) Graphics Technology. Get started today with the >> Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. >> http://p.sf.net/sfu/intelisp-dev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > From wmiller at sdr.com Thu Dec 2 22:27:30 2010 From: wmiller at sdr.com (Wes Miller) Date: Thu, 2 Dec 2010 13:27:30 -0800 (PST) Subject: [gst-devel] List All Properties of an Element In-Reply-To: <1291132150817-3065597.post@n4.nabble.com> References: <1291125744894-3065421.post@n4.nabble.com> <1291132150817-3065597.post@n4.nabble.com> Message-ID: <1291325250860-3070248.post@n4.nabble.com> Ok, drum roll, I got this working and made it "pretty" so I could share. The attached files are for gst_element_print_properties(). It is more or less a rework of the print_element_properties_info() in gst-inspect intended for in-line use. It's LGPL since that's what gst-inspect was. Comments and suggestions to me at wmillersdr.com. Happy Hanukkah, Wes Miller http://gstreamer-devel.966125.n4.nabble.com/file/n3070248/gst_element_print_properties.c gst_element_print_properties.c http://gstreamer-devel.966125.n4.nabble.com/file/n3070248/gst_element_print_properties.h gst_element_print_properties.h p.s. I've never uploaded files with this tool before. If they don't show up or aren't accessible, I'll post them to pastebin,com and update this forum. wjm -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/List-All-Properties-of-an-Element-tp3065421p3070248.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wmiller at sdr.com Thu Dec 2 22:40:07 2010 From: wmiller at sdr.com (Wes Miller) Date: Thu, 2 Dec 2010 13:40:07 -0800 (PST) Subject: [gst-devel] List All Properties of an Element In-Reply-To: <1291325250860-3070248.post@n4.nabble.com> References: <1291125744894-3065421.post@n4.nabble.com> <1291132150817-3065597.post@n4.nabble.com> <1291325250860-3070248.post@n4.nabble.com> Message-ID: <1291326007705-3070267.post@n4.nabble.com> Hey, I forgot to advertise... a call to gst_element_print_properties( GstElement *element ); returns something like the following example output for a gstrtpbin element. The output is about 110 columns wide. Not great for viewing in vi* unless you turn line wrap off. (if your viewer happens to use proportional fonts, try cut-n-pasting into another editor) <-->|<--- property name --->|<-- current value -->|<-------- type --------->|<----- range and default -----> | ELEMENT CLASS NAME | GstRtpBin | | | ELEMENT FACTORY NAME | gstrtpbin | RTP Bin | rw- | name | "rtpbin" | G_TYPE_STRING | null rw- | async-handling | false | G_TYPE_BOOLEAN | false rw- | latency | 1000 | G_TYPE_UINT | (0 - 4294967295) 200 rw- | sdes | GstStructure | Boxed pointer of type "GstStructure" rw- | do-lost | false | G_TYPE_BOOLEAN | false rw- | ignore-pt | false | G_TYPE_BOOLEAN | false rw- | autoremove | false | G_TYPE_BOOLEAN | false rw- | buffer-mode | 1, "slave" | Enum "RTPJitterBufferMode" : 1, "slave" In the function, the topmost three variable declarations are for c2w, c3w and c4w which control the width of columns 2, 3 and 4, respectively. The function header gives details on how the output is ordered and formatted. Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/List-All-Properties-of-an-Element-tp3065421p3070267.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bard_kuo at compalcomm.com Fri Dec 3 03:09:16 2010 From: bard_kuo at compalcomm.com (bard) Date: Thu, 2 Dec 2010 18:09:16 -0800 (PST) Subject: [gst-devel] out flush in gst-openmax In-Reply-To: References: <1291195356075-3066876.post@n4.nabble.com> <1291199687805-3067008.post@n4.nabble.com> Message-ID: <1291342156377-3070548.post@n4.nabble.com> Thank you Haakon and Michael. I don't know what paremeters/pipeline used , I downloaded it from website. bard at ubuntu:~/vmshare/video_sample/omxtestprogram$ file src13_hrc7_525_420_2.264 src13_hrc7_525_420_2.264: JVT NAL sequence, H.264 video, main @ L 30 By the way,I saw there is some configure needed from qcom's document about H264 NAL parser. But In gst-openmax ,it doesn't deal with it ? Michael Joachimiak wrote: > > I had the same problem yesterday. I feel that error reporting from OpenMAX > is way to short. > My problew was the stream size (width or height not divisible over 16). > What parameters/pipeline is used to encode src.264? What is the size of > the > stream? > > 2010/12/2 Haakon Sporsheim > >> 2010/12/1 Victor Manuel J?quez Leal : >> > On Wed, Dec 1, 2010 at 11:34 AM, bard wrote: >> >> >> >> Thank you vmjl , >> >> >> >> I will figure out what's wrong in OpenMax. >> >> >> >> So for gst-openmax , it doesn't need to modify in gstomx_base_filter.c >> >> pad_chain (). Right? >> > >> > In theory, if your omx component follows "correctly" the spec, no >> > changes are needed in gst-openmax >> > >> > vmjl >> > >> I've been trying to use gst-openmax on android utilizing the qualcomm >> snapdragon chip in HTC Passion/Google N1. I'm not sure if this is the >> same omx core as referenced by Bard. Anyway, I've found that >> gst-openmax has some shortcomings. First of all, responding to the >> event OMX_EventPortSettingsChanged. So I'm not sure whether or not >> gst-openmax "correctly" follows the OpenMAX IL spec. I'm not an expert >> in this area so you probably want to wait for an answer from one of >> the gst-openmax maintainers!? >> >> Haakon >> >> >> >> >> >> >> Bard >> >> -- >> >> View this message in context: >> http://gstreamer-devel.966125.n4.nabble.com/out-flush-in-gst-openmax-tp3066876p3067008.html >> >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> >> >> >> >> ------------------------------------------------------------------------------ >> >> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >> >> Tap into the largest installed PC base & get more eyes on your game by >> >> optimizing for Intel(R) Graphics Technology. Get started today with >> the >> >> Intel(R) Software Partner Program. Five $500 cash prizes are up for >> grabs. >> >> http://p.sf.net/sfu/intelisp-dev2dev >> >> _______________________________________________ >> >> gstreamer-devel mailing list >> >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> > >> > >> ------------------------------------------------------------------------------ >> > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >> > Tap into the largest installed PC base & get more eyes on your game by >> > optimizing for Intel(R) Graphics Technology. Get started today with the >> > Intel(R) Software Partner Program. Five $500 cash prizes are up for >> grabs. >> > http://p.sf.net/sfu/intelisp-dev2dev >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> >> >> ------------------------------------------------------------------------------ >> Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! >> Tap into the largest installed PC base & get more eyes on your game by >> optimizing for Intel(R) Graphics Technology. Get started today with the >> Intel(R) Software Partner Program. Five $500 cash prizes are up for >> grabs. >> http://p.sf.net/sfu/intelisp-dev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > Your Sincerely > Michael Joachimiak > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/out-flush-in-gst-openmax-tp3066876p3070548.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From yongnian.le at intel.com Fri Dec 3 04:09:23 2010 From: yongnian.le at intel.com (Le, Yongnian) Date: Fri, 3 Dec 2010 11:09:23 +0800 Subject: [gst-devel] Reduce GstXvImageSink rendering frequency when window is obscured Message-ID: Hi, I am working on video conference application using gstreamer framework. In video conference, there might be multiple screens for video rendering using XvImageSink. I found that although some window is obscured ( for example full screen mode), its related XvImageSink is still busy on calling gst_xvimagesink_show_frame for video rendering. I am wondering that we could have optimization to detect additional "Obscured" xevent and reduce the rendering frequency to very low value, like 1 fps, if such window is obscured. In this way, we can save precious resources for most applications, like CPU and power, at the same time, having minimal impact to end user. Does that approach make sense? What do you guys think of it? Thanks a lot for your feedback and comments. Best Regards Yongnian -------------- next part -------------- An HTML attachment was scrubbed... URL: From bing_shine at hotmail.com Fri Dec 3 06:00:57 2010 From: bing_shine at hotmail.com (=?gb2312?B?y+/J3LH4?=) Date: Fri, 3 Dec 2010 05:00:57 +0000 Subject: [gst-devel] deinterlace optimization in gst-plugins-good Message-ID: Hi, For some reasons in my application based on deinterlace plugin from gst-plugins-good, i need to deinterlace format of YUV i420 planar. And I prefer to use GreedyH method to deinterlace. But I find that deinterlace method from latest plugins-good versions does not implement MMX/SSE optimization for greedyh i420 scanline. Therefore the performance is unacceptable. My question is as follows, i'm new to deinterlace, and i only want to get greedyh or tomsmosomp mmx/sse optimization for i420 planar format. can you give me some suggestion, or someone can commit optimization for greedyh/tomsmocomp planar scanline function. thanks:), sincerely yous, sean -------------- next part -------------- An HTML attachment was scrubbed... URL: From vaish05_s at yahoo.com Fri Dec 3 09:55:11 2010 From: vaish05_s at yahoo.com (vaishnavi) Date: Fri, 3 Dec 2010 00:55:11 -0800 (PST) Subject: [gst-devel] Driver communication In-Reply-To: References: <1291274752540-3068601.post@n4.nabble.com> Message-ID: <1291366511087-3070786.post@n4.nabble.com> Thank you for the info...il look into it... -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Driver-communication-tp3068601p3070786.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Fri Dec 3 12:47:19 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 03 Dec 2010 11:47:19 +0000 Subject: [gst-devel] [gst-cvs] gstreamer: micro-optim: if (x) is cheaper than if (x > 0) for unsigned integers In-Reply-To: <20101203110404.52B5910057@kemper.freedesktop.org> References: <20101203110404.52B5910057@kemper.freedesktop.org> Message-ID: <1291376839.22392.20.camel@zingle> On Fri, 2010-12-03 at 03:04 -0800, Edward Hervey wrote: > Module: gstreamer > Branch: master > Commit: 6aa8ca37eeb9debfa6919741a023250bf278248f > URL: http://cgit.freedesktop.org/gstreamer/gstreamer/commit/?id=6aa8ca37eeb9debfa6919741a023250bf278248f > > Author: Edward Hervey > Date: Sat Apr 11 15:04:41 2009 +0200 > > micro-optim: if (x) is cheaper than if (x > 0) for unsigned integers I really really dislike this change. Is it really worth it? I think this compromises code readability too much. Cheers -Tim From bilboed at gmail.com Fri Dec 3 12:59:33 2010 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 03 Dec 2010 12:59:33 +0100 Subject: [gst-devel] [gst-cvs] gstreamer: micro-optim: if (x) is cheaper than if (x > 0) for unsigned integers In-Reply-To: <1291376839.22392.20.camel@zingle> References: <20101203110404.52B5910057@kemper.freedesktop.org> <1291376839.22392.20.camel@zingle> Message-ID: <1291377573.4291.19441.camel@localhost> On Fri, 2010-12-03 at 11:47 +0000, Tim-Philipp M?ller wrote: > On Fri, 2010-12-03 at 03:04 -0800, Edward Hervey wrote: > > > Module: gstreamer > > Branch: master > > Commit: 6aa8ca37eeb9debfa6919741a023250bf278248f > > URL: http://cgit.freedesktop.org/gstreamer/gstreamer/commit/?id=6aa8ca37eeb9debfa6919741a023250bf278248f > > > > Author: Edward Hervey > > Date: Sat Apr 11 15:04:41 2009 +0200 > > > > micro-optim: if (x) is cheaper than if (x > 0) for unsigned integers > > I really really dislike this change. Is it really worth it? I saw gcc asm where it was doing an expensive compare as opposed to a fast != 0. So yes, needed. Worth it ? Well I did title it "micro-optim". > > I think this compromises code readability too much. You mean if(x) is ... unreadable ? Isn't that C 101 ? Edward > > Cheers > -Tim > > From wmiller at sdr.com Fri Dec 3 13:34:01 2010 From: wmiller at sdr.com (Wes Miller) Date: Fri, 3 Dec 2010 04:34:01 -0800 (PST) Subject: [gst-devel] [gst-cvs] gstreamer: micro-optim: if (x) is cheaper than if (x > 0) for unsigned integers In-Reply-To: <1291377573.4291.19441.camel@localhost> References: <1291376839.22392.20.camel@zingle> <1291377573.4291.19441.camel@localhost> Message-ID: <1291379641290-3070998.post@n4.nabble.com> Take it from a visually impaired person, if (x) and if (x>0) are both quite readable compared to the slammed together, no whitespace, { on the end-of-line eye-tests some people churn out. Thank God for beautifiers. And besides, good practice says that ought to be if (0 2010-12-03 alexcuicn -------------- next part -------------- An HTML attachment was scrubbed... URL: From xavier.bestel at free.fr Fri Dec 3 14:00:04 2010 From: xavier.bestel at free.fr (Xavier Bestel) Date: Fri, 03 Dec 2010 14:00:04 +0100 Subject: [gst-devel] [gst-cvs] gstreamer: micro-optim: if (x) is cheaper than if (x > 0) for unsigned integers In-Reply-To: <1291377573.4291.19441.camel@localhost> References: <20101203110404.52B5910057@kemper.freedesktop.org> <1291376839.22392.20.camel@zingle> <1291377573.4291.19441.camel@localhost> Message-ID: <1291381204.18761.3.camel@skunk> On Fri, 2010-12-03 at 12:59 +0100, Edward Hervey wrote: > > > micro-optim: if (x) is cheaper than if (x > 0) for unsigned integers > > > > I really really dislike this change. Is it really worth it? > > I saw gcc asm where it was doing an expensive compare as opposed to a > fast != 0. So yes, needed. Worth it ? Well I did title it "micro-optim". I'd say the bug is with gcc, not gstreamer. A different compiler may generate correct code for this. Xav From chain at bsd.ee Fri Dec 3 16:23:00 2010 From: chain at bsd.ee (Tonu Jaansoo) Date: Fri, 3 Dec 2010 17:23:00 +0200 Subject: [gst-devel] glupload In-Reply-To: <4CF7DFC1.6050403@gmail.com> References: <4CF7DFC1.6050403@gmail.com> Message-ID: <163C242C-63EC-4F25-91C4-73C96A9CE945@bsd.ee> Hi! glReadPixels works fine for me, but thanks! Tonu On Dec 2, 2010, at 8:04 PM, Martin Bisson wrote: > A couple of years ago, I had problems with ATI drivers. It was under > Linux though, with a FireGL. A simple call to glReadPixels would cause > a kernel panic. Can you reproduce that problem with a hello world > program calling glReadPixels() after rendering a simple scene? > > Martin > > > On 12/2/2010 5:17 AM, Tonu Jaansoo wrote: >> Hi! >> >> I have a strange problem with glupload and a Late2007 imac with latest OS X. >> I get kernel panics quite randomly. Screen goes just blank or white. Logs show its bad ATI driver, but since >> i see this happen only when i mess around with glupload, i thought i might ask here. I tested same crashing software on a linux machine and it works fine there. >> Has anybody seen this and is there a workaround for that. I don't know how to debug kernel. >> >> Thanks. >> T?nu >> ------------------------------------------------------------------------------ >> Increase Visibility of Your 3D Game App& Earn a Chance To Win $500! >> Tap into the largest installed PC base& get more eyes on your game by >> optimizing for Intel(R) Graphics Technology. Get started today with the >> Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. >> http://p.sf.net/sfu/intelisp-dev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From tigrux at gmail.com Fri Dec 3 17:27:44 2010 From: tigrux at gmail.com (Sandino Flores Moreno) Date: Fri, 3 Dec 2010 10:27:44 -0600 Subject: [gst-devel] The 1.0 plan In-Reply-To: <1289307210.1963.174.camel@meany> References: <1289307210.1963.174.camel@meany> Message-ID: Something I'd like to see in the futire, is a better gobject-like use of callbacks. For example, instead of: void gst_pad_set_chain_function (GstPad *pad, GstPadChainFunction chain); GstFlowReturn (*GstPadChainFunction) (GstPad *pad, GstBuffer *buffer); I would prefer: void gst_pad_set_chain_function (GstPad *pad, GstPadChainFunction chain, gpointer user_data); GstFlowReturn (*GstPadChainFunction) (GstPad *pad, GstBuffer *buffer, gpointer user_data); This way, gstreamer would play more nicely with Vala and the bindings. Just my two cents. - Sandino On Tue, Nov 9, 2010 at 6:53 AM, Wim Taymans wrote: > Hello GStreamer hackers, > > As most of you know, we'll be starting the new 0.11 development this month. This > development effort should then eventually lead to a version 1.0 by the end of > 2011. ?We've collected a fair amount of desired changes and features [1], some of > them easy to implement and others not so much. > > Since the list of changes is rather large, we will have to have a strong focus > on the most pressing parts while making sure that the smaller changes can be > implemented on top of that. > > In this mail I would like to give a highlevel overview of the pressing problems > we would like to see fixed for 1.0. More detailed information can be found in > various places [1][2] and will be fleshed out even more as we go. > > The purpose is to give people an idea of the problems that will be fixed and > what new use case will become possible or easier. This list is mostly collected > from talking to people and the experiences of the past 5 years of GStreamer > 0.10 development. > > There are 4 areas that need improvements. Each of those areas can be further > subdivided in smaller subtasks [4]. > > 1) General cleanups. This includes small things like removing old or bad API > ? that we have hanging around, cleaning up structs, fixing the padding, adding > ? GstFlowReturn for events [6], etc.. > > 2) Performance improvements. The biggest one here is to rework the Caps system > ? to make it faster. We have several ideas here: incremental caps [3] and > ? reducing the amount of caps fields. Other ideas include changing the > ? datastructure for caps. We are also going to change reverse negotiation and > ? untie it from the allocation of data. > > 3) Extensible Buffer metadata. We're going to make it possible to dynamically > ? attach metadata to buffers. The goal is to make it much easier to integrate > ? with various hardware, DSPs and APIs like OpenMax, OpenGL, Cairo. Things like > ? strides, cropping and regions of interest should be looked into. We would > ? like to make sure that GStreamer elements can do zero-copy data passing for > ? the most common use cases. See [5] > > 4) Improve dynamic pipeline handling. Problems with events being lost and > ? newsegment accumulation cause difficulties when dynamically constructing > ? complicated pipelines. We would like to make the timing mode more > ? comprehensive and controllable from the application. > > I would like to ask everyone of you to see that this list is not missing > anything important that you think needs to be considered for 1.0. All ideas > are welcome and I would suggest to add them to the wiki [1]. Please try to focus > on new core features and improvements (bug fixes, documentation, new plugins, > performance improvements and other things that don't require API changes > continues as usual in the 0.10 branch). I would also like to see how you think > the API can be improved or how things can be done better API-wise. > > One of the big missing items on this list is improving the documentation. We've > been thinking again about making a book for 1.0. It always boils down to the > fact that most core developers would gladly write a chapter or two here and > there but the missing part, it seems, is someone who would take charge of > organizing and proofreading all this. If anyone feels like taking the lead > here, we would be very very happy hackers. > > In practice, I would like to encourage people to make git branches with > experiments and proposed solutions. I would also like to encourage you to add > your ideas and suggestions to the wiki or this mailing-list. ?We'll take 2 more > weeks to collect suggestions and to think about new ideas. Around the end of > November we should start the 0.11 branch to start merging branches and porting > elements. > > I would also like to do the weekly status updates of the 0.11 branch to let > people know where we are. > > Exciting times ahead! > > Wim > > > [1] http://gstreamer.freedesktop.org/wiki/ZeroPointEleven > [2] http://cgit.freedesktop.org/~wtay/gstreamer/tree/docs/random/plan-0.11.txt?h=working&id=5c05a15e7cd2327e98d0177fdce8e5be3215aa05 > [3] http://cgit.freedesktop.org/~ensonic/gstreamer/log/?h=lazycaps > [4] http://cgit.freedesktop.org/~wtay/gstreamer/tree/docs/random/use-cases-0.11.txt?h=working&id=5ca09f1421847264e69cb111b0887a00d8a58bf9 > [5] http://cgit.freedesktop.org/~wtay/gstreamer/log/?h=buffermeta > [6] http://cgit.freedesktop.org/~wtay/gstreamer/log/?h=events2 > > > > ------------------------------------------------------------------------------ > The Next 800 Companies to Lead America's Growth: New Video Whitepaper > David G. Thomson, author of the best-selling book "Blueprint to a > Billion" shares his insights and actions to help propel your > business during the next growth cycle. Listen Now! > http://p.sf.net/sfu/SAP-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From wb666greene at gmail.com Fri Dec 3 17:33:08 2010 From: wb666greene at gmail.com (wally_bkg) Date: Fri, 3 Dec 2010 08:33:08 -0800 (PST) Subject: [gst-devel] Which CODEC for mjpg? In-Reply-To: <4CF3F0DF.201@awpi.com> References: <4CF3F0DF.201@awpi.com> Message-ID: <1291393988296-3071342.post@n4.nabble.com> I have the opposite problem. I can make jpeg video files with: gst-launch v4l2src ! video/x-raw-yuv ! queue ! ffmpegcolorspace ! jpegenc ! avimux ! filesink location=mjpeg.avi and the file plays back fine in gstreamer (using Totum on Ubuntu 10.04) but when I copy it over to a windows system it doesn't play. There are several MJPEG playback codecs on the Windows system. Gspot Codec Information on Windows shows a valid playback pipeline, but errors in rendering. I tried Theoraenc instead with: gst-launch v4l2src ! queue ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=theora.ogv and again it plays fine in Totum but not on Windows (the Windows system can play back the ogg video sample files distributed with Ubuntu). -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Which-CODEC-for-mjpg-tp3064099p3071342.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wb666greene at gmail.com Fri Dec 3 18:09:17 2010 From: wb666greene at gmail.com (wally_bkg) Date: Fri, 3 Dec 2010 09:09:17 -0800 (PST) Subject: [gst-devel] The 1.0 plan In-Reply-To: <1289307210.1963.174.camel@meany> References: <1289307210.1963.174.camel@meany> Message-ID: <1291396157978-3071403.post@n4.nabble.com> Wim Taymans wrote: > > > One of the big missing items on this list is improving the documentation. > > Understatement of the year! The replies to this message alone are proof that information about the "proper" way to do things is sorely lacking. IMHO one of the worst parts of gstreamer is the v4l2src stuff, particularly selecting the proper video norms (seems about half the world will have video cards that default on power up to the wrong standard) and selecting the proper input. Is there even a way to do it via gst-launch? I also think the best documentation is a working sample code, with decent comments. The only way I could figure out how to interpret video data pixels in a buffer (either as a plugin or appsink) was to stumble around the source code for gstgamma and gstvideoscale. Of course the data you get in a buffer depends on the caps, but am I the only one who thinks there has to be a better way than: GstStructure *structure; gint width,height; // this seems rather ridiculous as a way to get info about interpreting the buffer, // but after inspecting gstvudeoscale.c appears to be what the gst designers intended :( data = GST_BUFFER_DATA (buf); size = GST_BUFFER_SIZE (buf); structure = gst_caps_get_structure (gst_buffer_get_caps(buf), 0); gst_structure_get_int (structure, "width", &width); gst_structure_get_int (structure, "height", &height); So I guess this comment just degenerates into another vote for having metadata in the buffers. OTOH I'd not be surprised if there is a better way that is undocumented and buried somewhere in the source tree I've not stumbled on to. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/The-1-0-plan-tp3033660p3071403.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From sebastian.droege at collabora.co.uk Fri Dec 3 18:17:51 2010 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Fri, 03 Dec 2010 18:17:51 +0100 Subject: [gst-devel] The 1.0 plan In-Reply-To: <1291396157978-3071403.post@n4.nabble.com> References: <1289307210.1963.174.camel@meany> <1291396157978-3071403.post@n4.nabble.com> Message-ID: <1291396671.3453.92.camel@odin.lan> On Fri, 2010-12-03 at 09:09 -0800, wally_bkg wrote: > I also think the best documentation is a working sample code, with decent > comments. The only way I could figure out how to interpret video data > pixels in a buffer (either as a plugin or appsink) was to stumble around the > source code for gstgamma and gstvideoscale. > > Of course the data you get in a buffer depends on the caps, but am I the > only one who thinks there has to be a better way than: > > [...] For this you should use the libgstvideo API to get the video format, width, height, rowstride, component widths/heights, component offsets, pixel strides, etc. from the caps: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstvideo.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: From ds at entropywave.com Fri Dec 3 20:17:06 2010 From: ds at entropywave.com (David Schleef) Date: Fri, 3 Dec 2010 14:17:06 -0500 Subject: [gst-devel] [gst-cvs] gstreamer: micro-optim: if (x) is cheaper than if (x > 0) for unsigned integers In-Reply-To: <1291377573.4291.19441.camel@localhost> References: <20101203110404.52B5910057@kemper.freedesktop.org> <1291376839.22392.20.camel@zingle> <1291377573.4291.19441.camel@localhost> Message-ID: <20101203191706.GA15328@cooker.entropywave.com> On Fri, Dec 03, 2010 at 12:59:33PM +0100, Edward Hervey wrote: > On Fri, 2010-12-03 at 11:47 +0000, Tim-Philipp M?ller wrote: > > On Fri, 2010-12-03 at 03:04 -0800, Edward Hervey wrote: > > > > > Module: gstreamer > > > Branch: master > > > Commit: 6aa8ca37eeb9debfa6919741a023250bf278248f > > > URL: http://cgit.freedesktop.org/gstreamer/gstreamer/commit/?id=6aa8ca37eeb9debfa6919741a023250bf278248f > > > > > > Author: Edward Hervey > > > Date: Sat Apr 11 15:04:41 2009 +0200 > > > > > > micro-optim: if (x) is cheaper than if (x > 0) for unsigned integers > > > > I really really dislike this change. Is it really worth it? > > I saw gcc asm where it was doing an expensive compare as opposed to a > fast != 0. So yes, needed. Worth it ? Well I did title it "micro-optim". What is an expensive compare? cmp $0, %eax and test %eax, %eax are both one cycle. It is somewhat amusing that gcc-4.5 generates different code for these, when it should know they are identical. gcc-4.4 doesn't, so I consider that a regression, but certainly not an important one. > Date: Sat Apr 11 15:04:41 2009 +0200 Hiding patches in our local tree, eh? David From wb666greene at gmail.com Fri Dec 3 20:36:34 2010 From: wb666greene at gmail.com (wally_bkg) Date: Fri, 3 Dec 2010 11:36:34 -0800 (PST) Subject: [gst-devel] The 1.0 plan In-Reply-To: <1291396671.3453.92.camel@odin.lan> References: <1289307210.1963.174.camel@meany> <1291396157978-3071403.post@n4.nabble.com> <1291396671.3453.92.camel@odin.lan> Message-ID: <1291404994856-3071651.post@n4.nabble.com> Sebastian Dr?ge-7 wrote: > > On Fri, 2010-12-03 at 09:09 -0800, wally_bkg wrote: > >> I also think the best documentation is a working sample code, with decent >> comments. The only way I could figure out how to interpret video data >> pixels in a buffer (either as a plugin or appsink) was to stumble around >> the >> source code for gstgamma and gstvideoscale. >> >> Of course the data you get in a buffer depends on the caps, but am I the >> only one who thinks there has to be a better way than: >> >> [...] > > For this you should use the libgstvideo API to get the video format, > width, height, rowstride, component widths/heights, component offsets, > pixel strides, etc. from the caps: > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstvideo.html > > Actually it was the nested function call to get the caps that seemed goofy to me. I still need to do that on the buffer to use most of these API functions. Unfortunately not all the API features I might like to use are in the 10.28 version that is in Ubuntu 10.04. Since I don't intend to maintain a distribution, I need to target something as a lowest common denominator (it was Ubuntu 8.04 when I first started learning about gstreamer). I'm still trying to decide if overall my problem is better solved by processing the video data by writing a plugin or using appsink & appsrc. I'm playing around right now with both approaches, but straight off the top using appsink & appsrc lets me create my application with a simple make file instead of needing all the automake stuff to build a plug in. Once I'd built the plugin template the first time, "make" was all I needed afterwards, but I'd sure hate to have start with virgin Ubuntu install, and the one file I modified for my "plugin" and have to show someone else how to build and install it again! -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/The-1-0-plan-tp3033660p3071651.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From gibrovacco at gmail.com Fri Dec 3 21:47:18 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Fri, 3 Dec 2010 22:47:18 +0200 Subject: [gst-devel] deinterlace optimization in gst-plugins-good In-Reply-To: References: Message-ID: Hi, 2010/12/3 ??? : > Hi, > For some reasons in my application based on deinterlace plugin from > gst-plugins-good, i need to deinterlace format of YUV i420 planar. And I > prefer to use GreedyH method to deinterlace. But I find that deinterlace > method from latest plugins-good versions does not implement MMX/SSE > optimization for greedyh i420 scanline. Therefore the performance is > unacceptable. > My question is as follows, i'm new to deinterlace, and i only want to get > greedyh or tomsmosomp mmx/sse optimization for i420 planar format. can you > give me some suggestion, or someone can commit optimization for > greedyh/tomsmocomp planar scanline function. > thanks:), sincerely yous, > sean I suggest you to give a look at the Oil Runtime Compiler (ORC) and the benefits it brings to GStreamer wrt SIMD instructions: http://code.entropywave.com/projects/orc/ http://gstreamer.freedesktop.org/wiki/GStreamerConference2010 (search for ORC) you would get code not only optimised for SSE, but many of the most common architectures in the wild. Regards > > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From wb666greene at gmail.com Fri Dec 3 22:53:55 2010 From: wb666greene at gmail.com (wally_bkg) Date: Fri, 3 Dec 2010 13:53:55 -0800 (PST) Subject: [gst-devel] Tell me about y4menc and other encoders. Message-ID: <1291413235446-3071830.post@n4.nabble.com> I'm working on a real time image processing system and I'd like to "Tee" off the input video and save it to a file for a variety of reasons -- archiving epochs of particular interest, re-analysis with algorithms that can't yet run in real time, system verification, etc. Previous work has shown that good quality compression (like with a commercial DVD Recorder) is acceptable (we've been using SVHS recorders for ~20 years :) ) I first tried: jpegenc, mpeg2enc, and theoraenc. I could eliminate mpeg2enc from a combination of too high CPU load and too low video quality (using only the defaults). If I'm wrong about mpeg2enc please set me straight about how to get good results from it. Here some sample pipelines gst-launch v4l2src ! video/x-raw-yuv, framerate=\(fraction\)30000/1001 ! tee name=t ! queue ! wally ! xvimagesink t. ! queue ! ffmpegcolorspace ! jpegenc ! avimux ! filesink location=mjpeg.avi gst-launch v4l2src ! video/x-raw-yuv, framerate=\(fraction\)30000/1001 ! tee name=t ! queue ! wally ! xvimagesink t. ! queue ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=theora.ogv The wally plug-in is a simple filter plug-in I wrote that does a simple transformation of the video data as a proxy for my image processing. Both pipelines seem to work OK but CPU usage seems high with theoraenc being about 40% more CPU but with about half the file size of jpegenc. My problem here is while the files play back fine in Totem, there is something wrong with them if I try to play them on a Windows system (that is set up for MJPEG and OGG/Theora). Not a show stopper, but I'd love clues as to what might be making these files not usable on Windows. Then I discovered y4menc: gst-launch v4l2src ! video/x-raw-yuv, framerate=\(fraction\)30000/1001 ! y4menc ! filesink location=y4menc.yuv while it gives about 30X the file size of theoraenc it uses very little CPU, less than half of the jpegenc pipeline. The doc at: http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-y4menc.html doesn't really tell me anything helpful. What are some of the pros and cons (other than gigantic files) of y4menc, and what is its intended purpose? I don't seem to be able to pipe it into one of the container formats like avimux, oggmux, or matroskamux. If I do "Open With Media Player" the file appears to play fine, although it seems playback in Totem doesn't seek. I can play it back with: gst-launch filesrc location=y4menc_big.yuv ! decodebin ! ffmpegcolorspace ! xvimagesink Having captured & played back an ~10 minute clip into a 9.7 GB file (apparently glitch free!) while composing this message I'm most impressed with this aspect of gstreamer performance! Are there better encoders in gstreamer that perhaps I don't have or know about? (I've installed about everything gst related that is available in the Ubuntu repos). -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Tell-me-about-y4menc-and-other-encoders-tp3071830p3071830.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From kapil.agl at gmail.com Sat Dec 4 07:26:05 2010 From: kapil.agl at gmail.com (Kapil Agrawal) Date: Sat, 4 Dec 2010 11:56:05 +0530 Subject: [gst-devel] The 1.0 plan In-Reply-To: References: <1289307210.1963.174.camel@meany> <1290685869.25170.39.camel@level.fluendo.fluendo.lan> Message-ID: I be more then happy to contribute. While giving gstreamer training in my organization, I realised the best they understand when proper set of examples are given as assignment & if we make them understand concepts on this basis. Also I think we need to make it very clear how do one debug, and understand the error, warnings etc in there application. People sometimes ask too ovious questions on irc, without even cheers - Kapil On Fri, Nov 26, 2010 at 4:47 PM, Marco Ballesio wrote: > Hi, > > On Thu, Nov 25, 2010 at 1:51 PM, Thomas Vander Stichele > wrote: > .. snip .. > > > > > Who else is interested in writing part of a GStreamer book? > > Networking for me, please! > > Regards, > Marco > > > > > Thomas > > > > -- > > Keep me in your bed all day > > Nothing heals me like you do > > -- > > URGent, best radio on the net - 24/7 ! > > http://urgent.fm/ > > > > > > > > > ------------------------------------------------------------------------------ > > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > > Tap into the largest installed PC base & get more eyes on your game by > > optimizing for Intel(R) Graphics Technology. Get started today with the > > Intel(R) Software Partner Program. Five $500 cash prizes are up for > grabs. > > http://p.sf.net/sfu/intelisp-dev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- www.mediamagictechnologies.com (Gstreamer, ffmpeg, Red5, Streaming) twitter handle: @gst_kaps http://www.linkedin.com/in/kapilagrawal -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Sat Dec 4 15:09:25 2010 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 04 Dec 2010 16:09:25 +0200 Subject: [gst-devel] The 1.0 plan In-Reply-To: <1291404994856-3071651.post@n4.nabble.com> References: <1289307210.1963.174.camel@meany> <1291396157978-3071403.post@n4.nabble.com> <1291396671.3453.92.camel@odin.lan> <1291404994856-3071651.post@n4.nabble.com> Message-ID: <4CFA4B95.7070701@hora-obscura.de> Am 03.12.2010 21:36, schrieb wally_bkg: > > > Sebastian Dr?ge-7 wrote: >> >> On Fri, 2010-12-03 at 09:09 -0800, wally_bkg wrote: >> >>> I also think the best documentation is a working sample code, with decent >>> comments. The only way I could figure out how to interpret video data >>> pixels in a buffer (either as a plugin or appsink) was to stumble around >>> the >>> source code for gstgamma and gstvideoscale. >>> >>> Of course the data you get in a buffer depends on the caps, but am I the >>> only one who thinks there has to be a better way than: >>> >>> [...] >> >> For this you should use the libgstvideo API to get the video format, >> width, height, rowstride, component widths/heights, component offsets, >> pixel strides, etc. from the caps: >> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstvideo.html >> >> > > Actually it was the nested function call to get the caps that seemed goofy > to me. I still need to do that on the buffer to use most of these API > functions. structure = gst_caps_get_structure (gst_buffer_get_caps(buf), 0); well read the docs, and you'll see that caps can have multimple structure entries (you specify 0 to get the first entry). > > Unfortunately not all the API features I might like to use are in the 10.28 > version that is in Ubuntu 10.04. Since I don't intend to maintain a > distribution, I need to target something as a lowest common denominator (it > was Ubuntu 8.04 when I first started learning about gstreamer). You can use the GStreamer developer PPA to have the latest releases on ubuntu. https://launchpad.net/~gstreamer-developers/+archive/ppa > > I'm still trying to decide if overall my problem is better solved by > processing the video data by writing a plugin or using appsink & appsrc. > I'm playing around right now with both approaches, but straight off the top > using appsink & appsrc lets me create my application with a simple make file > instead of needing all the automake stuff to build a plug in. Once I'd > built the plugin template the first time, "make" was all I needed > afterwards, but I'd sure hate to have start with virgin Ubuntu install, and > the one file I modified for my "plugin" and have to show someone else how to > build and install it again! > Maybe lets discuss this in a separate thread :) Stefan From bmschwar at fas.harvard.edu Sat Dec 4 17:38:14 2010 From: bmschwar at fas.harvard.edu (Benjamin M. Schwartz) Date: Sat, 04 Dec 2010 11:38:14 -0500 Subject: [gst-devel] Tell me about y4menc and other encoders. In-Reply-To: <1291413235446-3071830.post@n4.nabble.com> References: <1291413235446-3071830.post@n4.nabble.com> Message-ID: <4CFA6E76.1000003@fas.harvard.edu> On 12/03/2010 04:53 PM, wally_bkg wrote: > I first tried: jpegenc, mpeg2enc, and theoraenc. I could eliminate mpeg2enc > from a combination of too high CPU load and too low video quality (using > only the defaults). All of these elements take an argument that allows you to specify your desired bitrate/quality. There's nothing especially great about the defaults, which may be incomparable from one element to another. > Both > pipelines seem to work OK but CPU usage seems high with theoraenc being > about 40% more CPU but with about half the file size of jpegenc. This is not a surprise. Theora compresses video by comparing frames to each other, while JPEG compresses each frame independently. Theora will always achieve better quality in the same output filesize, but will use more CPU. My > problem here is while the files play back fine in Totem, there is something > wrong with them if I try to play them on a Windows system (that is set up > for MJPEG and OGG/Theora). If you want help debugging this you'll need to be more specific about the playback system and the problem. Latest VLC generally plays back Theora correctly under Windows. > Then I discovered y4menc: > while it gives about 30X the file size of theoraenc it uses very little CPU, > less than half of the jpegenc pipeline. y4m is uncompressed lossless raw video. Maybe you would enjoy http://xiph.org/video/vid1.shtml --Ben -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: OpenPGP digital signature URL: From ensonic at hora-obscura.de Sat Dec 4 23:31:09 2010 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sun, 05 Dec 2010 00:31:09 +0200 Subject: [gst-devel] Which CODEC for mjpg? In-Reply-To: <1291393988296-3071342.post@n4.nabble.com> References: <4CF3F0DF.201@awpi.com> <1291393988296-3071342.post@n4.nabble.com> Message-ID: <4CFAC12D.1050003@hora-obscura.de> Am 03.12.2010 18:33, schrieb wally_bkg: > > I have the opposite problem. I can make jpeg video files with: > > gst-launch v4l2src ! video/x-raw-yuv ! queue ! ffmpegcolorspace ! jpegenc ! > avimux ! filesink location=mjpeg.avi > > and the file plays back fine in gstreamer (using Totum on Ubuntu 10.04) but > when I copy it over to a windows system it doesn't play. There are several > MJPEG playback codecs on the Windows system. Gspot Codec Information on > Windows shows a valid playback pipeline, but errors in rendering. > > > I tried Theoraenc instead with: > > gst-launch v4l2src ! queue ! ffmpegcolorspace ! theoraenc ! oggmux ! > filesink location=theora.ogv > > and again it plays fine in Totum but not on Windows (the Windows system can > play back the ogg video sample files distributed with Ubuntu). > > > > Try gst-launch -e to make gst-launch sending and EOS on Ctrl-c (or use num-buffers on v4l2src). Stefan From thiago.sousa.santos at collabora.co.uk Sat Dec 4 06:30:58 2010 From: thiago.sousa.santos at collabora.co.uk (Thiago Sousa Santos) Date: Sat, 04 Dec 2010 02:30:58 -0300 Subject: [gst-devel] Camerabin2 Prototype and IRC Meeting In-Reply-To: <1291302452.7078.13.camel@blacksheep> References: <1291302452.7078.13.camel@blacksheep> Message-ID: <1291440658.3956.3.camel@blacksheep> On Thu, 2010-12-02 at 12:07 -0300, Thiago Sousa Santos wrote: > Hello, > > As mentioned some time ago, we started planning camerabin2. > > The good news is that a small prototype is ready at > http://gitorious.org/gstreamer-camerabin2/gst-plugins-bad > > It is as simple as it gets, being able to take pictures, record videos > and showing a viewfinder. Feedback, code review, patches, tests are > welcome. I'm currently trying to get the code in shape to push it as an > experimental plugin into -bad module. > > I'd like arrange meeting with people interested in camerabin2 to discuss > desired features and problems with this new design. I'm already poking > some people that mentioned interest to set up a date and time. We'd like > to do this soon, so expect it to be early next week (Monday or Tuesday). Based on the feedback I got from the interested people and their timezones: http://www.timeanddate.com/worldclock/meetingtime.html?day=7&month=12&year=2010&p1=101&p2=209&p3=64&p4=37 I guess 14PM UTC, Tuesday is a nice option. We'd meet on IRC on #camerabin2. Any major problems with this? > > Details on camerabin2 design are on the wiki: > http://www.gstreamer.net/wiki/CameraBin > > -- > Thiago > > > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From shankar.ganesh at ngintech.com Sun Dec 5 08:37:17 2010 From: shankar.ganesh at ngintech.com (shankar ganesh) Date: Sun, 5 Dec 2010 13:07:17 +0530 Subject: [gst-devel] Problem with mulicast streaming . Message-ID: Hello All, I am using the Gstreamer.0.10 version in a Intel_CE_2110_Media_Processor_Platform which uses the Gstreamer.0.10 libraries When I try to do a uncast UDP stream the platform streams successfully, but when I try to do a multicast udp stream the below errors are thrown from the platform Name: udpsrc0 Error quark - gst-resource-error-quark Error code 13 could not get/set settings from resource The ifconfig interface details in the platform is eth0 - up broadcast running multicast Below are the g_object-set calls. G_object-set (G_object (inputsrc),"multicast-group","239.1.1.1", mcastaddress, NULL); G_object-set (G_object(inputsrc),"multicast-interface","eth0",mcastaddress,NULL); I could see the Platform could not set/get the properties associated with gstream muticast. I am trying to understand where the problem could be,any inputs in these regards could be great help to me. Thanks and regards, shankar ganesh -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Sun Dec 5 12:37:18 2010 From: mcygogo at gmail.com (michael young) Date: Sun, 5 Dec 2010 19:37:18 +0800 Subject: [gst-devel] got some error when I test the FLV demuxer Message-ID: Hi all: I test the FLV demuxer like this: gst-launch -v filesrc location=./gsd1.flv ! flvdemux ! audioconvert ! autoaudiosink and this is the method provided by gstflvdemux.c. But I got the error like this: Setting pipeline to PAUSED ... Pipeline is PREROLLING ... /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:video: caps = video/x-h264, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:flvdemux0: Internal data stream error. Additional debug info: gstflvdemux.c(2282): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstFlvDemux:flvdemux0: stream stopped, reason not-linked ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:audio: caps = NULL /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:video: caps = NULL Freeing pipeline ... So , what's the information means? How I fix it ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From andreynech at googlemail.com Sun Dec 5 15:13:13 2010 From: andreynech at googlemail.com (Andrey Nechypurenko) Date: Sun, 5 Dec 2010 15:13:13 +0100 Subject: [gst-devel] dynamically change video resolution In-Reply-To: References: Message-ID: Hi, > ??? Now I want to change the resolution from D1->CIF on the fly, currently, > I do the following: > ??? 1. get the src pad of queue > ??? 2. block the pad > ??? 3. change the properties of resizer and the encoder. > ??? 4. unblock the pad In my project I was doing something similar. You can take a look at the function which adjust caps for the scaler element here: http://www.gitorious.org/veter/veter/blobs/master/src/vehicle/VideoSenderThread.cpp#line264 It assumes that there are scaler element named "qos-scaler" followed by caps filter named "qos-caps" . HTH, Andrey. From nico at inattendu.org Sun Dec 5 18:05:16 2010 From: nico at inattendu.org (Nicolas Bertrand) Date: Sun, 05 Dec 2010 21:05:16 +0400 Subject: [gst-devel] Camerabin2 Prototype and IRC Meeting In-Reply-To: <1291440658.3956.3.camel@blacksheep> References: <1291302452.7078.13.camel@blacksheep> <1291440658.3956.3.camel@blacksheep> Message-ID: <4CFBC64C.9050104@inattendu.org> > Based on the feedback I got from the interested people and their > timezones: > http://www.timeanddate.com/worldclock/meetingtime.html?day=7&month=12&year=2010&p1=101&p2=209&p3=64&p4=37 > > I guess 14PM UTC, Tuesday is a nice option. We'd meet on IRC on > #camerabin2. > > Any major problems with this? > I'll be also pleased to join thi IRC meeting. I have developed my "own" specific camerabin for luciole, and be very interested to participate. From bisht.sudarshan at gmail.com Sun Dec 5 19:33:59 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Mon, 6 Dec 2010 00:03:59 +0530 Subject: [gst-devel] got some error when I test the FLV demuxer In-Reply-To: References: Message-ID: This means .flv file ( which is a container format) has h264 video stream and you need to have appropriate h264 decoder plugin in order to decode the video stream. If your end goal is to play this file then you can use following pipeline ; gst-launch -v filesrc location=./gsd1.flv ! flvdemux name=demux demux. ! queue ! decodebin ! autovideosink demux. ! queue ! decodebin ! autoaudiosink On Sun, Dec 5, 2010 at 5:07 PM, michael young wrote: > Hi all: > I test the FLV demuxer like this: gst-launch -v filesrc > location=./gsd1.flv ! flvdemux ! audioconvert ! autoaudiosink > and this is the method provided by gstflvdemux.c. But I got the error like > this: > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:video: caps = > video/x-h264, pixel-aspect-ratio=(fraction)1/1, > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:flvdemux0: Internal > data stream error. > Additional debug info: > gstflvdemux.c(2282): gst_flv_demux_loop (): > /GstPipeline:pipeline0/GstFlvDemux:flvdemux0: > stream stopped, reason not-linked > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:audio: caps = NULL > /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:video: caps = NULL > Freeing pipeline ... > > So , what's the information means? How I fix it ? > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From webmail.hce at gmail.com Mon Dec 6 02:20:09 2010 From: webmail.hce at gmail.com (hce) Date: Mon, 6 Dec 2010 12:20:09 +1100 Subject: [gst-devel] Building gstreamer from source and plugin Message-ID: Hi, I've just built gstreamer-0.10.31 on a linux box (installed locally), but failed to run example program such as helloword. There was an error " could not find plugin mad". After checking to the source code, I found that mad was used by gst_element_factory_make ("mad", "decode"), I then installed both gst-plugins-base-0.10.31 and gst-plugins-good-0.10.26, and exported gst-plugins-base/Install/lib/pkgconfig to PKG_CONFIG_PATH. pkg-config --list-all | grep gstreamer shown gstreamer libraries, but there was no mad package. Where should I get mad plugin? Thank you. Kind regards, jupiter From thiagossantos at gmail.com Mon Dec 6 02:51:15 2010 From: thiagossantos at gmail.com (Thg.) Date: Sun, 5 Dec 2010 22:51:15 -0300 Subject: [gst-devel] Building gstreamer from source and plugin In-Reply-To: References: Message-ID: On Sun, Dec 5, 2010 at 10:20 PM, hce wrote: > Hi, > > I've just built gstreamer-0.10.31 on a linux box (installed locally), > but failed to run example program such as helloword. There was an > error " could not find plugin mad". After checking to the source code, > I found that mad was used by gst_element_factory_make ("mad", > "decode"), I then installed both gst-plugins-base-0.10.31 and > gst-plugins-good-0.10.26, and exported > gst-plugins-base/Install/lib/pkgconfig to PKG_CONFIG_PATH. > > pkg-config --list-all | grep gstreamer shown gstreamer libraries, but > there was no mad package. Where should I get mad plugin? > mad plugin is part of the gst-plugins-ugly module. You can list installed plugins using "gst-inspect" tool. > > Thank you. > > Kind regards, > > jupiter > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Thiago Sousa Santos -------------- next part -------------- An HTML attachment was scrubbed... URL: From braunsquared at gmail.com Mon Dec 6 02:53:42 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Sun, 5 Dec 2010 20:53:42 -0500 Subject: [gst-devel] Building gstreamer from source and plugin In-Reply-To: References: Message-ID: <28007555-268C-4A62-8013-0CA26CD20BB5@gmail.com> I believe the mad plugin is an -ugly or -bad plugin. You should be able to find out on the gstreamer site. Tim Sent from my iPhone On Dec 5, 2010, at 8:20 PM, hce wrote: > Hi, > > I've just built gstreamer-0.10.31 on a linux box (installed locally), > but failed to run example program such as helloword. There was an > error " could not find plugin mad". After checking to the source code, > I found that mad was used by gst_element_factory_make ("mad", > "decode"), I then installed both gst-plugins-base-0.10.31 and > gst-plugins-good-0.10.26, and exported > gst-plugins-base/Install/lib/pkgconfig to PKG_CONFIG_PATH. > > pkg-config --list-all | grep gstreamer shown gstreamer libraries, but > there was no mad package. Where should I get mad plugin? > > Thank you. > > Kind regards, > > jupiter > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From mcygogo at gmail.com Mon Dec 6 04:03:17 2010 From: mcygogo at gmail.com (michael) Date: Mon, 6 Dec 2010 11:03:17 +0800 Subject: [gst-devel] got some error when I test the FLV demuxer In-Reply-To: References: Message-ID: Hi sudarshan: I tried your method, but it's still can not work, there is no video and audio output. So It's that because I have on H264 plugin in my system? which gstreamer plugin I should install ? tks :) On Mon, Dec 6, 2010 at 2:33 AM, sudarshan bisht wrote: > This means .flv file ( which is a container format) has h264 video stream > and you need to have appropriate h264 decoder plugin in order to decode the > video stream. > > If your end goal is to play this file then you can use following pipeline ; > gst-launch -v filesrc location=./gsd1.flv ! flvdemux name=demux demux. ! > queue ! decodebin ! autovideosink demux. ! queue ! decodebin ! > autoaudiosink > > > > > > On Sun, Dec 5, 2010 at 5:07 PM, michael young wrote: > >> Hi all: >> I test the FLV demuxer like this: gst-launch -v filesrc >> location=./gsd1.flv ! flvdemux ! audioconvert ! autoaudiosink >> and this is the method provided by gstflvdemux.c. But I got the error >> like this: >> >> Setting pipeline to PAUSED ... >> Pipeline is PREROLLING ... >> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:video: caps = >> video/x-h264, pixel-aspect-ratio=(fraction)1/1, >> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:flvdemux0: Internal >> data stream error. >> Additional debug info: >> gstflvdemux.c(2282): gst_flv_demux_loop (): >> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0: >> stream stopped, reason not-linked >> ERROR: pipeline doesn't want to preroll. >> Setting pipeline to NULL ... >> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:audio: caps = NULL >> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:video: caps = NULL >> Freeing pipeline ... >> >> So , what's the information means? How I fix it ? >> >> >> ------------------------------------------------------------------------------ >> What happens now with your Lotus Notes apps - do you make another costly >> upgrade, or settle for being marooned without product support? Time to >> move >> off Lotus Notes and onto the cloud with Force.com, apps are easier to >> build, >> use, and manage than apps on traditional platforms. Sign up for the Lotus >> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > -- > Regards, > > Sudarshan Bisht > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Mon Dec 6 04:17:02 2010 From: mcygogo at gmail.com (michael) Date: Mon, 6 Dec 2010 11:17:02 +0800 Subject: [gst-devel] got some error when I test the FLV demuxer In-Reply-To: References: Message-ID: I got some info for 264 : $ gst-inspect-0.10 | grep 264 typefindfunctions: video/x-h264: h264, x264, 264 rtp: rtph264depay: RTP H264 depayloader rtp: rtph264pay: RTP H264 payloader h264parse: h264parse: H264Parse ffmpeg: ffmux_ipod: FFmpeg iPod H.264 MP4 format muxer ffmpeg: ffdec_h264: FFmpeg H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 decoder On Mon, Dec 6, 2010 at 11:03 AM, michael wrote: > Hi sudarshan: > I tried your method, but it's still can not work, there is no video and > audio output. > So It's that because I have on H264 plugin in my system? which gstreamer > plugin I > should install ? tks :) > > > On Mon, Dec 6, 2010 at 2:33 AM, sudarshan bisht > wrote: > >> This means .flv file ( which is a container format) has h264 video stream >> and you need to have appropriate h264 decoder plugin in order to decode the >> video stream. >> >> If your end goal is to play this file then you can use following pipeline >> ; >> gst-launch -v filesrc location=./gsd1.flv ! flvdemux name=demux demux. ! >> queue ! decodebin ! autovideosink demux. ! queue ! decodebin ! >> autoaudiosink >> >> >> >> >> >> On Sun, Dec 5, 2010 at 5:07 PM, michael young wrote: >> >>> Hi all: >>> I test the FLV demuxer like this: gst-launch -v filesrc >>> location=./gsd1.flv ! flvdemux ! audioconvert ! autoaudiosink >>> and this is the method provided by gstflvdemux.c. But I got the error >>> like this: >>> >>> Setting pipeline to PAUSED ... >>> Pipeline is PREROLLING ... >>> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:video: caps = >>> video/x-h264, pixel-aspect-ratio=(fraction)1/1, >>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:flvdemux0: >>> Internal data stream error. >>> Additional debug info: >>> gstflvdemux.c(2282): gst_flv_demux_loop (): >>> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0: >>> stream stopped, reason not-linked >>> ERROR: pipeline doesn't want to preroll. >>> Setting pipeline to NULL ... >>> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:audio: caps = NULL >>> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:video: caps = NULL >>> Freeing pipeline ... >>> >>> So , what's the information means? How I fix it ? >>> >>> >>> ------------------------------------------------------------------------------ >>> What happens now with your Lotus Notes apps - do you make another costly >>> upgrade, or settle for being marooned without product support? Time to >>> move >>> off Lotus Notes and onto the cloud with Force.com, apps are easier to >>> build, >>> use, and manage than apps on traditional platforms. Sign up for the Lotus >>> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> -- >> Regards, >> >> Sudarshan Bisht >> >> >> ------------------------------------------------------------------------------ >> What happens now with your Lotus Notes apps - do you make another costly >> upgrade, or settle for being marooned without product support? Time to >> move >> off Lotus Notes and onto the cloud with Force.com, apps are easier to >> build, >> use, and manage than apps on traditional platforms. Sign up for the Lotus >> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ved.kpl at gmail.com Mon Dec 6 05:07:33 2010 From: ved.kpl at gmail.com (ved kpl) Date: Mon, 6 Dec 2010 09:37:33 +0530 Subject: [gst-devel] got some error when I test the FLV demuxer In-Reply-To: References: Message-ID: try 1) gst-launch -v filesrc location=./gsd1.flv ! decodebin2 ! ffmpegcolorspace ! videoscale ! autovideosink -v 2) gst-launch -v filesrc location=./gsd1.flv ! decodebin2 name=bin ! ffmpegcolorspace ! videoscale ! autovideosink bin. ! audioconvert ! audioresample ! autoaudiosink -v maybe u need to install the audio/video decoder plugins... if u want this to work On Mon, Dec 6, 2010 at 8:47 AM, michael wrote: > I got some info ?for 264 : > $ gst-inspect-0.10 | grep 264 > typefindfunctions: video/x-h264: h264, x264, 264 > rtp: ?rtph264depay: RTP H264 depayloader > rtp: ?rtph264pay: RTP H264 payloader > h264parse: ?h264parse: H264Parse > ffmpeg: ?ffmux_ipod: FFmpeg iPod H.264 MP4 format muxer > ffmpeg: ?ffdec_h264: FFmpeg H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 > decoder > > On Mon, Dec 6, 2010 at 11:03 AM, michael wrote: >> >> Hi sudarshan: >> ?? ?I tried your method, but it's still can not work, there is no video >> and audio output. >> So It's that because I have on H264 plugin in my system? which gstreamer >> plugin I >> should install ? tks :) >> >> On Mon, Dec 6, 2010 at 2:33 AM, sudarshan bisht >> wrote: >>> >>> This means .flv file ( which is a container format) has h264 video stream >>> and you need to have appropriate h264 decoder plugin in order to decode the >>> video stream. >>> If your end goal is to play this file then you can use following pipeline >>> ; >>> gst-launch?-v filesrc location=./gsd1.flv ! flvdemux name=demux demux. ! >>> queue ! decodebin ! autovideosink ?demux. ! queue ! decodebin ! >>> autoaudiosink >>> >>> >>> >>> >>> On Sun, Dec 5, 2010 at 5:07 PM, michael young wrote: >>>> >>>> Hi all: >>>> ?? ?I test the FLV demuxer like this:?gst-launch -v filesrc >>>> location=./gsd1.flv ! flvdemux ! audioconvert ! autoaudiosink >>>> and this is the method provided by gstflvdemux.c. But I got the error >>>> like this: >>>> Setting pipeline to PAUSED ... >>>> Pipeline is PREROLLING ... >>>> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:video: caps = >>>> video/x-h264, pixel-aspect-ratio=(fraction)1/1, >>>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:flvdemux0: >>>> Internal data stream error. >>>> Additional debug info: >>>> gstflvdemux.c(2282): gst_flv_demux_loop (): >>>> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0: >>>> stream stopped, reason not-linked >>>> ERROR: pipeline doesn't want to preroll. >>>> Setting pipeline to NULL ... >>>> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:audio: caps = NULL >>>> /GstPipeline:pipeline0/GstFlvDemux:flvdemux0.GstPad:video: caps = NULL >>>> Freeing pipeline ... >>>> So , what's the information means? How I fix it ? >>>> >>>> ------------------------------------------------------------------------------ >>>> What happens now with your Lotus Notes apps - do you make another costly >>>> upgrade, or settle for being marooned without product support? Time to >>>> move >>>> off Lotus Notes and onto the cloud with Force.com, apps are easier to >>>> build, >>>> use, and manage than apps on traditional platforms. Sign up for the >>>> Lotus >>>> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>> >>> >>> >>> -- >>> Regards, >>> >>> Sudarshan Bisht >>> >>> >>> ------------------------------------------------------------------------------ >>> What happens now with your Lotus Notes apps - do you make another costly >>> upgrade, or settle for being marooned without product support? Time to >>> move >>> off Lotus Notes and onto the cloud with Force.com, apps are easier to >>> build, >>> use, and manage than apps on traditional platforms. Sign up for the Lotus >>> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >> > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From yogeshm.007 at gmail.com Mon Dec 6 06:24:34 2010 From: yogeshm.007 at gmail.com (Yogesh Marwaha) Date: Mon, 6 Dec 2010 10:54:34 +0530 Subject: [gst-devel] Please check this pipeline (regarding faac) Message-ID: Hi, I want to generate an aac file. Is the following pipeline correct (i.e. will it generate valid file): - gst-launch audiotestsrc wave=sine num-buffers=100 ! audioconvert ! faac outputformat=1 ! filesink location=sine.aac Regards, -- Yogesh M http://sparklemedia.sourceforge.net/ http://mazedaar.wordpress.com/ http://snakeeyes.wordpress.com/ From webmail.hce at gmail.com Mon Dec 6 06:24:41 2010 From: webmail.hce at gmail.com (hce) Date: Mon, 6 Dec 2010 16:24:41 +1100 Subject: [gst-devel] Building gstreamer from source and plugin In-Reply-To: References: Message-ID: > mad plugin is part of the gst-plugins-ugly module. > > You can list installed plugins using "gst-inspect" tool. Thanks. I installed gst-plugins-ugly in a non-standard local directory, and set --enable-mad. But it failed to build mad plugin as "libid3tag not available". So I built the libid3tag-0.15.0b in a non-standard local directory, and set --enable-id3tag --with-id3tag-prefix=/gstreamer/libid3tag-0.15.0b/Install, the gst-plugins-ugly displayed error of unknown tags, and suggested to "Package id3tag was not found in the pkg-config search path. Perhaps you should add the directory containing `id3tag.pc'". But, libid3tag-0.15.0b does not have id3tag.pc, only "libid3tag.a libid3tag.la libid3tag.so libid3tag.so.0 libid3tag.so.0.2.0" are in Install/lib. What are correct package tags I can add external packages to gst-plugins-ugly configuration? I guess I need download libmad-0.15.1b.tar.gz as well and add --with-mad=/gstreamer/libmad-0.15.1b/Install as well? Thank you. Kind regards, jupiter From ved.kpl at gmail.com Mon Dec 6 06:46:47 2010 From: ved.kpl at gmail.com (ved kpl) Date: Mon, 6 Dec 2010 11:16:47 +0530 Subject: [gst-devel] Please check this pipeline (regarding faac) In-Reply-To: References: Message-ID: yes, it should. On Mon, Dec 6, 2010 at 10:54 AM, Yogesh Marwaha wrote: > Hi, > > I want to generate an aac file. Is the following pipeline correct > (i.e. will it generate valid file): - > > gst-launch audiotestsrc wave=sine num-buffers=100 ! audioconvert ! > faac outputformat=1 ! filesink location=sine.aac > > > Regards, > > -- > Yogesh M > http://sparklemedia.sourceforge.net/ > http://mazedaar.wordpress.com/ > http://snakeeyes.wordpress.com/ > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From 4ernov at gmail.com Mon Dec 6 10:03:53 2010 From: 4ernov at gmail.com (4ernov) Date: Mon, 6 Dec 2010 12:03:53 +0300 Subject: [gst-devel] State change hangs if pipeline was changed in PAUSED Message-ID: Hello, I've already posted a couple of questions here concerning my case with duplicated video stream. Now it works quite right but some strange behavior still exists. When I change the pipeline while it's in PAUSED state and then try to resume it (setting state PLAYING with gst_element_set_state) it never continues to play anymore. gst_element_set_state() returns GST_STATE_CHANGE_ASYNC and gst_element_get_state() (as in documentation) waits for change to complete forever. Here's a test program: #include #include GstElement *_pipeline, *_fakesink, *_remotesink, *_mpoint, *_queue1, *_queue2, *_videobin, *_pipe, *gdppay; void create_pipeline_bin() { GstElement *tee, *queue1, *queue2, *valve, *xvimagesink, *audiosink; _pipeline = gst_element_factory_make("playbin", NULL); tee = gst_element_factory_make("tee", NULL); xvimagesink = gst_element_factory_make("ximagesink", NULL); GstPad* pad; _videobin = gst_bin_new("videobin"); _queue1 = queue1 = gst_element_factory_make("queue", NULL); _queue2 = queue2 = gst_element_factory_make("queue", NULL); valve = gst_element_factory_make("valve", NULL); _fakesink = gst_element_factory_make("fakesink", NULL); audiosink = gst_element_factory_make("fakesink", NULL); gst_object_ref(_fakesink); _mpoint = valve; gst_bin_add_many(GST_BIN(_videobin), tee, queue1, queue2, valve, xvimagesink, _fakesink, NULL); gst_element_link_many(tee, queue1, xvimagesink, NULL); gst_element_link_many(tee, queue2, valve, _fakesink, NULL); pad = gst_element_get_static_pad (tee, "sink"); gst_element_add_pad (_videobin, gst_ghost_pad_new ("sink", pad)); gst_object_unref (GST_OBJECT (pad)); g_object_set(G_OBJECT (_pipeline), "uri", "file:///home/alex/test.mp4", NULL); g_object_set(G_OBJECT (_pipeline), "video-sink", _videobin, NULL); g_object_set(G_OBJECT (_pipeline), "audio-sink", audiosink, NULL); gst_element_set_state (_pipeline, GST_STATE_PLAYING); } void create_remote_bin() { GstElement *videomixer, *ffmpegcolorspace; GstPad* pad; videomixer = gst_element_factory_make("videomixer", NULL); ffmpegcolorspace = gst_element_factory_make("ffmpegcolorspace", NULL); _pipe = gst_element_factory_make("xvimagesink", NULL); _remotesink = gst_bin_new("vmixerbin"); gst_bin_add_many(GST_BIN(_remotesink), videomixer, ffmpegcolorspace, _pipe, NULL); gst_element_link_many(videomixer, ffmpegcolorspace, _pipe, NULL); pad = gst_element_get_request_pad (videomixer, "sink_%d"); gst_element_add_pad (_remotesink, gst_ghost_pad_new ("sink", pad)); gst_object_unref (GST_OBJECT (pad)); gst_object_ref(_remotesink); } void connect_remote_client() { if (_remotesink && _mpoint && _videobin) { GstState state = GST_STATE(_pipeline); gst_element_unlink(_mpoint, _fakesink); gst_bin_remove(GST_BIN(_videobin), _fakesink); gst_element_set_state (_fakesink, GST_STATE_NULL); gst_bin_add(GST_BIN(_videobin), _remotesink); gst_element_link(_mpoint, _remotesink); gst_element_sync_state_with_parent(_remotesink); gst_element_set_state (_pipeline, state); } } void disconnect_remote_client() { if (_remotesink && _mpoint && _videobin && _fakesink) { GstState state = GST_STATE(_pipeline); gst_element_unlink(_mpoint, _remotesink); gst_bin_remove(GST_BIN(_videobin), _remotesink); gst_element_set_state (_remotesink, GST_STATE_NULL); gst_bin_add(GST_BIN(_videobin), _fakesink); gst_element_link(_mpoint, _fakesink); gst_element_sync_state_with_parent(_fakesink); gst_element_set_state (_pipeline, state); } } void pause_pipeline() { gst_element_set_state(_pipeline, GST_STATE_PAUSED); } void play_pipeline() { gst_element_set_state(_pipeline, GST_STATE_PLAYING); } gboolean connect_callback(gpointer) { connect_remote_client(); return FALSE; } gboolean disconnect_callback(gpointer) { disconnect_remote_client(); return FALSE; } gboolean pause_callback(gpointer) { pause_pipeline(); return FALSE; } gboolean play_callback(gpointer) { play_pipeline(); return FALSE; } int main(int argc, char *argv[]) { GMainLoop* loop = g_main_loop_new (NULL, FALSE); gst_init(&argc, &argv); create_remote_bin(); create_pipeline_bin(); g_timeout_add(7000, connect_callback, NULL); g_timeout_add(16000, pause_callback, NULL); g_timeout_add(17000, disconnect_callback, NULL); g_timeout_add(18000, play_callback, NULL); g_main_loop_run (loop); return 0; } Second xvimagesink connects successfully in 7 secs after start. Then in 16 sec the pipeline is set to PAUSED and in 17 sec second video sink is disconnected. But in 18 sec pipeline doesn't want to get to PLAYING state anymore. No messages is posted to output after trying to set the pipeline PLAYING state, even on DEBUG log level. One certain message is posted before it: GST_STATES gstelement.c:2515:gst_element_set_state_func: element was busy with async state change I should also notice that if I make pipeline changes in PLAYING state everything work flawlessly. Is it bug in GStreamer or should I make state change any other way? From shankar.ganesh at ngintech.com Mon Dec 6 10:21:08 2010 From: shankar.ganesh at ngintech.com (shankar ganesh) Date: Mon, 6 Dec 2010 14:51:08 +0530 Subject: [gst-devel] Problem with mulicast streaming . In-Reply-To: References: Message-ID: Hi All, Could any body help me out from the below problem , i got stuck with this for weeks. Thanks in advance Shankar ganesh On Sun, Dec 5, 2010 at 1:07 PM, shankar ganesh wrote: > Hello All, > > I am using the Gstreamer.0.10 version in a > > Intel_CE_2110_Media_Processor_Platform which uses the Gstreamer.0.10 > libraries > When I try to do a uncast UDP stream the platform streams successfully, but > when I try to > do a multicast udp stream the below errors are thrown from the platform > > Name: udpsrc0 > Error quark - gst-resource-error-quark > Error code 13 > could not get/set settings from resource > > The ifconfig interface details in the platform is > > eth0 - up broadcast running multicast > > Below are the g_object-set calls. > > G_object-set (G_object (inputsrc),"multicast-group","239.1.1.1", > mcastaddress, NULL); > > G_object-set > (G_object(inputsrc),"multicast-interface","eth0",mcastaddress,NULL); > > > I could see the Platform could not set/get the properties associated with > gstream muticast. > > I am trying to understand where the problem could be,any inputs in these > regards could be great help to me. > > Thanks and regards, > shankar ganesh > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gibrovacco at gmail.com Mon Dec 6 11:10:35 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Mon, 6 Dec 2010 12:10:35 +0200 Subject: [gst-devel] Problem with mulicast streaming . In-Reply-To: References: Message-ID: Hi Shankar, > > On Sun, Dec 5, 2010 at 1:07 PM, shankar ganesh > wrote: >> >> Hello All, >> >> I am using the Gstreamer.0.10 version in a >> >> Intel_CE_2110_Media_Processor_Platform which uses the Gstreamer.0.10 >> libraries >> When I try to do a uncast UDP stream the platform streams successfully, >> but when I try to >> do a multicast udp stream the below errors are thrown from the platform >> >> Name: udpsrc0 >> Error quark - gst-resource-error-quark >> Error code 13 >> could not get/set settings from resource I don't have much experience with multicast (just a few tests on desktop machines so far), btw maybe the following will help you. My suspect is that for some reasons your network iface doesn't support multicast or that the user you're running it from does not have enough privileges. What if you try and execute the pipeline from a privileged user (e.g. root)? What if you try in on another machine (e.g. a desktop)? Can you report the pipeline(s) you're running? Moreover, it would be helpful if you can report somewhere the debug output you get when setting for instance GST_DEBUG=udpsrc:5 If more than 20 lines, please report it to pastebin or as a txt attachment. Regards >> >> The ifconfig interface details in the platform is >> >> eth0 - up broadcast running multicast >> >> Below are the g_object-set calls. >> >> G_object-set (G_object (inputsrc),"multicast-group","239.1.1.1", >> mcastaddress, NULL); >> >> G_object-set >> (G_object(inputsrc),"multicast-interface","eth0",mcastaddress,NULL); >> >> I could see the Platform could not set/get the properties associated with >> gstream muticast. >> >> I am trying to understand where the problem could be,any inputs in these >> regards could be great help to me. >> >> Thanks and regards, >> shankar ganesh >> >> >> >> > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From bisht.sudarshan at gmail.com Mon Dec 6 13:36:54 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Mon, 6 Dec 2010 18:06:54 +0530 Subject: [gst-devel] Building gstreamer from source and plugin In-Reply-To: References: Message-ID: I would suggest you to install all the gstreamer packages and related libraries again in clean way. For this create a local directory gst_lib and install all the packages, libraries in this directory by following commands; ./configure --prefix=~path/gst_lib make make install (set these environment variables ) export PKG_CONFIG_PATH=$PKG_CONFIG_PATH:~path/gst_lib export LD_LIBRARY_PATH=$LD_LIBRARY_PATH:~path/gst_lib export GST_PLUGIN_PATH=~path/gst_lib On Mon, Dec 6, 2010 at 10:54 AM, hce wrote: > > mad plugin is part of the gst-plugins-ugly module. > > > > You can list installed plugins using "gst-inspect" tool. > > Thanks. I installed gst-plugins-ugly in a non-standard local > directory, and set --enable-mad. But it failed to build mad plugin as > "libid3tag not available". > > So I built the libid3tag-0.15.0b in a non-standard local directory, > and set --enable-id3tag > --with-id3tag-prefix=/gstreamer/libid3tag-0.15.0b/Install, the > gst-plugins-ugly displayed error of unknown tags, and suggested to > "Package id3tag was not found in the pkg-config search path. Perhaps > you should add the directory containing `id3tag.pc'". But, > libid3tag-0.15.0b does not have id3tag.pc, only "libid3tag.a > libid3tag.la libid3tag.so libid3tag.so.0 libid3tag.so.0.2.0" are in > Install/lib. > > What are correct package tags I can add external packages to > gst-plugins-ugly configuration? I guess I need download > libmad-0.15.1b.tar.gz as well and add > --with-mad=/gstreamer/libmad-0.15.1b/Install as well? > > Thank you. > > Kind regards, > > jupiter > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Mon Dec 6 16:08:47 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 06 Dec 2010 16:08:47 +0100 Subject: [gst-devel] 0.11 open for business Message-ID: <1291648127.2178.111.camel@meany> Hello Hackers, I pushed a branch called '0.11' in the gstreamer, gst-plugins-base,gst-plugins-good, gst-plugins-bad, gst-plugins-ugly, gst-ffmpeg, gst-python, gst-plugins-gl and gst-rtsp-server git modules. Some of the other modules, I hadn't checked out with write access, so those will be opened some other time... In order to get the development started, I propose to start with removing the deprecated API and fixing the padding of structures again. Also think about all methods and try to make them future proof. Remember that all of the current plans are kept in docs/random/plan-0.11.txt. When committing changes, always make sure that you keep the various modules in a compilabe state. When API changes, please write into the docs/random/porting-to-0.11.txt how one is supposed to port to your new API. Also write in the commit message that the API changed and briefly mention how to port things. The idea is that at the end of 0.11 we can compile a document with a complete porting guide. Compiling such a document later is going to be hard when there are no markers in the commit messages and no trail in the porting doc. So, let's start with cleaning up things and see how it goes. Hop in IRC and ask around if you want to help out or looking for things to do. Happy hacking, Wim Taymans I added From bard_kuo at compalcomm.com Wed Dec 1 03:11:16 2010 From: bard_kuo at compalcomm.com (bard) Date: Tue, 30 Nov 2010 18:11:16 -0800 (PST) Subject: [gst-devel] How to port gst-openmax Message-ID: <1291169476921-3066588.post@n4.nabble.com> Dear All, I am work on porting gst-openmax , but I didn't find some document described more specifically. Is there a document like porting guide ? or some flowchart available can make me figure out easily. Thanks, Bard -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-port-gst-openmax-tp3066588p3066588.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From jonathan.henson at innovisit.com Wed Dec 1 17:10:46 2010 From: jonathan.henson at innovisit.com (JonathanHenson) Date: Wed, 1 Dec 2010 08:10:46 -0800 (PST) Subject: [gst-devel] Trouble using x264enc with a tee Message-ID: <1291219846949-3067583.post@n4.nabble.com> I am writing a multiplexed video/audio streaming thread for a video server I am working on. I was having problems so I am testing, to make sure the pipeline works by knocking the data into a file. However, the end goal will be to use a multifdsink to send the stream to a socket. So, you will notice that the multifdsink element is actually a filesink for the time being. This pipeline works with other encoders, but when I use x264enc for the video encoder, the pipeline freezes and no data is put into the file. Also, there is a tee in both the audio and video portions of the pipeline so that other threads can grab the raw buffer if they need to access the data. That way only one thread is ever accessing the camera. If I remove the tee in the video pipeline, the pipeline works. Also I have tested that if I put an xvimagesink on both sides of the pipeline that both windows get the stream, so I am pretty sure that the tee is not the problem. Thanks, here is the class implementation. /* * H264Stream.cpp * * Created on: Nov 12, 2010 * Author: jonathan */ #include "H264Stream.h" H264Stream::H264Stream() : PThread (1000, NoAutoDeleteThread, HighestPriority, "H264Stream"), encoding(false) { //temporary setting of variables width = 352; height = 288; fps = 25; audioChannels = 2; audioSampleRate = 8000; bitWidth = 16; //create pipeline h264Pipeline = gst_pipeline_new("h264Pipeline"); //----------------------------------create videoPipe Elements------------------------------------------------------------------------------ //raw camera source v4l2Src = gst_element_factory_make("v4l2src", "v4l2Src"); //Text Filters chanNameFilter = gst_element_factory_make("textoverlay", "chanNameOverlay"); osdMessageFilter = gst_element_factory_make("textoverlay", "osdOverlay"); sessionTimerFilter = gst_element_factory_make("textoverlay", "sessionTimerOverlay"); //raw video caps GstCaps* rawVideoCaps = gst_caps_new_simple ("video/x-raw-yuv", "format", GST_TYPE_FOURCC, 0x30323449, "width", G_TYPE_INT, width, "height", G_TYPE_INT, height, "framerate", GST_TYPE_FRACTION, fps, 1, NULL); GstCaps* h264VideoCaps = gst_caps_new_simple ("video/x-h264","framerate", GST_TYPE_FRACTION, fps, 1, "width", G_TYPE_INT, width, "height", G_TYPE_INT, height, NULL); //video tee videoTee = gst_element_factory_make("tee", "videoTee"); //create tee src 1 receiver (videoSink) videoSink = gst_element_factory_make("appsink", "videoSink"); //create tee src 2 receiver (videoQueue) videoQueue = gst_element_factory_make("queue", "videoQueue"); videoAppSinkQueue = gst_element_factory_make("queue", "videoAppSinkQueue"); //create h264 Encoder videoEncoder = gst_element_factory_make("x264enc", "h264Enc"); //-----------------------------------------------------create audioPipe elements----------------------------------------------------------------------------- //create Alsa Source alsaSrc = gst_element_factory_make("alsasrc", "alsaSrc"); //create raw Audio Caps GstCaps* rawAudioCaps = gst_caps_new_simple("audio/x-raw-int", "channels", G_TYPE_INT, audioChannels, "rate", G_TYPE_INT, audioSampleRate, "width", G_TYPE_INT, bitWidth, "depth", G_TYPE_INT, bitWidth, "endianness", G_TYPE_INT, 1234, NULL); volume = gst_element_factory_make("volume", "volume"); //create audio tee soundTee = gst_element_factory_make("tee", "audioTee"); //create element to receive tee source #1 (audioSink) soundSink = gst_element_factory_make("appsink", "audioSink"); //create element to receive tee source #2 (audioQueue) soundQueue = gst_element_factory_make("queue", "audioQueue"); soundAppSinkQueue = gst_element_factory_make("queue", "soundAppSinkQueue"); //create an audio encoder to use when ready. soundEncoder = gst_element_factory_make("ffenc_mp2", "audioEncoder"); //-----------------------------------------------------Create Multiplexing Elements----------------------------------------------------------------------------- //create multiplexer (currently avi) multiplexer = gst_element_factory_make("avimux", "multiplexer"); //create multifdsink multifdSink = gst_element_factory_make("filesink", "multiFDSink"); g_object_set (G_OBJECT (multifdSink), "location", "/home/jonathan/test.avi" , NULL); //-----------------------------------------------------LINKERUP!---------------------------------------------------------------------------------------------- //add all elements(except for audio encoder as it isn't used yet) to the pipeline gst_bin_add_many (GST_BIN (h264Pipeline), v4l2Src, chanNameFilter, osdMessageFilter, sessionTimerFilter, videoQueue, videoAppSinkQueue, videoTee, videoSink, videoEncoder, alsaSrc, volume, soundTee, soundSink, soundQueue, soundAppSinkQueue, multiplexer, multifdSink, NULL); //link video source with text overlay surfaces bool link = gst_element_link_filtered(v4l2Src, chanNameFilter, rawVideoCaps); link = gst_element_link_filtered(chanNameFilter, osdMessageFilter, rawVideoCaps); link = gst_element_link_filtered(osdMessageFilter, sessionTimerFilter, rawVideoCaps); //link raw video with text to tee link = gst_element_link_filtered(sessionTimerFilter, videoTee, rawVideoCaps); //link video Tee to both videoSink and videoEncoder. To do this, we must request pads. //this pad is for the tee -> videoSink connection GstPad* videoSrcAppSinkPad = gst_element_get_request_pad(videoTee, "src%d"); //this pad is for the tee -> queue connection GstPad* videoSrcH264Pad = gst_element_get_request_pad(videoTee, "src%d"); //get static pads for the sinks receiving the tee GstPad* videoSinkAppSinkPad = gst_element_get_static_pad(videoAppSinkQueue, "sink"); GstPad* videoSinkH264Pad = gst_element_get_static_pad(videoQueue, "sink"); //link the pads GstPadLinkReturn padLink; padLink = gst_pad_link(videoSrcAppSinkPad, videoSinkAppSinkPad); padLink = gst_pad_link(videoSrcH264Pad, videoSinkH264Pad); gst_object_unref (GST_OBJECT (videoSrcAppSinkPad)); gst_object_unref (GST_OBJECT (videoSrcH264Pad)); gst_object_unref (GST_OBJECT (videoSinkAppSinkPad)); gst_object_unref (GST_OBJECT (videoSinkH264Pad)); link = gst_element_link_filtered(videoAppSinkQueue, videoSink, rawVideoCaps); link = gst_element_link_filtered(videoQueue, videoEncoder, rawVideoCaps); //We are done with the video part of the pipe for now. Now we link the sound elements together //link the alsa source to the volume element link = gst_element_link_filtered(alsaSrc, volume, rawAudioCaps); //link output from volume to soundTee link = gst_element_link_filtered(volume, soundTee, rawAudioCaps); //link audio Tee to both audioSink and multiplexer(when we do audio encoding we can do this with audioEncoder instead. To do this, we must request pads. //this pad is for the tee -> audioSink connection GstPad* audioSrcAppSinkPad = gst_element_get_request_pad(soundTee, "src%d"); //this pad is for the tee -> queue connection GstPad* audioSrcQueuePad = gst_element_get_request_pad(soundTee, "src%d"); //get pads for the sinks receiving the tee GstPad* audioSinkAppSinkPad = gst_element_get_static_pad(soundAppSinkQueue, "sink"); GstPad* audioSinkQueuePad = gst_element_get_static_pad(soundQueue, "sink"); //link the pads padLink = gst_pad_link(audioSrcAppSinkPad, audioSinkAppSinkPad); padLink = gst_pad_link(audioSrcQueuePad, audioSinkQueuePad); gst_object_unref (GST_OBJECT (audioSrcAppSinkPad)); gst_object_unref (GST_OBJECT (audioSrcQueuePad)); gst_object_unref (GST_OBJECT (audioSinkAppSinkPad)); gst_object_unref (GST_OBJECT (audioSinkQueuePad)); link = gst_element_link_filtered(soundAppSinkQueue, soundSink, rawAudioCaps); //Now we multiplex the two parallel streams to do this, we must request pads from the multiplexer. //this pad is for the audioQueue -> multiplex connection GstPad* audioSinkPad = gst_element_get_request_pad(multiplexer, "audio_%d"); //this pad is for the tee -> queue connection GstPad* videoSinkPad = gst_element_get_request_pad(multiplexer, "video_%d"); //get pads for the sources sending to multipexer GstPad* audioSrcPad = gst_element_get_static_pad(soundQueue, "src"); GstPad* videoSrcPad = gst_element_get_static_pad(videoEncoder, "src"); //do h264 caps negotiation //gst_pad_set_caps(videoSrcPad, h264VideoCaps); //gst_pad_set_caps(videoSinkPad, h264VideoCaps); //link the pads padLink = gst_pad_link(audioSrcPad, audioSinkPad); padLink = gst_pad_link(videoSrcPad, videoSinkPad); gst_object_unref (GST_OBJECT (audioSrcPad)); gst_object_unref (GST_OBJECT (audioSinkPad)); gst_object_unref (GST_OBJECT (videoSrcPad)); gst_object_unref (GST_OBJECT (videoSinkPad)); //finally we link the multiplexed stream to the multifdsink link = gst_element_link(multiplexer, multifdSink); gst_caps_unref(rawVideoCaps); gst_caps_unref(rawAudioCaps); gst_caps_unref(h264VideoCaps); } H264Stream::~H264Stream() { for(std::map::iterator pair = streamHandles.begin(); pair != streamHandles.end(); pair++) { g_signal_emit_by_name(multifdSink, "remove", pair->first, NULL); delete pair->second; } streamHandles.clear(); gst_element_set_state (h264Pipeline, GST_STATE_NULL); gst_object_unref (GST_OBJECT (h264Pipeline)); } void H264Stream::Main() { while(true) { PWaitAndSignal m(mutex); if(encoding) { OSDSettings osd; if(osd.getShowChanName()) { g_object_set (G_OBJECT (chanNameFilter), "silent", false , NULL); g_object_set (G_OBJECT (chanNameFilter), "text", osd.getChanName().c_str() , NULL); g_object_set (G_OBJECT (chanNameFilter), "halignment", osd.getChanNameHAlign() , NULL); g_object_set (G_OBJECT (chanNameFilter), "valignment", osd.getChanNameVAlign() , NULL); g_object_set (G_OBJECT (chanNameFilter), "wrap-mode", osd.getChanNameWordWrapMode() , NULL); g_object_set (G_OBJECT (chanNameFilter), "font-desc", osd.getChanNameFont().c_str() , NULL); g_object_set (G_OBJECT (chanNameFilter), "shaded-background", osd.getChanNameShadow() , NULL); } else { g_object_set (G_OBJECT (chanNameFilter), "text", "" , NULL); g_object_set (G_OBJECT (chanNameFilter), "silent", true , NULL); } if(osd.getShowOSDMessage()) { g_object_set (G_OBJECT (osdMessageFilter), "silent", false , NULL); g_object_set (G_OBJECT (osdMessageFilter), "text", osd.getOSDMessage().c_str() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "halignment", osd.getOSDMessageHAlign() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "valignment", osd.getOSDMessageVAlign() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "wrap-mode", osd.getOSDMessageWordWrapMode() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "font-desc", osd.getOSDMessageFont().c_str() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "shaded-background", osd.getOSDMessageShadow() , NULL); } else { g_object_set (G_OBJECT (osdMessageFilter), "text", "" , NULL); g_object_set (G_OBJECT (osdMessageFilter), "silent", true , NULL); } if(osd.getShowSessionTimer()) { g_object_set (G_OBJECT (sessionTimerFilter), "silent", false , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "text", osd.getSessionTimer().c_str() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "halignment", osd.getSessionTimerHAlign() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "valignment", osd.getSessionTimerVAlign() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "wrap-mode", osd.getSessionTimerWordWrapMode() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "font-desc", osd.getSessionTimerFont().c_str() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "shaded-background", osd.getSessionTimerShadow() , NULL); } else { g_object_set (G_OBJECT (sessionTimerFilter), "text", "" , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "silent", true , NULL); } this->Sleep(1000); } } } void H264Stream::RemoveStream(int handle) { if(handle != -1) { g_signal_emit_by_name(multifdSink, "remove", handle, G_TYPE_NONE); delete streamHandles[handle]; streamHandles.erase(handle); } if(!streamHandles.size()) StopEncoding(); } bool H264Stream::CheckAndBeginEncoding() { if(!encoding) { GstStateChangeReturn stateRet; stateRet = gst_element_set_state (h264Pipeline, GST_STATE_PLAYING); GstState state; stateRet = gst_element_get_state(h264Pipeline, &state, NULL, GST_SECOND); encoding = true; this->Restart(); return true; } else return true; } bool H264Stream::StopEncoding() { gst_element_set_state (h264Pipeline, GST_STATE_READY); encoding = false; return true; } int H264Stream::AddStreamOutput(string ip, string port) { PWaitAndSignal m(mutex); if(CheckAndBeginEncoding()) { ClientSocket* socket = new ClientSocket(ip, atoi(port.c_str())); int fd = socket->getDescriptor(); if(fd != -1) { //g_signal_emit_by_name(gst_app.multiFDSink, "add", fd, G_TYPE_NONE); streamHandles.insert(std::pair(fd, socket)); return fd; } } return -1; } GstBuffer* H264Stream::GetAudioBuffer() { PWaitAndSignal m(mutex); if (soundSink != NULL) { return gst_app_sink_pull_buffer (GST_APP_SINK (soundSink)); } return NULL; } GstBuffer* H264Stream::GetVideoBuffer() { PWaitAndSignal m(mutex); if (videoSink != NULL) { return gst_app_sink_pull_buffer (GST_APP_SINK (videoSink)); } return NULL; } GstCaps* H264Stream::GetCurrentAudioCaps() { PWaitAndSignal m(mutex); if (soundSink != NULL) { return gst_app_sink_get_caps (GST_APP_SINK (soundSink)); } return NULL; } GstCaps* H264Stream::GetCurrentVideoCaps() { PWaitAndSignal m(mutex); if (videoSink != NULL) { return gst_app_sink_get_caps (GST_APP_SINK (videoSink)); } return NULL; } bool H264Stream::SetSessionAudioCaps(GstCaps* caps) { PWaitAndSignal m(mutex); if (soundSink != NULL) { gst_app_sink_set_caps (GST_APP_SINK (soundSink), caps); gst_caps_unref(caps); return true; } return false; } bool H264Stream::SetSessionVideoCaps(GstCaps* caps) { PWaitAndSignal m(mutex); if (videoSink != NULL) { gst_app_sink_set_caps (GST_APP_SINK (videoSink), caps); gst_caps_unref(caps); return true; } return false; } void H264Stream::SetVolume(gfloat value) { g_object_set(G_OBJECT (volume), "volume", value, NULL); } Here is the class definition #ifndef H264STREAM_H_ #define H264STREAM_H_ #include #include #include #include #include #include #include #include #include #include #include #include "OSDSettings.h" #include "AudioSettings.h" #include "Communications.h" #include "common.h" #include "services.h" class H264Stream : public PThread { public: H264Stream(); virtual ~H264Stream(); /* * The user is responsible for renegotiating caps if they are different from the configuration file. i.e. after receiving H323 caps. * The user is also responsible for unrefing this buffer. */ GstBuffer* GetAudioBuffer(); /* * Current caps in case renegotiation is neccessary (for h323 and SIP caps negotiations) */ GstCaps* GetCurrentAudioCaps(); /* * Sets the caps for the Audio Buffer (for use by H323 and SIP server) */ bool SetSessionAudioCaps(GstCaps* caps); /* * The user is responsible for renegotiating caps if they are different from the configuration file. i.e. after receiving H323 caps. * The user is also responsible for unrefing this buffer. */ GstBuffer* GetVideoBuffer(); /* * Current caps in case renegotiation is neccessary (for h323 and SIP caps negotiations) */ GstCaps* GetCurrentVideoCaps(); /* * Sets the caps for the Video Buffer(for use by H323 and SIP server) */ bool SetSessionVideoCaps(GstCaps* caps); /* * Sends output stream to host at port */ int AddStreamOutput(string host, string port); /* * Remove file descriptor from output stream. */ void RemoveStream(int fd); void SetVolume(gfloat volume); bool CheckAndBeginEncoding(); protected: virtual void Main(); private: Ekiga::ServiceCore core; bool StopEncoding(); std::map streamHandles; unsigned size; unsigned height; unsigned width; unsigned fps; unsigned audioChannels; unsigned audioSampleRate; unsigned bitWidth; bool encoding; PMutex mutex; //pipeline GstElement *h264Pipeline; //Sound elements GstElement *alsaSrc, *volume, *soundTee, *soundSink, *soundAppSinkQueue, *soundQueue, *soundEncoder; //video elements GstElement *v4l2Src, *chanNameFilter, *osdMessageFilter, *sessionTimerFilter, *videoTee, *videoSink, *videoAppSinkQueue, *videoQueue, *videoEncoder; //multiplexed elements GstElement *multiplexer, *multifdSink; }; #endif /* H264STREAM_H_ */ -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Trouble-using-x264enc-with-a-tee-tp3067583p3067583.html Sent from the GStreamer-devel mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: From krkrams27 at gmail.com Fri Dec 3 07:15:40 2010 From: krkrams27 at gmail.com (rams k) Date: Fri, 3 Dec 2010 11:45:40 +0530 Subject: [gst-devel] ogg pipeline Message-ID: Hi, Can anybody let me know the correct pipeline to play the ogg(theora+vorbis) file. Im using the following pipeline but it is givng the error. $gst-launch filesrc location= ! qtdemux name=d d.video_00 ! queue ! theoradec ! xvimagesink d.audio_00 ! queue ! vorbisdec ! audioconvert ! audioresample ! alsasink (gst-launch-0.10:23114): GLib-WARNING **: g_set_prgname() called multiple times Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstQTDemux:d: This file is invalid and cannot be played. Additional debug info: qtdemux.c(420): gst_qtdemux_pull_atom (): /GstPipeline:pipeline0/GstQTDemux:d: atom has bogus size 1332176723 ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... rgds, KR -------------- next part -------------- An HTML attachment was scrubbed... URL: From redtux1 at gmail.com Sat Dec 4 14:36:19 2010 From: redtux1 at gmail.com (Mike Martin) Date: Sat, 4 Dec 2010 13:36:19 +0000 Subject: [gst-devel] hello, I do not want to receive these email for the gstreamer, can you tell me how to cannel it? thanks In-Reply-To: <201012032042113222187@163.com> References: <201012032042113222187@163.com> Message-ID: On 3 December 2010 12:42, alexcuicn wrote: > > > 2010-12-03 > ________________________________ > alexcuicn > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel god what does it take to click on a frigging link thats on every email sent by the list - grrrh From redtux1 at gmail.com Sat Dec 4 14:37:57 2010 From: redtux1 at gmail.com (Mike Martin) Date: Sat, 4 Dec 2010 13:37:57 +0000 Subject: [gst-devel] Which CODEC for mjpg? In-Reply-To: <1291393988296-3071342.post@n4.nabble.com> References: <4CF3F0DF.201@awpi.com> <1291393988296-3071342.post@n4.nabble.com> Message-ID: On 3 December 2010 16:33, wally_bkg wrote: > > I have the opposite problem. ?I can make jpeg video files with: > > gst-launch v4l2src ! video/x-raw-yuv ! queue ! ffmpegcolorspace ! jpegenc ! > avimux ! filesink location=mjpeg.avi > > and the file plays back fine in gstreamer (using Totum on Ubuntu 10.04) but > when I copy it over to a windows system it doesn't play. ?There are several > MJPEG playback codecs on the Windows system. Gspot Codec Information on > Windows shows a valid playback pipeline, but errors in rendering. > > > I tried Theoraenc instead with: > > gst-launch v4l2src ! ?queue ! ffmpegcolorspace ! theoraenc ! oggmux ! > filesink location=theora.ogv > > and again it plays fine in Totum but not on Windows (the Windows system can > play back the ogg video sample files distributed with Ubuntu). > > > what codec does it show as? > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Which-CODEC-for-mjpg-tp3064099p3071342.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From alex.launi at gmail.com Sun Dec 5 20:53:41 2010 From: alex.launi at gmail.com (Alex Launi) Date: Sun, 5 Dec 2010 14:53:41 -0500 Subject: [gst-devel] Need help with dvd playback Message-ID: Hi, I'm trying to get Banshee playing DVDs, but I'm having a whale of a time getting the pipeline to keep rolling and actually play DVDs. As I'm sure many of you are familiar with, Banshee handles most media types, and audio cds very well, yet for some reason the pipeline is stalling out with dvds. I can't figure out what the problem is, was hoping someone could help me debug the issue. We've got a fairly complex pipeline, but more or less it's playbin2 using gconf audio and video sinks. There are other elements for eq, visualizations, and etc. It looks like when you usually start playing a media file, gst goes through a series of internal state changes before the data actually starts flowing down the pipeline. Mine seems to be stalling out after the first state change. Basically what I'm doing is setting the playbin uri to dvd:// and the device property to /dev/sr0 (my disc drive), and then setting the state to PLAYING. Is there more I should need to do? That's all I'm aware of us doing for video files and audio files. Audio CDs work fine, and dvds play with gst-launch -v playbin2 uri=dvd:// Thanks for any help. If there's more info that would be useful please let me know and I'll provide it. -- --Alex Launi -------------- next part -------------- An HTML attachment was scrubbed... URL: From arun.raghavan at collabora.co.uk Mon Dec 6 16:44:12 2010 From: arun.raghavan at collabora.co.uk (Arun Raghavan) Date: Mon, 06 Dec 2010 21:14:12 +0530 Subject: [gst-devel] gst-plugins-good: "ext/pulse" directory is ignored on build In-Reply-To: References: Message-ID: <1291650252.13916.0.camel@snowflake> On Sun, 2010-11-28 at 11:22 +0500, ??????? ??????? wrote: > Hello Everyone! > > I want to build the gst-plugins-good project and it builds fine except > that "ext/pulse" directory is ignored (I made some changes exactly > there). How can I include this folder into the build process? > > I'm a newbie and this question should be easy but I got red eyes > trying to find the solution. I'm guessing the plugin was disabled during the configure stage, likely because of missing dependencies. Do you have the pulseaudio development packages for your distribution installed? -- Arun From olivier.aubert at liris.cnrs.fr Mon Dec 6 16:46:20 2010 From: olivier.aubert at liris.cnrs.fr (Olivier Aubert) Date: Mon, 06 Dec 2010 16:46:20 +0100 Subject: [gst-devel] Python gst without GTK In-Reply-To: <4CE438A4.2080602@tu-bs.de> References: <4CE438A4.2080602@tu-bs.de> Message-ID: <1291650380.3574.58.camel@abbeyroad.dnsalias.org> Hello You should run the mainloop from the gobject module: import gobject # Setup your gstreamer pipeline [...] # Handle keypress gobject.io_add_watch(sys.stdin, gobject.IO_IN, handle_keypress) # Run the mainloop gobject.MainLoop().run() Olivier On Wed, 2010-11-17 at 21:18 +0100, Jens Gottstein wrote: > Hi, > > I am trying to write a command line player using gstreamer in python > > In the example below, the on_message() is never called? > > Can anybody tell me what I did wrong? > > #!/usr/bin/python > > import pygst > pygst.require("0.10") > import gst > > def on_message(self, bus, message): > print "on_message" > > player = gst.element_factory_make("playbin2", "player") > bus = player.get_bus() > bus.add_signal_watch() > bus.connect("message", on_message) > player.set_property("uri", "file:///usr/share/sounds/KDE_Error_3.ogg") > player.set_state(gst.STATE_PLAYING) > > raw_input("anykey") > > > > I am using debian, GStreamer Python bindings 0.10.12 "A Wild Finish" > > Thanks a lot > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From olivier.aubert at liris.cnrs.fr Mon Dec 6 17:49:48 2010 From: olivier.aubert at liris.cnrs.fr (Olivier Aubert) Date: Mon, 06 Dec 2010 17:49:48 +0100 Subject: [gst-devel] 0.11 open for business In-Reply-To: <1291648127.2178.111.camel@meany> References: <1291648127.2178.111.camel@meany> Message-ID: <1291654188.3574.130.camel@abbeyroad.dnsalias.org> Hello I wanted to have a look at plan-0.11.txt and porting-to-0.11.txt, but cannot find them in any of the gstreamer, gst-plugins-base,gst-plugins-good, gst-plugins-bad, gst-plugins-ugly repositories, and googling for them returns nothing. Where are they located? Best regards, Olivier On Mon, 2010-12-06 at 16:08 +0100, Wim Taymans wrote: > Hello Hackers, > > I pushed a branch called '0.11' in the gstreamer, > gst-plugins-base,gst-plugins-good, gst-plugins-bad, gst-plugins-ugly, > gst-ffmpeg, gst-python, gst-plugins-gl and gst-rtsp-server git modules. > Some of the other modules, I hadn't checked out with write access, so > those will be opened some other time... > > In order to get the development started, I propose to start with > removing the deprecated API and fixing the padding of structures again. > Also think about all methods and try to make them future proof. Remember > that all of the current plans are kept in docs/random/plan-0.11.txt. > > When committing changes, always make sure that you keep the various > modules in a compilabe state. > > When API changes, please write into the docs/random/porting-to-0.11.txt > how one is supposed to port to your new API. Also write in the commit > message that the API changed and briefly mention how to port things. The > idea is that at the end of 0.11 we can compile a document with a > complete porting guide. Compiling such a document later is going to be > hard when there are no markers in the commit messages and no trail in > the porting doc. > > So, let's start with cleaning up things and see how it goes. Hop in IRC > and ask around if you want to help out or looking for things to do. > > Happy hacking, > Wim Taymans > > > > > > > > > > I added > > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From wim.taymans at gmail.com Mon Dec 6 18:04:15 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 06 Dec 2010 18:04:15 +0100 Subject: [gst-devel] 0.11 open for business In-Reply-To: <1291654188.3574.130.camel@abbeyroad.dnsalias.org> References: <1291648127.2178.111.camel@meany> <1291654188.3574.130.camel@abbeyroad.dnsalias.org> Message-ID: <1291655055.2178.128.camel@meany> On Mon, 2010-12-06 at 17:49 +0100, Olivier Aubert wrote: > Hello > > I wanted to have a look at plan-0.11.txt and porting-to-0.11.txt, but > cannot find them in any of the gstreamer, > gst-plugins-base,gst-plugins-good, gst-plugins-bad, gst-plugins-ugly > repositories, and googling for them returns nothing. > > Where are they located? http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/random/plan-0.11.txt?h=0.11 And http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/random/porting-to-0.11.txt?h=0.11 Wim > > Best regards, > Olivier > > On Mon, 2010-12-06 at 16:08 +0100, Wim Taymans wrote: > > Hello Hackers, > > > > I pushed a branch called '0.11' in the gstreamer, > > gst-plugins-base,gst-plugins-good, gst-plugins-bad, gst-plugins-ugly, > > gst-ffmpeg, gst-python, gst-plugins-gl and gst-rtsp-server git modules. > > Some of the other modules, I hadn't checked out with write access, so > > those will be opened some other time... > > > > In order to get the development started, I propose to start with > > removing the deprecated API and fixing the padding of structures again. > > Also think about all methods and try to make them future proof. Remember > > that all of the current plans are kept in docs/random/plan-0.11.txt. > > > > When committing changes, always make sure that you keep the various > > modules in a compilabe state. > > > > When API changes, please write into the docs/random/porting-to-0.11.txt > > how one is supposed to port to your new API. Also write in the commit > > message that the API changed and briefly mention how to port things. The > > idea is that at the end of 0.11 we can compile a document with a > > complete porting guide. Compiling such a document later is going to be > > hard when there are no markers in the commit messages and no trail in > > the porting doc. > > > > So, let's start with cleaning up things and see how it goes. Hop in IRC > > and ask around if you want to help out or looking for things to do. > > > > Happy hacking, > > Wim Taymans > > > > > > > > > > > > > > > > > > > > I added > > > > > > > > ------------------------------------------------------------------------------ > > What happens now with your Lotus Notes apps - do you make another costly > > upgrade, or settle for being marooned without product support? Time to move > > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > > use, and manage than apps on traditional platforms. Sign up for the Lotus > > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From thedogfarted at gmail.com Mon Dec 6 18:51:10 2010 From: thedogfarted at gmail.com (=?UTF-8?B?SsSBbmlzIFJ1a8WhxIFucw==?=) Date: Mon, 6 Dec 2010 19:51:10 +0200 Subject: [gst-devel] Gstreamer and Telos Iport In-Reply-To: <1290185926717-3050660.post@n4.nabble.com> References: <1290185926717-3050660.post@n4.nabble.com> Message-ID: Hi, On Fri, Nov 19, 2010 at 6:58 PM, majonezz wrote: > > Hello, please help me with gstreamer, becouse i can't make it working > together with Iport. > I have this device: http://telos-systems.com/zephyr/iport.htm . > It can produce an rtp audio stream - mp3 or aac. I want to decode this > stream with gstreamer. I have some experience with playing it in mplayer, > but mplayer have some long-playing synchro issues. > > I have a dump produced by gst-launch udpsrc port=5000 ?! filesink > location=dump.bin . > > Here it is: http://83.15.87.186/dump.bin > > Can somebody produce propper config for playing it? What is the pipeline you've tried, and what were the results? I can ask the guys working on the iPort (I'm employed at Telos) if they can give any help, but then again they're not familiar with GStreamer. And I'm in turn not familiar with iPort =/ Cheers, J?nis From sumit.git at gmail.com Mon Dec 6 18:59:31 2010 From: sumit.git at gmail.com (sumit kumar) Date: Mon, 6 Dec 2010 23:29:31 +0530 Subject: [gst-devel] GStreamer audio streaming on Windows In-Reply-To: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F16E6@SRV-VS06.TELEVIC.COM> References: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F16E6@SRV-VS06.TELEVIC.COM> Message-ID: Hi, try using directsoundsink...or if you are using autoaudiosink than give sync=false and see On Tue, Nov 30, 2010 at 8:14 PM, Callewaert Sven wrote: > Hi, > > > > I'm experimenting a bit with GStreamer (ossbuild 0.10.7) on Windows, but I > can't seem to make audio streaming between two computers work. All I hear at > the receiver side is a short beep followed by silence. > > > > This is the sender pipeline: > > gst-launch -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink > host=224.0.0.7 auto-multicast=true port=4444 > > > > This is the receiver pipeline: > > gst-launch -v udpsrc multicast-group=224.0.0.7 port=4444 > caps="application/x-rtp, media=(string)audio, channels=(int)1, > clock-rate=(int)44100, encoding-name=(string)L16" ! gstrtpbin ! rtpL16depay > ! audioconvert ! queue ! autoaudiosink > > > > I've already tried different queue settings and codecs. Same thing when I > try to stream an audio file, all I hear is about 1 second of it. What > could be the problem? > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dlafferty at gmail.com Mon Dec 6 18:54:00 2010 From: dlafferty at gmail.com (HaroldJRoth) Date: Mon, 6 Dec 2010 09:54:00 -0800 (PST) Subject: [gst-devel] how to flush a gstreamer pipeline? In-Reply-To: <1285313791321-2553146.post@n4.nabble.com> References: <1285313791321-2553146.post@n4.nabble.com> Message-ID: <1291658040326-3074973.post@n4.nabble.com> How about trying a gst_event_new_step ()? Seems to be able to clear the pipeline. At the very least, you could look at the source to see how it worked. However, it would be handier if a more experienced user could comment. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/how-to-flush-a-gstreamer-pipeline-tp2553146p3074973.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From hoyt6 at llnl.gov Mon Dec 6 19:12:41 2010 From: hoyt6 at llnl.gov (Hoyt, David) Date: Mon, 6 Dec 2010 10:12:41 -0800 Subject: [gst-devel] DirectDrawSink video flickering In-Reply-To: <1290529484831-3055749.post@n4.nabble.com> References: <1290529484831-3055749.post@n4.nabble.com> Message-ID: > Hi, > I'm new in GStreamer, and I'm developing a video player in C#. My player is > all most done, but I have a little problem. For my player I'm using window > form elements and the video is rendering on a panel, but when the video is > playing, it's flickering. That's interesting. I'm using it right now with no flickering. There's some when you resize a window, but I'm not entirely surprised by that. Did you build it yourself? If you're using ossbuild, try using d3dvideosink (direct 3d video sink) instead. It's still being prepared to be submitted upstream to gstreamer, but it's available in ossbuild. That may prove better than the aging directdrawsink. There's also dshowvideosink (direct show video sink). From bisht.sudarshan at gmail.com Mon Dec 6 19:57:18 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 7 Dec 2010 00:27:18 +0530 Subject: [gst-devel] Converting .264 elemntary stream to .Mp4 In-Reply-To: <1290595763181-3057059.post@n4.nabble.com> References: <1290595763181-3057059.post@n4.nabble.com> Message-ID: try , gst-launch filesrc location=<.264 file> ! h264parse ! ! qtmux ! filesink location= On Wed, Nov 24, 2010 at 4:19 PM, ramsk wrote: > > Hi, > Could you pls help in converting .264 multi-view encoded elementary stream > to .Mp4 container format using qtmux? > > Currently using the pipeline like this > gst-launch filesrc location=<.264 file> ! ! qtmux ! filesink > location= > > After getting the .MP4 file,If i decode with ffdec_h264 using the below > pipeline it is not decoding.It is going to pre-roll state and then to > playing state and closing the pipeline.Rendering is not happening > > gst-launch filesrc location=<.MP4 file> ! qtdemux ! fdec_h264! xvimagesink > > > Please help atleast in converting simple .264 file to .MP4 container > format. > > Thanking you in advance. > > rgds, > Rams > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/Converting-264-elemntary-stream-to-Mp4-tp3057059p3057059.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Mon Dec 6 20:10:36 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 7 Dec 2010 00:40:36 +0530 Subject: [gst-devel] configure gst-plugins-base In-Reply-To: <1289987281044-3046377.post@n4.nabble.com> References: <1289987281044-3046377.post@n4.nabble.com> Message-ID: Its not a problem, but a normal behavior. You need to install respective library packages to have those plugins. On Wed, Nov 17, 2010 at 3:18 PM, ice2000feng wrote: > > when I configure gst-plugins-base-0.10.30, there are much problems,: > > checking for a BSD-compatible install... /usr/bin/install -c > checking whether build environment is sane... yes > checking for a thread-safe mkdir -p... /bin/mkdir -p > checking for gawk... gawk > checking whether make sets $(MAKE)... yes > checking nano version... 0 (release) > checking whether to enable maintainer-specific portions of Makefiles... no > checking build system type... i686-pc-linux-gnu > checking host system type... i686-pc-linux-gnu > checking for style of include used by make... GNU > checking for gcc... gcc > checking whether the C compiler works... yes > checking for C compiler default output file name... a.out > checking for suffix of executables... > checking whether we are cross compiling... no > checking for suffix of object files... o > checking whether we are using the GNU C compiler... yes > checking whether gcc accepts -g... yes > checking for gcc option to accept ISO C89... none needed > checking dependency style of gcc... gcc3 > checking for a sed that does not truncate output... /bin/sed > checking for grep that handles long lines and -e... /bin/grep > checking for egrep... /bin/grep -E > checking for fgrep... /bin/grep -F > checking for ld used by gcc... /usr/bin/ld > checking if the linker (/usr/bin/ld) is GNU ld... yes > checking for BSD- or MS-compatible name lister (nm)... /usr/bin/nm -B > checking the name lister (/usr/bin/nm -B) interface... BSD nm > checking whether ln -s works... yes > checking the maximum length of command line arguments... 1572864 > checking whether the shell understands some XSI constructs... yes > checking whether the shell understands "+="... yes > checking for /usr/bin/ld option to reload object files... -r > checking for objdump... objdump > checking how to recognize dependent libraries... pass_all > checking for ar... ar > checking for strip... strip > checking for ranlib... ranlib > checking command to parse /usr/bin/nm -B output from gcc object... ok > checking how to run the C preprocessor... gcc -E > checking for ANSI C header files... yes > checking for sys/types.h... yes > checking for sys/stat.h... yes > checking for stdlib.h... yes > checking for string.h... yes > checking for memory.h... yes > checking for strings.h... yes > checking for inttypes.h... yes > checking for stdint.h... yes > checking for unistd.h... yes > checking for dlfcn.h... yes > checking for objdir... .libs > checking if gcc supports -fno-rtti -fno-exceptions... no > checking for gcc option to produce PIC... -fPIC -DPIC > checking if gcc PIC flag -fPIC -DPIC works... yes > checking if gcc static flag -static works... yes > checking if gcc supports -c -o file.o... yes > checking if gcc supports -c -o file.o... (cached) yes > checking whether the gcc linker (/usr/bin/ld) supports shared libraries... > yes > checking whether -lc should be explicitly linked in... no > checking dynamic linker characteristics... GNU/Linux ld.so > checking how to hardcode library paths into programs... immediate > checking for shl_load... no > checking for shl_load in -ldld... no > checking for dlopen... no > checking for dlopen in -ldl... yes > checking whether a program can dlopen itself... yes > checking whether a statically linked program can dlopen itself... no > checking whether stripping libraries is possible... yes > checking if libtool supports shared libraries... yes > checking whether to build shared libraries... yes > checking whether to build static libraries... yes > checking whether NLS is requested... yes > checking for msgfmt... /usr/bin/msgfmt > checking for gmsgfmt... /usr/bin/msgfmt > checking for xgettext... /usr/bin/xgettext > checking for msgmerge... /usr/bin/msgmerge > checking for ld used by GCC... /usr/bin/ld > checking if the linker (/usr/bin/ld) is GNU ld... yes > checking for shared library run path origin... done > checking for CFPreferencesCopyAppValue... no > checking for CFLocaleCopyCurrent... no > checking for GNU gettext in libc... yes > checking whether to use NLS... yes > checking where the gettext function comes from... libc > checking for pkg-config... /usr/bin/pkg-config > checking pkg-config is at least version 0.9.0... yes > checking for VALGRIND... no > no > configure: Using GStreamer Base Plug-ins source release as package name > configure: Using Unknown package origin as package origin > configure: Using /usr/local/libexec/gst-install-plugins-helper as plugin > install helper > > configure: *** checking feature: building of plug-ins with external deps > *** > configure: building external plug-ins > > configure: *** checking feature: building of experimental plug-ins *** > configure: not building experimental plug-ins > checking whether byte ordering is bigendian... no > checking for special C compiler options needed for large files... no > checking for _FILE_OFFSET_BITS value needed for large files... 64 > checking for gcc... (cached) gcc > checking whether we are using the GNU C compiler... (cached) yes > checking whether gcc accepts -g... (cached) yes > checking for gcc option to accept ISO C89... (cached) none needed > checking dependency style of gcc... (cached) gcc3 > checking for g++... g++ > checking whether we are using the GNU C++ compiler... yes > checking whether g++ accepts -g... yes > checking dependency style of g++... gcc3 > checking whether we are using the GNU C++ compiler... (cached) yes > checking whether g++ accepts -g... (cached) yes > checking dependency style of g++... (cached) gcc3 > checking how to run the C++ preprocessor... g++ -E > checking for ld used by g++... /usr/bin/ld > checking if the linker (/usr/bin/ld) is GNU ld... yes > checking whether the g++ linker (/usr/bin/ld) supports shared libraries... > yes > checking for g++ option to produce PIC... -fPIC -DPIC > checking if g++ PIC flag -fPIC -DPIC works... yes > checking if g++ static flag -static works... yes > checking if g++ supports -c -o file.o... yes > checking if g++ supports -c -o file.o... (cached) yes > checking whether the g++ linker (/usr/bin/ld) supports shared libraries... > yes > checking dynamic linker characteristics... GNU/Linux ld.so > checking how to hardcode library paths into programs... immediate > checking for g++... yes > checking how to run the C++ preprocessor... g++ -E > checking whether gcc and cc understand -c and -o together... yes > checking for valgrind... no > checking for gobject-introspection... no > checking whether to build gtk-doc documentation... no > checking for gtkdoc-check... no > configure: Looking for Python version >= 2.1 > checking for python... /usr/bin/python > checking "/usr/bin/python":... okay > checking local Python configuration... looks good > checking whether to build plugin documentation... no (gtk-doc disabled or > not available) > checking for cos in -lm... yes > checking for ANSI C header files... (cached) yes > checking malloc.h usability... yes > checking malloc.h presence... yes > checking for malloc.h... yes > checking sys/socket.h usability... yes > checking sys/socket.h presence... yes > checking for sys/socket.h... yes > checking winsock2.h usability... no > checking winsock2.h presence... no > checking for winsock2.h... no > checking for hstrerror... yes > checking process.h usability... no > checking process.h presence... no > checking for process.h... no > checking for sys/types.h... (cached) yes > checking sys/wait.h usability... yes > checking sys/wait.h presence... yes > checking for sys/wait.h... yes > checking for sys/stat.h... (cached) yes > checking for stdint types... stdint.h (shortcircuit) > make use of stdint.h in _stdint.h (assuming C99 compatible system) > checking for localtime_r... yes > checking for gmtime_r... yes > checking for cbrt... yes > checking for FIONREAD in sys/ioctl.h... yes > checking for _LARGEFILE_SOURCE value needed for large files... no > checking for lrint... yes > checking for lrintf... yes > checking whether gcc implements __PRETTY_FUNCTION__... yes > checking whether gcc implements __FUNCTION__... yes > checking whether gcc implements __func__... yes > checking for socket... yes > checking for gethostbyname... yes > checking for GLIB... yes > checking for LIBOIL... yes > checking for GST... yes > configure: using GStreamer tools in /usr/local/bin > configure: using GStreamer plug-ins in /usr/local/lib/gstreamer-0.10 > checking for GST_BASE... yes > checking for GST_GDP... yes > checking for GST_CONTROLLER... yes > checking for GST_CHECK... yes > checking for GTK... no > checking for GTK_X11... no > checking for QT... yes > configure: Using /usr/local/lib/gstreamer-0.10 as the plugin install > location > checking to see if compiler understands -Wall... yes > checking to see if compiler understands -Wdeclaration-after-statement... > yes > checking to see if compiler understands -Wvla... yes > checking to see if compiler understands -Wpointer-arith... yes > configure: set ERROR_CFLAGS to -Wall -Wdeclaration-after-statement -Wvla > -Wpointer-arith > checking to see if c++ compiler understands ... yes > checking to see if c++ compiler understands -Wall... yes > configure: set ERROR_CXXFLAGS to -Wall > configure: Using autoaudiosink as default audio sink > configure: Using alsasrc as default audio source > configure: Using autovideosink as default video sink > configure: Using v4l2src as default video source > configure: Using goom as default visualizer > checking which audio resample format to use for integer... auto > checking whether to use iso-codes if they are available... yes > checking whether iso-codes are available on this system... yes > checking whether iso-codes has iso-639 domain... yes > checking for iso-codes prefix... yes > checking ISO_CODES_PREFIX... /usr > checking ISO_639_DOMAIN... iso_639 > > configure: Checking libraries for plugins in sys/ > > checking for X... libraries , headers > > configure: *** checking feature: X libraries and plugins *** > configure: *** for plug-ins: ximagesink *** > checking for gethostbyname... (cached) yes > checking for connect... yes > checking for remove... yes > checking for shmat... yes > checking for IceConnectionNumber in -lICE... no > checking X11/Xlib.h usability... yes > checking X11/Xlib.h presence... yes > checking for X11/Xlib.h... yes > configure: *** These plugins will be built: ximagesink > > configure: *** checking feature: X11 XVideo extensions *** > configure: *** for plug-ins: xvimagesink *** > checking for XvQueryExtension in -lXv_pic... no > checking for XvQueryExtension in -lXv... no > configure: *** These plugins will not be built: xvimagesink > > configure: *** checking feature: X Shared Memory extension *** > checking for XShmAttach in -lXext... no > checking for XShmAttach in -lXextSam... no > > configure: *** checking feature: Video 4 Linux *** > configure: *** for plug-ins: video4linux *** > checking whether VID_TYPE_MPEG_ENCODER is declared... yes > configure: NO XVIDEO FOUND, VIDEO4LINUX WILL BE BUILT > configure: WITHOUT XOVERLAY SUPPORT > configure: *** These plugins will be built: video4linux > checking for GUDEV... no > > configure: Checking libraries for plugins in ext/ > > > configure: *** checking feature: ALSA *** > configure: *** for plug-ins: alsa *** > checking for ALSA... no > checking for ALSA CFLAGS... > checking for ALSA LDFLAGS... -lasound -lm -ldl -lpthread > checking for libasound headers version >= 0.9.1... not present. > configure: *** These plugins will not be built: alsa > > configure: *** checking feature: CDParanoia *** > configure: *** for plug-ins: cdparanoia *** > checking for cdda_open in -lcdda_interface... no > checking for cdda_open in -lcdda_interface... (cached) no > configure: *** These plugins will not be built: cdparanoia > > configure: *** checking feature: GNOME VFS *** > configure: *** for plug-ins: gnomevfs *** > checking for GNOME_VFS... no > configure: *** These plugins will not be built: gnomevfs > > configure: *** checking feature: integer vorbis plug-in *** > configure: *** for plug-ins: ivorbisdec *** > checking for IVORBIS... no > no > configure: No package 'vorbisidec' found > checking for vorbis_block_init in -lvorbisidec... no > configure: *** These plugins will not be built: ivorbisdec > > configure: *** checking feature: GIO library *** > configure: *** for plug-ins: gio *** > checking for GIO... yes > configure: *** These plugins will be built: gio > > configure: *** checking feature: libvisual visualization library *** > configure: *** for plug-ins: libvisual *** > checking for LIBVISUAL... no > no > configure: No package 'libvisual-0.4' found > checking for LIBVISUAL... no > no > configure: No package 'libvisual' found > configure: libvisual pluginsdir: > configure: *** These plugins will not be built: libvisual > > configure: *** checking feature: Xiph Ogg library *** > configure: *** for plug-ins: ogg *** > checking for OGG... no > no > configure: No package 'ogg' found > checking for Ogg... no > *** Could not run Ogg test program, checking why... > *** The test program failed to compile or link. See the file config.log for > the > *** exact error that occured. This usually means Ogg was incorrectly > installed > *** or that you have moved Ogg since it was installed. > configure: *** These plugins will not be built: ogg > > configure: *** checking feature: Pango font rendering *** > configure: *** for plug-ins: pango *** > checking for PANGO... no > no > configure: No package 'pango' found > No package 'pangocairo' found > configure: *** These plugins will not be built: pango > > configure: *** checking feature: Xiph Theora video codec *** > configure: *** for plug-ins: theora *** > checking for THEORA... no > no > configure: No package 'theoradec' found > No package 'theoraenc' found > configure: *** These plugins will not be built: theora > > configure: *** checking feature: Xiph Vorbis audio codec *** > configure: *** for plug-ins: vorbis *** > checking for VORBIS... no > no > configure: No package 'vorbis' found > No package 'vorbisenc' found > checking for Vorbis... no > *** Could not run Vorbis test program, checking why... > *** The test program failed to compile or link. See the file config.log for > the > *** exact error that occured. This usually means Vorbis was incorrectly > installed > *** or that you have moved Vorbis since it was installed. > configure: *** These plugins will not be built: vorbis > checking for FT2... no > checking for freetype-config... no > checking for FreeType - version >= 2.0.9... no > *** The freetype-config script installed by FreeType 2 could not be found. > *** If FreeType 2 was installed in PREFIX, make sure PREFIX/bin is in > *** your path, or set the FT2_CONFIG environment variable to the > *** full path to freetype-config. > checking for sed... /bin/sed > configure: creating ./config.status > config.status: creating Makefile > config.status: creating common/shave > config.status: creating common/shave-libtool > config.status: creating gst-plugins-base.spec > config.status: creating gst/Makefile > config.status: creating gst/adder/Makefile > config.status: creating gst/app/Makefile > config.status: creating gst/audioconvert/Makefile > config.status: creating gst/audiorate/Makefile > config.status: creating gst/audiotestsrc/Makefile > config.status: creating gst/ffmpegcolorspace/Makefile > config.status: creating gst/gdp/Makefile > config.status: creating gst/playback/Makefile > config.status: creating gst/audioresample/Makefile > config.status: creating gst/subparse/Makefile > config.status: creating gst/tcp/Makefile > config.status: creating gst/typefind/Makefile > config.status: creating gst/videotestsrc/Makefile > config.status: creating gst/videorate/Makefile > config.status: creating gst/videoscale/Makefile > config.status: creating gst/volume/Makefile > config.status: creating sys/Makefile > config.status: creating sys/ximage/Makefile > config.status: creating sys/xvimage/Makefile > config.status: creating sys/v4l/Makefile > config.status: creating ext/Makefile > config.status: creating ext/alsa/Makefile > config.status: creating ext/cdparanoia/Makefile > config.status: creating ext/gnomevfs/Makefile > config.status: creating ext/gio/Makefile > config.status: creating ext/libvisual/Makefile > config.status: creating ext/ogg/Makefile > config.status: creating ext/pango/Makefile > config.status: creating ext/theora/Makefile > config.status: creating ext/vorbis/Makefile > config.status: creating gst-libs/Makefile > config.status: creating gst-libs/gst/Makefile > config.status: creating gst-libs/gst/audio/Makefile > config.status: creating gst-libs/gst/app/Makefile > config.status: creating gst-libs/gst/cdda/Makefile > config.status: creating gst-libs/gst/fft/Makefile > config.status: creating gst-libs/gst/floatcast/Makefile > config.status: creating gst-libs/gst/interfaces/Makefile > config.status: creating gst-libs/gst/netbuffer/Makefile > config.status: creating gst-libs/gst/riff/Makefile > config.status: creating gst-libs/gst/rtp/Makefile > config.status: creating gst-libs/gst/rtsp/Makefile > config.status: creating gst-libs/gst/sdp/Makefile > config.status: creating gst-libs/gst/tag/Makefile > config.status: creating gst-libs/gst/pbutils/Makefile > config.status: creating gst-libs/gst/video/Makefile > config.status: creating tools/Makefile > config.status: creating pkgconfig/Makefile > config.status: creating pkgconfig/gstreamer-audio.pc > config.status: creating pkgconfig/gstreamer-audio-uninstalled.pc > config.status: creating pkgconfig/gstreamer-app.pc > config.status: creating pkgconfig/gstreamer-app-uninstalled.pc > config.status: creating pkgconfig/gstreamer-cdda.pc > config.status: creating pkgconfig/gstreamer-cdda-uninstalled.pc > config.status: creating pkgconfig/gstreamer-fft.pc > config.status: creating pkgconfig/gstreamer-fft-uninstalled.pc > config.status: creating pkgconfig/gstreamer-floatcast.pc > config.status: creating pkgconfig/gstreamer-floatcast-uninstalled.pc > config.status: creating pkgconfig/gstreamer-interfaces.pc > config.status: creating pkgconfig/gstreamer-interfaces-uninstalled.pc > config.status: creating pkgconfig/gstreamer-netbuffer.pc > config.status: creating pkgconfig/gstreamer-netbuffer-uninstalled.pc > config.status: creating pkgconfig/gstreamer-pbutils.pc > config.status: creating pkgconfig/gstreamer-pbutils-uninstalled.pc > config.status: creating pkgconfig/gstreamer-riff.pc > config.status: creating pkgconfig/gstreamer-riff-uninstalled.pc > config.status: creating pkgconfig/gstreamer-rtp.pc > config.status: creating pkgconfig/gstreamer-rtp-uninstalled.pc > config.status: creating pkgconfig/gstreamer-rtsp.pc > config.status: creating pkgconfig/gstreamer-rtsp-uninstalled.pc > config.status: creating pkgconfig/gstreamer-sdp.pc > config.status: creating pkgconfig/gstreamer-sdp-uninstalled.pc > config.status: creating pkgconfig/gstreamer-tag.pc > config.status: creating pkgconfig/gstreamer-tag-uninstalled.pc > config.status: creating pkgconfig/gstreamer-video.pc > config.status: creating pkgconfig/gstreamer-video-uninstalled.pc > config.status: creating pkgconfig/gstreamer-plugins-base.pc > config.status: creating pkgconfig/gstreamer-plugins-base-uninstalled.pc > config.status: creating tests/Makefile > config.status: creating tests/check/Makefile > config.status: creating tests/examples/Makefile > config.status: creating tests/examples/app/Makefile > config.status: creating tests/examples/dynamic/Makefile > config.status: creating tests/examples/overlay/Makefile > config.status: creating tests/examples/seek/Makefile > config.status: creating tests/examples/volume/Makefile > config.status: creating tests/examples/snapshot/Makefile > config.status: creating tests/examples/gio/Makefile > config.status: creating tests/examples/v4l/Makefile > config.status: creating tests/icles/Makefile > config.status: creating docs/Makefile > config.status: creating docs/design/Makefile > config.status: creating docs/libs/Makefile > config.status: creating docs/plugins/Makefile > config.status: creating docs/version.entities > config.status: creating po/Makefile.in > config.status: creating common/Makefile > config.status: creating common/m4/Makefile > config.status: creating m4/Makefile > config.status: creating config.h > config.status: config.h is unchanged > config.status: executing depfiles commands > config.status: executing libtool commands > config.status: executing po-directories commands > config.status: creating po/POTFILES > config.status: creating po/Makefile > config.status: executing _stdint.h commands > config.status: creating _stdint.h : _GST_PLUGINS_BASE__STDINT_H > config.status: _stdint.h is unchanged > configure: *** Plug-ins without external dependencies that will be built: > adder > app > audioconvert > audiorate > audioresample > audiotestsrc > ffmpegcolorspace > gdp > playback > subparse > tcp > typefind > videorate > videoscale > videotestsrc > volume > > configure: *** Plug-ins without external dependencies that will NOT be > built: > > configure: *** Plug-ins with dependencies that will be built: > gio > video4linux > ximagesink > > configure: *** Plug-ins with dependencies that will NOT be built: > alsa > cdparanoia > gnomevfs > ivorbisdec > libvisual > ogg > pango > theora > vorbis > xvimagesink > > > how to resolve the problem? > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/configure-gst-plugins-base-tp3046377p3046377.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Mon Dec 6 20:19:03 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 7 Dec 2010 00:49:03 +0530 Subject: [gst-devel] ogg pipeline In-Reply-To: References: Message-ID: Are you able to play this file with other players ? - If no then file has some problem. And one more thing you can try just to narrow down the issue, use following pipeline; $gst-launch filesrc location= ! qtdemux name=d d.video_00 ! queue ! fakesink d.audio_00 ! queue ! fakesink Does this run without any error ? On Fri, Dec 3, 2010 at 11:45 AM, rams k wrote: > Hi, > > Can anybody let me know the correct pipeline to play the ogg(theora+vorbis) > file. > > Im using the following pipeline but it is givng the error. > > $gst-launch filesrc location= ! qtdemux name=d d.video_00 ! > queue ! theoradec ! xvimagesink d.audio_00 ! queue ! vorbisdec ! > audioconvert ! audioresample ! alsasink > > (gst-launch-0.10:23114): GLib-WARNING **: g_set_prgname() called multiple > times > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > ERROR: from element /GstPipeline:pipeline0/GstQTDemux:d: This file is > invalid and cannot be played. > Additional debug info: > qtdemux.c(420): gst_qtdemux_pull_atom (): > /GstPipeline:pipeline0/GstQTDemux:d: > atom has bogus size 1332176723 > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > Freeing pipeline ... > > > rgds, > KR > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Mon Dec 6 20:28:54 2010 From: msmith at xiph.org (Michael Smith) Date: Mon, 6 Dec 2010 11:28:54 -0800 Subject: [gst-devel] ogg pipeline In-Reply-To: References: Message-ID: You have an ogg file. You're using a quicktime demuxer. It should not be a surprise that it says that it's an invalid file. Try this instead: gst-launch-0.10 playbin2 uri=file:///path/to/file.ogg Mike On Thu, Dec 2, 2010 at 10:15 PM, rams k wrote: > Hi, > > Can anybody let me know the correct pipeline to play the ogg(theora+vorbis) > file. > > Im using the following pipeline but it is givng the error. > > $gst-launch filesrc location= ! qtdemux name=d d.video_00 ! queue > !? theoradec ! xvimagesink d.audio_00 ! queue ! vorbisdec? ! audioconvert ! > audioresample ! alsasink > > (gst-launch-0.10:23114): GLib-WARNING **: g_set_prgname() called multiple > times > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > ERROR: from element /GstPipeline:pipeline0/GstQTDemux:d: This file is > invalid and cannot be played. > Additional debug info: > qtdemux.c(420): gst_qtdemux_pull_atom (): > /GstPipeline:pipeline0/GstQTDemux:d: > atom has bogus size 1332176723 > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > Freeing pipeline ... > > > rgds, > KR > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From gibrovacco at gmail.com Mon Dec 6 21:17:35 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Mon, 6 Dec 2010 22:17:35 +0200 Subject: [gst-devel] Trouble using x264enc with a tee In-Reply-To: <1291219846949-3067583.post@n4.nabble.com> References: <1291219846949-3067583.post@n4.nabble.com> Message-ID: Hi, it looks like your email client ate up all of the newlines.. useless to say how hard it is to read its outcomes :) btw, some hints below.. On Wed, Dec 1, 2010 at 6:10 PM, JonathanHenson wrote: > I am writing a multiplexed video/audio streaming thread for a video server I > am working on. I was having problems so I am testing, to make sure the > pipeline works by knocking the data into a file. However, the end goal will > be to use a multifdsink to send the stream to a socket. So, you will notice > that the multifdsink element is actually a filesink for the time being. ok so it looks like you're currently using a filesink, aren't you? > This > pipeline works with other encoders, but when I use x264enc for the video > encoder, the pipeline freezes and no data is put into the file. Also, there > is a tee in both the audio and video portions of the pipeline so that other > threads can grab the raw buffer if they need to access the data. the default question under this condition is.. are you putting a queue element after one of the two ends in the queue? The rationale behind this is that you need to split the threads for the two downstream pipelines. > That way > only one thread is ever accessing the camera. If I remove the tee in the > video pipeline, the pipeline works. +1 for my comment of above > Also I have tested that if I put an > xvimagesink on both sides of the pipeline that both windows get the stream, > so I am pretty sure that the tee is not the problem. actually, it may be if not properly used :) sorry for not commenting on the remaining part, but I'm just over a party for the Finnish Independence Day and it's too hard to read considering the % of alcohol in my blood. Hopefully my comments of above will give you an hint. Please attach again your sources and output either as separate files or in pastebin if you need further help. Regards From gibrovacco at gmail.com Mon Dec 6 21:31:41 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Mon, 6 Dec 2010 22:31:41 +0200 Subject: [gst-devel] Figure out performance with Gstreamer on ARM processor In-Reply-To: <1C29F06818DB1A4C8819EAB61C0855D3064512@039-SN1MPN1-001.039d.mgd.msft.net> References: <1C29F06818DB1A4C8819EAB61C0855D3064512@039-SN1MPN1-001.039d.mgd.msft.net> Message-ID: Hi, this appears to be a re-posted message duplicated from: http://permalink.gmane.org/gmane.comp.video.gstreamer.devel/32134 (or something like that). Btw, some on-the-fly comments below.. On Mon, Nov 22, 2010 at 5:09 PM, Vahe Gerald-B06319 wrote: > Hello List, > > > > I am running FFMPEG codecs on an ARM processor.? In an attempt to figure out > performance, I am trying to encode the videotestsrc with the MPEG4 software > encoder. generally speaking, as ARM architectures come with HW cores for video codecs you should you them when available. Anyway, it's still possible your HW is missing them (I wonder which weird HW it is in such a case though :/). > > > > I initially pulled together this pipeline, which worked: > > gst-launch -e videotestsrc ! video/x-raw-yuv, framerate=15/1, > width=320, height=240 ! ffenc_mpeg4 ! avimux ! filesink location=test.avi > > > > > > However, I notice that the processor is over its limit.? In an attempt to > figure out how under real time it is, I figured I would create a RAW file on > the target with a known length.? (IE---10 seconds) > > > > Then, ENCODE that file with a separate command.?? That way, I could see if > it takes 12 seconds or 20 seconds to encode.? (And I will know how close the > processor is to handling it real time.) > > > > To create the RAW file, I pulled together this pipeline, which appeared to > work fine: > > gst-launch -e videotestsrc ! video/x-raw-yuv framerate=15/1, width=320, > height=240 ! filesink location=testraw.avi > > > > > > Now, I was thinking I could use this command to ?TIME? how long it would > take to encode the file.? However, I get a bunch of errors.? (Below) > > > > time gst-launch -e filesrc location=testraw.avi ! ffenc_mpeg4 ! avimux ! > filesink location=test.avi you're missing (at least) the avidemux element here (between the filesrc and the encoder), then maybe you'll need to put a capsfilter (not sure about how avi deals with raw formats). > > > > Any Idea what I am doing wrong or what these errors mean?? I also tried > AVIMUX and DEMUX in their perspective places, but got similar errors. > well, if you tried to put and avidemux between the source and the encoder and you still got an error, please post it here. Regards > > > Any Help or suggestions would be GREATLY appreciated! > > > > -gerry > > > > > > > > ** (gst-launch-0.10:2013): CRITICAL **: gst_ffmpegenc_chain_video: assertion > `frame_size == GST_BUFFER_SIZE (inbuf)' failed > > Pipeline is PREROLLING ... > > ERROR: from element /GstPipeline:pipeline0/GstFileSrc:filesrc0: Internal > data flow error. > > Additional debug info: > > gstbasesrc.c(2378): gst_base_src_loop (): > /GstPipeline:pipeline0/GstFileSrc:filesrc0: > > streaming task paused, reason error (-5) > > ERROR: pipeline doesn't want to preroll. > > Setting pipeline to NULL ... > > Freeing pipeline ... > > root at freescale ~$ > > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From gibrovacco at gmail.com Mon Dec 6 21:35:45 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Mon, 6 Dec 2010 22:35:45 +0200 Subject: [gst-devel] Gstreamer and Telos Iport In-Reply-To: <1290185926717-3050660.post@n4.nabble.com> References: <1290185926717-3050660.post@n4.nabble.com> Message-ID: Hi, On Fri, Nov 19, 2010 at 6:58 PM, majonezz wrote: > > Hello, please help me with gstreamer, becouse i can't make it working > together with Iport. > I have this device: http://telos-systems.com/zephyr/iport.htm . > It can produce an rtp audio stream - mp3 or aac. I want to decode this > stream with gstreamer. I have some experience with playing it in mplayer, > but mplayer have some long-playing synchro issues. > > I have a dump produced by gst-launch udpsrc port=5000 ?! filesink > location=dump.bin . > > Here it is: http://83.15.87.186/dump.bin > > Can somebody produce propper config for playing it? This is more than a good starting point (see the "Example pipelines" section): http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-gstrtpbin.html If the camera is only streaming video you can mangle away the audio-related pipelines. Regards > Thanks in advance. > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Gstreamer-and-Telos-Iport-tp3050660p3050660.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From gibrovacco at gmail.com Mon Dec 6 21:46:45 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Mon, 6 Dec 2010 22:46:45 +0200 Subject: [gst-devel] Need help with dvd playback In-Reply-To: References: Message-ID: Hi, On Sun, Dec 5, 2010 at 9:53 PM, Alex Launi wrote: > Hi, I'm trying to get Banshee playing DVDs, but I'm having a whale of a time > getting the pipeline to keep rolling and actually play DVDs. As I'm sure > many of you are familiar with, Banshee handles most media types, and audio > cds very well, yet for some reason the pipeline is stalling out with dvds. I > can't figure out what the problem is, was hoping someone could help me debug > the issue. We've got a fairly complex pipeline, but more or less it's > playbin2 using gconf audio and video sinks. There are other elements for eq, > visualizations, and etc. > It looks like when you usually start playing a media file, gst goes through > a series of internal state changes before the data actually starts flowing > down the pipeline. Mine seems to be stalling out after the first state > change. > Basically what I'm doing is setting the playbin uri to dvd:// and the device which element is the "device" property related to? > property to /dev/sr0 (my disc drive), and then setting the state to PLAYING. > Is there more I should need to do? That's all I'm aware of us doing for > video files and audio files. Audio CDs work fine, and dvds play with > gst-launch -v playbin2 uri=dvd:// posting a backtrace of the stalling point may be of help here. You may get it running the process under gdb and then asking for a backtrace when the pipeline gets stuck. Alternatively, you can send a SIGABRT to the stuck application, collect the core and the perform a post-mortem analysis. Regards > > Thanks for any help. If there's more info that would be useful please let me > know and I'll provide it. > > -- > --Alex Launi > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From dingchengus1 at yahoo.com Mon Dec 6 21:57:29 2010 From: dingchengus1 at yahoo.com (ding huah cheng) Date: Mon, 6 Dec 2010 12:57:29 -0800 (PST) Subject: [gst-devel] gst-rtsp-server example problem Message-ID: <1291669049685-3075311.post@n4.nabble.com> Hi, On the latest Ubuntu, I have GStreamer, base and good plugin loaded, gst-rtsp-server is compiled without problem. At first, I use this command "gst-launch playbin uri="file://$PWD/test.mp4" to make sure GStreamer is functioning. then I open 2 terminals: terminal 1: ./test-mp4 test.mp4 terminal 2: gst-launch playbin uri="rtsp://127.0.0.1:8554/test" This setup never work, terminal 1 always come back with lots error message: ------------------------------------------------------------------------ ** Message: client 0x818aec0: received a request ** Message: found media 0x81d1940 for url abspath /test ** Message: enter mainloop ** Message: found stream 0 with payloader 0x8290008 ** Message: found stream 1 with payloader 0x82940a0 ** Message: constructed media 0x8299008 for url /test ** Message: preparing media 0x8299008 (lt-test-mp4:5764): GLib-GObject-CRITICAL **: g_object_set: assertion `G_IS_OBJECT (object)' failed (lt-test-mp4:5764): GStreamer-CRITICAL **: gst_bin_add: assertion `GST_IS_ELEMENT (element)' failed ...... ... ------------------------------------------------------------------------- Am I missing something? Thanks, -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/gst-rtsp-server-example-problem-tp3075311p3075311.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wmiller at sdr.com Mon Dec 6 22:07:45 2010 From: wmiller at sdr.com (Wes Miller) Date: Mon, 6 Dec 2010 13:07:45 -0800 (PST) Subject: [gst-devel] GstRtpBin Internals In-Reply-To: References: <1289851876558-3043711.post@n4.nabble.com> Message-ID: <1291669665943-3075323.post@n4.nabble.com> Marco, I have spent weeks fighting sound quality issues and so missed your wonderful reply. Thanks for all the help. I'd forgotten about cscope and usually discount source code as a source of help since most people seem to think comments are somebody else's responsibility. Merry Christmas, Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/GstRtpBin-Internals-tp3043711p3075323.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From thedogfarted at gmail.com Mon Dec 6 23:14:38 2010 From: thedogfarted at gmail.com (=?UTF-8?B?SsSBbmlzIFJ1a8WhxIFucw==?=) Date: Tue, 7 Dec 2010 00:14:38 +0200 Subject: [gst-devel] Gstreamer and Telos Iport In-Reply-To: References: <1290185926717-3050660.post@n4.nabble.com> Message-ID: On Mon, Dec 6, 2010 at 10:35 PM, Marco Ballesio wrote: > If the camera is only streaming video you can mangle away the > audio-related pipelines. Actually, it is not a camera, it is streaming only audio. I guess the problem the op is having is that he doesn't know the parameters needed for rtpbin. To the op: a Wireshark capture (pcap file) would be more useful than "dump" from udpsrc ! filesink. From S.Callewaert at TELEVIC.com Mon Dec 6 23:25:53 2010 From: S.Callewaert at TELEVIC.com (Callewaert Sven) Date: Mon, 6 Dec 2010 23:25:53 +0100 Subject: [gst-devel] GStreamer audio streaming on Windows Message-ID: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1CB2@SRV-VS06.TELEVIC.COM> Hi, I'm experimenting a bit with GStreamer (ossbuild 0.10.7) on Windows, but I can't seem to make audio streaming between two computers work. All I hear at the receiver side is a short beep followed by silence. This is the sender pipeline: gst-launch -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink host=224.0.0.7 auto-multicast=true port=4444 This is the receiver pipeline: gst-launch -v udpsrc multicast-group=224.0.0.7 port=4444 caps="application/x-rtp, media=(string)audio, channels=(int)1, clock-rate=(int)44100, encoding-name=(string)L16" ! gstrtpbin ! rtpL16depay ! audioconvert ! queue ! autoaudiosink I've already tried different queue settings and codecs. Same thing when I try to stream an audio file, all I hear is about 1 second of it. What could be the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: From abarajase at gmail.com Tue Dec 7 00:34:26 2010 From: abarajase at gmail.com (Hector Barajas) Date: Mon, 6 Dec 2010 17:34:26 -0600 Subject: [gst-devel] mmap support on gstalsasink Message-ID: Hi everyone, Im trying to run and audio device with mmap support, I already did that with pulseaudio but it adds overheads and therefore I still need to reuse the cpu consumption. By looking at the cvs repository it does look like mmap was an option in gstreamer several years ago, and it disappeared in version 1.40 of gstalsasink.c. Does anyone know why this support was removed ? And how difficult it would be to add it again ? Thanks for your feedback. Regards, Hector Barajas -------------- next part -------------- An HTML attachment was scrubbed... URL: From webmail.hce at gmail.com Tue Dec 7 01:23:40 2010 From: webmail.hce at gmail.com (hce) Date: Tue, 7 Dec 2010 11:23:40 +1100 Subject: [gst-devel] Building gstreamer from source and plugin In-Reply-To: References: Message-ID: On Mon, Dec 6, 2010 at 11:36 PM, sudarshan bisht wrote: > I would suggest you to install all the gstreamer packages and related > libraries again in clean way. Thanks Sudarshan. From webmail.hce at gmail.com Tue Dec 7 01:31:10 2010 From: webmail.hce at gmail.com (hce) Date: Tue, 7 Dec 2010 11:31:10 +1100 Subject: [gst-devel] audio control Message-ID: Hi, I've looked at Gstreamer API and example code, I thought there might have APIs to pause audio, get / set audio position so an application can control the audio forwards / backwards, but I could not find the APIs I am looking for. Which audio APIs can be used for above control? Is there any examples for calling APIs to control audio forward / backward? Thank you. Kind regards, jupiter From braunsquared at gmail.com Tue Dec 7 02:02:55 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Mon, 6 Dec 2010 20:02:55 -0500 Subject: [gst-devel] audio control In-Reply-To: References: Message-ID: Jupiter, Play/pause is controlled by setting the pipeline state. Seeks are performed using messages. I would suggest that you take a look at the application developer guide on the gstreamer website. It contains all the information you're looking for. Somebody please correct me if I'm wrong. Hope this helps. Tim Sent from my iPhone On Dec 6, 2010, at 7:31 PM, hce wrote: > Hi, > > I've looked at Gstreamer API and example code, I thought there might > have APIs to pause audio, get / set audio position so an application > can control the audio forwards / backwards, but I could not find the > APIs I am looking for. Which audio APIs can be used for above control? > Is there any examples for calling APIs to control audio forward / > backward? > > Thank you. > > Kind regards, > > jupiter > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ved.kpl at gmail.com Tue Dec 7 04:50:58 2010 From: ved.kpl at gmail.com (ved kpl) Date: Tue, 7 Dec 2010 09:20:58 +0530 Subject: [gst-devel] audio control In-Reply-To: References: Message-ID: *seeks are perfomed using events (not messages) On Tue, Dec 7, 2010 at 6:32 AM, Timothy Braun wrote: > Jupiter, > Play/pause is controlled by setting the pipeline state. Seeks are performed using messages. I would suggest that you take a look at the application developer guide on the gstreamer website. It contains all the information you're looking for. > > Somebody please correct me if I'm wrong. > > Hope this helps. > > Tim > > Sent from my iPhone > > On Dec 6, 2010, at 7:31 PM, hce wrote: > >> Hi, >> >> I've looked at Gstreamer API and example code, I thought there might >> have APIs to pause audio, get / set audio position so an application >> can control the audio forwards / backwards, but I could not find the >> APIs I am looking for. Which audio APIs can be used for above control? >> Is there any examples for calling APIs to control audio forward / >> backward? >> >> Thank you. >> >> Kind regards, >> >> jupiter >> >> ------------------------------------------------------------------------------ >> What happens now with your Lotus Notes apps - do you make another costly >> upgrade, or settle for being marooned without product support? Time to move >> off Lotus Notes and onto the cloud with Force.com, apps are easier to build, >> use, and manage than apps on traditional platforms. Sign up for the Lotus >> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From dingchengus1 at yahoo.com Tue Dec 7 06:22:13 2010 From: dingchengus1 at yahoo.com (ding cheng) Date: Mon, 6 Dec 2010 21:22:13 -0800 (PST) Subject: [gst-devel] gst-rtsp-server example problem In-Reply-To: <1291669049685-3075311.post@n4.nabble.com> Message-ID: <857954.68560.qm@web120014.mail.ne1.yahoo.com> Problem solved. my mp4 file format is wrong. Also, plugins bad and ugly are also needed. --- On Mon, 12/6/10, ding huah cheng wrote: From: ding huah cheng Subject: [gst-devel] gst-rtsp-server example problem To: gstreamer-devel at lists.sourceforge.net Date: Monday, December 6, 2010, 12:57 PM Hi, On the latest Ubuntu, I have GStreamer, base and good plugin loaded, gst-rtsp-server is compiled without problem. At first, I use this command "gst-launch playbin uri="file://$PWD/test.mp4" to make sure GStreamer is functioning. then I open 2 terminals: terminal 1: ./test-mp4 test.mp4 terminal 2: gst-launch playbin uri="rtsp://127.0.0.1:8554/test" This setup never work, terminal 1 always come back with lots error message: ------------------------------------------------------------------------ ** Message: client 0x818aec0: received a request ** Message: found media 0x81d1940 for url abspath /test ** Message: enter mainloop ** Message: found stream 0 with payloader 0x8290008 ** Message: found stream 1 with payloader 0x82940a0 ** Message: constructed media 0x8299008 for url /test ** Message: preparing media 0x8299008 (lt-test-mp4:5764): GLib-GObject-CRITICAL **: g_object_set: assertion `G_IS_OBJECT (object)' failed (lt-test-mp4:5764): GStreamer-CRITICAL **: gst_bin_add: assertion `GST_IS_ELEMENT (element)' failed ...... ... ------------------------------------------------------------------------- Am I missing something? Thanks, -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/gst-rtsp-server-example-problem-tp3075311p3075311.html Sent from the GStreamer-devel mailing list archive at Nabble.com. ------------------------------------------------------------------------------ What happens now with your Lotus Notes apps - do you make another costly upgrade, or settle for being marooned without product support? Time to move off Lotus Notes and onto the cloud with Force.com, apps are easier to build, use, and manage than apps on traditional platforms. Sign up for the Lotus Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Tue Dec 7 08:08:13 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 7 Dec 2010 12:38:13 +0530 Subject: [gst-devel] audio control In-Reply-To: References: Message-ID: these links may be helpful for you; http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/chapter-queryevents.html#section-querying http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-eventsseek.html On Tue, Dec 7, 2010 at 9:20 AM, ved kpl wrote: > *seeks are perfomed using events (not messages) > > On Tue, Dec 7, 2010 at 6:32 AM, Timothy Braun > wrote: > > Jupiter, > > Play/pause is controlled by setting the pipeline state. Seeks are > performed using messages. I would suggest that you take a look at the > application developer guide on the gstreamer website. It contains all the > information you're looking for. > > > > Somebody please correct me if I'm wrong. > > > > Hope this helps. > > > > Tim > > > > Sent from my iPhone > > > > On Dec 6, 2010, at 7:31 PM, hce wrote: > > > >> Hi, > >> > >> I've looked at Gstreamer API and example code, I thought there might > >> have APIs to pause audio, get / set audio position so an application > >> can control the audio forwards / backwards, but I could not find the > >> APIs I am looking for. Which audio APIs can be used for above control? > >> Is there any examples for calling APIs to control audio forward / > >> backward? > >> > >> Thank you. > >> > >> Kind regards, > >> > >> jupiter > >> > >> > ------------------------------------------------------------------------------ > >> What happens now with your Lotus Notes apps - do you make another costly > >> upgrade, or settle for being marooned without product support? Time to > move > >> off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > >> use, and manage than apps on traditional platforms. Sign up for the > Lotus > >> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > > What happens now with your Lotus Notes apps - do you make another costly > > upgrade, or settle for being marooned without product support? Time to > move > > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > > use, and manage than apps on traditional platforms. Sign up for the Lotus > > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From sumit.git at gmail.com Tue Dec 7 08:17:33 2010 From: sumit.git at gmail.com (sumit kumar) Date: Tue, 7 Dec 2010 12:47:33 +0530 Subject: [gst-devel] GStreamer audio streaming on Windows In-Reply-To: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1CB2@SRV-VS06.TELEVIC.COM> References: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1CB2@SRV-VS06.TELEVIC.COM> Message-ID: Hi, Try with directsoundsink.. else try setting sync=false for autoaudiosink -Sumit On Tue, Dec 7, 2010 at 3:55 AM, Callewaert Sven wrote: > Hi, > > > > I'm experimenting a bit with GStreamer (ossbuild 0.10.7) on Windows, but I > can't seem to make audio streaming between two computers work. All I hear at > the receiver side is a short beep followed by silence. > > > > This is the sender pipeline: > > gst-launch -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink > host=224.0.0.7 auto-multicast=true port=4444 > > > > This is the receiver pipeline: > > gst-launch -v udpsrc multicast-group=224.0.0.7 port=4444 > caps="application/x-rtp, media=(string)audio, channels=(int)1, > clock-rate=(int)44100, encoding-name=(string)L16" ! gstrtpbin ! rtpL16depay > ! audioconvert ! queue ! autoaudiosink > > > > I've already tried different queue settings and codecs. Same thing when I > try to stream an audio file, all I hear is about 1 second of it. What > could be the problem? > > > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From S.Callewaert at TELEVIC.com Tue Dec 7 09:53:10 2010 From: S.Callewaert at TELEVIC.com (Callewaert Sven) Date: Tue, 7 Dec 2010 09:53:10 +0100 Subject: [gst-devel] GStreamer audio streaming on Windows In-Reply-To: References: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1CB2@SRV-VS06.TELEVIC.COM> Message-ID: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1CD1@SRV-VS06.TELEVIC.COM> Replacing the autoaudiosink element with directsoundsink did the trick. Now I hear a continuous beep but there is a lot of distortion on it. When I stream the microphone by replacing audiotestsrc with directsoundsrc on the sender side, I also hear a lot of distortion. The distortion however only occurs when the microphone captures moderate to loud noises. There is no distortion with quiet noises. Could it be a signed/unsigned issue? Can I force the sender the use signed=false in some way? From: sumit kumar [mailto:sumit.git at gmail.com] Sent: dinsdag 7 december 2010 8:18 To: Discussion of the development of GStreamer Subject: Re: [gst-devel] GStreamer audio streaming on Windows Hi, Try with directsoundsink.. else try setting sync=false for autoaudiosink -Sumit On Tue, Dec 7, 2010 at 3:55 AM, Callewaert Sven > wrote: Hi, I'm experimenting a bit with GStreamer (ossbuild 0.10.7) on Windows, but I can't seem to make audio streaming between two computers work. All I hear at the receiver side is a short beep followed by silence. This is the sender pipeline: gst-launch -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink host=224.0.0.7 auto-multicast=true port=4444 This is the receiver pipeline: gst-launch -v udpsrc multicast-group=224.0.0.7 port=4444 caps="application/x-rtp, media=(string)audio, channels=(int)1, clock-rate=(int)44100, encoding-name=(string)L16" ! gstrtpbin ! rtpL16depay ! audioconvert ! queue ! autoaudiosink I've already tried different queue settings and codecs. Same thing when I try to stream an audio file, all I hear is about 1 second of it. What could be the problem? ------------------------------------------------------------------------------ What happens now with your Lotus Notes apps - do you make another costly upgrade, or settle for being marooned without product support? Time to move off Lotus Notes and onto the cloud with Force.com, apps are easier to build, use, and manage than apps on traditional platforms. Sign up for the Lotus Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From mengdong.lin at intel.com Tue Dec 7 09:57:06 2010 From: mengdong.lin at intel.com (Lin, Mengdong) Date: Tue, 7 Dec 2010 16:57:06 +0800 Subject: [gst-devel] How can a parser plug-in support "gsttagreader" interface? Message-ID: <749B9D3DBF0F054390025D9EAFF47F223149FDD6@shsmsx501.ccr.corp.intel.com> Is there an example how a base or good plug-in support this interface? Many thanks! Amanda -------------- next part -------------- An HTML attachment was scrubbed... URL: From vaish05_s at yahoo.com Tue Dec 7 11:31:39 2010 From: vaish05_s at yahoo.com (vaishnavi) Date: Tue, 7 Dec 2010 02:31:39 -0800 (PST) Subject: [gst-devel] Using BaseTransform class Message-ID: <1291717899514-3076117.post@n4.nabble.com> Hi, I'm using BaseTransform as parent class. I'm using transform_ip to communicate with the driver. The incoming buffer will be passed to driver and read back from driver the processed data into the same buffer. The issue is that it is the buffer contents coming from driver is not pushed out. The contents to be wriiten and read out of driver is verified using print statements..driver works fine..but somehow the data read from driver is not getting pushed. I have include the code i have used for transform_ip and prepare_output_buffer Can you tel me why the contents are not getting changed. static GstFlowReturn gst_dummy_prepare_output_buffer (GstBaseTransform * trans, GstBuffer * in_buf, gint out_size, GstCaps * out_caps, GstBuffer ** out_buf) { GstDummy *dummy = GST_DUMMY (trans); *out_buf = gst_buffer_ref (in_buf); *out_buf=gst_buffer_copy(dummy->cpybuf); return GST_FLOW_OK; } static GstFlowReturn gst_dummy_transform_ip (GstBaseTransform * trans, GstBuffer * buf) { GstFlowReturn ret = GST_FLOW_OK; guint size =1024; gint n,x; GstDummy *dummy = GST_DUMMY (trans); g_print("pf_handle_ip%d\n",dummy->pf_handle); g_print("the first few bytes of buffer%d\t%d\t%d\t",buf->data[0],buf->data[1],buf->data[2]); GstClockTime runtimestamp = G_GINT64_CONSTANT (0); g_print("size before allocation%d\n",GST_BUFFER_SIZE(buf)); dummy->cpybuf=gst_buffer_copy(buf); g_print("the first few bytes of cpy buffer%d\t%d\t%d\t",dummy->cpybuf->data[0],dummy->cpybuf->data[1],dummy->cpybuf->data[2]); n=write(dummy->pf_handle,GST_BUFFER_DATA(dummy->cpybuf),size); if(n<0) { g_print("could not write; the contents of cpybuf %d and n is %d\n,GST_BUFFER_DATA(cpybuf),n"); } x=read(dummy->pf_handle,GST_BUFFER_DATA(dummy->cpybuf),n); g_print("the first few bytes of buffer after read%d\t%d\t%d\t",dummy->cpybuf->data[0],dummy->cpybuf->data[1],dummy->cpybuf->data[2]); if(x<0) { g_print("could not read\n"); } if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) gst_object_sync_values (G_OBJECT (dummy), GST_BUFFER_TIMESTAMP (buf)); if (dummy->silent == FALSE) g_print ("I'm plugged, therefore I'm in Dummy TC.\n"); buf=gst_buffer_copy(dummy->cpybuf); g_print("the first few bytes of buffer befor return%d\t%d\t%d\t",buf->data[0],buf->data[1],buf->data[2]); return ret; } -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Using-BaseTransform-class-tp3076117p3076117.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From vaish05_s at yahoo.com Tue Dec 7 11:44:24 2010 From: vaish05_s at yahoo.com (vaishnavi) Date: Tue, 7 Dec 2010 02:44:24 -0800 (PST) Subject: [gst-devel] Transform function not working Message-ID: <1291718664740-3076135.post@n4.nabble.com> Hi, I'm using basetransform as parent class for my plugin. I have implemented transform function,set_caps,get_unit_size and prepare_output_buffer of parent class. Here im taking the in_buf contents passing to driver and reading the processed data to out_buf. The driver works fine... But i get error and warning saying the caps are not negotiated. This happens when i launch the application. The caps are set to any. If i only use transform function the i get an error,after implementing set_caps and get_unit_size i receive warning. The warning message is WARN basetransform gstbasetransform.c:1969:gst_base_transform_handle_buffer: error: not negotiated 0:00:00.076602538 2989 0x9b73038 WARN basetransform gstbasetransform.c:1969:gst_base_transform_handle_buffer: error: not negotiated WARN basesrc gstbasesrc.c:2507:gst_base_src_loop: error: Internal data flow error. 0:00:00.076898043 2989 0x9b73038 WARN basesrc gstbasesrc.c:2507:gst_base_src_loop: error: streaming task paused, reason not-negotiated (-4) static gboolean gst_dummy_get_unit_size (GstBaseTransform * btrans, GstCaps * caps, guint * size) { *size=1; return TRUE; } static GstFlowReturn gst_dummy_transform (GstBaseTransform * trans, GstBuffer * inbuf, GstBuffer * outbuf) { GstFlowReturn ret = GST_FLOW_OK; return ret; } static gboolean gst_dummy_set_caps(GstBaseTransform *trans, GstCaps *incaps, GstCaps *outcaps) { outcaps=incaps; return TRUE; } static GstFlowReturn gst_dummy_prepare_output_buffer (GstBaseTransform * trans, GstBuffer * in_buf, gint out_size, GstCaps * out_caps, GstBuffer ** out_buf) { GstDummy *dummy = GST_DUMMY (trans); guint size; gint n,x; out_caps=GST_CAPS_ANY; GstClockTime runtimestamp = G_GINT64_CONSTANT (0); g_print("The input buffer contents are%d\t%d\t%d\t",in_buf->data[0],in_buf->data[1],in_buf->data[2]); size=GST_BUFFER_SIZE(in_buf); g_print("size of inbuf is %d\n",size); gst_buffer_make_writable(dummy->cpybuf); n=write(dummy->pf_handle,GST_BUFFER_DATA(in_buf),size); if(n<0) { g_print("could not write\n"); } x=read(dummy->pf_handle,GST_BUFFER_DATA(dummy->cpybuf),n); if(x<0) { g_print("could not read\n"); } else { g_print("the size of output buffer is %d\n",x); } if (dummy->silent == FALSE) g_print ("I'm plugged, therefore I'm in Dummy TC.\n"); *out_buf=gst_buffer_copy(dummy->cpybuf); gst_dummy_set_caps(trans,GST_PAD_CAPS(dummy->srcpad),GST_PAD_CAPS(dummy->sinkpad)); return GST_FLOW_OK; } -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Transform-function-not-working-tp3076135p3076135.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wim.taymans at gmail.com Tue Dec 7 12:38:47 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 07 Dec 2010 12:38:47 +0100 Subject: [gst-devel] 0.11 development Message-ID: <1291721927.2178.155.camel@meany> Hello, This mail is just to clarify some rules for the new 0.11 branch: 1- Development of plugins and new features continues on the master branch. Everything that can possibly go into 0.10 should go to master first. This can include cleanups that are needed in the 0.11 branch such as removal of deprecated methods. 2- We periodically merge master into 0.11. I would like to see frequent merges. When you commit something to 0.11, try to merge the changes on master when possible. 3- At the end of the development cycle we want to switch the 0.11 and master branch and prepare for release. We don't want to do a massive merge at that point, so we need the small incremental merges. 4- changes to 0.11 that could end up in 0.10 as well can be cherry picked when needed. Note that this case should be very usual because it means that you failed to follow rule 1. The end result is that all the new features get developed and tested in 0.10 and eventually also end up in 0.11. Merges will probably get more difficult when the code bases are getting too different, when we get to that stage we will re-evaluate how to maintain the two branches. It could be that we only do a weekly merge then or even less frequent, we'll have to see. Wim From bilboed at gmail.com Tue Dec 7 13:25:39 2010 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 07 Dec 2010 13:25:39 +0100 Subject: [gst-devel] 0.11 development In-Reply-To: <1291721927.2178.155.camel@meany> References: <1291721927.2178.155.camel@meany> Message-ID: <1291724739.1589.4.camel@localhost> On Tue, 2010-12-07 at 12:38 +0100, Wim Taymans wrote: > Hello, A word of caution regarding "git push" for those with commit access. If you're tracking both master and 0.11, running "git push" without any arguments implies that you want to push *all* locally tracked branches to the *origin* repo. Better avoid regretting it by always specifying at least the branch name you wish to push. => git push > > Wim > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From t.i.m at zen.co.uk Tue Dec 7 13:43:59 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 07 Dec 2010 12:43:59 +0000 Subject: [gst-devel] 0.11 development In-Reply-To: <1291724739.1589.4.camel@localhost> References: <1291721927.2178.155.camel@meany> <1291724739.1589.4.camel@localhost> Message-ID: <1291725839.15436.13.camel@zingle> On Tue, 2010-12-07 at 13:25 +0100, Edward Hervey wrote: > A word of caution regarding "git push" for those with commit access. > > If you're tracking both master and 0.11, running "git push" without > any arguments implies that you want to push *all* locally tracked > branches to the *origin* repo. > > Better avoid regretting it by always specifying at least the branch > name you wish to push. > > => git push Two more things worth mentioning: 1. there's git push --dry-run 2. it's possible to change the default behaviour of git push to something more conservative: git config --global push.default tracking or git config --global push.default current (see man git-config) Cheers -Tim From wl2776 at gmail.com Tue Dec 7 13:48:11 2010 From: wl2776 at gmail.com (wl2776) Date: Tue, 7 Dec 2010 04:48:11 -0800 (PST) Subject: [gst-devel] DirectDrawSink video flickering In-Reply-To: <1290529484831-3055749.post@n4.nabble.com> References: <1290529484831-3055749.post@n4.nabble.com> Message-ID: <1291726091132-3076296.post@n4.nabble.com> This could be caused by the oddities of your video subsystem. We have a similar case. Another application extensively using DirectDraw, causes flickering on one computer and doesn't cause on several others. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/DirectDrawSink-video-flickering-tp3055749p3076296.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From sumit.git at gmail.com Tue Dec 7 14:41:29 2010 From: sumit.git at gmail.com (sumit kumar) Date: Tue, 7 Dec 2010 19:11:29 +0530 Subject: [gst-devel] GStreamer audio streaming on Windows In-Reply-To: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1CD1@SRV-VS06.TELEVIC.COM> References: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1CB2@SRV-VS06.TELEVIC.COM> <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1CD1@SRV-VS06.TELEVIC.COM> Message-ID: try setting sync=false for directsoundsink properties. On Tue, Dec 7, 2010 at 2:23 PM, Callewaert Sven wrote: > Replacing the autoaudiosink element with directsoundsink did the trick. Now > I hear a continuous beep but there is a lot of distortion on it. When I > stream the microphone by replacing audiotestsrc with directsoundsrc on the > sender side, I also hear a lot of distortion. The distortion however only > occurs when the microphone captures moderate to loud noises. There is no > distortion with quiet noises. Could it be a signed/unsigned issue? Can I > force the sender the use signed=false in some way? > > > > *From:* sumit kumar [mailto:sumit.git at gmail.com] > *Sent:* dinsdag 7 december 2010 8:18 > *To:* Discussion of the development of GStreamer > *Subject:* Re: [gst-devel] GStreamer audio streaming on Windows > > > > Hi, > > > > Try with directsoundsink.. else try setting sync=false for autoaudiosink > > > > -Sumit > > On Tue, Dec 7, 2010 at 3:55 AM, Callewaert Sven > wrote: > > Hi, > > > > I'm experimenting a bit with GStreamer (ossbuild 0.10.7) on Windows, but I > can't seem to make audio streaming between two computers work. All I hear at > the receiver side is a short beep followed by silence. > > > > This is the sender pipeline: > > gst-launch -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink > host=224.0.0.7 auto-multicast=true port=4444 > > > > This is the receiver pipeline: > > gst-launch -v udpsrc multicast-group=224.0.0.7 port=4444 > caps="application/x-rtp, media=(string)audio, channels=(int)1, > clock-rate=(int)44100, encoding-name=(string)L16" ! gstrtpbin ! rtpL16depay > ! audioconvert ! queue ! autoaudiosink > > > > I've already tried different queue settings and codecs. Same thing when I > try to stream an audio file, all I hear is about 1 second of it. What > could be the problem? > > > > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jonathan.henson at innovisit.com Mon Dec 6 21:40:26 2010 From: jonathan.henson at innovisit.com (JonathanHenson) Date: Mon, 6 Dec 2010 12:40:26 -0800 (PST) Subject: [gst-devel] Trouble using x264enc with a tee In-Reply-To: References: <1291219846949-3067583.post@n4.nabble.com> Message-ID: <1291668026288-3075279.post@n4.nabble.com> Here is the code. I actually have started using ffmux_asf untill I can get the x264enc to work. However, this is the same exact pipeline except I am using the asf encoders and muxers instead of x264 and avi. However, now I have the problem of when I use filesink with a location, the file is perfect (though with no seeking--don't know what that is about), however, when I use this code, the client on the other end of the socket can't play the file. I added a standard file to the multifdsink as a test but it isn't receiving any output. Thanks for your reply, I hope your head gets better. /* * H264Stream.cpp * * Created on: Nov 12, 2010 * Author: jonathan */ #include "H264Stream.h" #include "VideoInput.h" int fileD; H264Stream::H264Stream() : PThread (1000, NoAutoDeleteThread, HighestPriority, "H264Stream"), encoding(false) { //temporary setting of variables width = 352; height = 288; fps = 25; audioChannels = 2; audioSampleRate = 8000; bitWidth = 16; GError* error = NULL; gchar* command = NULL; command = g_strdup_printf ("v4l2src ! video/x-raw-yuv, format=(fourcc)I420, width=%d, height=%d, framerate=(fraction)%d/1 !" " videobalance name=VideoBalance ! textoverlay name=chanNameFilter ! textoverlay name=osdMessageFilter ! textoverlay name=sessionTimerOverlay ! " "tee name=t ! queue ! appsink name=videoSink t. ! queue ! ffenc_wmv2 name=videoEncoder me-method=5 ! amux. alsasrc ! " "audio/x-raw-int, depth=%d, width=%d, channels=2, endianness=1234, rate=%d, signed=true ! volume name=volumeFilter ! " "tee name=souTee ! queue ! appsink name=soundSink souTee. ! queue ! ffenc_wmav2 ! amux. ffmux_asf name=amux ! multifdsink name=multifdsink", width, height, fps, bitWidth, bitWidth, audioSampleRate); g_print ("Pipeline: %s\n", command); h264Pipeline = gst_parse_launch (command, &error); if(error != NULL) std::cout << error->message << "\n"; chanNameFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), "chanNameFilter"); osdMessageFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), "osdMessageFilter"); sessionTimerFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), "sessionTimerOverlay"); videoBalance = gst_bin_get_by_name (GST_BIN (h264Pipeline), "VideoBalance"); videoEncoder = gst_bin_get_by_name (GST_BIN (h264Pipeline), "videoEncoder"); volume = gst_bin_get_by_name (GST_BIN (h264Pipeline), "volumeFilter"); multifdSink = gst_bin_get_by_name (GST_BIN (h264Pipeline), "multifdsink"); soundSink = gst_bin_get_by_name (GST_BIN (h264Pipeline), "soundSink"); } H264Stream::~H264Stream() { for(std::map::iterator pair = streamHandles.begin(); pair != streamHandles.end(); pair++) { g_signal_emit_by_name(multifdSink, "remove", pair->first, NULL); delete pair->second; } streamHandles.clear(); gst_element_set_state (h264Pipeline, GST_STATE_NULL); gst_object_unref (GST_OBJECT (h264Pipeline)); } void H264Stream::Main() { while(true) { PWaitAndSignal m(mutex); if(encoding) { OSDSettings osd; if(osd.getShowChanName()) { g_object_set (G_OBJECT (chanNameFilter), "silent", false , NULL); g_object_set (G_OBJECT (chanNameFilter), "text", osd.getChanName().c_str() , NULL); g_object_set (G_OBJECT (chanNameFilter), "halignment", osd.getChanNameHAlign() , NULL); g_object_set (G_OBJECT (chanNameFilter), "valignment", osd.getChanNameVAlign() , NULL); g_object_set (G_OBJECT (chanNameFilter), "wrap-mode", osd.getChanNameWordWrapMode() , NULL); g_object_set (G_OBJECT (chanNameFilter), "font-desc", osd.getChanNameFont().c_str() , NULL); g_object_set (G_OBJECT (chanNameFilter), "shaded-background", osd.getChanNameShadow() , NULL); } else { g_object_set (G_OBJECT (chanNameFilter), "text", "" , NULL); g_object_set (G_OBJECT (chanNameFilter), "silent", true , NULL); } if(osd.getShowOSDMessage()) { g_object_set (G_OBJECT (osdMessageFilter), "silent", false , NULL); g_object_set (G_OBJECT (osdMessageFilter), "text", osd.getOSDMessage().c_str() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "halignment", osd.getOSDMessageHAlign() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "valignment", osd.getOSDMessageVAlign() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "wrap-mode", osd.getOSDMessageWordWrapMode() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "font-desc", osd.getOSDMessageFont().c_str() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "shaded-background", osd.getOSDMessageShadow() , NULL); } else { g_object_set (G_OBJECT (osdMessageFilter), "text", "" , NULL); g_object_set (G_OBJECT (osdMessageFilter), "silent", true , NULL); } if(osd.getShowSessionTimer()) { g_object_set (G_OBJECT (sessionTimerFilter), "silent", false , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "text", osd.getSessionTimer().c_str() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "halignment", osd.getSessionTimerHAlign() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "valignment", osd.getSessionTimerVAlign() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "wrap-mode", osd.getSessionTimerWordWrapMode() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "font-desc", osd.getSessionTimerFont().c_str() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "shaded-background", osd.getSessionTimerShadow() , NULL); } else { g_object_set (G_OBJECT (sessionTimerFilter), "text", "" , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "silent", true , NULL); } this->Sleep(1000); } } } void H264Stream::RemoveStream(int handle) { if(handle != -1) { g_signal_emit_by_name(multifdSink, "remove", handle, G_TYPE_NONE); delete streamHandles[handle]; streamHandles.erase(handle); g_signal_emit_by_name(multifdSink, "remove", fileD, G_TYPE_NONE); close(fileD); } if(!streamHandles.size()) StopEncoding(); } bool H264Stream::CheckAndBeginEncoding() { if(!encoding) { GstStateChangeReturn stateRet; stateRet = gst_element_set_state (h264Pipeline, GST_STATE_PLAYING); GstState state; stateRet = gst_element_get_state(h264Pipeline, &state, NULL, GST_SECOND); encoding = true; this->Restart(); return true; } else return true; } bool H264Stream::StopEncoding() { gst_element_set_state (h264Pipeline, GST_STATE_READY); encoding = false; return true; } int H264Stream::AddStreamOutput(string ip, string port) { PWaitAndSignal m(mutex); if(CheckAndBeginEncoding()) { fileD = open("/home/jonathan/anotherTest.wmv", O_RDWR | O_APPEND | O_CREAT, 0666); if(fileD != -1) { g_signal_emit_by_name(multifdSink, "add", fileD, G_TYPE_NONE); //streamHandles.insert(std::pair(fd, socket)); } ClientSocket* socket = new ClientSocket(ip, atoi(port.c_str())); int fd = socket->getDescriptor(); if(fd != -1) { g_signal_emit_by_name(multifdSink, "add", fd, G_TYPE_NONE); streamHandles.insert(std::pair(fd, socket)); return fd; } } return -1; } GstBuffer* H264Stream::GetAudioBuffer() { PWaitAndSignal m(mutex); if (soundSink != NULL) { return gst_app_sink_pull_buffer (GST_APP_SINK (soundSink)); } return NULL; } GstBuffer* H264Stream::GetVideoBuffer() { PWaitAndSignal m(mutex); if (videoSink != NULL) { return gst_app_sink_pull_buffer (GST_APP_SINK (videoSink)); } return NULL; } GstCaps* H264Stream::GetCurrentAudioCaps() { PWaitAndSignal m(mutex); if (soundSink != NULL) { return gst_app_sink_get_caps (GST_APP_SINK (soundSink)); } return NULL; } GstCaps* H264Stream::GetCurrentVideoCaps() { PWaitAndSignal m(mutex); if (videoSink != NULL) { return gst_app_sink_get_caps (GST_APP_SINK (videoSink)); } return NULL; } bool H264Stream::SetSessionAudioCaps(GstCaps* caps) { PWaitAndSignal m(mutex); if (soundSink != NULL) { gst_app_sink_set_caps (GST_APP_SINK (soundSink), caps); gst_caps_unref(caps); return true; } return false; } bool H264Stream::SetSessionVideoCaps(GstCaps* caps) { PWaitAndSignal m(mutex); if (videoSink != NULL) { gst_app_sink_set_caps (GST_APP_SINK (videoSink), caps); gst_caps_unref(caps); return true; } return false; } void H264Stream::SetVolume(gfloat value) { g_object_set(G_OBJECT (volume), "volume", value, NULL); } bool H264Stream::SetSaturation(double color) { g_object_set(G_OBJECT (videoBalance), "saturation", color, NULL); return true; } bool H264Stream::SetBrightness(double brightness) { g_object_set(G_OBJECT (videoBalance), "brightness", brightness, NULL); return true; } bool H264Stream::SetHue(double hue) { g_object_set(G_OBJECT (videoBalance), "hue", hue, NULL); return true; } bool H264Stream::SetContrast(double contrast) { g_object_set(G_OBJECT (videoBalance), "contrast", contrast, NULL); return true; } -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Trouble-using-x264enc-with-a-tee-tp3067583p3075279.html Sent from the GStreamer-devel mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jonathan.henson at innovisit.com Mon Dec 6 21:41:23 2010 From: jonathan.henson at innovisit.com (JonathanHenson) Date: Mon, 6 Dec 2010 12:41:23 -0800 (PST) Subject: [gst-devel] Trouble using x264enc with a tee In-Reply-To: <1291668026288-3075279.post@n4.nabble.com> References: <1291219846949-3067583.post@n4.nabble.com> <1291668026288-3075279.post@n4.nabble.com> Message-ID: <1291668083262-3075282.post@n4.nabble.com> damn it, this damn thing keeps screwing up, let me try again. Here is the code. I actually have started using ffmux_asf untill I can get the x264enc to work. However, this is the same exact pipeline except I am using the asf encoders and muxers instead of x264 and avi. However, now I have the problem of when I use filesink with a location, the file is perfect (though with no seeking--don't know what that is about), however, when I use this code, the client on the other end of the socket can't play the file. I added a standard file to the multifdsink as a test but it isn't receiving any output. Thanks for your reply, I hope your head gets better. /* * H264Stream.cpp * * Created on: Nov 12, 2010 * Author: jonathan */ #include "H264Stream.h" #include "VideoInput.h" int fileD; H264Stream::H264Stream() : PThread (1000, NoAutoDeleteThread, HighestPriority, "H264Stream"), encoding(false) { //temporary setting of variables width = 352; height = 288; fps = 25; audioChannels = 2; audioSampleRate = 8000; bitWidth = 16; GError* error = NULL; gchar* command = NULL; command = g_strdup_printf ("v4l2src ! video/x-raw-yuv, format=(fourcc)I420, width=%d, height=%d, framerate=(fraction)%d/1 !" " videobalance name=VideoBalance ! textoverlay name=chanNameFilter ! textoverlay name=osdMessageFilter ! textoverlay name=sessionTimerOverlay ! " "tee name=t ! queue ! appsink name=videoSink t. ! queue ! ffenc_wmv2 name=videoEncoder me-method=5 ! amux. alsasrc ! " "audio/x-raw-int, depth=%d, width=%d, channels=2, endianness=1234, rate=%d, signed=true ! volume name=volumeFilter ! " "tee name=souTee ! queue ! appsink name=soundSink souTee. ! queue ! ffenc_wmav2 ! amux. ffmux_asf name=amux ! multifdsink name=multifdsink", width, height, fps, bitWidth, bitWidth, audioSampleRate); g_print ("Pipeline: %s\n", command); h264Pipeline = gst_parse_launch (command, &error); if(error != NULL) std::cout << error->message << "\n"; chanNameFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), "chanNameFilter"); osdMessageFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), "osdMessageFilter"); sessionTimerFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), "sessionTimerOverlay"); videoBalance = gst_bin_get_by_name (GST_BIN (h264Pipeline), "VideoBalance"); videoEncoder = gst_bin_get_by_name (GST_BIN (h264Pipeline), "videoEncoder"); volume = gst_bin_get_by_name (GST_BIN (h264Pipeline), "volumeFilter"); multifdSink = gst_bin_get_by_name (GST_BIN (h264Pipeline), "multifdsink"); soundSink = gst_bin_get_by_name (GST_BIN (h264Pipeline), "soundSink"); } H264Stream::~H264Stream() { for(std::map::iterator pair = streamHandles.begin(); pair != streamHandles.end(); pair++) { g_signal_emit_by_name(multifdSink, "remove", pair->first, NULL); delete pair->second; } streamHandles.clear(); gst_element_set_state (h264Pipeline, GST_STATE_NULL); gst_object_unref (GST_OBJECT (h264Pipeline)); } void H264Stream::Main() { while(true) { PWaitAndSignal m(mutex); if(encoding) { OSDSettings osd; if(osd.getShowChanName()) { g_object_set (G_OBJECT (chanNameFilter), "silent", false , NULL); g_object_set (G_OBJECT (chanNameFilter), "text", osd.getChanName().c_str() , NULL); g_object_set (G_OBJECT (chanNameFilter), "halignment", osd.getChanNameHAlign() , NULL); g_object_set (G_OBJECT (chanNameFilter), "valignment", osd.getChanNameVAlign() , NULL); g_object_set (G_OBJECT (chanNameFilter), "wrap-mode", osd.getChanNameWordWrapMode() , NULL); g_object_set (G_OBJECT (chanNameFilter), "font-desc", osd.getChanNameFont().c_str() , NULL); g_object_set (G_OBJECT (chanNameFilter), "shaded-background", osd.getChanNameShadow() , NULL); } else { g_object_set (G_OBJECT (chanNameFilter), "text", "" , NULL); g_object_set (G_OBJECT (chanNameFilter), "silent", true , NULL); } if(osd.getShowOSDMessage()) { g_object_set (G_OBJECT (osdMessageFilter), "silent", false , NULL); g_object_set (G_OBJECT (osdMessageFilter), "text", osd.getOSDMessage().c_str() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "halignment", osd.getOSDMessageHAlign() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "valignment", osd.getOSDMessageVAlign() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "wrap-mode", osd.getOSDMessageWordWrapMode() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "font-desc", osd.getOSDMessageFont().c_str() , NULL); g_object_set (G_OBJECT (osdMessageFilter), "shaded-background", osd.getOSDMessageShadow() , NULL); } else { g_object_set (G_OBJECT (osdMessageFilter), "text", "" , NULL); g_object_set (G_OBJECT (osdMessageFilter), "silent", true , NULL); } if(osd.getShowSessionTimer()) { g_object_set (G_OBJECT (sessionTimerFilter), "silent", false , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "text", osd.getSessionTimer().c_str() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "halignment", osd.getSessionTimerHAlign() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "valignment", osd.getSessionTimerVAlign() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "wrap-mode", osd.getSessionTimerWordWrapMode() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "font-desc", osd.getSessionTimerFont().c_str() , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "shaded-background", osd.getSessionTimerShadow() , NULL); } else { g_object_set (G_OBJECT (sessionTimerFilter), "text", "" , NULL); g_object_set (G_OBJECT (sessionTimerFilter), "silent", true , NULL); } this->Sleep(1000); } } } void H264Stream::RemoveStream(int handle) { if(handle != -1) { g_signal_emit_by_name(multifdSink, "remove", handle, G_TYPE_NONE); delete streamHandles[handle]; streamHandles.erase(handle); g_signal_emit_by_name(multifdSink, "remove", fileD, G_TYPE_NONE); close(fileD); } if(!streamHandles.size()) StopEncoding(); } bool H264Stream::CheckAndBeginEncoding() { if(!encoding) { GstStateChangeReturn stateRet; stateRet = gst_element_set_state (h264Pipeline, GST_STATE_PLAYING); GstState state; stateRet = gst_element_get_state(h264Pipeline, &state, NULL, GST_SECOND); encoding = true; this->Restart(); return true; } else return true; } bool H264Stream::StopEncoding() { gst_element_set_state (h264Pipeline, GST_STATE_READY); encoding = false; return true; } int H264Stream::AddStreamOutput(string ip, string port) { PWaitAndSignal m(mutex); if(CheckAndBeginEncoding()) { fileD = open("/home/jonathan/anotherTest.wmv", O_RDWR | O_APPEND | O_CREAT, 0666); if(fileD != -1) { g_signal_emit_by_name(multifdSink, "add", fileD, G_TYPE_NONE); //streamHandles.insert(std::pair(fd, socket)); } ClientSocket* socket = new ClientSocket(ip, atoi(port.c_str())); int fd = socket->getDescriptor(); if(fd != -1) { g_signal_emit_by_name(multifdSink, "add", fd, G_TYPE_NONE); streamHandles.insert(std::pair(fd, socket)); return fd; } } return -1; } GstBuffer* H264Stream::GetAudioBuffer() { PWaitAndSignal m(mutex); if (soundSink != NULL) { return gst_app_sink_pull_buffer (GST_APP_SINK (soundSink)); } return NULL; } GstBuffer* H264Stream::GetVideoBuffer() { PWaitAndSignal m(mutex); if (videoSink != NULL) { return gst_app_sink_pull_buffer (GST_APP_SINK (videoSink)); } return NULL; } GstCaps* H264Stream::GetCurrentAudioCaps() { PWaitAndSignal m(mutex); if (soundSink != NULL) { return gst_app_sink_get_caps (GST_APP_SINK (soundSink)); } return NULL; } GstCaps* H264Stream::GetCurrentVideoCaps() { PWaitAndSignal m(mutex); if (videoSink != NULL) { return gst_app_sink_get_caps (GST_APP_SINK (videoSink)); } return NULL; } bool H264Stream::SetSessionAudioCaps(GstCaps* caps) { PWaitAndSignal m(mutex); if (soundSink != NULL) { gst_app_sink_set_caps (GST_APP_SINK (soundSink), caps); gst_caps_unref(caps); return true; } return false; } bool H264Stream::SetSessionVideoCaps(GstCaps* caps) { PWaitAndSignal m(mutex); if (videoSink != NULL) { gst_app_sink_set_caps (GST_APP_SINK (videoSink), caps); gst_caps_unref(caps); return true; } return false; } void H264Stream::SetVolume(gfloat value) { g_object_set(G_OBJECT (volume), "volume", value, NULL); } bool H264Stream::SetSaturation(double color) { g_object_set(G_OBJECT (videoBalance), "saturation", color, NULL); return true; } bool H264Stream::SetBrightness(double brightness) { g_object_set(G_OBJECT (videoBalance), "brightness", brightness, NULL); return true; } bool H264Stream::SetHue(double hue) { g_object_set(G_OBJECT (videoBalance), "hue", hue, NULL); return true; } bool H264Stream::SetContrast(double contrast) { g_object_set(G_OBJECT (videoBalance), "contrast", contrast, NULL); return true; } -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Trouble-using-x264enc-with-a-tee-tp3067583p3075282.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From raymond.frst at gmail.com Tue Dec 7 11:49:11 2010 From: raymond.frst at gmail.com (Raymond Frost) Date: Tue, 7 Dec 2010 02:49:11 -0800 (PST) Subject: [gst-devel] Problem with flacdec and ismd_audio_sink Message-ID: <1291718951876-3076140.post@n4.nabble.com> Hi, Im working on a intel CE4150 board with gstreamer and i have issues of extremely low volume whenever i use flacdec along with ismd_audio_sink basically my setup is filesrc ! queue ! flacdec ! audioconvert ! ismd_audio_sink i have no issues when using other combinations such as filesrc ! queue ! flacdec ! audioconvert ! alsasink [same flac playback with a usb uDAC] or filesrc ! queue ! oggparse ! vorbsdec ! audioconvert ! ismd_audio_sink [same device setup for vorbis playback] anybody can shed some light as in what did i miss in setting up a flac pipeline? Thanks in advance -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Problem-with-flacdec-and-ismd-audio-sink-tp3076140p3076140.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Tue Dec 7 16:31:35 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 07 Dec 2010 15:31:35 +0000 Subject: [gst-devel] Problem with flacdec and ismd_audio_sink In-Reply-To: <1291718951876-3076140.post@n4.nabble.com> References: <1291718951876-3076140.post@n4.nabble.com> Message-ID: <1291735895.15436.15.camel@zingle> On Tue, 2010-12-07 at 02:49 -0800, Raymond Frost wrote: > Im working on a intel CE4150 board with gstreamer > and i have issues of extremely low volume whenever i use flacdec along with > ismd_audio_sink. Basically my setup is > > filesrc ! queue ! flacdec ! audioconvert ! ismd_audio_sink > > i have no issues when using other combinations such as (snip) So what format is being negotiated between flacdec and your sink? (pass -v to gst-launch) What if you use e.g. ... ! flacdec ! audioconvert ! \ audio/x-raw-int,width=16,depth=16,channels=2 ! \ ismd_audio_sink for example (try other combinations too, maybe the same as are used with the other formats). Cheers -Tim From wb666greene at gmail.com Tue Dec 7 16:44:06 2010 From: wb666greene at gmail.com (wally_bkg) Date: Tue, 7 Dec 2010 07:44:06 -0800 (PST) Subject: [gst-devel] Tell me about y4menc and other encoders. In-Reply-To: <4CFA6E76.1000003@fas.harvard.edu> References: <1291413235446-3071830.post@n4.nabble.com> <4CFA6E76.1000003@fas.harvard.edu> Message-ID: <1291736646321-3076605.post@n4.nabble.com> Benjamin Schwartz wrote: > > On 12/03/2010 04:53 PM, wally_bkg wrote: >> I first tried: jpegenc, mpeg2enc, and theoraenc. I could eliminate >> mpeg2enc >> from a combination of too high CPU load and too low video quality (using >> only the defaults). > > All of these elements take an argument that allows you to specify your > desired bitrate/quality. There's nothing especially great about the > defaults, which may be incomparable from one element to another. > I guess this is why video editors like Pitivi generally do a crappy job of actually producing output -- expecting either the application developer, or worse the end user, to be expert in encoding parameters instead of providing reasonable defaults. The default quality for theoraenc and jpegenc were usable, the defaults fro mpeg2enc were so far from usable I wouldn't know where to start changing things. Benjamin Schwartz wrote: > >> problem here is while the files play back fine in Totem, there is >> something >> wrong with them if I try to play them on a Windows system (that is set up >> for MJPEG and OGG/Theora). > > If you want help debugging this you'll need to be more specific about the > playback system and the problem. Latest VLC generally plays back Theora > correctly under Windows. > I was using the K-lite Codec pack and "Media Player Classic" on Windows. As I said, the system plays every MJPEG and OGV file I'd thrown at it until these samples I created with gstreamer. I downloaded VLC-win32 1.1.5 and it plays the OGV file back OK. I guess my Codec pack is out of date. I was worried something was wrong with the file headers But VLC doesn't play the MJPEG file, so maybe something is not setup correctly for avimux. Benjamin Schwartz wrote: > >> Then I discovered y4menc: >> while it gives about 30X the file size of theoraenc it uses very little >> CPU, >> less than half of the jpegenc pipeline. > > y4m is uncompressed lossless raw video. > --Ben > I was aware of this, but since a y4mdec seems to be missing I was wondering how it was intended to be used since I couldn't seem to find a video file format that would "wrap" it so seeking would work. As I said, decodebin let me play it back, got some playback speed glitches, but the images all looked good and in proper sequence -- I was quite impressed with the handling of an ~10GB file on my 64-bit system. Thanks for the link. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Tell-me-about-y4menc-and-other-encoders-tp3071830p3076605.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wb666greene at gmail.com Tue Dec 7 16:54:12 2010 From: wb666greene at gmail.com (wally_bkg) Date: Tue, 7 Dec 2010 07:54:12 -0800 (PST) Subject: [gst-devel] Which CODEC for mjpg? In-Reply-To: <4CFAC12D.1050003@hora-obscura.de> References: <4CF3F0DF.201@awpi.com> <1291393988296-3071342.post@n4.nabble.com> <4CFAC12D.1050003@hora-obscura.de> Message-ID: <1291737252433-3076624.post@n4.nabble.com> Stefan Kost wrote: > > Am 03.12.2010 18:33, schrieb wally_bkg: >> >> I have the opposite problem. I can make jpeg video files with: >> >> gst-launch v4l2src ! video/x-raw-yuv ! queue ! ffmpegcolorspace ! jpegenc >> ! >> avimux ! filesink location=mjpeg.avi >> >> and the file plays back fine in gstreamer (using Totum on Ubuntu 10.04) >> but >> when I copy it over to a windows system it doesn't play. There are >> several >> MJPEG playback codecs on the Windows system. Gspot Codec Information on >> Windows shows a valid playback pipeline, but errors in rendering. >> >> >> I tried Theoraenc instead with: >> >> gst-launch v4l2src ! queue ! ffmpegcolorspace ! theoraenc ! oggmux ! >> filesink location=theora.ogv >> >> and again it plays fine in Totum but not on Windows (the Windows system >> can >> play back the ogg video sample files distributed with Ubuntu). >> > > Try gst-launch -e to make gst-launch sending and EOS on Ctrl-c (or use > num-buffers on v4l2src). > > Stefan > Bingo! That was what was missing. Seems the OGV on Windows issue is my K-lite Codec pack is not up to date. VLC-win32 v1.1.5 plays it back fine. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Which-CODEC-for-mjpg-tp3064099p3076624.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wb666greene at gmail.com Tue Dec 7 17:18:53 2010 From: wb666greene at gmail.com (wally_bkg) Date: Tue, 7 Dec 2010 08:18:53 -0800 (PST) Subject: [gst-devel] v4l2src fails to change video norm from PAL to NTSC Message-ID: <1291738733147-3076696.post@n4.nabble.com> This worked with v4l2src version 0.10.7 on Ubuntu 8.04, but when I moved to Ubuntu 10.04 64-bit which has v4l2src version 0.10.21 the norm doesn't change and the video window displays only an apparently static green screen. When I close the video window to exit the program, everything works fine the next times as the norm is NTSC then. It seems about half my video cards default to PAL after a restart, so I need to figure out why this no longer works. Here is the norm setting code: gst_element_set_state (pipeline, GST_STATE_PAUSED); GstTuner *tuner = GST_TUNER( (GstTuner *) source); // get a pointer to access the tuner interface GstTunerNorm *prevNorm=gst_tuner_get_norm(tuner); g_print("Previous Norm: %s\n", prevNorm->label); GstTunerNorm *ntsc=gst_tuner_find_norm_by_name (tuner, "NTSC"); // most have no NTSC-M :( if(ntsc) goto norm_set; ntsc=gst_tuner_find_norm_by_name (tuner, "NTSC-M"); // stupid card, no NTSC :( norm_set: if(ntsc) gst_tuner_set_norm (tuner, ntsc); else g_error("Can't set NTSC!\n"); g_print("Current Norm: %s\n\n", ntsc->label); -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/v4l2src-fails-to-change-video-norm-from-PAL-to-NTSC-tp3076696p3076696.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From pgousseau at hotmail.com Tue Dec 7 18:39:38 2010 From: pgousseau at hotmail.com (goussepi) Date: Tue, 7 Dec 2010 09:39:38 -0800 (PST) Subject: [gst-devel] What plugin elements to implement for a multi input frame/audio grabber ? Message-ID: <1291743578959-3076865.post@n4.nabble.com> Hello, I am looking at implemeting a plugin for a multi inputs audio and frame grabber (eg 8 composite inputs and 8 audio analogue inputs). It would have to allow each input to have different pipelines connected to it. The hardware is a single device with 8 mmaped video buffer and 8 mmaped audio buffers which all share the same configuration setup when opening the device. Should i look at implementing a source plugin with 1 audio and 1 video source pad which will randomly return a buffer from each 8 channel, but then how do i redirect it to the appropriate pipeline ? Or instanciate 8 source elements for video and 8 source elements for audio, but only 1 element initialise the hardware which seems ugly ? Regards, Pierre -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-plugin-elements-to-implement-for-a-multi-input-frame-audio-grabber-tp3076865p3076865.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From acandido at hi-iberia.es Tue Dec 7 18:54:25 2010 From: acandido at hi-iberia.es (Andres Gonzalez) Date: Tue, 07 Dec 2010 18:54:25 +0100 Subject: [gst-devel] Simple ogg pipeline doesn't work Message-ID: <4CFE74D1.5090903@hi-iberia.es> Hi, I have written a simple python program that is supossed to copy an ogg file, using gstreamer. The original ogg file weights 662135 bytes, but the program doesn't write more than 46365. Why does this program get blocked, any idea? Thanks, Andr?s -------------- next part -------------- A non-text attachment was scrubbed... Name: copyogg.py Type: text/x-python Size: 2952 bytes Desc: not available URL: From yogeshm.007 at gmail.com Tue Dec 7 19:11:45 2010 From: yogeshm.007 at gmail.com (Yogesh Marwaha) Date: Tue, 7 Dec 2010 23:41:45 +0530 Subject: [gst-devel] Please check this pipeline (regarding faac) In-Reply-To: References: Message-ID: <201012072341.46331.yogeshm.007@gmail.com> Thanks for the reply. Following is the pipeline I actually tried: - gst-launch cdiocddasrc track=5 device=/dev/cdrom ! audioconvert ! faac outputformat=1 ! filesink location=track5.aac Resultant file is playable by gstreamer (tested through sparklemedia, my own homemade) but its length is not accurately cauculated by gstreamer, it keeps on changing as the file is playedback. (Is something like xingmux applicable here also?). If mp4mux is added to the pipeline (with outputformat=0) it is played fine and length is calculated/shown correctly). Amarok (with xine backend) is unable to play track5.aac, it actually even refuse to add it to playlist. mplayer (through mediacoder/wine) is also unable to play track5.aac (though if it is contained in mp4 it plays fine). Your comments are welcome. Regards, Yogesh Marwaha On Monday 06 Dec 2010 11:16:47 am ved kpl wrote: > yes, it should. > > On Mon, Dec 6, 2010 at 10:54 AM, Yogesh Marwaha wrote: > > Hi, > > > > I want to generate an aac file. Is the following pipeline correct > > (i.e. will it generate valid file): - > > > > gst-launch audiotestsrc wave=sine num-buffers=100 ! audioconvert ! > > faac outputformat=1 ! filesink location=sine.aac > > > > > > Regards, > > > > -- > > Yogesh M > > http://sparklemedia.sourceforge.net/ > > http://mazedaar.wordpress.com/ > > http://snakeeyes.wordpress.com/ > > > > ------------------------------------------------------------------------- > > ----- What happens now with your Lotus Notes apps - do you make another > > costly upgrade, or settle for being marooned without product support? > > Time to move off Lotus Notes and onto the cloud with Force.com, apps are > > easier to build, use, and manage than apps on traditional platforms. > > Sign up for the Lotus Notes Migration Kit to learn more. > > http://p.sf.net/sfu/salesforce-d2d > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > --------------------------------------------------------------------------- > --- What happens now with your Lotus Notes apps - do you make another > costly upgrade, or settle for being marooned without product support? Time > to move off Lotus Notes and onto the cloud with Force.com, apps are easier > to build, use, and manage than apps on traditional platforms. Sign up for > the Lotus Notes Migration Kit to learn more. > http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From gibrovacco at gmail.com Tue Dec 7 19:51:26 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Tue, 7 Dec 2010 20:51:26 +0200 Subject: [gst-devel] Trouble using x264enc with a tee In-Reply-To: <1291668083262-3075282.post@n4.nabble.com> References: <1291219846949-3067583.post@n4.nabble.com> <1291668026288-3075279.post@n4.nabble.com> <1291668083262-3075282.post@n4.nabble.com> Message-ID: Hi, On Mon, Dec 6, 2010 at 10:41 PM, JonathanHenson wrote: > > damn it, this damn thing keeps screwing up, let me try again. > this is better.. > Here is the code. I actually have started using ffmux_asf untill I can get > the x264enc to work. However, this is the same exact pipeline except I am > using the asf encoders and muxers instead of x264 and avi. However, now I > have the problem of when I use filesink with a location, the file is perfect > (though with no seeking--don't know what that is about), however, when I use > this code, the client on the other end of the socket can't play the file. What if you store the file in one pass (not using a socket) and AFTER it you try to play? I know it's not exactly as in your requirements, but it may help understanding the following.. Are there any reasons why you cannot use a streaming protocol for this? Usually a muxer cannot store a complete set of information into a file until it gets an EOS (that is, unless the player is using some heuristics, it's rarely possible to properly play a file unless it has been completely stored). Because of this, streaming protocols (e.g. RTP) are imo more suitable even for inter-process streaming on the same machine. If that's an option I would suggest you not to involve network-aimed components such as a jitter buffer: very basic udp/(de)payloader pipelines over the loopback interface can do the job. > added a standard file to the multifdsink as a test but it isn't receiving > any output. Thanks for your reply, I hope your head gets better. > yep, definitely better now ;). Btw I think that a named pipe with a simple filesink should be the same.. > /* > ?* H264Stream.cpp > ?* > ?* ?Created on: Nov 12, 2010 > ?* ? ? ?Author: jonathan > ?*/ > > #include "H264Stream.h" > #include "VideoInput.h" > > int fileD; > > H264Stream::H264Stream() : PThread (1000, NoAutoDeleteThread, > HighestPriority, "H264Stream"), > ? ? ? ?encoding(false) > { > ? ? ? ?//temporary setting of variables > ? ? ? ?width = 352; > ? ? ? ?height = 288; > ? ? ? ?fps = 25; > > ? ? ? ?audioChannels = 2; > ? ? ? ?audioSampleRate = 8000; > ? ? ? ?bitWidth = 16; > > > ? ? ? ?GError* error = NULL; > ? ? ? ?gchar* command = NULL; > > ? ? ? ?command = g_strdup_printf ("v4l2src ! video/x-raw-yuv, format=(fourcc)I420, > width=%d, height=%d, framerate=(fraction)%d/1 !" > ? ? ? ? ? ? ? ? ? ? ? ?" videobalance name=VideoBalance ! textoverlay name=chanNameFilter ! > textoverlay name=osdMessageFilter ! textoverlay name=sessionTimerOverlay ! " > ? ? ? ? ? ? ? ? ? ? ? ?"tee name=t ! queue ! appsink name=videoSink t. ! queue ! ffenc_wmv2 I see you're using appsrc and appsink. Usually, the first good question in such a case is: "do I really need such elements in my pipeline?". The answer depends on your use case and requirements :). Note that if your app is not producing and consuming buffers in the proper way you may run into troubles. The behaviour will e.g. differ depending on whether you're in pull or push mode. See the elements' documentation for more details. > name=videoEncoder me-method=5 ! amux. ?alsasrc ! " > ? ? ? ? ? ? ? ? ? ? ? ?"audio/x-raw-int, depth=%d, width=%d, channels=2, endianness=1234, > rate=%d, signed=true ! volume name=volumeFilter ! " > ? ? ? ? ? ? ? ? ? ? ? ?"tee name=souTee ! queue ! appsink name=soundSink souTee. ! queue ! maybe you don't need this second queue (not sure). > ffenc_wmav2 ! amux. ffmux_asf name=amux ! multifdsink name=multifdsink", > ? ? ? ? ? ? ? ? ? ? ? ? width, height, fps, bitWidth, bitWidth, audioSampleRate); here, instead of a mux and multifdsink, you should really try with payloaders and udpsinks (if you want to setup a media producer/consumer processes pair). Regards > > ? g_print ("Pipeline: %s\n", command); > ? ? ? ?h264Pipeline = gst_parse_launch (command, &error); > > ? ? ? ?if(error != NULL) > ? ? ? ?std::cout << error->message << "\n"; > > ? ? ? ?chanNameFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), > "chanNameFilter"); > ? ? ? ?osdMessageFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), > "osdMessageFilter"); > ? ? ? ?sessionTimerFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), > "sessionTimerOverlay"); > ? ? ? ?videoBalance = gst_bin_get_by_name (GST_BIN (h264Pipeline), > "VideoBalance"); > ? ? ? ?videoEncoder = gst_bin_get_by_name (GST_BIN (h264Pipeline), > "videoEncoder"); > ? ? ? ?volume = gst_bin_get_by_name (GST_BIN (h264Pipeline), "volumeFilter"); > ? ? ? ?multifdSink = gst_bin_get_by_name (GST_BIN (h264Pipeline), "multifdsink"); > ? ? ? ?soundSink = gst_bin_get_by_name (GST_BIN (h264Pipeline), "soundSink"); > } > > H264Stream::~H264Stream() > { > ? ? ? ?for(std::map::iterator pair = streamHandles.begin(); > pair != streamHandles.end(); pair++) > ? ? ? ?{ > ? ? ? ? ? ? ? ?g_signal_emit_by_name(multifdSink, "remove", pair->first, NULL); > ? ? ? ? ? ? ? ?delete pair->second; > ? ? ? ?} > > ? ? ? ?streamHandles.clear(); > > ? ? ? ?gst_element_set_state (h264Pipeline, GST_STATE_NULL); > ? ? ? ?gst_object_unref (GST_OBJECT (h264Pipeline)); > } > > void H264Stream::Main() > { > ? ? ? ?while(true) > ? ? ? ?{ > ? ? ? ? ? ? ? ?PWaitAndSignal m(mutex); > ? ? ? ? ? ? ? ?if(encoding) > ? ? ? ? ? ? ? ?{ > ? ? ? ? ? ? ? ? ?OSDSettings osd; > > ? ? ? ? ? ? ? ? ?if(osd.getShowChanName()) > ? ? ? ? ? ? ? ? ?{ > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (chanNameFilter), "silent", false , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (chanNameFilter), "text", > osd.getChanName().c_str() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (chanNameFilter), "halignment", > osd.getChanNameHAlign() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (chanNameFilter), "valignment", > osd.getChanNameVAlign() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (chanNameFilter), "wrap-mode", > osd.getChanNameWordWrapMode() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (chanNameFilter), "font-desc", > osd.getChanNameFont().c_str() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (chanNameFilter), "shaded-background", > osd.getChanNameShadow() , NULL); > ? ? ? ? ? ? ? ? ?} > ? ? ? ? ? ? ? ? ?else > ? ? ? ? ? ? ? ? ?{ > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (chanNameFilter), "text", "" , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (chanNameFilter), "silent", true , NULL); > ? ? ? ? ? ? ? ? ?} > > ? ? ? ? ? ? ? ? ?if(osd.getShowOSDMessage()) > ? ? ? ? ? ? ? ? ?{ > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (osdMessageFilter), "silent", false , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (osdMessageFilter), "text", > osd.getOSDMessage().c_str() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (osdMessageFilter), "halignment", > osd.getOSDMessageHAlign() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (osdMessageFilter), "valignment", > osd.getOSDMessageVAlign() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (osdMessageFilter), "wrap-mode", > osd.getOSDMessageWordWrapMode() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (osdMessageFilter), "font-desc", > osd.getOSDMessageFont().c_str() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (osdMessageFilter), "shaded-background", > osd.getOSDMessageShadow() , NULL); > ? ? ? ? ? ? ? ? ?} > ? ? ? ? ? ? ? ? ?else > ? ? ? ? ? ? ? ? ?{ > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (osdMessageFilter), "text", "" , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (osdMessageFilter), "silent", true , NULL); > ? ? ? ? ? ? ? ? ?} > > ? ? ? ? ? ? ? ? ?if(osd.getShowSessionTimer()) > ? ? ? ? ? ? ? ? ?{ > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (sessionTimerFilter), "silent", false , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (sessionTimerFilter), "text", > osd.getSessionTimer().c_str() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (sessionTimerFilter), "halignment", > osd.getSessionTimerHAlign() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (sessionTimerFilter), "valignment", > osd.getSessionTimerVAlign() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (sessionTimerFilter), "wrap-mode", > osd.getSessionTimerWordWrapMode() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (sessionTimerFilter), "font-desc", > osd.getSessionTimerFont().c_str() , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (sessionTimerFilter), "shaded-background", > osd.getSessionTimerShadow() , NULL); > > ? ? ? ? ? ? ? ? ?} > ? ? ? ? ? ? ? ? ?else > ? ? ? ? ? ? ? ? ?{ > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (sessionTimerFilter), "text", "" , NULL); > ? ? ? ? ? ? ? ? ? ? ? ? ?g_object_set (G_OBJECT (sessionTimerFilter), "silent", true , NULL); > ? ? ? ? ? ? ? ? ?} > > ? ? ? ? ? ? ? ? ? ? ? ?this->Sleep(1000); > ? ? ? ? ? ? ? ?} > ? ? ? ?} > } > > void H264Stream::RemoveStream(int handle) > { > ? ? ? ?if(handle != -1) > ? ? ? ?{ > ? ? ? ? ? ? ? ?g_signal_emit_by_name(multifdSink, "remove", handle, G_TYPE_NONE); > ? ? ? ? ? ? ? ?delete streamHandles[handle]; > ? ? ? ? ? ? ? ?streamHandles.erase(handle); > > ? ? ? ? ? ? ? ?g_signal_emit_by_name(multifdSink, "remove", fileD, G_TYPE_NONE); > ? ? ? ? ? ? ? ?close(fileD); > ? ? ? ?} > > ? ? ? ?if(!streamHandles.size()) > ? ? ? ? ? ? ? ?StopEncoding(); > } > > bool H264Stream::CheckAndBeginEncoding() > { > ? ? ? ?if(!encoding) > ? ? ? ?{ > ? ? ? ? ? ? ? ?GstStateChangeReturn stateRet; > ? ? ? ? ? ? ? ?stateRet = gst_element_set_state (h264Pipeline, GST_STATE_PLAYING); > > ? ? ? ? ? ? ? ?GstState state; > > ? ? ? ? ? ? ? ?stateRet = gst_element_get_state(h264Pipeline, &state, NULL, GST_SECOND); > ? ? ? ? ? ? ? ?encoding = true; > ? ? ? ? ? ? ? ?this->Restart(); > ? ? ? ? ? ? ? ?return true; > ? ? ? ?} > ? ? ? ?else > ? ? ? ? ? ? ? ?return true; > } > > bool H264Stream::StopEncoding() > { > ? ? ? ?gst_element_set_state (h264Pipeline, GST_STATE_READY); > > ? ? ? ?encoding = false; > ? ? ? ?return true; > } > > int H264Stream::AddStreamOutput(string ip, string port) > { > ? ? ? ?PWaitAndSignal m(mutex); > ? ? ? ?if(CheckAndBeginEncoding()) > ? ? ? ?{ > ? ? ? ? ? ? ? ?fileD = open("/home/jonathan/anotherTest.wmv", O_RDWR | O_APPEND | > O_CREAT, 0666); > > ? ? ? ? ? ? ? ?if(fileD != -1) > ? ? ? ? ? ? ? ?{ > ? ? ? ? ? ? ? ? ? ? ? ?g_signal_emit_by_name(multifdSink, "add", fileD, G_TYPE_NONE); > ? ? ? ? ? ? ? ? ? ? ? ?//streamHandles.insert(std::pair(fd, socket)); > ? ? ? ? ? ? ? ?} > > ? ? ? ? ? ? ? ?ClientSocket* socket = new ClientSocket(ip, atoi(port.c_str())); > > ? ? ? ? ? ? ? ?int fd = socket->getDescriptor(); > > ? ? ? ? ? ? ? ?if(fd != -1) > ? ? ? ? ? ? ? ?{ > ? ? ? ? ? ? ? ? ? ? ? ?g_signal_emit_by_name(multifdSink, "add", fd, G_TYPE_NONE); > ? ? ? ? ? ? ? ? ? ? ? ?streamHandles.insert(std::pair(fd, socket)); > ? ? ? ? ? ? ? ? ? ? ? ?return fd; > ? ? ? ? ? ? ? ?} > > > ? ? ? ?} > ? ? ? ?return -1; > } > > GstBuffer* H264Stream::GetAudioBuffer() > { > ? ? ? ?PWaitAndSignal m(mutex); > > ? ? ? ? if (soundSink != NULL) { > ? ? ? ? ? ? ? ? return gst_app_sink_pull_buffer (GST_APP_SINK (soundSink)); > ? ? ? ? } > ? ? ? ? return NULL; > } > > GstBuffer* H264Stream::GetVideoBuffer() > { > ? ? ? ?PWaitAndSignal m(mutex); > > ? ? ? ? if (videoSink != NULL) { > ? ? ? ? ? ? ? ? return gst_app_sink_pull_buffer (GST_APP_SINK (videoSink)); > ? ? ? ? } > ? ? ? ? return NULL; > } > > GstCaps* H264Stream::GetCurrentAudioCaps() > { > ? ? ? ?PWaitAndSignal m(mutex); > > ? ? ? ? if (soundSink != NULL) { > ? ? ? ? ? ? ? ? return gst_app_sink_get_caps (GST_APP_SINK (soundSink)); > ? ? ? ? } > ? ? ? ? return NULL; > } > > GstCaps* H264Stream::GetCurrentVideoCaps() > { > ? ? ? ?PWaitAndSignal m(mutex); > > ? ? ? ? if (videoSink != NULL) { > ? ? ? ? ? ? ? ? return gst_app_sink_get_caps (GST_APP_SINK (videoSink)); > ? ? ? ? } > ? ? ? ? return NULL; > } > > bool H264Stream::SetSessionAudioCaps(GstCaps* caps) > { > ? ? ? ? PWaitAndSignal m(mutex); > > ? ? ? ? if (soundSink != NULL) { > ? ? ? ? ? ? ? ? gst_app_sink_set_caps (GST_APP_SINK (soundSink), caps); > ? ? ? ? ? ? ? ? gst_caps_unref(caps); > ? ? ? ? ? ? ? ? return true; > ? ? ? ? } > ? ? ? ? return false; > } > > bool H264Stream::SetSessionVideoCaps(GstCaps* caps) > { > ? ? ? ? PWaitAndSignal m(mutex); > > ? ? ? ? if (videoSink != NULL) { > ? ? ? ? ? ? ? ? gst_app_sink_set_caps (GST_APP_SINK (videoSink), caps); > ? ? ? ? ? ? ? ? gst_caps_unref(caps); > ? ? ? ? ? ? ? ? return true; > ? ? ? ? } > ? ? ? ? return false; > } > > void H264Stream::SetVolume(gfloat value) > { > ? ? ? ?g_object_set(G_OBJECT (volume), "volume", value, NULL); > } > > bool H264Stream::SetSaturation(double color) > { > ? ? ? ?g_object_set(G_OBJECT (videoBalance), "saturation", color, NULL); > > ? ? ? ?return true; > } > > bool H264Stream::SetBrightness(double brightness) > { > ? ? ? ?g_object_set(G_OBJECT (videoBalance), "brightness", brightness, NULL); > > ? ? ? ?return true; > } > > bool H264Stream::SetHue(double hue) > { > ? ? ? ?g_object_set(G_OBJECT (videoBalance), "hue", hue, NULL); > > ? ? ? ?return true; > } > > bool H264Stream::SetContrast(double contrast) > { > ? ? ? ?g_object_set(G_OBJECT (videoBalance), "contrast", contrast, NULL); > > ? ? ? ?return true; > } > > > > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Trouble-using-x264enc-with-a-tee-tp3067583p3075282.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From t.i.m at zen.co.uk Tue Dec 7 20:22:58 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 07 Dec 2010 19:22:58 +0000 Subject: [gst-devel] Trouble using x264enc with a tee In-Reply-To: <1291668083262-3075282.post@n4.nabble.com> References: <1291219846949-3067583.post@n4.nabble.com> <1291668026288-3075279.post@n4.nabble.com> <1291668083262-3075282.post@n4.nabble.com> Message-ID: <1291749778.15436.20.camel@zingle> On Mon, 2010-12-06 at 12:41 -0800, JonathanHenson wrote: Hi, > Here is the code. I actually have started using ffmux_asf untill I can get > the x264enc to work. However, this is the same exact pipeline except I am > using the asf encoders and muxers instead of x264 and avi. However, now I > have the problem of when I use filesink with a location, the file is perfect > (though with no seeking--don't know what that is about), however, when I use > this code, the client on the other end of the socket can't play the file. I > added a standard file to the multifdsink as a test but it isn't receiving > any output. Didn't look into it in too much detail, but most likely the problem is that the other queues in the non-x264enc branches are too small. x264enc has a very high latency by default (60-80 frames IIRC), so will need a lot of input buffers before it outputs its first buffer (which is what any muxer would wait for). Either increase or unset queue limits in the other branches (e.g. queue max-size-bytes=0 max-size-time=0 max-size-buffers=0), or use multiqueue or use x264enc tune=zerolatency (this leads to much lower quality output though). The live source in your pipeline will limit dataflow upstream of the queues, so you shouldn't have a problem with queues growing endlessly until you run out of memory if you remove all size limits. Cheers -Tim > /* > * H264Stream.cpp > * > * Created on: Nov 12, 2010 > * Author: jonathan > */ > > #include "H264Stream.h" > #include "VideoInput.h" > > int fileD; > > H264Stream::H264Stream() : PThread (1000, NoAutoDeleteThread, > HighestPriority, "H264Stream"), > encoding(false) > { > //temporary setting of variables > width = 352; > height = 288; > fps = 25; > > audioChannels = 2; > audioSampleRate = 8000; > bitWidth = 16; > > > GError* error = NULL; > gchar* command = NULL; > > command = g_strdup_printf ("v4l2src ! video/x-raw-yuv, format=(fourcc)I420, > width=%d, height=%d, framerate=(fraction)%d/1 !" > " videobalance name=VideoBalance ! textoverlay name=chanNameFilter ! > textoverlay name=osdMessageFilter ! textoverlay name=sessionTimerOverlay ! " > "tee name=t ! queue ! appsink name=videoSink t. ! queue ! ffenc_wmv2 > name=videoEncoder me-method=5 ! amux. alsasrc ! " > "audio/x-raw-int, depth=%d, width=%d, channels=2, endianness=1234, > rate=%d, signed=true ! volume name=volumeFilter ! " > "tee name=souTee ! queue ! appsink name=soundSink souTee. ! queue ! > ffenc_wmav2 ! amux. ffmux_asf name=amux ! multifdsink name=multifdsink", > width, height, fps, bitWidth, bitWidth, audioSampleRate); > > g_print ("Pipeline: %s\n", command); > h264Pipeline = gst_parse_launch (command, &error); > > if(error != NULL) > std::cout << error->message << "\n"; > > chanNameFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), > "chanNameFilter"); > osdMessageFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), > "osdMessageFilter"); > sessionTimerFilter = gst_bin_get_by_name (GST_BIN (h264Pipeline), > "sessionTimerOverlay"); > videoBalance = gst_bin_get_by_name (GST_BIN (h264Pipeline), > "VideoBalance"); > videoEncoder = gst_bin_get_by_name (GST_BIN (h264Pipeline), > "videoEncoder"); > volume = gst_bin_get_by_name (GST_BIN (h264Pipeline), "volumeFilter"); > multifdSink = gst_bin_get_by_name (GST_BIN (h264Pipeline), "multifdsink"); > soundSink = gst_bin_get_by_name (GST_BIN (h264Pipeline), "soundSink"); > } > > H264Stream::~H264Stream() > { > for(std::map::iterator pair = streamHandles.begin(); > pair != streamHandles.end(); pair++) > { > g_signal_emit_by_name(multifdSink, "remove", pair->first, NULL); > delete pair->second; > } > > streamHandles.clear(); > > gst_element_set_state (h264Pipeline, GST_STATE_NULL); > gst_object_unref (GST_OBJECT (h264Pipeline)); > } > > void H264Stream::Main() > { > while(true) > { > PWaitAndSignal m(mutex); > if(encoding) > { > OSDSettings osd; > > if(osd.getShowChanName()) > { > g_object_set (G_OBJECT (chanNameFilter), "silent", false , NULL); > g_object_set (G_OBJECT (chanNameFilter), "text", > osd.getChanName().c_str() , NULL); > g_object_set (G_OBJECT (chanNameFilter), "halignment", > osd.getChanNameHAlign() , NULL); > g_object_set (G_OBJECT (chanNameFilter), "valignment", > osd.getChanNameVAlign() , NULL); > g_object_set (G_OBJECT (chanNameFilter), "wrap-mode", > osd.getChanNameWordWrapMode() , NULL); > g_object_set (G_OBJECT (chanNameFilter), "font-desc", > osd.getChanNameFont().c_str() , NULL); > g_object_set (G_OBJECT (chanNameFilter), "shaded-background", > osd.getChanNameShadow() , NULL); > } > else > { > g_object_set (G_OBJECT (chanNameFilter), "text", "" , NULL); > g_object_set (G_OBJECT (chanNameFilter), "silent", true , NULL); > } > > if(osd.getShowOSDMessage()) > { > g_object_set (G_OBJECT (osdMessageFilter), "silent", false , NULL); > g_object_set (G_OBJECT (osdMessageFilter), "text", > osd.getOSDMessage().c_str() , NULL); > g_object_set (G_OBJECT (osdMessageFilter), "halignment", > osd.getOSDMessageHAlign() , NULL); > g_object_set (G_OBJECT (osdMessageFilter), "valignment", > osd.getOSDMessageVAlign() , NULL); > g_object_set (G_OBJECT (osdMessageFilter), "wrap-mode", > osd.getOSDMessageWordWrapMode() , NULL); > g_object_set (G_OBJECT (osdMessageFilter), "font-desc", > osd.getOSDMessageFont().c_str() , NULL); > g_object_set (G_OBJECT (osdMessageFilter), "shaded-background", > osd.getOSDMessageShadow() , NULL); > } > else > { > g_object_set (G_OBJECT (osdMessageFilter), "text", "" , NULL); > g_object_set (G_OBJECT (osdMessageFilter), "silent", true , NULL); > } > > if(osd.getShowSessionTimer()) > { > g_object_set (G_OBJECT (sessionTimerFilter), "silent", false , NULL); > g_object_set (G_OBJECT (sessionTimerFilter), "text", > osd.getSessionTimer().c_str() , NULL); > g_object_set (G_OBJECT (sessionTimerFilter), "halignment", > osd.getSessionTimerHAlign() , NULL); > g_object_set (G_OBJECT (sessionTimerFilter), "valignment", > osd.getSessionTimerVAlign() , NULL); > g_object_set (G_OBJECT (sessionTimerFilter), "wrap-mode", > osd.getSessionTimerWordWrapMode() , NULL); > g_object_set (G_OBJECT (sessionTimerFilter), "font-desc", > osd.getSessionTimerFont().c_str() , NULL); > g_object_set (G_OBJECT (sessionTimerFilter), "shaded-background", > osd.getSessionTimerShadow() , NULL); > > } > else > { > g_object_set (G_OBJECT (sessionTimerFilter), "text", "" , NULL); > g_object_set (G_OBJECT (sessionTimerFilter), "silent", true , NULL); > } > > this->Sleep(1000); > } > } > } > > void H264Stream::RemoveStream(int handle) > { > if(handle != -1) > { > g_signal_emit_by_name(multifdSink, "remove", handle, G_TYPE_NONE); > delete streamHandles[handle]; > streamHandles.erase(handle); > > g_signal_emit_by_name(multifdSink, "remove", fileD, G_TYPE_NONE); > close(fileD); > } > > if(!streamHandles.size()) > StopEncoding(); > } > > bool H264Stream::CheckAndBeginEncoding() > { > if(!encoding) > { > GstStateChangeReturn stateRet; > stateRet = gst_element_set_state (h264Pipeline, GST_STATE_PLAYING); > > GstState state; > > stateRet = gst_element_get_state(h264Pipeline, &state, NULL, GST_SECOND); > encoding = true; > this->Restart(); > return true; > } > else > return true; > } > > bool H264Stream::StopEncoding() > { > gst_element_set_state (h264Pipeline, GST_STATE_READY); > > encoding = false; > return true; > } > > int H264Stream::AddStreamOutput(string ip, string port) > { > PWaitAndSignal m(mutex); > if(CheckAndBeginEncoding()) > { > fileD = open("/home/jonathan/anotherTest.wmv", O_RDWR | O_APPEND | > O_CREAT, 0666); > > if(fileD != -1) > { > g_signal_emit_by_name(multifdSink, "add", fileD, G_TYPE_NONE); > //streamHandles.insert(std::pair(fd, socket)); > } > > ClientSocket* socket = new ClientSocket(ip, atoi(port.c_str())); > > int fd = socket->getDescriptor(); > > if(fd != -1) > { > g_signal_emit_by_name(multifdSink, "add", fd, G_TYPE_NONE); > streamHandles.insert(std::pair(fd, socket)); > return fd; > } > > > } > return -1; > } > > GstBuffer* H264Stream::GetAudioBuffer() > { > PWaitAndSignal m(mutex); > > if (soundSink != NULL) { > return gst_app_sink_pull_buffer (GST_APP_SINK (soundSink)); > } > return NULL; > } > > GstBuffer* H264Stream::GetVideoBuffer() > { > PWaitAndSignal m(mutex); > > if (videoSink != NULL) { > return gst_app_sink_pull_buffer (GST_APP_SINK (videoSink)); > } > return NULL; > } > > GstCaps* H264Stream::GetCurrentAudioCaps() > { > PWaitAndSignal m(mutex); > > if (soundSink != NULL) { > return gst_app_sink_get_caps (GST_APP_SINK (soundSink)); > } > return NULL; > } > > GstCaps* H264Stream::GetCurrentVideoCaps() > { > PWaitAndSignal m(mutex); > > if (videoSink != NULL) { > return gst_app_sink_get_caps (GST_APP_SINK (videoSink)); > } > return NULL; > } > > bool H264Stream::SetSessionAudioCaps(GstCaps* caps) > { > PWaitAndSignal m(mutex); > > if (soundSink != NULL) { > gst_app_sink_set_caps (GST_APP_SINK (soundSink), caps); > gst_caps_unref(caps); > return true; > } > return false; > } > > bool H264Stream::SetSessionVideoCaps(GstCaps* caps) > { > PWaitAndSignal m(mutex); > > if (videoSink != NULL) { > gst_app_sink_set_caps (GST_APP_SINK (videoSink), caps); > gst_caps_unref(caps); > return true; > } > return false; > } > > void H264Stream::SetVolume(gfloat value) > { > g_object_set(G_OBJECT (volume), "volume", value, NULL); > } > > bool H264Stream::SetSaturation(double color) > { > g_object_set(G_OBJECT (videoBalance), "saturation", color, NULL); > > return true; > } > > bool H264Stream::SetBrightness(double brightness) > { > g_object_set(G_OBJECT (videoBalance), "brightness", brightness, NULL); > > return true; > } > > bool H264Stream::SetHue(double hue) > { > g_object_set(G_OBJECT (videoBalance), "hue", hue, NULL); > > return true; > } > > bool H264Stream::SetContrast(double contrast) > { > g_object_set(G_OBJECT (videoBalance), "contrast", contrast, NULL); > > return true; > } > > > > > From wb666greene at gmail.com Wed Dec 8 00:31:51 2010 From: wb666greene at gmail.com (wally_bkg) Date: Tue, 7 Dec 2010 15:31:51 -0800 (PST) Subject: [gst-devel] V4L2src plugin does not call VIDIOC_S_INPUT???? In-Reply-To: <1287049451686-2995076.post@n4.nabble.com> References: <1287049451686-2995076.post@n4.nabble.com> Message-ID: <1291764711921-3077443.post@n4.nabble.com> I believe you need to use the v4l2src tuner interface. A sample can be found here: http://gstreamer-devel.966125.n4.nabble.com/Problem-Trying-To-Use-v4l2src-td973410.html#a973414 I've still not got it to work 100% as the apparently bogus error message is gone with the newer gstreamer build, but now the switch from PAL to NTSC doesn't work see: http://gstreamer-devel.966125.n4.nabble.com/v4l2src-fails-to-change-video-norm-from-PAL-to-NTSC-td3076696.html Seems seems half the video cards I have available for testing default to PAL on powerup. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/V4L2src-plugin-does-not-call-VIDIOC-S-INPUT-tp2995076p3077443.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wb666greene at gmail.com Wed Dec 8 00:39:43 2010 From: wb666greene at gmail.com (wally_bkg) Date: Tue, 7 Dec 2010 15:39:43 -0800 (PST) Subject: [gst-devel] v4l2src fails to change video norm from PAL to NTSC In-Reply-To: <1291738733147-3076696.post@n4.nabble.com> References: <1291738733147-3076696.post@n4.nabble.com> Message-ID: <1291765183409-3077453.post@n4.nabble.com> Looks like another undocumented or unintentional change to gstreamer, need to set pipeline to READY instead of PAUSED now before using the interface. Change: gst_element_set_state (pipeline, GST_STATE_PAUSED); To: gst_element_set_state (pipeline, GST_STATE_READY); And the PAL to NTSC switch happens correctly now. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/v4l2src-fails-to-change-video-norm-from-PAL-to-NTSC-tp3076696p3077453.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Wed Dec 8 00:56:19 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 07 Dec 2010 23:56:19 +0000 Subject: [gst-devel] v4l2src fails to change video norm from PAL to NTSC In-Reply-To: <1291765183409-3077453.post@n4.nabble.com> References: <1291738733147-3076696.post@n4.nabble.com> <1291765183409-3077453.post@n4.nabble.com> Message-ID: <1291766179.20631.31.camel@zingle> On Tue, 2010-12-07 at 15:39 -0800, wally_bkg wrote: > Looks like another undocumented or unintentional change to gstreamer, need to > set pipeline to READY instead of PAUSED now before using the interface. > > Change: > > gst_element_set_state (pipeline, GST_STATE_PAUSED); > > To: > > gst_element_set_state (pipeline, GST_STATE_READY); > > And the PAL to NTSC switch happens correctly now. I believe this change (open device when going to READY, not PAUSED) was intentional, to align v4l2src with other elements (opening the device in the NULL->READY state change), but by itself shouldn't break anything. It shouldn't make a difference at the v4l2src level, since the device will go to READY state before going to PAUSED state. The reason this breaks for you between 0.10.7 and is probably that you're dealing with the pipeline as a whole, and latency handling was introduced around 0.10.12-0.10.14, which introduced behavioural incompatibilities for some cases which were thought not to work properly before anyway (there was a transition period with warnings for a few releases, but that doesn't help if you upgrade from an ancient release to a very new one of course). Cheers -Tim From jonathan.henson at innovisit.com Tue Dec 7 20:12:50 2010 From: jonathan.henson at innovisit.com (JonathanHenson) Date: Tue, 7 Dec 2010 11:12:50 -0800 (PST) Subject: [gst-devel] Trouble using x264enc with a tee In-Reply-To: References: <1291219846949-3067583.post@n4.nabble.com> <1291668026288-3075279.post@n4.nabble.com> <1291668083262-3075282.post@n4.nabble.com> Message-ID: <1291749170799-3077032.post@n4.nabble.com> "What if you store the file in one pass (not using a socket) and AFTER it you try to play? I know it's not exactly as in your requirements, but it may help understanding the following.. " If i use filesink instead of multifdsink, I can play the file back fine. I am using the multifdsink so that I can stream to an indefinite number of clients over a TCP/IP socket. However, I just realized, that to receive the stream on the C#.NET side, I will need to use RTP (of which I know very little about--I did read the spec though). "Are there any reasons why you cannot use a streaming protocol for this? " Yes, I don't know shit about RTP. I need some help on the gstreamer side. It seems that I am not supposed to multiplex the stream before sending it in RTP but send to separate streams. I currently have: command = g_strdup_printf ("v4l2src ! video/x-raw-yuv, format=(fourcc)I420, width=%d, height=%d, framerate=(fraction)%d/1 !" " videobalance name=VideoBalance ! textoverlay name=chanNameFilter ! textoverlay name=osdMessageFilter ! textoverlay name=sessionTimerOverlay ! " "tee name=t ! queue ! appsink name=videoSink t. ! queue ! ffenc_wmv1 name=videoEncoder me-method=5 ! amux. alsasrc ! " "audio/x-raw-int, depth=%d, width=%d, channels=2, endianness=1234, rate=%d, signed=true ! volume name=volumeFilter ! " "tee name=souTee ! queue ! appsink name=soundSink souTee. ! queue ! ffenc_wmav1 ! amux. asfmux name=amux ! rtpasfpay ! multifdsink name=multifdsink", width, height, fps, bitWidth, bitWidth, audioSampleRate); I think I would need so use gstrtpbin instead of the multifdsink and get rid of the muxing? "see you're using appsrc and appsink. Usually, the first good question in such a case is: "do I really need such elements in my pipeline?". The answer depends on your use case and requirements :). Note that if your app is not producing and consuming buffers in the proper way you may run into troubles. The behaviour will e.g. differ depending on whether you're in pull or push mode. See the elements' documentation for more details. " I am using the appsinks because this app is using OPAL in other threads to answer H323 and SIP requests and it needs the raw data buffer. This thread is used for a client control computer and SDK which will monitor sessions, and make recordings of the sessions (i.e. a Windows Server 2008 Web Server using ASP.NET/ C#.net with an SDK I have written to talk to this device.) Do you have a better approach in mind? The OPAL thread grabs this buffer when it needs it. Thank you once again, Jonathan -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Trouble-using-x264enc-with-a-tee-tp3067583p3077032.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From raymond.frst at gmail.com Wed Dec 8 10:35:09 2010 From: raymond.frst at gmail.com (Raymond Frost) Date: Wed, 8 Dec 2010 01:35:09 -0800 (PST) Subject: [gst-devel] Problem with flacdec and ismd_audio_sink In-Reply-To: <1291735895.15436.15.camel@zingle> References: <1291718951876-3076140.post@n4.nabble.com> <1291735895.15436.15.camel@zingle> Message-ID: <1291800909157-3077927.post@n4.nabble.com> Tim-Philipp M?ller-2 wrote: > > On Tue, 2010-12-07 at 02:49 -0800, Raymond Frost wrote: > >> Im working on a intel CE4150 board with gstreamer >> and i have issues of extremely low volume whenever i use flacdec along >> with >> ismd_audio_sink. Basically my setup is >> >> filesrc ! queue ! flacdec ! audioconvert ! ismd_audio_sink >> >> i have no issues when using other combinations such as (snip) > > So what format is being negotiated between flacdec and your sink? (pass > -v to gst-launch) > > What if you use e.g. > > ... ! flacdec ! audioconvert ! \ > audio/x-raw-int,width=16,depth=16,channels=2 ! \ > ismd_audio_sink > > for example (try other combinations too, maybe the same as are used with > the other formats). > > Cheers > -Tim > > well, i have been trying something similar before, it goes something like the codes below, *note that I'm using C audio_caps = gst_caps_new_simple("audio/x-raw-int", "width", G_TYPE_INT, 16, "rate", G_TYPE_INT, 48000, "channels", G_TYPE_INT, 2, "endianness", G_TYPE_INT, 1234, "depth", G_TYPE_INT, 24, "signed", G_TYPE_BOOLEAN, true, NULL); g_assert(audio_caps); gst_pad_set_caps(audioconvert_src_pad, audio_caps); gst_caps_unref(audio_caps); I run this before linking up the pipeline, and so far i have tried numerous setting combinations, so far i have no positive improvements yet. so far only by replacing the audiosink to alsasink, i can get good volume with the same setting on all the other part of the pipeline. i guess this part of setting could be the only connection to this problem. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Problem-with-flacdec-and-ismd-audio-sink-tp3076140p3077927.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From c.r.n.a at wanadoo.fr Wed Dec 8 13:04:05 2010 From: c.r.n.a at wanadoo.fr (Nicolas) Date: Wed, 08 Dec 2010 13:04:05 +0100 Subject: [gst-devel] Fakesink and GstNavigation Message-ID: <4CFF7435.9020000@wanadoo.fr> Hello, I created a "playbin2" and a "fakesink" (I render the video in an evas image) Now my problem is that "fakesink" has no "GstNavigation". How can I do to implement the "GstNavigation" to "Fakesink" ? Thank you in advance Sincerely, Nicolas. From gibrovacco at gmail.com Wed Dec 8 14:13:26 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 8 Dec 2010 15:13:26 +0200 Subject: [gst-devel] Trouble using x264enc with a tee In-Reply-To: <1291749170799-3077032.post@n4.nabble.com> References: <1291219846949-3067583.post@n4.nabble.com> <1291668026288-3075279.post@n4.nabble.com> <1291668083262-3075282.post@n4.nabble.com> <1291749170799-3077032.post@n4.nabble.com> Message-ID: Hi, On Tue, Dec 7, 2010 at 9:12 PM, JonathanHenson wrote: > > "What if you store the file in one pass (not using a socket) and AFTER > it you try to play? I know it's not exactly as in your requirements, > but it may help understanding the following.. " > > If i use filesink instead of multifdsink, I can play the file back fine. I > am using the multifdsink so that I can stream to an indefinite number of > clients over a TCP/IP socket. However, I just realized, that to receive the > stream on the C#.NET side, I will need to use RTP (of which I know very > little about--I did read the spec though). Ok so this confirms my suspects -and you understood the issue properly, as I read below ;) -. Please note that GStreamer DOES support multicast streams. > > "Are there any reasons why you cannot use a streaming protocol for this? " > > Yes, I don't know shit about RTP. I need some help on the gstreamer side. It > seems that I am not supposed to multiplex the stream before sending it in > RTP but send to separate streams. Yep, you don't need to mux anything when dealing with multimedia streaming. Here are some pretty good examples that maybe you've already seen: http://library.gnome.org/devel/gst-plugins-libs/unstable/gst-plugins-good-plugins-gstrtpbin.html See the "example pipelines" section. > I currently have: > > command = g_strdup_printf ("v4l2src ! video/x-raw-yuv, format=(fourcc)I420, > width=%d, height=%d, framerate=(fraction)%d/1 !" > ? ? ? ? ? ? ? ? ? ? ? ?" videobalance name=VideoBalance ! textoverlay name=chanNameFilter ! > textoverlay name=osdMessageFilter ! textoverlay name=sessionTimerOverlay ! " > ? ? ? ? ? ? ? ? ? ? ? ?"tee name=t ! queue ! appsink name=videoSink t. ! queue ! ffenc_wmv1 > name=videoEncoder me-method=5 ! amux. ?alsasrc ! " > ? ? ? ? ? ? ? ? ? ? ? ?"audio/x-raw-int, depth=%d, width=%d, channels=2, endianness=1234, > rate=%d, signed=true ! volume name=volumeFilter ! " > ? ? ? ? ? ? ? ? ? ? ? ?"tee name=souTee ! queue ! appsink name=soundSink souTee. ! queue ! > ffenc_wmav1 ! amux. asfmux name=amux ! rtpasfpay ! multifdsink > name=multifdsink", > ? ? ? ? ? ? ? ? ? ? ? ? width, height, fps, bitWidth, bitWidth, audioSampleRate); > > I think I would need so use gstrtpbin instead of the multifdsink and get rid > of the muxing? yes, you should plug a gstrtpbin (if you're going to stream over a network), as described in the "Encode and payload H263 video captured from a v4l2src. Encode and payload AMR audio generated from audiotestsrc" example in gstrtpbin. > > "see you're using appsrc and appsink. Usually, the first good > question in such a case is: "do I really need such elements in my > pipeline?". The answer depends on your use case and requirements :). > > Note that if your app is not producing and consuming buffers in the > proper way you may run into troubles. The behaviour will e.g. differ > depending on whether you're in pull or push mode. See the elements' > documentation for more details. " > > I am using the appsinks because this app is using OPAL in other threads to > answer H323 and SIP requests and it needs the raw data buffer. This thread > is used for a client control computer and SDK which will monitor sessions, > and make recordings of the sessions (i.e. a Windows Server 2008 Web Server > using ASP.NET/ C#.net with an SDK I have written to talk to this device.) Do > you have a better approach in mind? well, you may want to use telepathy and farsight/stream engine for this. the "framework": http://telepathy.freedesktop.org/wiki/ the "SIP backend" (well, you also need libsofiasip): http://git.collabora.co.uk/?p=telepathy-sofiasip.git the "streaming backend" (using GStreamer) http://farsight.freedesktop.org/wiki/ After installing the proper plugins, SIP support is great, I can grant it. I've read here and there about H323 support, but I've never tested it personally and don't know where sources can be found (I've heard about a mod_opal but never seen it). The only drawback may be about the portability of the stack across platforms (as it appears you're using MS stuff), so I suggest you to check on the #farsight channel on freenode abot this issue. If the effort is too hard, maybe you solution will be easier to deploy. Btw I've no religious arguments against appsink/appsource and it appear your case may well justify their usage. Regards > > The OPAL thread grabs this buffer when it needs it. > > Thank you once again, > > Jonathan > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Trouble-using-x264enc-with-a-tee-tp3067583p3077032.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ikt011 at gmail.com Wed Dec 8 15:08:28 2010 From: ikt011 at gmail.com (Kocsis Tibor) Date: Wed, 8 Dec 2010 15:08:28 +0100 Subject: [gst-devel] rtspsrc can't connect to network camera Message-ID: Hi, I have a network camera and i wan't to use rtspsrc to watch live, but it can't connect to the camera. The last few lines are the following before I have the "not linked (-1)" error message: 0:00:00.857076440 7086 0x9ab0800 DEBUG rtspsrc gstrtspsrc.c:2197:request_pt_map: getting pt map for pt 100 in session 0 0:00:00.857117205 7086 0x9ab0800 DEBUG rtspsrc gstrtspsrc.c:2197:request_pt_map: getting pt map for pt 111 in session 0 0:00:00.857424306 7086 0x9ad2508 DEBUG rtspsrc gstrtspsrc.c:2129:new_session_pad: got new session pad 0:00:00.857531911 7086 0x9ad2508 DEBUG rtspsrc gstrtspsrc.c:2137:new_session_pad: stream: 0, SSRC 708331394, PT 111 0:00:00.857818625 7086 0x9ad2508 DEBUG rtspsrc gstrtspsrc.c:2161:new_session_pad: stream 0x9aa7b48, container 0, disabled 0, added 1 0:00:00.857919871 7086 0x9ad2508 DEBUG rtspsrc gstrtspsrc.c:2161:new_session_pad: stream 0x9a9b310, container 0, disabled 0, added 0 WARNING: from element /GstPipeline:pipeline0/GstFakeSink:fakesink0: Internal data flow problem. Additional debug info: gstbasesink.c(3446): gst_base_sink_chain_unlocked (): /GstPipeline:pipeline0/GstFakeSink:fakesink0: Received buffer without a new-segment. Assuming timestamps start from 0. 0:00:00.858316430 7086 0x9a9ed78 DEBUG rtspsrc gstrtspsrc.c:1991:gst_rtspsrc_handle_src_query: pad rtspsrc0:recv_rtp_src_0_708331394_111 received query latency 0:00:00.858416389 7086 0x9a9ed78 DEBUG rtspsrc gstrtspsrc.c:1875:gst_rtspsrc_handle_src_event: pad rtspsrc0:recv_rtp_src_0_708331394_111 received event latency 0:00:02.856541066 7086 0x9ad2508 DEBUG rtspsrc gstrtspsrc.c:2129:new_session_pad: got new session pad 0:00:02.856568323 7086 0x9ad2508 DEBUG rtspsrc gstrtspsrc.c:2137:new_session_pad: stream: 0, SSRC 708331394, PT 100 0:00:02.856657855 7086 0x9ad2508 DEBUG rtspsrc gstrtspsrc.c:2161:new_session_pad: stream 0x9aa7b48, container 0, disabled 0, added 1 0:00:02.856669701 7086 0x9ad2508 DEBUG rtspsrc gstrtspsrc.c:2161:new_session_pad: stream 0x9a9b310, container 0, disabled 0, added 0 0:00:02.857156131 7086 0x9ab0800 DEBUG rtspsrc gstrtspsrc.c:6060:gst_rtspsrc_handle_message: got error from udpsrc0 0:00:02.857172483 7086 0x9ab0800 DEBUG rtspsrc gstrtspsrc.c:6074:gst_rtspsrc_handle_message: combined flows: not-linked ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0: Internal data flow error. Additional debug info: gstbasesrc.c(2563): gst_base_src_loop (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0: streaming task paused, reason not-linked (-1) Execution ended after 2704644757 n VLC and mplayer can work with the camera, only gstreamer fails. It is possible to have some problem with the "PT 100"? For h264 the payload is usually 96, but this camera sends the following sdp back: medias: media 0: media: 'video' port: '0' num_ports: '4294967295' proto: 'RTP/AVP' formats: format '100' information: '(NULL)' bandwidths: type: 'AS' bandwidth: '1800' key: type: '(NULL)' data: '(NULL)' attributes: attribute 'framerate' : '12.5' attribute 'quality' : '8' attribute 'control' : 'trackID=1' attribute 'rtpmap' : '100 H264/90000' attribute 'fmtp' : '100 profile-level-id=42e01f; Reso=320:240; Scanning=0; TVSystem=pal; CameraMode=standard; BitRateMode=vbr; Gop=30; AspectRatio=4:3; packetization-mode=1; sprop-parameter-sets=Z0LwFJGwUH7AW4KAgKAAAH0gAAw1EIAAAAAAAAA=,aM44gA==' media 1: media: 'data' port: '0' num_ports: '4294967295' proto: 'RTP/AVP' formats: format '111' information: '(NULL)' key: type: '(NULL)' data: '(NULL)' attributes: attribute 'rtpmap' : '111 octet-stream/1' attribute 'fmtp' : '111 Mac=08007b889ebe; Model=VCC-HD2300P; TargetBitRate=900; FirmVer=010103; CameraSeries=2;' Any ideas? Thanks Tibor From gibrovacco at gmail.com Wed Dec 8 15:52:13 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 8 Dec 2010 16:52:13 +0200 Subject: [gst-devel] rtspsrc can't connect to network camera In-Reply-To: References: Message-ID: Hi, On Wed, Dec 8, 2010 at 4:08 PM, Kocsis Tibor wrote: > Hi, > > I have a network camera and i wan't to use rtspsrc to watch live, but > it can't connect to the camera. The last few lines are the following > before I have the "not linked (-1)" error message: it appears some of the elements in the pipeline are not negotiating proper caps. Maybe you're just missing an ffmpegcolorspace element between the video decoder and the sink, maybe the pipeline is completely wrong or maybe you're missing a plugin, but you should get an explicit message from gst-launch in such a case -if you're using gst-launch-. You can understand more about caps negotiation adding the -v command-line option to gst-launch. ..snip.. > > VLC and mplayer can work with the camera, only gstreamer fails. It is > possible to have some problem with the "PT 100"? For h264 the payload all the dynamic payload types are supported as specified in rfc1890, table 2. The value "100" does not have any particular meaning wrt that. > is usually 96, but this camera sends the following sdp back: > > ?medias: > ?media 0: > ? media: ? ? ? 'video' > ? port: ? ? ? ?'0' > ? num_ports: ? '4294967295' > ? proto: ? ? ? 'RTP/AVP' > ? formats: > ? ?format ?'100' > ? information: '(NULL)' > ? bandwidths: > ? ?type: ? ? ? ? 'AS' > ? ?bandwidth: ? ?'1800' > ? key: > ? ?type: ? ? ? '(NULL)' > ? ?data: ? ? ? '(NULL)' > ? attributes: > ? ?attribute 'framerate' : '12.5' > ? ?attribute 'quality' : '8' > ? ?attribute 'control' : 'trackID=1' > ? ?attribute 'rtpmap' : '100 H264/90000' > ? ?attribute 'fmtp' : '100 profile-level-id=42e01f; Reso=320:240; > Scanning=0; TVSystem=pal; CameraMode=standard; BitRateMode=vbr; > Gop=30; AspectRatio=4:3; packetization-mode=1; > sprop-parameter-sets=Z0LwFJGwUH7AW4KAgKAAAH0gAAw1EIAAAAAAAAA=,aM44gA==' > ?media 1: > ? media: ? ? ? 'data' > ? port: ? ? ? ?'0' > ? num_ports: ? '4294967295' > ? proto: ? ? ? 'RTP/AVP' > ? formats: > ? ?format ?'111' > ? information: '(NULL)' > ? key: > ? ?type: ? ? ? '(NULL)' > ? ?data: ? ? ? '(NULL)' > ? attributes: > ? ?attribute 'rtpmap' : '111 octet-stream/1' > ? ?attribute 'fmtp' : '111 Mac=08007b889ebe; Model=VCC-HD2300P; > TargetBitRate=900; FirmVer=010103; CameraSeries=2;' Nothing awfully wrong here imo. Can you post the pipeline you're using? Have you tried with a simple: gst-launch playbin2 uri=rtsp://camera_address ? Regards > > > Any ideas? > > Thanks > Tibor > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From gibrovacco at gmail.com Wed Dec 8 16:04:52 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 8 Dec 2010 17:04:52 +0200 Subject: [gst-devel] What plugin elements to implement for a multi input frame/audio grabber ? In-Reply-To: <1291743578959-3076865.post@n4.nabble.com> References: <1291743578959-3076865.post@n4.nabble.com> Message-ID: Hi, On Tue, Dec 7, 2010 at 7:39 PM, goussepi wrote: > > Hello, > > I am looking at implemeting a plugin for a multi inputs audio and frame > grabber (eg 8 composite inputs and 8 audio analogue inputs). > > It would have to allow each input to have different pipelines connected to > it. > > The hardware is a single device with 8 mmaped video buffer and 8 mmaped > audio buffers which all share the same configuration setup when opening the > device. > > Should i look at implementing a source plugin with 1 audio and 1 video > source pad which will randomly return a buffer from each 8 channel, but then > how do i redirect it to the appropriate pipeline ? > > Or instanciate 8 source elements for video and 8 source elements for audio, > but only 1 element initialise the hardware which seems ugly ? Just an idea: you may implement your own custom element which initialises the HW and with 8 source pads. If your HW has a driver with a standard interface (e.g. v4l) you may check whether a gstreamer source exists supporting it. See here for more details: http://www.gstreamer.net/data/doc/gstreamer/head/pwg/html/section-base-src.html Regards > > Regards, > > Pierre > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-plugin-elements-to-implement-for-a-multi-input-frame-audio-grabber-tp3076865p3076865.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From wb666greene at gmail.com Wed Dec 8 16:56:57 2010 From: wb666greene at gmail.com (wally_bkg) Date: Wed, 8 Dec 2010 07:56:57 -0800 (PST) Subject: [gst-devel] Tell me about y4menc and other encoders. Followup In-Reply-To: <1291413235446-3071830.post@n4.nabble.com> References: <1291413235446-3071830.post@n4.nabble.com> Message-ID: <1291823817245-3078454.post@n4.nabble.com> Just to close the thread in case someone else has the same issue, the motion jpeg avi issue is solved by adding a -e to the gst-launch command so a Ctrl-C to stop the recording sends an EOS to the elements (gleaned from another thread) which seems to cause the avimux to do something to the file that is needed on Windows, but not on Linux (at least if playing back with gstreamer via Totem). I updated my K-lite Codecs on Windows but the theoraenc->oggmux ogv file still doesn't play beyond the first few frames. Adding the -e made no difference, although either ogv file plays on Windows using vlc-win32-1.1.5. The ogg/theora video example files distributed with Ubuntu all play fine on my windows system so it seems something is different between the theoraenc->oggmux file and the Ubuntu distributed samples that vlc understands, but breaks the most recent K-lite theora/ogg codec. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Tell-me-about-y4menc-and-other-encoders-tp3071830p3078454.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From rafael.lmsousa at gmail.com Wed Dec 8 18:04:11 2010 From: rafael.lmsousa at gmail.com (Rafael Sousa) Date: Wed, 8 Dec 2010 13:04:11 -0400 Subject: [gst-devel] problem with h.264 stream (blurring video) Message-ID: Hi all, I'm using the lastest versions of gstreamer installed from git repository. I've installed the gstreamer, gst-plugins-bad, gst-plugins-good, gst-plugins-ugly, gst-ffmpeg and I've installed the lastest build of x264 too from git. The problem is that now my pipeline gives as result a blurring video, that before didn't occur. I don't know what to do, anyone can help? My pipeline works to read a YUV file at the sender, encodes it in H.264, and sends it to receiver using RTP in encoding time. At the receiver, it receives the stream and stores it in YUV format again for posterior process. The pipeline is as follows: Sender: "gstrtpbin name=rtpbin filesrc location=%s blocksize=%d ! video/x-raw-yuv, width=%d, height=%d, framerate=(fraction)15/1, format=(fourcc)I420 ! x264enc, key-int-max=30, key-int-min=15, intra-refresh=true ! rtph264pay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink host=%s port=5000 rtpbin.send_rtcp_src_0 ! udpsink host=%s port=5001 sync=false async=false udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 " Receiver: "gstrtpbin name=rtpbin udpsrc caps=\"application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" port=5000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! ffdec_h264 ! filesink location=%s udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false" I don't know why the video is blurring since I've tested it localhost, without loss rates. thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From bossart.nospam at gmail.com Wed Dec 8 18:13:45 2010 From: bossart.nospam at gmail.com (pl bossart) Date: Wed, 8 Dec 2010 11:13:45 -0600 Subject: [gst-devel] X-lpcm cap and format? Message-ID: Maybe it's a stupid question but does anyone know the difference between audio/x-lpcm and audio/x-raw-int? I am trying to understand what mpegtsmux does, and was wondering what this x-lpcm cap on the sink pad is. The only explanation I could think of was lpcm is the wording used for DVD audio program streams, is there some kind of implicit link here? Can't we mux regular PCM into an MPEG transport stream? Thanks, -Pierre From ikt011 at gmail.com Wed Dec 8 18:49:32 2010 From: ikt011 at gmail.com (Kocsis Tibor) Date: Wed, 8 Dec 2010 18:49:32 +0100 Subject: [gst-devel] rtspsrc can't connect to network camera In-Reply-To: References: Message-ID: Hi, I used a very simple pipeline: gst-launch rtspsrc location="..." ! fakesink sync=true As far as I know this means that the negotiation problem is at one of the rtspsrc's elements: 0:00:02.857172483 7086 0x9ab0800 DEBUG rtspsrc gstrtspsrc.c:6074:gst_rtspsrc_handle_message: combined flows: not-linked ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0: Internal data flow error. Additional debug info: gstbasesrc.c(2563): gst_base_src_loop (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0: streaming task paused, reason not-linked (-1) I'll make a test tomorrow with playbin2, but I guess the result will be the same as above. Regards Tibor On Wed, Dec 8, 2010 at 3:52 PM, Marco Ballesio wrote: > Hi, > > On Wed, Dec 8, 2010 at 4:08 PM, Kocsis Tibor wrote: >> Hi, >> >> I have a network camera and i wan't to use rtspsrc to watch live, but >> it can't connect to the camera. The last few lines are the following >> before I have the "not linked (-1)" error message: > > it appears some of the elements in the pipeline are not negotiating > proper caps. Maybe you're just missing an ffmpegcolorspace element > between the video decoder and the sink, maybe the pipeline is > completely wrong or maybe you're missing a plugin, but you should get > an explicit message from gst-launch in such a case -if you're using > gst-launch-. > > You can understand more about caps negotiation adding the -v > command-line option to gst-launch. > > ..snip.. > >> >> VLC and mplayer can work with the camera, only gstreamer fails. It is >> possible to have some problem with the "PT 100"? For h264 the payload > > all the dynamic payload types are supported as specified in rfc1890, > table 2. The value "100" does not have any particular meaning wrt > that. > >> is usually 96, but this camera sends the following sdp back: >> >> ?medias: >> ?media 0: >> ? media: ? ? ? 'video' >> ? port: ? ? ? ?'0' >> ? num_ports: ? '4294967295' >> ? proto: ? ? ? 'RTP/AVP' >> ? formats: >> ? ?format ?'100' >> ? information: '(NULL)' >> ? bandwidths: >> ? ?type: ? ? ? ? 'AS' >> ? ?bandwidth: ? ?'1800' >> ? key: >> ? ?type: ? ? ? '(NULL)' >> ? ?data: ? ? ? '(NULL)' >> ? attributes: >> ? ?attribute 'framerate' : '12.5' >> ? ?attribute 'quality' : '8' >> ? ?attribute 'control' : 'trackID=1' >> ? ?attribute 'rtpmap' : '100 H264/90000' >> ? ?attribute 'fmtp' : '100 profile-level-id=42e01f; Reso=320:240; >> Scanning=0; TVSystem=pal; CameraMode=standard; BitRateMode=vbr; >> Gop=30; AspectRatio=4:3; packetization-mode=1; >> sprop-parameter-sets=Z0LwFJGwUH7AW4KAgKAAAH0gAAw1EIAAAAAAAAA=,aM44gA==' >> ?media 1: >> ? media: ? ? ? 'data' >> ? port: ? ? ? ?'0' >> ? num_ports: ? '4294967295' >> ? proto: ? ? ? 'RTP/AVP' >> ? formats: >> ? ?format ?'111' >> ? information: '(NULL)' >> ? key: >> ? ?type: ? ? ? '(NULL)' >> ? ?data: ? ? ? '(NULL)' >> ? attributes: >> ? ?attribute 'rtpmap' : '111 octet-stream/1' >> ? ?attribute 'fmtp' : '111 Mac=08007b889ebe; Model=VCC-HD2300P; >> TargetBitRate=900; FirmVer=010103; CameraSeries=2;' > > Nothing awfully wrong here imo. Can you post the pipeline you're > using? Have you tried with a simple: > > gst-launch playbin2 uri=rtsp://camera_address > > ? > > Regards > >> >> >> Any ideas? >> >> Thanks >> Tibor >> >> ------------------------------------------------------------------------------ >> What happens now with your Lotus Notes apps - do you make another costly >> upgrade, or settle for being marooned without product support? Time to move >> off Lotus Notes and onto the cloud with Force.com, apps are easier to build, >> use, and manage than apps on traditional platforms. Sign up for the Lotus >> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From msmith at xiph.org Wed Dec 8 19:21:17 2010 From: msmith at xiph.org (Michael Smith) Date: Wed, 8 Dec 2010 10:21:17 -0800 Subject: [gst-devel] X-lpcm cap and format? In-Reply-To: References: Message-ID: audio/x-lpcm is DVD-style PCM data - it's arranged differently from normal interleaved PCM samples, and carries some additional metadata usually. It's essentially still raw PCM - just with a more complex interleaving used. dvdlpcmdec can convert it to normal PCM. I don't think we have a corresponding encoder, but the format is fairly trivial, so writing one shouldn't be more than a couple of hours work. Mike On Wed, Dec 8, 2010 at 9:13 AM, pl bossart wrote: > Maybe it's a stupid question but does anyone know the difference > between audio/x-lpcm and audio/x-raw-int? > I am trying to understand what mpegtsmux does, and was wondering what > this x-lpcm cap on the sink pad is. The only explanation I could think > of was lpcm is the wording used for DVD audio program streams, is > there some kind of implicit link here? Can't we mux regular PCM into > an MPEG transport stream? > Thanks, > -Pierre > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ensonic at hora-obscura.de Wed Dec 8 20:08:16 2010 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 08 Dec 2010 21:08:16 +0200 Subject: [gst-devel] Fakesink and GstNavigation In-Reply-To: <4CFF7435.9020000@wanadoo.fr> References: <4CFF7435.9020000@wanadoo.fr> Message-ID: <4CFFD7A0.9050201@hora-obscura.de> Am 08.12.2010 14:04, schrieb Nicolas: > Hello, > > I created a "playbin2" and a "fakesink" (I render the video in an evas > image) > Now my problem is that "fakesink" has no "GstNavigation". > How can I do to implement the "GstNavigation" to "Fakesink" ? That is the wrong approach. You rather want to implement a evasvideosink and implement the navigation iface there. Stefan > > Thank you in advance > Sincerely, > Nicolas. > From thiago.sousa.santos at collabora.co.uk Wed Dec 8 20:13:53 2010 From: thiago.sousa.santos at collabora.co.uk (Thiago Sousa Santos) Date: Wed, 08 Dec 2010 16:13:53 -0300 Subject: [gst-devel] camerabin2 is now on -bad Message-ID: <1291835633.16533.15.camel@blacksheep> Hello, for those not on the commits mailing list, camerabin2 prototype is now on the gst-plugins-bad module on Freedesktop (official repository). It is marked as experimental and in order to build it and its tests/example you need to pass '--enable-experimental' to configure. There is still lots of work to do, if you'd like to contribute you can put patches on https://bugzilla.gnome.org/ . In case you don't know exactly what/where to help, ping me (thiagoss) on #gstreamer and I can show a TODO list. The more people we can gather developing it from the start, the better code gets and we make sure it will cover more use cases from different applications. If you prefer to use gitorious, I'll keep my 'work in progress' branch on http://gitorious.org/gstreamer-camerabin2/gst-plugins-bad , so feel free to fork from there and send pull requests. I'll review from there and then push upstream. Regarding the meeting we had yesterday, I've uploaded the logs here: http://people.collabora.co.uk/~thiagoss/camerabin2_20101207.log -- Thiago From gibrovacco at gmail.com Wed Dec 8 21:02:12 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 8 Dec 2010 22:02:12 +0200 Subject: [gst-devel] rtspsrc can't connect to network camera In-Reply-To: References: Message-ID: On 12/8/10, Kocsis Tibor wrote: > Hi, > > I used a very simple pipeline: > gst-launch rtspsrc location="..." ! fakesink sync=true > > As far as I know this means that the negotiation problem is at one of > the rtspsrc's elements: that may not be true. A caps negotiation issue may trigger an error from the source element as the one you're seeing. I suggest you to add the -v option o gst-launch in order to see how the negotiations occur. Regards > > 0:00:02.857172483 7086 0x9ab0800 DEBUG rtspsrc > gstrtspsrc.c:6074:gst_rtspsrc_handle_message: combined > flows: not-linked > ERROR: from element > /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0: Internal > data flow error. > Additional debug info: > gstbasesrc.c(2563): gst_base_src_loop (): > /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0: > streaming task paused, reason not-linked (-1) > > I'll make a test tomorrow with playbin2, but I guess the result will > be the same as above. > > Regards > Tibor > > > > On Wed, Dec 8, 2010 at 3:52 PM, Marco Ballesio wrote: >> Hi, >> >> On Wed, Dec 8, 2010 at 4:08 PM, Kocsis Tibor wrote: >>> Hi, >>> >>> I have a network camera and i wan't to use rtspsrc to watch live, but >>> it can't connect to the camera. The last few lines are the following >>> before I have the "not linked (-1)" error message: >> >> it appears some of the elements in the pipeline are not negotiating >> proper caps. Maybe you're just missing an ffmpegcolorspace element >> between the video decoder and the sink, maybe the pipeline is >> completely wrong or maybe you're missing a plugin, but you should get >> an explicit message from gst-launch in such a case -if you're using >> gst-launch-. >> >> You can understand more about caps negotiation adding the -v >> command-line option to gst-launch. >> >> ..snip.. >> >>> >>> VLC and mplayer can work with the camera, only gstreamer fails. It is >>> possible to have some problem with the "PT 100"? For h264 the payload >> >> all the dynamic payload types are supported as specified in rfc1890, >> table 2. The value "100" does not have any particular meaning wrt >> that. >> >>> is usually 96, but this camera sends the following sdp back: >>> >>> medias: >>> media 0: >>> media: 'video' >>> port: '0' >>> num_ports: '4294967295' >>> proto: 'RTP/AVP' >>> formats: >>> format '100' >>> information: '(NULL)' >>> bandwidths: >>> type: 'AS' >>> bandwidth: '1800' >>> key: >>> type: '(NULL)' >>> data: '(NULL)' >>> attributes: >>> attribute 'framerate' : '12.5' >>> attribute 'quality' : '8' >>> attribute 'control' : 'trackID=1' >>> attribute 'rtpmap' : '100 H264/90000' >>> attribute 'fmtp' : '100 profile-level-id=42e01f; Reso=320:240; >>> Scanning=0; TVSystem=pal; CameraMode=standard; BitRateMode=vbr; >>> Gop=30; AspectRatio=4:3; packetization-mode=1; >>> sprop-parameter-sets=Z0LwFJGwUH7AW4KAgKAAAH0gAAw1EIAAAAAAAAA=,aM44gA==' >>> media 1: >>> media: 'data' >>> port: '0' >>> num_ports: '4294967295' >>> proto: 'RTP/AVP' >>> formats: >>> format '111' >>> information: '(NULL)' >>> key: >>> type: '(NULL)' >>> data: '(NULL)' >>> attributes: >>> attribute 'rtpmap' : '111 octet-stream/1' >>> attribute 'fmtp' : '111 Mac=08007b889ebe; Model=VCC-HD2300P; >>> TargetBitRate=900; FirmVer=010103; CameraSeries=2;' >> >> Nothing awfully wrong here imo. Can you post the pipeline you're >> using? Have you tried with a simple: >> >> gst-launch playbin2 uri=rtsp://camera_address >> >> ? >> >> Regards >> >>> >>> >>> Any ideas? >>> >>> Thanks >>> Tibor >>> >>> ------------------------------------------------------------------------------ >>> What happens now with your Lotus Notes apps - do you make another costly >>> upgrade, or settle for being marooned without product support? Time to >>> move >>> off Lotus Notes and onto the cloud with Force.com, apps are easier to >>> build, >>> use, and manage than apps on traditional platforms. Sign up for the Lotus >>> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >> >> ------------------------------------------------------------------------------ >> What happens now with your Lotus Notes apps - do you make another costly >> upgrade, or settle for being marooned without product support? Time to >> move >> off Lotus Notes and onto the cloud with Force.com, apps are easier to >> build, >> use, and manage than apps on traditional platforms. Sign up for the Lotus >> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From n770galaxy at gmail.com Wed Dec 8 21:30:56 2010 From: n770galaxy at gmail.com (Josep Torra) Date: Wed, 08 Dec 2010 21:30:56 +0100 Subject: [gst-devel] Problem with flacdec and ismd_audio_sink In-Reply-To: <1291800909157-3077927.post@n4.nabble.com> References: <1291718951876-3076140.post@n4.nabble.com> <1291735895.15436.15.camel@zingle> <1291800909157-3077927.post@n4.nabble.com> Message-ID: <4CFFEB00.8060303@gmail.com> On 12/8/10 10:35 AM, Raymond Frost wrote: > > > Tim-Philipp M?ller-2 wrote: >> >> On Tue, 2010-12-07 at 02:49 -0800, Raymond Frost wrote: >> >>> Im working on a intel CE4150 board with gstreamer >>> and i have issues of extremely low volume whenever i use flacdec along >>> with >>> ismd_audio_sink. Basically my setup is >>> >>> filesrc ! queue ! flacdec ! audioconvert ! ismd_audio_sink >>> >>> i have no issues when using other combinations such as (snip) >> >> So what format is being negotiated between flacdec and your sink? (pass >> -v to gst-launch) >> >> What if you use e.g. >> >> ... ! flacdec ! audioconvert ! \ >> audio/x-raw-int,width=16,depth=16,channels=2 ! \ >> ismd_audio_sink >> >> for example (try other combinations too, maybe the same as are used with >> the other formats). >> >> Cheers >> -Tim >> >> > > well, > i have been trying something similar before, > it goes something like the codes below, > *note that I'm using C > > audio_caps = gst_caps_new_simple("audio/x-raw-int", > "width", G_TYPE_INT, 16, > "rate", G_TYPE_INT, 48000, > "channels", G_TYPE_INT, 2, > "endianness", G_TYPE_INT, 1234, > "depth", G_TYPE_INT, 24, > "signed", G_TYPE_BOOLEAN, true, > NULL); > g_assert(audio_caps); > gst_pad_set_caps(audioconvert_src_pad, audio_caps); > gst_caps_unref(audio_caps); > > > > I run this before linking up the pipeline, > and so far i have tried numerous setting combinations, > so far i have no positive improvements yet. > > so far only by replacing the audiosink to alsasink, > i can get good volume with the same setting on all the other part of the > pipeline. > i guess this part of setting could be the only connection to this problem. > Having depth > width doesn't make sense. Try with 16 bits on both. Cheers Josep From radivojejovanovic at gmail.com Wed Dec 8 21:53:59 2010 From: radivojejovanovic at gmail.com (Radivoje Jovanovic) Date: Wed, 8 Dec 2010 12:53:59 -0800 Subject: [gst-devel] GST_FLOW_WRONG_STATE Message-ID: Hello, I have a video sink and ALSA driver to be used to play videos on the platform. If I play video or audio by them self everything works fine, but if I specify the whole pipeline for video and audio, the video decoder (in pipeline right before the sink) comes back with the message: "The decoded frame did not successfully push out to downstream element" The message comes because the gst_pad_push have returned with GST_FLOW_WRONG_STATE instead GST_FLOW_OK. Any idea why this might happen? -------------- next part -------------- An HTML attachment was scrubbed... URL: From gibrovacco at gmail.com Wed Dec 8 22:38:31 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 8 Dec 2010 23:38:31 +0200 Subject: [gst-devel] mmap support on gstalsasink In-Reply-To: References: Message-ID: Hi, On Tue, Dec 7, 2010 at 1:34 AM, Hector Barajas wrote: > Hi everyone, > Im trying to run and audio device with mmap support, I already did that with > pulseaudio but it adds overheads and therefore I still need to reuse the cpu > consumption. > > By looking at the cvs repository it does look like mmap was an option in > gstreamer several years ago, and it disappeared in version 1.40 of > gstalsasink.c. Does anyone know why this support was removed ? And how > difficult it would be to add it again ? > interesting question, imo worth a discussion. Digging in gst-plugins-base, mmap support for ALSA has been removed with commit: 851547e3212a49fd60e86cd8707e06ecbaed22ee With the rationale "Implement alsasink with simple open/write/close API". As, on one hand, I can understand the improved simplicity of the element, I think on the other that we've lost a premium feature especially for embedded devices where, for one reason or another, we want to reduce the memory footprint and/or latencies to the bare bone but we want to still use GStreamer. Does anybody agree with me? Is this the case to file a bug/enhancement request or do we have a replacement element hidden somewhere (I didn't find any)? Regards > Thanks for your feedback. > > Regards, > Hector Barajas > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From pgousseau at hotmail.com Thu Dec 9 00:26:24 2010 From: pgousseau at hotmail.com (goussepi) Date: Wed, 8 Dec 2010 15:26:24 -0800 (PST) Subject: [gst-devel] What plugin elements to implement for a multi input frame/audio grabber ? In-Reply-To: References: <1291743578959-3076865.post@n4.nabble.com> Message-ID: <1291850784206-3079324.post@n4.nabble.com> Marco Ballesio wrote: > > Hi, > > On Tue, Dec 7, 2010 at 7:39 PM, goussepi wrote: >> >> Hello, >> >> I am looking at implemeting a plugin for a multi inputs audio and frame >> grabber (eg 8 composite inputs and 8 audio analogue inputs). >> >> It would have to allow each input to have different pipelines connected >> to >> it. >> >> The hardware is a single device with 8 mmaped video buffer and 8 mmaped >> audio buffers which all share the same configuration setup when opening >> the >> device. >> >> Should i look at implementing a source plugin with 1 audio and 1 video >> source pad which will randomly return a buffer from each 8 channel, but >> then >> how do i redirect it to the appropriate pipeline ? >> >> Or instanciate 8 source elements for video and 8 source elements for >> audio, >> but only 1 element initialise the hardware which seems ugly ? > > Just an idea: you may implement your own custom element which > initialises the HW and with 8 source pads. If your HW has a driver > with a standard interface (e.g. v4l) you may check whether a gstreamer > source exists supporting it. > > See here for more details: > http://www.gstreamer.net/data/doc/gstreamer/head/pwg/html/section-base-src.html > > Regards > >> >> Regards, >> >> Pierre >> -- >> View this message in context: >> http://gstreamer-devel.966125.n4.nabble.com/What-plugin-elements-to-implement-for-a-multi-input-frame-audio-grabber-tp3076865p3076865.html >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> >> ------------------------------------------------------------------------------ >> What happens now with your Lotus Notes apps - do you make another costly >> upgrade, or settle for being marooned without product support? Time to >> move >> off Lotus Notes and onto the cloud with Force.com, apps are easier to >> build, >> use, and manage than apps on traditional platforms. Sign up for the Lotus >> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to > move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > Thanks i ll experiment with that, and is it possible to connect/disconnect different pipelines to each source pads of such an element ? And doing it dynamically so a user can decide to change the pipeline connected to one of the source pad while the others pipelines keep running ? Regards, Pierre -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-plugin-elements-to-implement-for-a-multi-input-frame-audio-grabber-tp3076865p3079324.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Thu Dec 9 01:05:43 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 09 Dec 2010 00:05:43 +0000 Subject: [gst-devel] GST_FLOW_WRONG_STATE In-Reply-To: References: Message-ID: <1291853143.3531.43.camel@zingle> On Wed, 2010-12-08 at 12:53 -0800, Radivoje Jovanovic wrote: > I have a video sink and ALSA driver to be used to play videos on the > platform. If I play video or audio by them self everything works fine, > but if I specify the whole pipeline for video and audio, the video > decoder (in pipeline right before the sink) comes back with the > message: > "The decoded frame did not successfully push out to downstream > element" > The message comes because the gst_pad_push have returned with > GST_FLOW_WRONG_STATE instead GST_FLOW_OK. > > Any idea why this might happen? It usually helps if you post your exact pipeline and/or any code snippets that go with it. FLOW_WRONG_STATE is what you get when a pad is flushing, which may be normal (happens during a flushing seek, to make the old streaming thread stop) or because you forgot to set an element into PAUSED/PLAYING state (e.g. because you added it from a pad-added or new-decoded-pad callback or so). The GST_DEBUG=*:5 log might give you more information (just grep for wrong-state and read the lines before that). Cheers -Tim From radivojejovanovic at gmail.com Thu Dec 9 01:56:03 2010 From: radivojejovanovic at gmail.com (Radivoje Jovanovic) Date: Wed, 8 Dec 2010 16:56:03 -0800 Subject: [gst-devel] GST_FLOW_WRONG_STATE In-Reply-To: <1291853143.3531.43.camel@zingle> References: <1291853143.3531.43.camel@zingle> Message-ID: So the pipeline I am using is: gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue ! mpeg4dec ! MY_VIDEO_SINK t. ! queue ! faad ! alsasink This pipeline will hang and after CTRL-C the mpeg4dec will show the message that is the result of calling gst_push_pad. This gst_push_pad will return GST_FLOW_WRONG_STATE. If I run gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue ! mpeg4dec ! MY_VIDEO_SINK the video will play just fine If I run: gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue ! faad ! alsasink the audio from the audio will play just fine. I am not sure which code would be helpful since I have a lot of it. Thanks for the GST_DEBUG info. Ogi On Wed, Dec 8, 2010 at 4:05 PM, Tim-Philipp M?ller wrote: > On Wed, 2010-12-08 at 12:53 -0800, Radivoje Jovanovic wrote: > > > I have a video sink and ALSA driver to be used to play videos on the > > platform. If I play video or audio by them self everything works fine, > > but if I specify the whole pipeline for video and audio, the video > > decoder (in pipeline right before the sink) comes back with the > > message: > > "The decoded frame did not successfully push out to downstream > > element" > > The message comes because the gst_pad_push have returned with > > GST_FLOW_WRONG_STATE instead GST_FLOW_OK. > > > > Any idea why this might happen? > > It usually helps if you post your exact pipeline and/or any code > snippets that go with it. > > FLOW_WRONG_STATE is what you get when a pad is flushing, which may be > normal (happens during a flushing seek, to make the old streaming thread > stop) or because you forgot to set an element into PAUSED/PLAYING state > (e.g. because you added it from a pad-added or new-decoded-pad callback > or so). > > The GST_DEBUG=*:5 log might give you more information (just grep for > wrong-state and read the lines before that). > > Cheers > -Tim > > > > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ds at entropywave.com Thu Dec 9 02:02:09 2010 From: ds at entropywave.com (David Schleef) Date: Wed, 8 Dec 2010 20:02:09 -0500 Subject: [gst-devel] mmap support on gstalsasink In-Reply-To: References: Message-ID: <20101209010209.GA22402@cooker.entropywave.com> On Wed, Dec 08, 2010 at 11:38:31PM +0200, Marco Ballesio wrote: > Hi, > > On Tue, Dec 7, 2010 at 1:34 AM, Hector Barajas wrote: > > Hi everyone, > > Im trying to run and audio device with mmap support, I already did that with > > pulseaudio but it adds overheads and therefore I still need to reuse the cpu > > consumption. > > > > By looking at the cvs repository it does look like mmap was an option in > > gstreamer several years ago, and it disappeared in version 1.40 of > > gstalsasink.c. Does anyone know why this support was removed ? And how > > difficult it would be to add it again ? > > > > interesting question, imo worth a discussion. Digging in > gst-plugins-base, mmap support for ALSA has been removed with commit: > > 851547e3212a49fd60e86cd8707e06ecbaed22ee > > With the rationale "Implement alsasink with simple open/write/close API". > > As, on one hand, I can understand the improved simplicity of the > element, I think on the other that we've lost a premium feature > especially for embedded devices where, for one reason or another, we > want to reduce the memory footprint and/or latencies to the bare bone > but we want to still use GStreamer. > > Does anybody agree with me? Is this the case to file a bug/enhancement > request or do we have a replacement element hidden somewhere (I didn't > find any)? If you want it, it would be better to write a patch than file an enhancement request. Implementing it would require implementing an alsa ring buffer to use in the mmap case, switching alsasink to the GstBaseAudioSink class, and copying over the software ring buffer code from GstAudioSink class for the non-mmap case. And then either a) testing the hell out of it, or b) writing the patches in such a way that it's obvious that no bugs are being introduced for the default case. David From ved.kpl at gmail.com Thu Dec 9 05:14:10 2010 From: ved.kpl at gmail.com (ved kpl) Date: Thu, 9 Dec 2010 09:44:10 +0530 Subject: [gst-devel] GST_FLOW_WRONG_STATE In-Reply-To: References: <1291853143.3531.43.camel@zingle> Message-ID: On Thu, Dec 9, 2010 at 6:26 AM, Radivoje Jovanovic wrote: > So the pipeline I am using is: > gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue? ! > mpeg4dec ! MY_VIDEO_SINK t. ! queue ! faad ! alsasink > This pipeline will hang and after CTRL-C the mpeg4dec will show the message > that is the result of calling gst_push_pad. This gst_push_pad will return > GST_FLOW_WRONG_STATE. > > If I run gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue > ! mpeg4dec ! MY_VIDEO_SINK > the video will play just fine > > If I run: > gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue ! faad ! > alsasink > the audio from the audio will play just fine. > > I am not sure which code would be helpful since I have a lot of it. > Thanks for the GST_DEBUG info. > > Ogi > > > On Wed, Dec 8, 2010 at 4:05 PM, Tim-Philipp M?ller wrote: >> >> On Wed, 2010-12-08 at 12:53 -0800, Radivoje Jovanovic wrote: >> >> > I have a video sink and ALSA driver to be used to play videos on the >> > platform. If I play video or audio by them self everything works fine, >> > but if I specify the whole pipeline for video and audio, the video >> > decoder (in pipeline right before the sink) comes back with the >> > message: >> > "The decoded frame did not successfully push out to downstream >> > element" >> > The message comes because the gst_pad_push have returned with >> > GST_FLOW_WRONG_STATE instead GST_FLOW_OK. >> > >> > Any idea why this might happen? >> >> It usually helps if you post your exact pipeline and/or any code >> snippets that go with it. >> >> FLOW_WRONG_STATE is what you get when a pad is flushing, which may be >> normal (happens during a flushing seek, to make the old streaming thread >> stop) or because you forgot to set an element into PAUSED/PLAYING state >> (e.g. because you added it from a pad-added or new-decoded-pad callback >> or so). >> >> The GST_DEBUG=*:5 log might give you more information (just grep for >> wrong-state and read the lines before that). >> >> Cheers >> ?-Tim >> >> >> >> >> ------------------------------------------------------------------------------ >> This SF Dev2Dev email is sponsored by: >> >> WikiLeaks The End of the Free Internet >> http://p.sf.net/sfu/therealnews-com >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From ved.kpl at gmail.com Thu Dec 9 05:27:42 2010 From: ved.kpl at gmail.com (ved kpl) Date: Thu, 9 Dec 2010 09:57:42 +0530 Subject: [gst-devel] GST_FLOW_WRONG_STATE In-Reply-To: References: <1291853143.3531.43.camel@zingle> Message-ID: Hi, One possible reason could be that one of the queues is getting full while the other remains empty because of (a) continuous audio/video buffers for a considerably longer time, longer than the what the queues can hold. (b) high latency at the your video decoder (mpeg4dec). Hence the streaming thread is blocked & one of the sinks is not able to commit the state to PAUSED. (because it has not received a buffer yet). Yo can also set async=FALSE on the sinks and see. (not recommended) You can try disabling the the max time & max bytes properties of the queue. Try the following pipeline gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue max-size-time=0 max-size-bytes=0 ! mpeg4dec ! MY_VIDEO_SINK t. ! queue max-size-time=0 max-size-bytes=0 ! faad ! alsasink -v Note that this will increase the mem usage.. and even more if u set max-size-buffers=0 as well. (queue without any limits!) Vikram On Thu, Dec 9, 2010 at 9:44 AM, ved kpl wrote: > On Thu, Dec 9, 2010 at 6:26 AM, Radivoje Jovanovic > wrote: >> So the pipeline I am using is: >> gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue? ! >> mpeg4dec ! MY_VIDEO_SINK t. ! queue ! faad ! alsasink >> This pipeline will hang and after CTRL-C the mpeg4dec will show the message >> that is the result of calling gst_push_pad. This gst_push_pad will return >> GST_FLOW_WRONG_STATE. >> >> If I run gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue >> ! mpeg4dec ! MY_VIDEO_SINK >> the video will play just fine >> >> If I run: >> gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue ! faad ! >> alsasink >> the audio from the audio will play just fine. >> >> I am not sure which code would be helpful since I have a lot of it. >> Thanks for the GST_DEBUG info. >> >> Ogi >> >> >> On Wed, Dec 8, 2010 at 4:05 PM, Tim-Philipp M?ller wrote: >>> >>> On Wed, 2010-12-08 at 12:53 -0800, Radivoje Jovanovic wrote: >>> >>> > I have a video sink and ALSA driver to be used to play videos on the >>> > platform. If I play video or audio by them self everything works fine, >>> > but if I specify the whole pipeline for video and audio, the video >>> > decoder (in pipeline right before the sink) comes back with the >>> > message: >>> > "The decoded frame did not successfully push out to downstream >>> > element" >>> > The message comes because the gst_pad_push have returned with >>> > GST_FLOW_WRONG_STATE instead GST_FLOW_OK. >>> > >>> > Any idea why this might happen? >>> >>> It usually helps if you post your exact pipeline and/or any code >>> snippets that go with it. >>> >>> FLOW_WRONG_STATE is what you get when a pad is flushing, which may be >>> normal (happens during a flushing seek, to make the old streaming thread >>> stop) or because you forgot to set an element into PAUSED/PLAYING state >>> (e.g. because you added it from a pad-added or new-decoded-pad callback >>> or so). >>> >>> The GST_DEBUG=*:5 log might give you more information (just grep for >>> wrong-state and read the lines before that). >>> >>> Cheers >>> ?-Tim >>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> This SF Dev2Dev email is sponsored by: >>> >>> WikiLeaks The End of the Free Internet >>> http://p.sf.net/sfu/therealnews-com >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> ------------------------------------------------------------------------------ >> This SF Dev2Dev email is sponsored by: >> >> WikiLeaks The End of the Free Internet >> http://p.sf.net/sfu/therealnews-com >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > From rob at ti.com Thu Dec 9 05:28:03 2010 From: rob at ti.com (Clark, Rob) Date: Wed, 8 Dec 2010 22:28:03 -0600 Subject: [gst-devel] 0.11 development In-Reply-To: <1291721927.2178.155.camel@meany> References: <1291721927.2178.155.camel@meany> Message-ID: On Tue, Dec 7, 2010 at 5:38 AM, Wim Taymans wrote: > Hello, > > This mail is just to clarify some rules for the new 0.11 branch: > > 1- Development of plugins and new features continues on the master > ? branch. Everything that can possibly go into 0.10 should go to > ? master first. This can include cleanups that are needed in the 0.11 > ? branch such as removal of deprecated methods. > any opinions on v4l2/v4l2sink patches? We've got xoverlay, navigation, interlaced support (in addition to the rowstride stuff which is only for 0.11). Some of the patches are already in bugzilla. I can freshen up the patches for 0.10 if there is interest, or I can just focus on 0.11. BR, -R From S.Callewaert at TELEVIC.com Thu Dec 9 09:33:53 2010 From: S.Callewaert at TELEVIC.com (Callewaert Sven) Date: Thu, 9 Dec 2010 09:33:53 +0100 Subject: [gst-devel] GStreamer audio streaming on Windows References: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1CB2@SRV-VS06.TELEVIC.COM> Message-ID: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1EF9@SRV-VS06.TELEVIC.COM> The problem of the distortion on the audio stream appears to be linked to the rtpL16pay payloader. When I use the rtpmp4apay payloader the problem disappears. From: Callewaert Sven Sent: dinsdag 7 december 2010 9:53 To: 'Discussion of the development of GStreamer' Subject: RE: [gst-devel] GStreamer audio streaming on Windows Replacing the autoaudiosink element with directsoundsink did the trick. Now I hear a continuous beep but there is a lot of distortion on it. When I stream the microphone by replacing audiotestsrc with directsoundsrc on the sender side, I also hear a lot of distortion. The distortion however only occurs when the microphone captures moderate to loud noises. There is no distortion with quiet noises. Could it be a signed/unsigned issue? Can I force the sender the use signed=false in some way? From: sumit kumar [mailto:sumit.git at gmail.com] Sent: dinsdag 7 december 2010 8:18 To: Discussion of the development of GStreamer Subject: Re: [gst-devel] GStreamer audio streaming on Windows Hi, Try with directsoundsink.. else try setting sync=false for autoaudiosink -Sumit On Tue, Dec 7, 2010 at 3:55 AM, Callewaert Sven > wrote: Hi, I'm experimenting a bit with GStreamer (ossbuild 0.10.7) on Windows, but I can't seem to make audio streaming between two computers work. All I hear at the receiver side is a short beep followed by silence. This is the sender pipeline: gst-launch -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink host=224.0.0.7 auto-multicast=true port=4444 This is the receiver pipeline: gst-launch -v udpsrc multicast-group=224.0.0.7 port=4444 caps="application/x-rtp, media=(string)audio, channels=(int)1, clock-rate=(int)44100, encoding-name=(string)L16" ! gstrtpbin ! rtpL16depay ! audioconvert ! queue ! autoaudiosink I've already tried different queue settings and codecs. Same thing when I try to stream an audio file, all I hear is about 1 second of it. What could be the problem? ------------------------------------------------------------------------------ What happens now with your Lotus Notes apps - do you make another costly upgrade, or settle for being marooned without product support? Time to move off Lotus Notes and onto the cloud with Force.com, apps are easier to build, use, and manage than apps on traditional platforms. Sign up for the Lotus Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From S.Callewaert at TELEVIC.com Thu Dec 9 09:41:41 2010 From: S.Callewaert at TELEVIC.com (Callewaert Sven) Date: Thu, 9 Dec 2010 09:41:41 +0100 Subject: [gst-devel] Distortion when using directsoundsrc References: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1CB2@SRV-VS06.TELEVIC.COM> Message-ID: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1F03@SRV-VS06.TELEVIC.COM> On my Vista computer I hear some distortions on the audio that is captured from my microphone. The distortion is a periodical noisy crack. Playing around with the buffer-time parameter changes the number of cracks, but it doesn't seem possible to make them go away. I use the ossbuild of gstreamer version 0.10.7. On my Windows 7 computer there is no problem. This is the pipeline: gst-launch directsoundsrc ! directsoundsink Is there any way to get rid of the distortion? -------------- next part -------------- An HTML attachment was scrubbed... URL: From gibrovacco at gmail.com Thu Dec 9 10:08:33 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Thu, 9 Dec 2010 11:08:33 +0200 Subject: [gst-devel] What plugin elements to implement for a multi input frame/audio grabber ? In-Reply-To: <1291850784206-3079324.post@n4.nabble.com> References: <1291743578959-3076865.post@n4.nabble.com> <1291850784206-3079324.post@n4.nabble.com> Message-ID: Hi, On Thu, Dec 9, 2010 at 1:26 AM, goussepi wrote: ..snip.. >> >> > > Thanks i ll experiment with that, and is it possible to connect/disconnect > different pipelines to each source pads of such an element ? And doing it > dynamically so a user can decide to change the pipeline connected to one of > the source pad while the others pipelines keep running ? A few notes: - If it's about topologically separated pipelines there shouldn't be any problems. - If you need this because of dynamic changes in the pipeline during signalling phases, maybe the feature is already provided by Farsight. Check, for instance, at: http://git.collabora.co.uk/?p=farsight2.git;a=blob;f=gst-libs/gst/farsight/fs-stream.c;h=128122221e99f6c80c91583e2eb8cf9ed6d3b58f;hb=HEAD#l298 Telepathy Stream Engine (lightweight relative of Farsight) is the place where I got to know how the thing works for sink elements ("valve" approach). - If this still doesn't work, maybe using an input-selector may help you, but I've never tried it, so take it as "Just An Idea (R)". - Discussing it on the Farsight IRC (freenode) may clarify which features are supported (or not). Regards > > Regards, > > Pierre From gurpreet at tataelxsi.co.in Thu Dec 9 10:04:37 2010 From: gurpreet at tataelxsi.co.in (Gurpreet) Date: Thu, 09 Dec 2010 14:34:37 +0530 Subject: [gst-devel] Sequence of calls in Gstreamer Application Message-ID: <4D009BA5.3040803@tataelxsi.co.in> Hi All... I am new to Gstreamer. I am writing a Gstreamer application which will have 1. filesrc 2. Aac Encoder 3. filesink Can anybody tell me what will be the sequence of Calls in Application ( from initilization to end ). Thanks Gurpreet -------------- next part -------------- An HTML attachment was scrubbed... URL: From gibrovacco at gmail.com Thu Dec 9 10:45:35 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Thu, 9 Dec 2010 11:45:35 +0200 Subject: [gst-devel] mmap support on gstalsasink In-Reply-To: <20101209010209.GA22402@cooker.entropywave.com> References: <20101209010209.GA22402@cooker.entropywave.com> Message-ID: Hi, On Thu, Dec 9, 2010 at 3:02 AM, David Schleef wrote: > On Wed, Dec 08, 2010 at 11:38:31PM +0200, Marco Ballesio wrote: >> Hi, >> >> On Tue, Dec 7, 2010 at 1:34 AM, Hector Barajas wrote: >> > Hi everyone, >> > Im trying to run and audio device with mmap support, I already did that with >> > pulseaudio but it adds overheads and therefore I still need to reuse the cpu >> > consumption. >> > >> > By looking at the cvs repository it does look like mmap was an option in >> > gstreamer several years ago, and it disappeared in version 1.40 of >> > gstalsasink.c. Does anyone know why this support was removed ? And how >> > difficult it would be to add it again ? >> > >> >> interesting question, imo worth a discussion. Digging in >> gst-plugins-base, mmap support for ALSA has been removed with commit: >> >> 851547e3212a49fd60e86cd8707e06ecbaed22ee >> >> With the rationale "Implement alsasink with simple open/write/close API". >> >> As, on one hand, I can understand the improved simplicity of the >> element, I think on the other that we've lost a premium feature >> especially for embedded devices where, for one reason or another, we >> want to reduce the memory footprint and/or latencies to the bare bone >> but we want to still use GStreamer. >> >> Does anybody agree with me? Is this the case to file a bug/enhancement >> request or do we have a replacement element hidden somewhere (I didn't >> find any)? > > If you want it, it would be better to write a patch than file an > enhancement request. The intent of my reply is to understand whether the community has interest in bringing back the feature or if the "status quo" is preferred. The tone of the reply makes me understand we're in the second case. Sure I (or another spare developer) could write such a patch, but as long as there's no interest from the community it would be pretty hard to get anything into the official repositories, especially when we're talking about gst-plugins-base. > > Implementing it would require implementing an alsa ring buffer to > use in the mmap case, switching alsasink to the GstBaseAudioSink > class, and copying over the software ring buffer code from > GstAudioSink class for the non-mmap case. If modifying the alsasink is somewhere close to the doomsday, why not to write a separate element? > And then either a) testing > the hell out of it, or b) writing the patches in such a way that > it's obvious that no bugs are being introduced for the default case. or c) adding the new element to gst-plugins bad and let nature follow its own course. Regards > > > > David > > > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From wim.taymans at gmail.com Thu Dec 9 10:58:35 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Thu, 09 Dec 2010 10:58:35 +0100 Subject: [gst-devel] mmap support on gstalsasink In-Reply-To: References: <20101209010209.GA22402@cooker.entropywave.com> Message-ID: <1291888715.1995.9.camel@meany> On Thu, 2010-12-09 at 11:45 +0200, Marco Ballesio wrote: > Hi, > > On Thu, Dec 9, 2010 at 3:02 AM, David Schleef wrote: > > If you want it, it would be better to write a patch than file an > > enhancement request. > > The intent of my reply is to understand whether the community has > interest in bringing back the feature or if the "status quo" is > preferred. The tone of the reply makes me understand we're in the > second case. Most distros and embedded devices go to pulseaudio nowadays. Plain alsa has very few use cases left. One of the reasons mmap support was removed is because it didn't work so well on too many cards back in the day. Now that pulseaudio has tried to shake out most of the bugs it might work a bit better. > > Sure I (or another spare developer) could write such a patch, but as > long as there's no interest from the community it would be pretty hard > to get anything into the official repositories, especially when we're > talking about gst-plugins-base. I guess a good patch would be accepted. > > > > > Implementing it would require implementing an alsa ring buffer to > > use in the mmap case, switching alsasink to the GstBaseAudioSink > > class, and copying over the software ring buffer code from > > GstAudioSink class for the non-mmap case. You can actually just implement the ringbuffer and keep extending from GstAudioSink. The existing code path is not modified when you don't provide a ringbuffer. You would always need the existing code as a fallback because some devices can't be opened in mmap mode at all. > > If modifying the alsasink is somewhere close to the doomsday, why not > to write a separate element? You don't have to. Writing a new element would be painful because then you would need to copy existing alsasink code for the fallback and then you can just as well add the new code to existing alsasink. > > > And then either a) testing > > the hell out of it, or b) writing the patches in such a way that > > it's obvious that no bugs are being introduced for the default case. > > or c) adding the new element to gst-plugins bad and let nature follow > its own course. That would cause more trouble than needed. Wim > > Regards > > > > > > > > > David > > > > > > > > ------------------------------------------------------------------------------ > > This SF Dev2Dev email is sponsored by: > > > > WikiLeaks The End of the Free Internet > > http://p.sf.net/sfu/therealnews-com > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From gibrovacco at gmail.com Thu Dec 9 13:13:28 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Thu, 9 Dec 2010 14:13:28 +0200 Subject: [gst-devel] mmap support on gstalsasink In-Reply-To: <1291888715.1995.9.camel@meany> References: <20101209010209.GA22402@cooker.entropywave.com> <1291888715.1995.9.camel@meany> Message-ID: Hi, thanks for the comments, they definitely give me good hints for the right direction to take. A few words below. On Thu, Dec 9, 2010 at 11:58 AM, Wim Taymans wrote: > On Thu, 2010-12-09 at 11:45 +0200, Marco Ballesio wrote: >> Hi, >> >> On Thu, Dec 9, 2010 at 3:02 AM, David Schleef wrote: > > > >> > If you want it, it would be better to write a patch than file an >> > enhancement request. >> >> The intent of my reply is to understand whether the community has >> interest in bringing back the feature or if the "status quo" is >> preferred. The tone of the reply makes me understand we're in the >> second case. > > Most distros and embedded devices go to pulseaudio nowadays. Plain alsa > has very few use cases left. I would say that some _really_ embedded, single-task devices (classes fairly lower than, e.g., an N900) may still not be able to deal with pulseaudio. Even though we may address them as "niche products", the market is still full of those. > > One of the reasons mmap support was removed is because it didn't work so > well on too many cards back in the day. Now that pulseaudio has tried to > shake out most of the bugs it might work a bit better. > >> >> Sure I (or another spare developer) could write such a patch, but as >> long as there's no interest from the community it would be pretty hard >> to get anything into the official repositories, especially when we're >> talking about gst-plugins-base. > > I guess a good patch would be accepted. "good" is subjective ;). Even if it may sound the same, I would better say "properly reviewed and agreed", where the process involves a discussion on Bugzilla -or wherever else-. Unfortunately or luckily, for such a delicate component certain activities are required from people who could not have time to follow the issue, so this may require a really long time to get the patches accepted (the "hardiness" of above). This is why I'm searching for community involvement before me/whoever else starts with the actual development. Comments from both Wim and David are an excellent starting point in giving me the pulse (thank you guys). > >> >> > >> > Implementing it would require implementing an alsa ring buffer to >> > use in the mmap case, switching alsasink to the GstBaseAudioSink >> > class, and copying over the software ring buffer code from >> > GstAudioSink class for the non-mmap case. > > You can actually just implement the ringbuffer and keep extending from > GstAudioSink. The existing code path is not modified when you don't > provide a ringbuffer. You would always need the existing code as a > fallback because some devices can't be opened in mmap mode at all. > >> >> If modifying the alsasink is somewhere close to the doomsday, ?why not >> to write a separate element? > > You don't have to. Writing a new element would be painful because then > you would need to copy existing alsasink code for the fallback and then > you can just as well add the new code to existing alsasink. Roget that :). Regards > >> >> > And then either a) testing >> > the hell out of it, or b) writing the patches in such a way that >> > it's obvious that no bugs are being introduced for the default case. >> >> or c) adding the new element to gst-plugins bad and let nature follow >> its own course. > > That would cause more trouble than needed. > > Wim > >> >> Regards >> >> > >> > >> > >> > David >> > >> > >> > >> > ------------------------------------------------------------------------------ >> > This SF Dev2Dev email is sponsored by: >> > >> > WikiLeaks The End of the Free Internet >> > http://p.sf.net/sfu/therealnews-com >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> >> ------------------------------------------------------------------------------ >> This SF Dev2Dev email is sponsored by: >> >> WikiLeaks The End of the Free Internet >> http://p.sf.net/sfu/therealnews-com >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From marasu2010 at gmail.com Thu Dec 9 13:54:38 2010 From: marasu2010 at gmail.com (tamil arasu) Date: Thu, 9 Dec 2010 18:24:38 +0530 Subject: [gst-devel] Reducing the time in GST-registry updating and parsing the command line arguments In-Reply-To: References: Message-ID: Hi All, We are using the DM365 target board. I wrote one Audio streaming application.I have attached that source code.This application takes three second to execute.Its taking two second to execute gst_init(&argc,&argv) API ,We used the Libraries mentioned below gst-plugins-good-0.10.14, gst-plugins-base-0.10.23, gstreamer-0.10.23,gst-plugins-bad-0.10.13, liboil ,libxml , libiconv Is it possible to get executed in one second. Can anyone help on this issue regards, Arasu.M On Sun, Nov 28, 2010 at 6:00 PM, Marco Ballesio wrote: > Hi, > > as usual, a late reply. Here anyway my 0.05 ?.. > > On Mon, Nov 22, 2010 at 4:42 PM, tamil arasu wrote: > > HI All, > > > > We are using the gstreamer for capturing the video .we are > > using packages mentioned below > > > > gst-plugins-good-0.10.14, gst-plugins-base-0.10.23, gstreamer-0.10.23, > > gst-plugins-bad-0.10.13 > > > > we are using the pipeline for capturing the video mentioned below > > > > export GST_REGISTRY_UPDATE=no > > export GST_REGISTRY_FORK=no > > > > I don't think it is necessary to set "GST_REGISTRY_FORK=no" in order > to get any improvement. Actually, imo setting this would make things > worse in your case (see below). > > > gst-launch rtspsrc location=rtsp://192.168.1.168:8554/mpeg4 latency=100 > > name=demux demux. ! queue ! rtpmp4vdepay ! mpeg4videoparse ! > dmaidec_mpeg4 ! > > TIDmaiVideoSink videoOutput=LCD videoStd=320X240 demux. ! queue ! > > rtppcmudepay ! mulawdec ! alsasink > > > > When we launch the above mentiond pileline on the command line,we are > able > > to display video on Lcd.The output is > > > > Setting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > DavinciDisplay DavinciDisplay.1: Before finishing with S_FMT: > > layer.pix_fmt.bytesperline = 320, > > layer.pix_fmt.width = 320, > > layer.pix_fmt.height = 240, > > layer.pix_fmt.sizeimage =115200 > > New clock: GstSyDavinciDisplay DavinciDisplay.1: pixfmt->width = 320, > > layer->layer_info.config.line_length= 320 > > stemClock > > > > Once I run the above command ,its taking 2 second to get the first > message > > (Setting pipeline to PAUSED ...). > > one second for update the registry, > > it's strange, as you've set GST_REGISTRY_UPDATE to "no". How did you > understand this latency is caused when updating the registry? > > In gstregistry.c, setting GST_REGISTRY_FORK=no forces the scan helper > state to "REGISTRY_SCAN_HELPER_NOT_STARTED" in opposition to > "REGISTRY_SCAN_HELPER_DISABLED". > > Just to say that, probably, disabling forks you did preempts any > settings in GST_REGISTRY_UPDATE and triggers a registry generation > anyway. > > > anothe one second for parsing the command > > line arguments.(gst_parse_launchv) > > > > you'll get rid of this by building the pipeline "manually" in a custom > application. See here for more help: > > http://www.gstreamer.net/data/doc/gstreamer/head/manual/html/index.html > > > how to reduce the this 2 second.Can anyone help me on this issue? > > See above. Probably some extra-latency is caused by the video decoder > buffering a fixed amount of memory but, as long as your source is in > the local filesystem, its effect would be negligible. > > Regards > > > > > > > thanks and regards, > > > > Arasu.M > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > Beautiful is writing same markup. Internet Explorer 9 supports > > standards for HTML5, CSS3, SVG 1.1, ECMAScript5, and DOM L2 & L3. > > Spend less time writing and rewriting code and more time creating great > > experiences on the web. Be a part of the beta today > > http://p.sf.net/sfu/msIE9-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Audio.c Type: text/x-csrc Size: 2950 bytes Desc: not available URL: From mq_han at hotmail.com Thu Dec 9 14:22:46 2010 From: mq_han at hotmail.com (mingqian Han) Date: Thu, 9 Dec 2010 21:22:46 +0800 Subject: [gst-devel] examples for demux Message-ID: Hi all, I have read the plugin writer's guide, but I still cannot figure out how to write a demux plugin. Is there some examples to get started? Thanks & regards, Rafael -------------- next part -------------- An HTML attachment was scrubbed... URL: From mq_han at hotmail.com Thu Dec 9 14:27:10 2010 From: mq_han at hotmail.com (mingqian Han) Date: Thu, 9 Dec 2010 21:27:10 +0800 Subject: [gst-devel] ways to pause the pipeline, something i am confused. Message-ID: Hi all, Say, I am going to pause the pipeline, in order to do some realtime change of some elements. I can set the pipeline to PAUSED state to achieve it, and I can also block some pads to pause the pipeline from playing. What is the difference? Or limitations? Thanks and regards, Rafael -------------- next part -------------- An HTML attachment was scrubbed... URL: From lijinsyam at gmail.com Thu Dec 9 15:15:11 2010 From: lijinsyam at gmail.com (LIJIN SYAM K) Date: Thu, 9 Dec 2010 19:45:11 +0530 Subject: [gst-devel] Splitting Video in a Recording Pipeline Message-ID: Hi I am,trying to implement a recorder that will split a larger media file or a live source in to smaller chunks. First i created a pipeline to record the steam from filesrc and rtspsrc . For Implementing the file change i put a callback in the encoder element which keep on checking the size of the file created by the pipeline. When the filesize is reached the predefined size then i will put the pipeline in to GST_STATE_PAUSE. Then i move the recorded file to the destination .,after that that the state of the pipeline is set to GST_STATE_NULL. and i unref the main pipeline.Here the first video is created and after that the same pipeline is created and the video is recorded . I can Implement file change in this way..But is this the right way to Implement my idea..? Pipeline Created --->> GST_STATE_READY---->> GST_STATE_PAUSE ---->>GST_STATE_PLAYING-----{ after a predefined file size splitting file } -->> GST_STATE_PAUSE--->> GST_STATE_READY-->>GST_STATE_NULL-->>unref the pipeline After this the same method is repeated to create the second chunk. Is there any better way to Implement idea...? Regards LijiN -------------- next part -------------- An HTML attachment was scrubbed... URL: From wl2776 at gmail.com Thu Dec 9 17:05:40 2010 From: wl2776 at gmail.com (wl2776) Date: Thu, 9 Dec 2010 08:05:40 -0800 (PST) Subject: [gst-devel] rtspsrc receives tons of qos messages. What does it mean? Message-ID: <1291910740810-3080425.post@n4.nabble.com> I have a playbin2 pipeline with the rtspsrc source. While reading rtsp stream it outputs tons of messages like "rtspsrc gstrtspsrc.c:1895:gst_rtspsrc_handle_src_event: pad source:stream0 received event qos" What should I do with them? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/rtspsrc-receives-tons-of-qos-messages-What-does-it-mean-tp3080425p3080425.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wl2776 at gmail.com Thu Dec 9 17:09:53 2010 From: wl2776 at gmail.com (wl2776) Date: Thu, 9 Dec 2010 08:09:53 -0800 (PST) Subject: [gst-devel] RTSPsrc slow transition PLAYING->PAUSE in windows. Message-ID: <1291910993402-3080442.post@n4.nabble.com> I use rtspsrc inside playbin2 to receive RTSP streams. Transition of playbin2 from playing to paused in windows takes about 30-40 seconds, while in linux it's almost immediate. What's the problem with win32? Example: 0:00:24.109375000 DEBUG rtspsrc gstrtspsrc.c:6104:gst_rtspsrc_pause: wait for connection idle 0:01:02.031250000 DEBUG rtspsrc gstrtspsrc.c:3729:gst_rtspsrc_loop_udp: got interrupted: stop connection flush How can I speed up it? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/RTSPsrc-slow-transition-PLAYING-PAUSE-in-windows-tp3080442p3080442.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wl2776 at gmail.com Thu Dec 9 18:39:36 2010 From: wl2776 at gmail.com (wl2776) Date: Thu, 9 Dec 2010 09:39:36 -0800 (PST) Subject: [gst-devel] RTSPsrc slow transition PLAYING->PAUSE in windows. In-Reply-To: <1291910993402-3080442.post@n4.nabble.com> References: <1291910993402-3080442.post@n4.nabble.com> Message-ID: <1291916376658-3080611.post@n4.nabble.com> Seems like its a bug. I've filed bug report https://bugzilla.gnome.org/show_bug.cgi?id=636893 Will be happy if anyone could find some time to look into this and fix it. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/RTSPsrc-slow-transition-PLAYING-PAUSE-in-windows-tp3080442p3080611.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From braunsquared at gmail.com Thu Dec 9 19:07:22 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Thu, 9 Dec 2010 13:07:22 -0500 Subject: [gst-devel] Intermittent Blank Video Message-ID: <4570B114-E742-48F8-9413-151C670CA4A1@gmail.com> Hello Everyone, So I have a gnlcomposition containing 6 gnlsources and a videomixer in gnloperation. Everything is running wonderfully, except, on occasion, one or two of the videos will be empty. It's not file related since the offending videos will change between launches. I'm thinking it's something related to an async message being dropped or flushing pad, but I'm not sure what to look for or how to resolve it. Any guidance would be lovely. The pipeline will complete successfully and no errors are being thrown, which is the part that baffles me. Each gnlsource consists of a bin which resembles: filesrc ! decodebin2 ! ffmpegcolorspace ! videoscale ! videorate ! video/x-raw-yuv,format=Y444,framerate=30/1,width=240,height=240 Thanks again, Tim From bilboed at gmail.com Thu Dec 9 19:51:58 2010 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 09 Dec 2010 19:51:58 +0100 Subject: [gst-devel] rtspsrc receives tons of qos messages. What does it mean? In-Reply-To: <1291910740810-3080425.post@n4.nabble.com> References: <1291910740810-3080425.post@n4.nabble.com> Message-ID: <1291920718.2584.1.camel@deumeu> On Thu, 2010-12-09 at 08:05 -0800, wl2776 wrote: > I have a playbin2 pipeline with the rtspsrc source. > While reading rtsp stream it outputs tons of messages like > "rtspsrc gstrtspsrc.c:1895:gst_rtspsrc_handle_src_event: pad > source:stream0 received event qos" > > What should I do with them? That's perfectly normal. A live videosink will push QoS messages upstream for every buffer it receives. Edward From c.r.n.a at wanadoo.fr Thu Dec 9 20:46:37 2010 From: c.r.n.a at wanadoo.fr (Nicolas) Date: Thu, 09 Dec 2010 20:46:37 +0100 Subject: [gst-devel] Fakesink and GstNavigation In-Reply-To: <4CFFD7A0.9050201@hora-obscura.de> References: <4CFF7435.9020000@wanadoo.fr> <4CFFD7A0.9050201@hora-obscura.de> Message-ID: <4D01321D.5010308@wanadoo.fr> Hello, Thanks for your response. Ok, I have the gstreamer plugin and i implemented the navigation interface. My problem is I don't know what to put in the function: static void evas_video_sink_navigation_send_event (GstNavigation * navigation, GstStructure * structure) What is require in this function ? Thanks in advance, Nicolas. >> Hello, >> >> I created a "playbin2" and a "fakesink" (I render the video in an evas >> image) >> Now my problem is that "fakesink" has no "GstNavigation". >> How can I do to implement the "GstNavigation" to "Fakesink" ? >> > That is the wrong approach. You rather want to implement a evasvideosink and > implement the navigation iface there. > > Stefan From radivojejovanovic at gmail.com Thu Dec 9 21:09:52 2010 From: radivojejovanovic at gmail.com (Radivoje Jovanovic) Date: Thu, 9 Dec 2010 12:09:52 -0800 Subject: [gst-devel] GST_FLOW_WRONG_STATE In-Reply-To: References: <1291853143.3531.43.camel@zingle> Message-ID: It seems like we are on a good path. after I tried the pipeline: gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue max-size-time=0 max-size-bytes=0 ! mpeg4dec ! MY_VIDEO_SINK t. ! queue ! faad ! alsasink (so I only give time and bytes for video side) the video and audio worked just fine. Unfortunately this trick does not work for the other decoder I have which decodes h264 videos. Even with the pipeline: gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue max-size-time=0 max-size-bytes=0 ! h264dec ! MY_VIDEO_SINK t. ! queue ! faad ! alsasink The pipeline is stuck. This all has nothing to do with GST_FLOW_WRONG_STATE since this error pops up all the time after CTRL-C operation. Any idea why would h264 get stuck? Another question is if I am giving too much memory to gstreamer with max-size-time=0 max-size-bytes=0? Cheers Ogi On Wed, Dec 8, 2010 at 8:27 PM, ved kpl wrote: > Hi, > > One possible reason could be that one of the queues is getting full > while the other remains empty > because of > (a) continuous audio/video buffers for a considerably longer time, > longer than the what the queues can hold. > (b) high latency at the your video decoder (mpeg4dec). > Hence the streaming thread is blocked & one of the sinks is not able > to commit the state to PAUSED. > (because it has not received a buffer yet). > > Yo can also set async=FALSE on the sinks and see. (not recommended) > > You can try disabling the the max time & max bytes properties of the queue. > Try the following pipeline > > gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue > max-size-time=0 max-size-bytes=0 ! mpeg4dec ! MY_VIDEO_SINK t. ! queue > max-size-time=0 max-size-bytes=0 ! faad ! alsasink -v > > Note that this will increase the mem usage.. and even more if u set > max-size-buffers=0 as well. (queue without any limits!) > > Vikram > > On Thu, Dec 9, 2010 at 9:44 AM, ved kpl wrote: > > On Thu, Dec 9, 2010 at 6:26 AM, Radivoje Jovanovic > > wrote: > >> So the pipeline I am using is: > >> gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue ! > >> mpeg4dec ! MY_VIDEO_SINK t. ! queue ! faad ! alsasink > >> This pipeline will hang and after CTRL-C the mpeg4dec will show the > message > >> that is the result of calling gst_push_pad. This gst_push_pad will > return > >> GST_FLOW_WRONG_STATE. > >> > >> If I run gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! > queue > >> ! mpeg4dec ! MY_VIDEO_SINK > >> the video will play just fine > >> > >> If I run: > >> gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue ! > faad ! > >> alsasink > >> the audio from the audio will play just fine. > >> > >> I am not sure which code would be helpful since I have a lot of it. > >> Thanks for the GST_DEBUG info. > >> > >> Ogi > >> > >> > >> On Wed, Dec 8, 2010 at 4:05 PM, Tim-Philipp M?ller > wrote: > >>> > >>> On Wed, 2010-12-08 at 12:53 -0800, Radivoje Jovanovic wrote: > >>> > >>> > I have a video sink and ALSA driver to be used to play videos on the > >>> > platform. If I play video or audio by them self everything works > fine, > >>> > but if I specify the whole pipeline for video and audio, the video > >>> > decoder (in pipeline right before the sink) comes back with the > >>> > message: > >>> > "The decoded frame did not successfully push out to downstream > >>> > element" > >>> > The message comes because the gst_pad_push have returned with > >>> > GST_FLOW_WRONG_STATE instead GST_FLOW_OK. > >>> > > >>> > Any idea why this might happen? > >>> > >>> It usually helps if you post your exact pipeline and/or any code > >>> snippets that go with it. > >>> > >>> FLOW_WRONG_STATE is what you get when a pad is flushing, which may be > >>> normal (happens during a flushing seek, to make the old streaming > thread > >>> stop) or because you forgot to set an element into PAUSED/PLAYING state > >>> (e.g. because you added it from a pad-added or new-decoded-pad callback > >>> or so). > >>> > >>> The GST_DEBUG=*:5 log might give you more information (just grep for > >>> wrong-state and read the lines before that). > >>> > >>> Cheers > >>> -Tim > >>> > >>> > >>> > >>> > >>> > ------------------------------------------------------------------------------ > >>> This SF Dev2Dev email is sponsored by: > >>> > >>> WikiLeaks The End of the Free Internet > >>> http://p.sf.net/sfu/therealnews-com > >>> _______________________________________________ > >>> gstreamer-devel mailing list > >>> gstreamer-devel at lists.sourceforge.net > >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >> > >> > >> > ------------------------------------------------------------------------------ > >> This SF Dev2Dev email is sponsored by: > >> > >> WikiLeaks The End of the Free Internet > >> http://p.sf.net/sfu/therealnews-com > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >> > >> > > > > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From thaytan at noraisin.net Thu Dec 9 23:13:53 2010 From: thaytan at noraisin.net (Jan Schmidt) Date: Fri, 10 Dec 2010 09:13:53 +1100 Subject: [gst-devel] Fakesink and GstNavigation In-Reply-To: <4D01321D.5010308@wanadoo.fr> References: <4CFF7435.9020000@wanadoo.fr> <4CFFD7A0.9050201@hora-obscura.de> <4D01321D.5010308@wanadoo.fr> Message-ID: <1291932833.20840.53.camel@fancy-ubuntu> On Thu, 2010-12-09 at 20:46 +0100, Nicolas wrote: > Hello, > > Thanks for your response. > Ok, I have the gstreamer plugin and i implemented the navigation interface. > > My problem is I don't know what to put in the function: > > static void evas_video_sink_navigation_send_event (GstNavigation * > navigation, GstStructure * structure) > > What is require in this function ? Here's one example: http://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/sys/xvimage/xvimagesink.c#n2730 The xvimagesink implementation is about as complex as it gets, because the xvimagesink supports scaling and offsetting of the video into the output surface. The navigation_send_event implementation reverses those operations, so it takes mouse clicks in the window coordinates, and converts it back to a mouse click coordinate within the original video. - Jan. > > Thanks in advance, > Nicolas. > > > >> Hello, > >> > >> I created a "playbin2" and a "fakesink" (I render the video in an evas > >> image) > >> Now my problem is that "fakesink" has no "GstNavigation". > >> How can I do to implement the "GstNavigation" to "Fakesink" ? > >> > > That is the wrong approach. You rather want to implement a evasvideosink and > > implement the navigation iface there. > > > > Stefan > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Jan Schmidt From ved.kpl at gmail.com Fri Dec 10 04:45:45 2010 From: ved.kpl at gmail.com (ved kpl) Date: Fri, 10 Dec 2010 09:15:45 +0530 Subject: [gst-devel] GST_FLOW_WRONG_STATE In-Reply-To: References: <1291853143.3531.43.camel@zingle> Message-ID: On Fri, Dec 10, 2010 at 1:39 AM, Radivoje Jovanovic wrote: > It seems like we are on a good path. after I tried the pipeline: > ?gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue > max-size-time=0 max-size-bytes=0 ! mpeg4dec ! MY_VIDEO_SINK t. ! queue > ? ! faad ! alsasink > ?? (so I only give time and bytes for video side) the video and audio worked > just fine. > > Unfortunately this trick does not work for the other decoder I have which > decodes h264 videos. > > Even with the pipeline: > ?gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue > max-size-time=0 max-size-bytes=0 ! h264dec ! MY_VIDEO_SINK t. ! queue > ? ! faad ! alsasink > The pipeline is stuck. This all has nothing to do with GST_FLOW_WRONG_STATE > since this error pops up all the time after CTRL-C operation. Any idea why > would h264 get stuck? To start off, try setting max-size-time/bytes/buffers=0 on the both the queues (no limits) If it works, it indicates that your h264dec has a very high latency..and you can narrow to the problem from here. > Another question is if I am giving too much memory to gstreamer with > max-size-time=0 max-size-bytes=0? Well, It would depend on your codec details (resilution, etc), environment/platform. You could possibly track down the issue. > Cheers > Ogi > > > On Wed, Dec 8, 2010 at 8:27 PM, ved kpl wrote: >> >> Hi, >> >> One possible reason could be that one of the queues is getting full >> while the other remains empty >> because of >> (a) continuous audio/video buffers for a considerably longer time, >> longer than the what the queues can hold. >> (b) high latency at the your video decoder (mpeg4dec). >> Hence the streaming thread is blocked & one of the sinks is not able >> to commit the state to PAUSED. >> (because it has not received a buffer yet). >> >> Yo can also set async=FALSE on the sinks and see. (not recommended) >> >> You can try disabling the the max time & max bytes properties of the >> queue. >> Try the following pipeline >> >> ?gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue >> max-size-time=0 max-size-bytes=0 ! mpeg4dec ! MY_VIDEO_SINK t. ! queue >> max-size-time=0 max-size-bytes=0 ! faad ! alsasink -v >> >> Note that this will increase the mem usage.. and even more if u set >> max-size-buffers=0 as well. (queue without any limits!) >> >> Vikram >> >> On Thu, Dec 9, 2010 at 9:44 AM, ved kpl wrote: >> > On Thu, Dec 9, 2010 at 6:26 AM, Radivoje Jovanovic >> > wrote: >> >> So the pipeline I am using is: >> >> gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue? ! >> >> mpeg4dec ! MY_VIDEO_SINK t. ! queue ! faad ! alsasink >> >> This pipeline will hang and after CTRL-C the mpeg4dec will show the >> >> message >> >> that is the result of calling gst_push_pad. This gst_push_pad will >> >> return >> >> GST_FLOW_WRONG_STATE. >> >> >> >> If I run gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! >> >> queue >> >> ! mpeg4dec ! MY_VIDEO_SINK >> >> the video will play just fine >> >> >> >> If I run: >> >> gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue ! >> >> faad ! >> >> alsasink >> >> the audio from the audio will play just fine. >> >> >> >> I am not sure which code would be helpful since I have a lot of it. >> >> Thanks for the GST_DEBUG info. >> >> >> >> Ogi >> >> >> >> >> >> On Wed, Dec 8, 2010 at 4:05 PM, Tim-Philipp M?ller >> >> wrote: >> >>> >> >>> On Wed, 2010-12-08 at 12:53 -0800, Radivoje Jovanovic wrote: >> >>> >> >>> > I have a video sink and ALSA driver to be used to play videos on the >> >>> > platform. If I play video or audio by them self everything works >> >>> > fine, >> >>> > but if I specify the whole pipeline for video and audio, the video >> >>> > decoder (in pipeline right before the sink) comes back with the >> >>> > message: >> >>> > "The decoded frame did not successfully push out to downstream >> >>> > element" >> >>> > The message comes because the gst_pad_push have returned with >> >>> > GST_FLOW_WRONG_STATE instead GST_FLOW_OK. >> >>> > >> >>> > Any idea why this might happen? >> >>> >> >>> It usually helps if you post your exact pipeline and/or any code >> >>> snippets that go with it. >> >>> >> >>> FLOW_WRONG_STATE is what you get when a pad is flushing, which may be >> >>> normal (happens during a flushing seek, to make the old streaming >> >>> thread >> >>> stop) or because you forgot to set an element into PAUSED/PLAYING >> >>> state >> >>> (e.g. because you added it from a pad-added or new-decoded-pad >> >>> callback >> >>> or so). >> >>> >> >>> The GST_DEBUG=*:5 log might give you more information (just grep for >> >>> wrong-state and read the lines before that). >> >>> >> >>> Cheers >> >>> ?-Tim >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> ------------------------------------------------------------------------------ >> >>> This SF Dev2Dev email is sponsored by: >> >>> >> >>> WikiLeaks The End of the Free Internet >> >>> http://p.sf.net/sfu/therealnews-com >> >>> _______________________________________________ >> >>> gstreamer-devel mailing list >> >>> gstreamer-devel at lists.sourceforge.net >> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> >> >> >> >> ------------------------------------------------------------------------------ >> >> This SF Dev2Dev email is sponsored by: >> >> >> >> WikiLeaks The End of the Free Internet >> >> http://p.sf.net/sfu/therealnews-com >> >> _______________________________________________ >> >> gstreamer-devel mailing list >> >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> >> > >> >> >> ------------------------------------------------------------------------------ >> This SF Dev2Dev email is sponsored by: >> >> WikiLeaks The End of the Free Internet >> http://p.sf.net/sfu/therealnews-com >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From markus.t.vartiainen at nokia.com Fri Dec 10 09:47:05 2010 From: markus.t.vartiainen at nokia.com (Vartiainen Markus.T (Nokia-MS/Tampere)) Date: Fri, 10 Dec 2010 10:47:05 +0200 Subject: [gst-devel] camerabin2 is now on -bad In-Reply-To: <1291835633.16533.15.camel@blacksheep> References: <1291835633.16533.15.camel@blacksheep> Message-ID: <4D01E909.8090304@nokia.com> On 12/08/2010 09:13 PM, ext Thiago Sousa Santos wrote: > Hello, > > for those not on the commits mailing list, camerabin2 prototype is now > on the gst-plugins-bad module on Freedesktop (official repository). It > is marked as experimental and in order to build it and its tests/example > you need to pass '--enable-experimental' to configure. > > There is still lots of work to do, if you'd like to contribute you can > put patches on https://bugzilla.gnome.org/ . In case you don't know > exactly what/where to help, ping me (thiagoss) on #gstreamer and I can > show a TODO list. The more people we can gather developing it from the > start, the better code gets and we make sure it will cover more use > cases from different applications. > > If you prefer to use gitorious, I'll keep my 'work in progress' branch > on http://gitorious.org/gstreamer-camerabin2/gst-plugins-bad , so feel > free to fork from there and send pull requests. I'll review from there > and then push upstream. > > Regarding the meeting we had yesterday, I've uploaded the logs here: > http://people.collabora.co.uk/~thiagoss/camerabin2_20101207.log > > -- > Thiago Hi, Thanks for this heads-up. The IRC meeting log enlightens nicely the context we're talking about, but it also makes it obvious that it is not an easy task to respond to all varying needs and hardware/use-case combinations. (Btw, are the actual meeting notes / TODO items available publicly somewhere?) I'll throw some more ingredients into the soup regarding what I discussed with Tommi My?h?nen (main contributor to camera source on Nokia side). In camerabin, with the single pad camera source, it has been obvious that we can only get either viewfinder frames, video frames or still images out of the camera source, one at a time. This imposes some limitations, but also provides a simple abstraction of what a camera source can provide. In camerabin2, with the three-pad camera source, it is not obvious any more what combinations of simultaneous streams can be expected out from the camera source at the same time, and how differently the streams in different source pads can be configured, and how much that might affect performance. This depends at least to some extent to the capabilities of the underlying hardware. At least from the mobile basic camera application point of view, these capabilities are directly related to what can be feasibly provided for the application user. Some capabilities affect the feature set, while some might affect the quality of the features. Our conclusion was that a way to provide this information to the camerabin2 user is necessary. The questions regarding this, could be: 1. What are the features that all camera sources are expected to provide by default? 2. What are the optional features that should be reported by the camera source? 3. What is the interface/API/language to access this information? 4. Whether to provide an extension method for this information, and how to implement it? (5. How much slack we are willing to give in the specification, i.e. how much the camerabin2 user "just needs to know" in each case, depending on the environment and use-case?) I'm trying to come up with some of the answers myself, once I learn a bit more around the subject. Regards, Markus Vartiainen From josel.segura at gmx.es Fri Dec 10 11:41:12 2010 From: josel.segura at gmx.es (=?ISO-8859-1?B?Sm9z6Q==?= Luis Segura Lucas) Date: Fri, 10 Dec 2010 11:41:12 +0100 Subject: [gst-devel] Select input on a v4l2 device Message-ID: <20101210114112.7648092b@peter> Good morning from Spain! I'm using Gstreamer with some v4l2 devices. Most of them only have one input and I have no problem with them, but I need to use a video capture card and this hardware has 2 inputs at least (S-Video and a composite analogical video). If I use this device, by default it uses the composite analogical input, but I need to be able to select the input I want. I inspect the properties of the v4l2src element and I didn't find anything related. Do you know any way to select the input of my v4l2src using the Gstreamer properties? Regards and thanks in advance. From mjoachimiak at gmail.com Fri Dec 10 12:06:16 2010 From: mjoachimiak at gmail.com (Michael Joachimiak) Date: Fri, 10 Dec 2010 13:06:16 +0200 Subject: [gst-devel] Select input on a v4l2 device In-Reply-To: <20101210114112.7648092b@peter> References: <20101210114112.7648092b@peter> Message-ID: device parameter. device=/dev/video0 and so on. 2010/12/10 Jos? Luis Segura Lucas > Good morning from Spain! > > I'm using Gstreamer with some v4l2 devices. Most of them only have one > input and I have no problem with them, but I need to use a video > capture card and this hardware has 2 inputs at least (S-Video and a > composite analogical video). > > If I use this device, by default it uses the composite analogical > input, but I need to be able to select the input I want. > > I inspect the properties of the v4l2src element and I didn't find > anything related. Do you know any way to select the input of my v4l2src > using the Gstreamer properties? > > Regards and thanks in advance. > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Your Sincerely Michael Joachimiak -------------- next part -------------- An HTML attachment was scrubbed... URL: From josel.segura at gmx.es Fri Dec 10 12:09:51 2010 From: josel.segura at gmx.es (=?ISO-8859-1?B?Sm9z6Q==?= Luis Segura Lucas) Date: Fri, 10 Dec 2010 12:09:51 +0100 Subject: [gst-devel] Select input on a v4l2 device In-Reply-To: References: <20101210114112.7648092b@peter> Message-ID: <20101210120951.270b66c0@peter> El Fri, 10 Dec 2010 13:06:16 +0200 Michael Joachimiak escribi?: > device parameter. > device=/dev/video0 and so on. No, that's the problem. It's only one device (/dev/video0). I want to select the input on that device, no using another device... From t.i.m at zen.co.uk Fri Dec 10 12:24:08 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 10 Dec 2010 11:24:08 +0000 Subject: [gst-devel] Select input on a v4l2 device In-Reply-To: <20101210120951.270b66c0@peter> References: <20101210114112.7648092b@peter> <20101210120951.270b66c0@peter> Message-ID: <1291980248.24811.0.camel@zingle> On Fri, 2010-12-10 at 12:09 +0100, Jos? Luis Segura Lucas wrote: > No, that's the problem. It's only one device (/dev/video0). I want to > select the input on that device, no using another device... Have you tried the GstTuner interface? (Can't use it from gst-launch) Set v4l2src to READY state, then you should be able to use the tuner interface on it. Cheers -Tim From tristan at sat.qc.ca Fri Dec 10 13:35:02 2010 From: tristan at sat.qc.ca (Tristan Matthews) Date: Fri, 10 Dec 2010 07:35:02 -0500 Subject: [gst-devel] Select input on a v4l2 device In-Reply-To: <1291980248.24811.0.camel@zingle> References: <20101210114112.7648092b@peter> <20101210120951.270b66c0@peter> <1291980248.24811.0.camel@zingle> Message-ID: Otherwise you can always use v4l2-ctl, i.e. v4l2-ctl --set-input 1 v4l2-ctl is part of the ivtv-utils package. -t 2010/12/10 Tim-Philipp M?ller > On Fri, 2010-12-10 at 12:09 +0100, Jos? Luis Segura Lucas wrote: > > > No, that's the problem. It's only one device (/dev/video0). I want to > > select the input on that device, no using another device... > > Have you tried the GstTuner interface? (Can't use it from gst-launch) > > Set v4l2src to READY state, then you should be able to use the tuner > interface on it. > > Cheers > -Tim > > > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Tristan Matthews email: tristan at sat.qc.ca web: http://tristanswork.blogspot.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.henningsson at canonical.com Fri Dec 10 14:57:51 2010 From: david.henningsson at canonical.com (David Henningsson) Date: Fri, 10 Dec 2010 14:57:51 +0100 Subject: [gst-devel] [pulseaudio-discuss] [PATCH 0/3] Fighting rewinds In-Reply-To: References: <4D00DFAF.3000807@canonical.com> Message-ID: <4D0231DF.7040008@canonical.com> [adding gstreamer-devel to the discussion] On 2010-12-09 17:18, pl bossart wrote: >> However, the problem is quite complex and there does not seem to be one >> perfect fix, it's more of an optimisation problem. GStreamer in particular >> sends out many small data packages, and PulseAudio does not handle that very >> well. > > That's the default behavior, but you can cut the traffic by using the > latency-time property in pulsesink. This makes sure you send bigger > buffers, up to the 64k limit that PulseAudio has internally. Thanks, that's good to know. Now, I'm just playing a simple audio file with totem, so obviously we want high latency and large packet sizes, but I'm not a gstreamer developer - do you have an idea of what can be at fault here? Should totem specify a big packet size (regardless of sink?), or can we just change the default packet size to something big for people not requesting anything in particular? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic From rob at ti.com Fri Dec 10 15:49:40 2010 From: rob at ti.com (Clark, Rob) Date: Fri, 10 Dec 2010 08:49:40 -0600 Subject: [gst-devel] camerabin2 is now on -bad In-Reply-To: <4D01E909.8090304@nokia.com> References: <1291835633.16533.15.camel@blacksheep> <4D01E909.8090304@nokia.com> Message-ID: On Fri, Dec 10, 2010 at 2:47 AM, Vartiainen Markus.T (Nokia-MS/Tampere) < markus.t.vartiainen at nokia.com> wrote: > On 12/08/2010 09:13 PM, ext Thiago Sousa Santos wrote: > > Hello, > > > > for those not on the commits mailing list, camerabin2 prototype is now > > on the gst-plugins-bad module on Freedesktop (official repository). It > > is marked as experimental and in order to build it and its tests/example > > you need to pass '--enable-experimental' to configure. > > > > There is still lots of work to do, if you'd like to contribute you can > > put patches on https://bugzilla.gnome.org/ . In case you don't know > > exactly what/where to help, ping me (thiagoss) on #gstreamer and I can > > show a TODO list. The more people we can gather developing it from the > > start, the better code gets and we make sure it will cover more use > > cases from different applications. > > > > If you prefer to use gitorious, I'll keep my 'work in progress' branch > > on http://gitorious.org/gstreamer-camerabin2/gst-plugins-bad , so feel > > free to fork from there and send pull requests. I'll review from there > > and then push upstream. > > > > Regarding the meeting we had yesterday, I've uploaded the logs here: > > http://people.collabora.co.uk/~thiagoss/camerabin2_20101207.log > > > > -- > > Thiago > > Hi, > > Thanks for this heads-up. The IRC meeting log enlightens nicely the > context we're talking about, but it also makes it obvious that it is not > an easy task to respond to all varying needs and hardware/use-case > combinations. > > (Btw, are the actual meeting notes / TODO items available publicly > somewhere?) > > I'll throw some more ingredients into the soup regarding what I > discussed with Tommi My?h?nen (main contributor to camera source on > Nokia side). > > In camerabin, with the single pad camera source, it has been obvious > that we can only get either viewfinder frames, video frames or still > images out of the camera source, one at a time. This imposes some > limitations, but also provides a simple abstraction of what a camera > source can provide. > > In camerabin2, with the three-pad camera source, it is not obvious any > more what combinations of simultaneous streams can be expected out from > the camera source at the same time, and how differently the streams in > different source pads can be configured, and how much that might affect > performance. This depends at least to some extent to the capabilities of > the underlying hardware. > I think in image capture mode, the camsrcbin would be expected to push on vf pad and img pad. In video capture mode, vf pad and vid pad. (It would be nice to somehow, if the camera supports it, have image+video capture mode where you are pushing on all three pads.) Internally, within camsrcbin implementation, I think it would be implementation dependent if you are actually pushing the same captured frame from the camera on multiple pads (perhaps after a vscale), or if you are just reconfiguring the camera device and capturing successive frames at different resolutions. BR, -R > > At least from the mobile basic camera application point of view, these > capabilities are directly related to what can be feasibly provided for > the application user. Some capabilities affect the feature set, while > some might affect the quality of the features. > > Our conclusion was that a way to provide this information to the > camerabin2 user is necessary. > > The questions regarding this, could be: > > 1. What are the features that all camera sources are expected to provide > by default? > > 2. What are the optional features that should be reported by the camera > source? > > 3. What is the interface/API/language to access this information? > > 4. Whether to provide an extension method for this information, and how > to implement it? > > (5. How much slack we are willing to give in the specification, i.e. how > much the camerabin2 user "just needs to know" in each case, depending on > the environment and use-case?) > > I'm trying to come up with some of the answers myself, once I learn a > bit more around the subject. > > Regards, > > Markus Vartiainen > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rafael.lmsousa at gmail.com Fri Dec 10 16:05:25 2010 From: rafael.lmsousa at gmail.com (Rafael Sousa) Date: Fri, 10 Dec 2010 11:05:25 -0400 Subject: [gst-devel] x264 build 110 incompatible with gstreamer? Message-ID: Hi all, I'm trying to encode a yuv video to h.264, for this, I installed the lastest builds of gstreamer, plugins-bad ugly and good, gst-ffmpeg and x264 (build 110). But when I play the encoded video the image is blurred. I tried to downgrade the build of x264 to build 85 and it worked fine. There is some incompatibility with the gstreamer and x264's lastest build? I'm using ubuntu 10.04. thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From bossart.nospam at gmail.com Fri Dec 10 16:37:06 2010 From: bossart.nospam at gmail.com (pl bossart) Date: Fri, 10 Dec 2010 09:37:06 -0600 Subject: [gst-devel] [pulseaudio-discuss] [PATCH 0/3] Fighting rewinds In-Reply-To: <4D0231DF.7040008@canonical.com> References: <4D00DFAF.3000807@canonical.com> <4D0231DF.7040008@canonical.com> Message-ID: >>> However, the problem is quite complex and there does not seem to be one >>> perfect fix, it's more of an optimisation problem. GStreamer in >>> particular >>> sends out many small data packages, and PulseAudio does not handle that >>> very >>> well. >> >> That's the default behavior, but you can cut the traffic by using the >> latency-time property in pulsesink. This makes sure you send bigger >> buffers, up to the 64k limit that PulseAudio has internally. > > Thanks, that's good to know. Now, I'm just playing a simple audio file with > totem, so obviously we want high latency and large packet sizes, but I'm not > a gstreamer developer - do you have an idea of what can be at fault here? > Should totem specify a big packet size (regardless of sink?), or can we just > change the default packet size to something big for people not requesting > anything in particular? The default pulsesink settings for latency/buffering are inherited from a base class. I would agree with you that they should be large by default and decreased explicitly when the application wants low-latency (gaming, speech), but the opposite is done... It's my understanding that higher-level components in the Meego/Qt stack will specify bigger buffers/higher latencies, but for people who use plain vanilla gstreamer optimizing for power requires a bit of knowledge and manual configurations. This is unfortunate since every single player will need to repeat this configuration. -Pierre From wb666greene at gmail.com Fri Dec 10 18:00:55 2010 From: wb666greene at gmail.com (wally_bkg) Date: Fri, 10 Dec 2010 09:00:55 -0800 (PST) Subject: [gst-devel] Select input on a v4l2 device In-Reply-To: <1291980248.24811.0.camel@zingle> References: <20101210114112.7648092b@peter> <20101210120951.270b66c0@peter> <1291980248.24811.0.camel@zingle> Message-ID: <1292000455372-3082321.post@n4.nabble.com> Tim-Philipp M?ller-2 wrote: > > On Fri, 2010-12-10 at 12:09 +0100, Jos? Luis Segura Lucas wrote: > >> No, that's the problem. It's only one device (/dev/video0). I want to >> select the input on that device, no using another device... > > Have you tried the GstTuner interface? (Can't use it from gst-launch) > > Set v4l2src to READY state, then you should be able to use the tuner > interface on it. > > Cheers > -Tim > > Some sample code of how to do this can be found here: A sample can be found here: http://gstreamer-devel.966125.n4.nabble.com/Problem-Trying-To-Use-v4l2src-td973410.html#a973414 Just be aware that it sets the pipeline to READY instead of PAUSED which worked for an older version of gstreamer. The ugliness of v4l2 is exposed GST_TUNER and I pretty much have had to add a new string to look for every time I've gotten a new capture card to test. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Select-input-on-a-v4l2-device-tp3081723p3082321.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wb666greene at gmail.com Fri Dec 10 18:13:21 2010 From: wb666greene at gmail.com (wally_bkg) Date: Fri, 10 Dec 2010 09:13:21 -0800 (PST) Subject: [gst-devel] Why does this video format conversion fail? In-Reply-To: <1291922502531-3080822.post@n4.nabble.com> References: <1291922502531-3080822.post@n4.nabble.com> Message-ID: <1292001201150-3082344.post@n4.nabble.com> wally_bkg wrote: > > I need to do it in C but I can illustrate the problem simply with a few > gst-lanuch commands. > > Basically I have a USB video capture device (Hauppauge WinTV-HVR 950Q) > that works with gstreamer if I simply do: > > gst-launch v4l2src device=/dev/video2 ! xvimagesink > > However I'm having trouble figuring out a caps filter to use that will let > me get the buffers in a yuv type format. > > > On a normal capture card if I do: > > gst-launch v4l2src device=/dev/video1 ! video/x-raw-yuv, > framerate=\(fraction\)30000/1001, width=640, height=480 ! xvimagesink > > It works fine, but change to /dev/video2 (the USB device) I get: > > Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Could not > negotiate format > Additional debug info: > gstbasesrc.c(2719): gst_base_src_start (): > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: > Check your filtered caps, if any > > > > So I tried using ffmpegcolorspace to convert: > > gst-launch v4l2src device=/dev/video1 ! ffmpegcolorspace ! > video/x-raw-yuv, framerate=\(fraction\)30000/1001, width=640, height=480 ! > ffmpegcolorspace ! xvimagesink > > And get the same error message as without the ffmpegcolorspace elements > between the capsfilter. > > > One of my main reasons for trying to use gstreamer is to let it do the > heavy lifting of dealing with video input and output. At the end of the > day all I want from the appsink element is a pointer to the video data in > a format documented well enough that I can pull out a 640x480 intensity > (grayscale) image. > > Up to getting this device, setting the caps to { video/x-raw-yuv, > framerate=\(fraction\)30000/1001, width=640, height=480 } has worked fine > for all the capture cards I've tried, and obviously needing to deal with > only a single raw format in my code simplifies it greatly. > > > I'm having trouble in my C code to extract the caps that get negotiated if > I leave out the capsfilter from my pipeline. Any samples out there of how > to do it? > > I figured out how to extract the caps. When using /dev/video1 (saa713x card) the "default" Buffer caps: video/x-raw-gray, bpp=(int)8, framerate=(fraction)30000/1001, width=(int)704, height=(int)480 When uisng /dev/video2 (the 950Q USB device) Buffer caps: video/x-raw-rgb, bpp=(int)24, depth=(int)24, red_mask=(int)255, green_mask=(int)65280, blue_mask=(int)16711680, endianness=(int)4321, framerate=(fraction)30000/1001, width=(int)720, height=(int)480 But this doesn't give me any clues as to why ffmpegcolor space can't convert the rgb caps to the yuv or grey caps I'd prefer to use. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Why-does-this-video-format-conversion-fail-tp3080822p3082344.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From braunsquared at gmail.com Fri Dec 10 18:57:07 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Fri, 10 Dec 2010 12:57:07 -0500 Subject: [gst-devel] Why does this video format conversion fail? In-Reply-To: <1292001201150-3082344.post@n4.nabble.com> References: <1291922502531-3080822.post@n4.nabble.com> <1292001201150-3082344.post@n4.nabble.com> Message-ID: Wally, Theres other parts to the negotiation besides color space. Framerate and size are also considered. You may want to make it resemble: gst-launch v4l2src device=/dev/video1 ! ffmpegcolorspace ! videoscale ! videorate ! video/x-raw-yuv, framerate=\(fraction\)30000/1001, width=640, height=480 ! ffmpegcolorspace ! xvimagesink I don't think the second ffmpegcolorspace is needed either. If you run this with GST_DEBUG=GST_CAPS:3, you will see the logs of the negotiations and you can hopefully decipher whats happening. Hope this helps. Tim On Fri, Dec 10, 2010 at 12:13 PM, wally_bkg wrote: > > > wally_bkg wrote: > > > > I need to do it in C but I can illustrate the problem simply with a few > > gst-lanuch commands. > > > > Basically I have a USB video capture device (Hauppauge WinTV-HVR 950Q) > > that works with gstreamer if I simply do: > > > > gst-launch v4l2src device=/dev/video2 ! xvimagesink > > > > However I'm having trouble figuring out a caps filter to use that will > let > > me get the buffers in a yuv type format. > > > > > > On a normal capture card if I do: > > > > gst-launch v4l2src device=/dev/video1 ! video/x-raw-yuv, > > framerate=\(fraction\)30000/1001, width=640, height=480 ! xvimagesink > > > > It works fine, but change to /dev/video2 (the USB device) I get: > > > > Setting pipeline to PAUSED ... > > ERROR: Pipeline doesn't want to pause. > > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Could not > > negotiate format > > Additional debug info: > > gstbasesrc.c(2719): gst_base_src_start (): > > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: > > Check your filtered caps, if any > > > > > > > > So I tried using ffmpegcolorspace to convert: > > > > gst-launch v4l2src device=/dev/video1 ! ffmpegcolorspace ! > > video/x-raw-yuv, framerate=\(fraction\)30000/1001, width=640, height=480 > ! > > ffmpegcolorspace ! xvimagesink > > > > And get the same error message as without the ffmpegcolorspace elements > > between the capsfilter. > > > > > > One of my main reasons for trying to use gstreamer is to let it do the > > heavy lifting of dealing with video input and output. At the end of the > > day all I want from the appsink element is a pointer to the video data in > > a format documented well enough that I can pull out a 640x480 intensity > > (grayscale) image. > > > > Up to getting this device, setting the caps to { video/x-raw-yuv, > > framerate=\(fraction\)30000/1001, width=640, height=480 } has worked fine > > for all the capture cards I've tried, and obviously needing to deal with > > only a single raw format in my code simplifies it greatly. > > > > > > I'm having trouble in my C code to extract the caps that get negotiated > if > > I leave out the capsfilter from my pipeline. Any samples out there of > how > > to do it? > > > > > > I figured out how to extract the caps. > > When using /dev/video1 (saa713x card) the "default" Buffer caps: > video/x-raw-gray, bpp=(int)8, framerate=(fraction)30000/1001, > width=(int)704, height=(int)480 > > When uisng /dev/video2 (the 950Q USB device) Buffer caps: video/x-raw-rgb, > bpp=(int)24, depth=(int)24, red_mask=(int)255, green_mask=(int)65280, > blue_mask=(int)16711680, endianness=(int)4321, > framerate=(fraction)30000/1001, width=(int)720, height=(int)480 > > But this doesn't give me any clues as to why ffmpegcolor space can't > convert > the rgb caps to the yuv or grey caps I'd prefer to use. > > > > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/Why-does-this-video-format-conversion-fail-tp3080822p3082344.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From josel.segura at gmx.es Fri Dec 10 19:44:04 2010 From: josel.segura at gmx.es (=?ISO-8859-1?B?Sm9z6Q==?= Luis Segura Lucas) Date: Fri, 10 Dec 2010 19:44:04 +0100 Subject: [gst-devel] Select input on a v4l2 device In-Reply-To: <1292000455372-3082321.post@n4.nabble.com> References: <20101210114112.7648092b@peter> <20101210120951.270b66c0@peter> <1291980248.24811.0.camel@zingle> <1292000455372-3082321.post@n4.nabble.com> Message-ID: <20101210194404.779e01e4@peter> Thanks for all your answers, there are very helpful :-) I will try Monday, at work :-) Regards El Fri, 10 Dec 2010 09:00:55 -0800 (PST) wally_bkg escribi?: > > > Tim-Philipp M?ller-2 wrote: > > > > On Fri, 2010-12-10 at 12:09 +0100, Jos? Luis Segura Lucas wrote: > > > >> No, that's the problem. It's only one device (/dev/video0). I want > >> to select the input on that device, no using another device... > > > > Have you tried the GstTuner interface? (Can't use it from > > gst-launch) > > > > Set v4l2src to READY state, then you should be able to use the tuner > > interface on it. > > > > Cheers > > -Tim > > > > > > Some sample code of how to do this can be found here: > > A sample can be found here: > > http://gstreamer-devel.966125.n4.nabble.com/Problem-Trying-To-Use-v4l2src-td973410.html#a973414 > > > Just be aware that it sets the pipeline to READY instead of PAUSED > which worked for an older version of gstreamer. > > The ugliness of v4l2 is exposed GST_TUNER and I pretty much have had > to add a new string to look for every time I've gotten a new capture > card to test. > > > From bisht.sudarshan at gmail.com Fri Dec 10 20:13:45 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Sat, 11 Dec 2010 00:43:45 +0530 Subject: [gst-devel] Sequence of calls in Gstreamer Application In-Reply-To: <4D009BA5.3040803@tataelxsi.co.in> References: <4D009BA5.3040803@tataelxsi.co.in> Message-ID: Take a look at this link; http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/chapter-helloworld.html#section-helloworld This Hello-World application has filesrc , demuxer, decoder and audiosink. But your application does not have demuxer so you don't need to use on_pad_added callback . All the elements can be linked statically. On Thu, Dec 9, 2010 at 2:34 PM, Gurpreet wrote: > Hi All... > > I am new to Gstreamer. I am writing a Gstreamer application which will have > > 1. filesrc > 2. Aac Encoder > 3. filesink > > Can anybody tell me what will be the sequence of Calls in Application ( > from initilization to end ). > > Thanks > Gurpreet > > > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From wb666greene at gmail.com Fri Dec 10 22:05:43 2010 From: wb666greene at gmail.com (wally_bkg) Date: Fri, 10 Dec 2010 13:05:43 -0800 (PST) Subject: [gst-devel] Why does this video format conversion fail? In-Reply-To: References: <1291922502531-3080822.post@n4.nabble.com> <1292001201150-3082344.post@n4.nabble.com> Message-ID: <1292015143269-3082708.post@n4.nabble.com> Timothy Braun wrote: > > Wally, > Theres other parts to the negotiation besides color space. Framerate > and > size are also considered. You may want to make it resemble: > > gst-launch v4l2src device=/dev/video1 ! ffmpegcolorspace ! videoscale ! > videorate ! video/x-raw-yuv, framerate=\(fraction\)30000/1001, width=640, > height=480 ! ffmpegcolorspace ! xvimagesink > > Thanks, adding videoscale was all that was required, and you are correct the second ffmpegcolorspace is not needed. This one works: gst-launch v4l2src device=/dev/video2 ! ffmpegcolorspace ! videoscale ! video/x-raw-yuv, framerate=\(fraction\)30000/1001, width=640, height=480 ! xvimagesink The PCI cards all can do a yuv format, but apparently the USB Hauppauge 950Q can only do rgb formats (I'm using only the S-video or Composite SD TV inputs it has). The second ffmpegcolorspace element is only needed if I want to use raw-gray. Which brings up another question is it more effecient to pass yuv buffers around that contain 50% more data than will be used (intensity only algorithm) or have a second ffmpegcolorspace in the pipeline? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Why-does-this-video-format-conversion-fail-tp3080822p3082708.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ensonic at hora-obscura.de Fri Dec 10 23:22:06 2010 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 11 Dec 2010 00:22:06 +0200 Subject: [gst-devel] Reducing the time in GST-registry updating and parsing the command line arguments In-Reply-To: References: Message-ID: <4D02A80E.9040304@hora-obscura.de> Am 09.12.2010 14:54, schrieb tamil arasu: > Hi All, > > We are using the DM365 target board. I wrote one Audio streaming > application.I have attached that source code.This application takes three second > to execute.Its taking two second to execute gst_init(&argc,&argv) API ,We used > the Libraries mentioned below > > gst-plugins-good-0.10.14, gst-plugins-base-0.10.23, > gstreamer-0.10.23,gst-plugins-bad-0.10.13, liboil ,libxml , libiconv You don't need libxml2 (and especialy not libxml) for gstreamer. > > Is it possible to get executed in one second. If its still taking that long even with export GST_REGISTRY_UPDATE=no you need to profile what takes most of the time in gst_init(). Try to have a new glib, as a while ago (~ half year) many gobject performance improvements where released. You can probably speed up the time a bit, but removing plugins you won't need. Stefan > > Can anyone help on this issue > > > regards, > > Arasu.M > > > > > > > > On Sun, Nov 28, 2010 at 6:00 PM, Marco Ballesio > wrote: > > Hi, > > as usual, a late reply. Here anyway my 0.05 ?.. > > On Mon, Nov 22, 2010 at 4:42 PM, tamil arasu > wrote: > > HI All, > > > > We are using the gstreamer for capturing the video .we are > > using packages mentioned below > > > > gst-plugins-good-0.10.14, gst-plugins-base-0.10.23, gstreamer-0.10.23, > > gst-plugins-bad-0.10.13 > > > > we are using the pipeline for capturing the video mentioned below > > > > export GST_REGISTRY_UPDATE=no > > export GST_REGISTRY_FORK=no > > > > I don't think it is necessary to set "GST_REGISTRY_FORK=no" in order > to get any improvement. Actually, imo setting this would make things > worse in your case (see below). > > > gst-launch rtspsrc location=rtsp://192.168.1.168:8554/mpeg4 > latency=100 > > name=demux demux. ! queue ! rtpmp4vdepay ! mpeg4videoparse ! dmaidec_mpeg4 ! > > TIDmaiVideoSink videoOutput=LCD videoStd=320X240 demux. ! queue ! > > rtppcmudepay ! mulawdec ! alsasink > > > > When we launch the above mentiond pileline on the command line,we are able > > to display video on Lcd.The output is > > > > Setting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > DavinciDisplay DavinciDisplay.1: Before finishing with S_FMT: > > layer.pix_fmt.bytesperline = 320, > > layer.pix_fmt.width = 320, > > layer.pix_fmt.height = 240, > > layer.pix_fmt.sizeimage =115200 > > New clock: GstSyDavinciDisplay DavinciDisplay.1: pixfmt->width = 320, > > layer->layer_info.config.line_length= 320 > > stemClock > > > > Once I run the above command ,its taking 2 second to get the first message > > (Setting pipeline to PAUSED ...). > > one second for update the registry, > > it's strange, as you've set GST_REGISTRY_UPDATE to "no". How did you > understand this latency is caused when updating the registry? > > In gstregistry.c, setting GST_REGISTRY_FORK=no forces the scan helper > state to "REGISTRY_SCAN_HELPER_NOT_STARTED" in opposition to > "REGISTRY_SCAN_HELPER_DISABLED". > > Just to say that, probably, disabling forks you did preempts any > settings in GST_REGISTRY_UPDATE and triggers a registry generation > anyway. > > > anothe one second for parsing the command > > line arguments.(gst_parse_launchv) > > > > you'll get rid of this by building the pipeline "manually" in a custom > application. See here for more help: > > http://www.gstreamer.net/data/doc/gstreamer/head/manual/html/index.html > > > how to reduce the this 2 second.Can anyone help me on this issue? > > See above. Probably some extra-latency is caused by the video decoder > buffering a fixed amount of memory but, as long as your source is in > the local filesystem, its effect would be negligible. > > Regards > > > > > > > thanks and regards, > > > > Arasu.M > > > > > > > > > > > > ------------------------------------------------------------------------------ > > Beautiful is writing same markup. Internet Explorer 9 supports > > standards for HTML5, CSS3, SVG 1.1, ECMAScript5, and DOM L2 & L3. > > Spend less time writing and rewriting code and more time creating great > > experiences on the web. Be a part of the beta today > > http://p.sf.net/sfu/msIE9-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > > > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From shamun.toha at gmail.com Fri Dec 10 23:24:15 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Fri, 10 Dec 2010 23:24:15 +0100 Subject: [gst-devel] [NEVER WORKS] - /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Message-ID: Dear Experts, I was trying a lot to get this work. But still have same issue with OGGDEMUG. I also tried with simple codecs like SPEEX but its always same error as following. Please advise. My system info: ================== [root at example ~]# uname -a Linux example 2.6.32.12-114.fc12.i686 #1 SMP Tue Apr 27 21:29:58 UTC 2010 i686 i686 i386 GNU/Linux Gstreamer info: ============= [root at example ~]# yum install gstreamer Loaded plugins: presto, refresh-packagekit Setting up Install Process Package gstreamer-0.10.29-1.fc12.i686 already installed and latest version Nothing to do [root at example ~]# CLIENT - ERROR/BUG ================== [root at example ~]# gst-launch -v udpsrc multicast-group=127.0.0.1 port=4444 ! oggdemux ! vorbisdec ! audioconvert ! alsasink Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 421c8315 detected WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 421c8315 detected WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 421c8315 detected ^CCaught interrupt -- handling interrupt. Interrupt: Stopping pipeline ... Execution ended after 1746771949 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... [root at example ~]# SERVER - OK ============= [sun at example tmp]$ gst-launch -v autoaudiosrc num-buffers=100 ! speexenc ! oggmux ! udpsink host=127.0.0.1 auto-multicast=true port=4444 Setting pipeline to PAUSED ... /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: actual-buffer-time = 23777210 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: actual-latency-time = 115986 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstAudioSrcClock /GstPipeline:pipeline0/GstSpeexEnc:speexenc0: last-message = "Warning: speex is optimized for 8, 16 and 32 KHz" /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:src: caps = audio/x-speex, rate=(int)44100, channels=(int)2, streamheader=(buffer)< 5370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1996515720: caps = audio/x-speex, rate=(int)44100, channels=(int)2, streamheader=(buffer)< 5370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = application/ogg, streamheader=(buffer)< 4f676753000200000000000000008869007700000000aa8326b101505370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, 4f6767530000000000000000000088690077010000000c96225e01281f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = application/ogg, streamheader=(buffer)< 4f676753000200000000000000008869007700000000aa8326b101505370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, 4f6767530000000000000000000088690077010000000c96225e01281f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > WARNING: from element /GstPipeline:pipeline0/GstUDPSink:udpsink0: Internal data flow problem. Additional debug info: gstbasesink.c(3343): gst_base_sink_chain_unlocked (): /GstPipeline:pipeline0/GstUDPSink:udpsink0: Received buffer without a new-segment. Assuming timestamps start from 0. ^CCaught interrupt -- handling interrupt. Interrupt: Stopping pipeline ... Execution ended after 1126778907597 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1996515720: caps = NULL /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = NULL /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:src: caps = NULL /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: caps = NULL /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: caps = NULL Setting pipeline to NULL ... Freeing pipeline ... [sun at example tmp]$ clear [sun at example tmp]$ gst-launch -v autoaudiosrc ! audioconvert ! vorbisenc ! oggmux ! udpsink host=127.0.0.1 auto-multicast=true port=4444 Setting pipeline to PAUSED ... /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: actual-buffer-time = 23777210 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: actual-latency-time = 115986 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, channels=(int)1 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, channels=(int)1 Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstAudioSrcClock /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, channels=(int)1 /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, channels=(int)1 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0: caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, channels=(int)1 /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0: last-message = "encoding at quality level 0.30" /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0.GstPad:sink: caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, channels=(int)1 /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0.GstPad:src: caps = audio/x-vorbis, streamheader=(buffer)< 01766f72626973000000000144ac0000000000008038010000000000b801, 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> WARNING: from element /GstPipeline:pipeline0/GstUDPSink:udpsink0: Internal data flow problem. Additional debug info: gstbasesink.c(3343): gst_base_sink_chain_unlocked (): /GstPipeline:pipeline0/GstUDPSink:udpsink0: Received buffer without a new-segment. Assuming timestamps start from 0. Thanks & Regards Shamun -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Fri Dec 10 23:26:31 2010 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 11 Dec 2010 00:26:31 +0200 Subject: [gst-devel] [pulseaudio-discuss] [PATCH 0/3] Fighting rewinds In-Reply-To: References: <4D00DFAF.3000807@canonical.com> <4D0231DF.7040008@canonical.com> Message-ID: <4D02A917.7040007@hora-obscura.de> Am 10.12.2010 17:37, schrieb pl bossart: >>>> However, the problem is quite complex and there does not seem to be one >>>> perfect fix, it's more of an optimisation problem. GStreamer in >>>> particular >>>> sends out many small data packages, and PulseAudio does not handle that >>>> very >>>> well. >>> >>> That's the default behavior, but you can cut the traffic by using the >>> latency-time property in pulsesink. This makes sure you send bigger >>> buffers, up to the 64k limit that PulseAudio has internally. >> >> Thanks, that's good to know. Now, I'm just playing a simple audio file with >> totem, so obviously we want high latency and large packet sizes, but I'm not >> a gstreamer developer - do you have an idea of what can be at fault here? >> Should totem specify a big packet size (regardless of sink?), or can we just >> change the default packet size to something big for people not requesting >> anything in particular? > > The default pulsesink settings for latency/buffering are inherited > from a base class. I would agree with you that they should be large by > default and decreased explicitly when the application wants > low-latency (gaming, speech), but the opposite is done... > It's my understanding that higher-level components in the Meego/Qt > stack will specify bigger buffers/higher latencies, but for people who > use plain vanilla gstreamer optimizing for power requires a bit of > knowledge and manual configurations. This is unfortunate since every > single player will need to repeat this configuration. > -Pierre We could consider tweaking the settings on the high-level bins. I mean after all playbin2 is used for playback of media, so it could select a high latency (at least a higher default). Imho farsight would alreday set a low latency on the pulsesrc/sink. Stefan From braunsquared at gmail.com Fri Dec 10 23:37:23 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Fri, 10 Dec 2010 17:37:23 -0500 Subject: [gst-devel] [NEVER WORKS] - /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. In-Reply-To: References: Message-ID: I'm not an expert, but have you tried running it with GST_DEBUG=3 (or even 4) to see if any more details get dropped to the console? Tim On Fri, Dec 10, 2010 at 5:24 PM, Shamun toha md wrote: > Dear Experts, > > I was trying a lot to get this work. But still have same issue with > OGGDEMUG. I also tried with simple codecs like SPEEX but its always same > error as following. Please advise. > > My system info: > ================== > [root at example ~]# uname -a > Linux example 2.6.32.12-114.fc12.i686 #1 SMP Tue Apr 27 21:29:58 UTC 2010 > i686 i686 i386 GNU/Linux > > Gstreamer info: > ============= > [root at example ~]# yum install gstreamer > Loaded plugins: presto, refresh-packagekit > Setting up Install Process > Package gstreamer-0.10.29-1.fc12.i686 already installed and latest version > Nothing to do > [root at example ~]# > > > > CLIENT - ERROR/BUG > ================== > > [root at example ~]# gst-launch -v udpsrc multicast-group=127.0.0.1 port=4444 > ! oggdemux ! vorbisdec ! audioconvert ! alsasink > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > not decode stream. > Additional debug info: > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > unknown ogg pad for serial 421c8315 detected > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > not decode stream. > Additional debug info: > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > unknown ogg pad for serial 421c8315 detected > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > not decode stream. > Additional debug info: > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > unknown ogg pad for serial 421c8315 detected > ^CCaught interrupt -- handling interrupt. > Interrupt: Stopping pipeline ... > Execution ended after 1746771949 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > Freeing pipeline ... > [root at example ~]# > > > > SERVER - OK > ============= > > > [sun at example tmp]$ gst-launch -v autoaudiosrc num-buffers=100 ! speexenc ! > oggmux ! udpsink host=127.0.0.1 auto-multicast=true port=4444 > Setting pipeline to PAUSED ... > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > actual-buffer-time = 23777210 > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > actual-latency-time = 115986 > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: caps > = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstAudioSrcClock > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0: last-message = "Warning: > speex is optimized for 8, 16 and 32 KHz" > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:sink: caps = > audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, > depth=(int)16, rate=(int)44100, channels=(int)2 > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0: > caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:src: caps = > audio/x-speex, rate=(int)44100, channels=(int)2, streamheader=(buffer)< > 5370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1996515720: caps = > audio/x-speex, rate=(int)44100, channels=(int)2, streamheader=(buffer)< > 5370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = > application/ogg, streamheader=(buffer)< > 4f676753000200000000000000008869007700000000aa8326b101505370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > 4f6767530000000000000000000088690077010000000c96225e01281f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = > application/ogg, streamheader=(buffer)< > 4f676753000200000000000000008869007700000000aa8326b101505370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > 4f6767530000000000000000000088690077010000000c96225e01281f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > WARNING: from element /GstPipeline:pipeline0/GstUDPSink:udpsink0: Internal > data flow problem. > Additional debug info: > gstbasesink.c(3343): gst_base_sink_chain_unlocked (): > /GstPipeline:pipeline0/GstUDPSink:udpsink0: > Received buffer without a new-segment. Assuming timestamps start from 0. > ^CCaught interrupt -- handling interrupt. > Interrupt: Stopping pipeline ... > Execution ended after 1126778907597 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = NULL > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1996515720: caps = > NULL > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = NULL > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:src: caps = NULL > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:sink: caps = NULL > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: caps > = NULL > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > caps = NULL > Setting pipeline to NULL ... > Freeing pipeline ... > [sun at example tmp]$ clear > > [sun at example tmp]$ gst-launch -v autoaudiosrc ! audioconvert ! vorbisenc ! > oggmux ! udpsink host=127.0.0.1 auto-multicast=true port=4444 > Setting pipeline to PAUSED ... > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > actual-buffer-time = 23777210 > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > actual-latency-time = 115986 > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: caps > = audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, > channels=(int)1 > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, > rate=(int)44100, channels=(int)1 > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstAudioSrcClock > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = > audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, > channels=(int)1 > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = > audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, > channels=(int)1 > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0: > caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, > rate=(int)44100, channels=(int)1 > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0: last-message = "encoding at > quality level 0.30" > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0.GstPad:sink: caps = > audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, > channels=(int)1 > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0.GstPad:src: caps = > audio/x-vorbis, streamheader=(buffer)< > 01766f72626973000000000144ac0000000000008038010000000000b801, > 03766f726269731d000000586970682e4f7267206c6962566f7262697320492032303039303730390000000001, > 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> > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1109164821: caps = > audio/x-vorbis, streamheader=(buffer)< > 01766f72626973000000000144ac0000000000008038010000000000b801, > 03766f726269731d000000586970682e4f7267206c6962566f7262697320492032303039303730390000000001, > 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> > > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = > application/ogg, streamheader=(buffer)< > 4f6767530002000000000000000015831c42000000004904efb7011e01766f72626973000000000144ac0000000000008038010000000000b801, > 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> > > WARNING: from element /GstPipeline:pipeline0/GstUDPSink:udpsink0: Internal > data flow problem. > Additional debug info: > gstbasesink.c(3343): gst_base_sink_chain_unlocked (): > /GstPipeline:pipeline0/GstUDPSink:udpsink0: > Received buffer without a new-segment. Assuming timestamps start from 0. > > > > > Thanks & Regards > Shamun > > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jedierikb at gmail.com Sat Dec 11 05:52:11 2010 From: jedierikb at gmail.com (Erik Blankinship) Date: Fri, 10 Dec 2010 23:52:11 -0500 Subject: [gst-devel] stopping video sets the background to black -- can I change this color? In-Reply-To: <4CCF2F66.2080304@hora-obscura.de> References: <4CCD8619.6030408@hora-obscura.de> <4CCF2F66.2080304@hora-obscura.de> Message-ID: I opened this bug on this issue: https://bugzilla.gnome.org/show_bug.cgi?id=635800 Tonight with the help of friendly folk on irc I was able to compile xvimagesink with the changes I think are needed. (I am on fedora 11) I seem to have created two files: - libgstxvimagesink.la - libgstxvimagesink_la-xvimagesink.lo Is that it? I thought I compiling would create a .so file.. If it worked, I have a followup question: I would like to use my newly compiled version of xvimagesink in a pygst project I distribute on the OLPC XO laptops (Which may not get upgraded gstreamer for a long time). How do I reference my bundled xvimagesink files in my pipelines? Thank you. It is fun and educational to learn about the gstreamer build process. On Mon, Nov 1, 2010 at 5:21 PM, Stefan Kost wrote: > Am 31.10.2010 21:18, schrieb Erik Blankinship: > > > > > > On Sun, Oct 31, 2010 at 11:07 AM, Stefan Kost > > wrote: > > > > hi, > > > > Am 31.10.2010 03:55, schrieb Erik Blankinship: > > > When I set my pipeline to state null (i.e. pressing stop), my > xvimagesink (or > > > ximagesink) changes to the color black. > > > > > > Can I set this color? > > > > > > > No, you would need to turn off some things on xvimagesink and manage > them > > yourself. I don't remember exactly which one: "autopaint-colorkey", > > "handle-expose" - you could check the source of the plugin. > > > > > > Just toggling those parameters had no effect (except, sometimes the video > > wouldn't show up at all). And just setting "colorkey" didn't seem to do > > anything either (but I was real excited for a minute hoping it would > work). > > Since we implemented the "colorkey" property, only a few drivers actually > do > something on it. Anyway that not what you where looking for, as the > colorkey is > a color you *don't* want to see. > > > > > Is the idea to listen for x expose events and hide the component? > > I said, please check the source: gst_xvimagesink_xwindow_clear() is > painting the > window black when going from READY_TO_NULL. It's something we should make > optional I think (and remove at some point). Its also inconsistent as apps > would > need to make their video window black initially if they want it to be > black. > File a bug? > > Stefan > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jedierikb at gmail.com Sat Dec 11 06:29:09 2010 From: jedierikb at gmail.com (Erik Blankinship) Date: Sat, 11 Dec 2010 00:29:09 -0500 Subject: [gst-devel] stopping video sets the background to black -- can I change this color? In-Reply-To: References: <4CCD8619.6030408@hora-obscura.de> <4CCF2F66.2080304@hora-obscura.de> Message-ID: > > > Tonight with the help > of friendly folk on irc I was able to compile xvimagesink with the changes I > think are needed. (I am on fedora 11) > > I seem to have created two files: > > - libgstxvimagesink.la > - libgstxvimagesink_la-xvimagesink.lo > > Is that it? I thought I compiling would create a .so file.. > > I found my compiled libgstxvimagesink.so file :-) And when I replace it in /usr/lib/ it solves my bug. However, I cannot replace this file on everyone's machine. How do I reference this modified plugin in my pipelines? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Sat Dec 11 07:17:11 2010 From: mcygogo at gmail.com (michael) Date: Sat, 11 Dec 2010 14:17:11 +0800 Subject: [gst-devel] How I test the FLV demux plugin Message-ID: Hi all: Now I am learning some about the flv plugin--flvdemux, I try to demux my flv test file like this: gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! queue ! filesink location=test.aes But when I run it , I just get the information like this: Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal data stream error. Additional debug info: gstflvdemux.c(2283): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstFlvDemux:demux: stream stopped, reason not-linked ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... I just want test the flvdemux plugin , So How I fixed it? tks:) -------------- next part -------------- An HTML attachment was scrubbed... URL: From jedierikb at gmail.com Sat Dec 11 07:33:48 2010 From: jedierikb at gmail.com (Erik Blankinship) Date: Sat, 11 Dec 2010 01:33:48 -0500 Subject: [gst-devel] stopping video sets the background to black -- can I change this color? In-Reply-To: References: <4CCD8619.6030408@hora-obscura.de> <4CCF2F66.2080304@hora-obscura.de> Message-ID: > > > I found my compiled libgstxvimagesink.so file :-) And when I replace it in > /usr/lib/ it solves my bug. > > However, I cannot replace this file on everyone's machine. How do I > reference this modified plugin in my pipelines? > [Sorry for flooding the list.] Some more research ( http://stackoverflow.com/questions/2120444/gstreamer-plugin-search-path ) suggests I need to set GST_PLUGIN_PATH in my activity. I am having a hard time figuring out how to do this with pygst. Any pygst experts out there who can nudge me in the right direction? -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Sat Dec 11 09:33:08 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Sat, 11 Dec 2010 14:03:08 +0530 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: References: Message-ID: Could you run the same pipeline with -v option and paste the log here ? On Sat, Dec 11, 2010 at 11:47 AM, michael wrote: > Hi all: > Now I am learning some about the flv plugin--flvdemux, I try to demux > my flv test file like this: > gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 ! > queue ! filesink location=test.ves demux.audio_00 ! queue ! filesink > location=test.aes > But when I run it , I just get the information like this: > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal data > stream error. > Additional debug info: > gstflvdemux.c(2283): gst_flv_demux_loop (): > /GstPipeline:pipeline0/GstFlvDemux:demux: > stream stopped, reason not-linked > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > Freeing pipeline ... > > I just want test the flvdemux plugin , So How I fixed it? tks:) > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Sat Dec 11 09:45:40 2010 From: bilboed at gmail.com (Edward Hervey) Date: Sat, 11 Dec 2010 09:45:40 +0100 Subject: [gst-devel] [NEVER WORKS] - /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. In-Reply-To: References: Message-ID: <1292057140.2568.5.camel@deumeu> On Fri, 2010-12-10 at 23:24 +0100, Shamun toha md wrote: > Dear Experts, > > I was trying a lot to get this work. But still have same issue with > OGGDEMUG. I also tried with simple codecs like SPEEX but its always > same error as following. Please advise. > > [root at example ~]# gst-launch -v udpsrc multicast-group=127.0.0.1 > port=4444 ! oggdemux ! vorbisdec ! audioconvert ! alsasink > > [sun at example tmp]$ gst-launch -v autoaudiosrc num-buffers=100 ! > speexenc ! oggmux ! udpsink host=127.0.0.1 auto-multicast=true > port=4444 > > [sun at example tmp]$ gst-launch -v autoaudiosrc ! audioconvert ! > vorbisenc ! oggmux ! udpsink host=127.0.0.1 auto-multicast=true > port=4444 > You're sending data over UDP which, unlike TCP, does not guarantee: * order of receiving packets * packets arriving at all So basically your client pipeline is receiving random junk, I wouldn't expect that to work. Switch to using either : * TCP elements (Which won't do multicast) * or RTP (de)payloaders * or just use gst-rtsp-server on your server and playbin2 on your client Edward > > Thanks & Regards > Shamun > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From mcygogo at gmail.com Sat Dec 11 09:49:37 2010 From: mcygogo at gmail.com (michael) Date: Sat, 11 Dec 2010 16:49:37 +0800 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: References: Message-ID: Hi sudarshan: I run it with the -v option, and the result like this: Setting pipeline to PAUSED ... Pipeline is PREROLLING ... /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = video/x-h264, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal data stream error. Additional debug info: gstflvdemux.c(2283): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstFlvDemux:demux: stream stopped, reason not-linked ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL Freeing pipeline ... I have found that the in function: gst_flv_demux_parse_tag_video() , the call gst_pad_push(demux->video_pad, outbuf) returned the -1; On Sat, Dec 11, 2010 at 4:33 PM, sudarshan bisht wrote: > Could you run the same pipeline with -v option and paste the log here ? > > > On Sat, Dec 11, 2010 at 11:47 AM, michael wrote: > >> Hi all: >> Now I am learning some about the flv plugin--flvdemux, I try to demux >> my flv test file like this: >> gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 >> ! queue ! filesink location=test.ves demux.audio_00 ! queue ! filesink >> location=test.aes >> But when I run it , I just get the information like this: >> Setting pipeline to PAUSED ... >> Pipeline is PREROLLING ... >> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal >> data stream error. >> Additional debug info: >> gstflvdemux.c(2283): gst_flv_demux_loop (): >> /GstPipeline:pipeline0/GstFlvDemux:demux: >> stream stopped, reason not-linked >> ERROR: pipeline doesn't want to preroll. >> Setting pipeline to NULL ... >> Freeing pipeline ... >> >> I just want test the flvdemux plugin , So How I fixed it? tks:) >> >> >> ------------------------------------------------------------------------------ >> Oracle to DB2 Conversion Guide: Learn learn about native support for >> PL/SQL, >> new data types, scalar functions, improved concurrency, built-in packages, >> OCI, SQL*Plus, data movement tools, best practices and more. >> http://p.sf.net/sfu/oracle-sfdev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > -- > Regards, > > Sudarshan Bisht > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shamun.toha at gmail.com Sat Dec 11 10:09:23 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sat, 11 Dec 2010 10:09:23 +0100 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 42 In-Reply-To: References: Message-ID: YES, now i tried debug 4/5 for specific oggdemux getting as following. And more details log 5 here: https://gist.github.com/737273 Debug 4: [sun at example ~]$ gst-launch -v --gst-debug=oggdemux:4 udpsrc multicast-group=127.0.0.1 port=4444 ! oggdemux ! vorbisdec ! audioconvert ! alsasink 0:00:00.029828546 3578 0x9035050 DEBUG oggdemux gstoggdemux.c:3310:gst_ogg_demux_plugin_init: binding text domain gst-plugins-base-0.10 to locale dir /usr/share/locale Setting pipeline to PAUSED ... 0:00:00.049287241 3578 0x9035050 DEBUG oggdemux gstoggdemux.c:3213:gst_ogg_demux_sink_activate: activating push Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock 0:00:00.502757325 3578 0x9192d38 DEBUG oggdemux gstoggdemux.c:1275:gst_ogg_demux_sink_event: got a new segment event 0:00:00.502912789 3578 0x9192d38 DEBUG oggdemux gstoggdemux.c:2864:gst_ogg_demux_chain: chain 0:00:00.502958464 3578 0x9192d38 DEBUG oggdemux gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 3996 bytes 0:00:00.503076704 3578 0x9192d38 WARN oggdemux gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown ogg pad for serial 20a147e6 detected WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 20a147e6 detected 0:00:01.022867801 3578 0x9192d38 DEBUG oggdemux gstoggdemux.c:2864:gst_ogg_demux_chain: chain 0:00:01.022950352 3578 0x9192d38 DEBUG oggdemux gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4107 bytes 0:00:01.023052948 3578 0x9192d38 WARN oggdemux gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown ogg pad for serial 20a147e6 detected WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 20a147e6 detected 0:00:01.508798084 3578 0x9192d38 DEBUG oggdemux gstoggdemux.c:2864:gst_ogg_demux_chain: chain 0:00:01.508856191 3578 0x9192d38 DEBUG oggdemux gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4283 bytes 0:00:01.508906336 3578 0x9192d38 WARN oggdemux gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown ogg pad for serial 20a147e6 detected WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 20a147e6 detected ^CCaught interrupt -- handling interrupt. Interrupt: Stopping pipeline ... Execution ended after 1749126474 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... [sun at example ~]$ Debug 5: [sun at example ~]$ gst-launch -v --gst-debug=oggdemux:5 udpsrc multicast-group=127.0.0.1 port=4444 ! oggdemux ! vorbisdec ! audioconvert ! alsasink 0:00:00.029476132 3581 0x9eee050 DEBUG oggdemux gstoggdemux.c:3310:gst_ogg_demux_plugin_init: binding text domain gst-plugins-base-0.10 to locale dir /usr/share/locale Setting pipeline to PAUSED ... 0:00:00.063592227 3581 0x9eee050 DEBUG oggdemux gstoggdemux.c:3213:gst_ogg_demux_sink_activate: activating push Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock 0:00:00.448037917 3581 0xa03ba28 DEBUG oggdemux gstoggdemux.c:1275:gst_ogg_demux_sink_event: got a new segment event 0:00:00.448196035 3581 0xa03ba28 DEBUG oggdemux gstoggdemux.c:2864:gst_ogg_demux_chain: chain 0:00:00.448242129 3581 0xa03ba28 DEBUG oggdemux gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4304 bytes 0:00:00.448371823 3581 0xa03ba28 LOG oggdemux gstoggdemux.c:2769:gst_ogg_demux_handle_page: processing ogg page (serial 20a147e6, pageno 643, granulepos 14205248, bos 0) 0:00:00.448430070 3581 0xa03ba28 WARN oggdemux gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown ogg pad for serial 20a147e6 detected WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 20a147e6 detected 0:00:00.959745085 3581 0xa03ba28 DEBUG oggdemux gstoggdemux.c:2864:gst_ogg_demux_chain: chain 0:00:00.959842373 3581 0xa03ba28 DEBUG oggdemux gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 3985 bytes 0:00:00.959935260 3581 0xa03ba28 LOG oggdemux gstoggdemux.c:2769:gst_ogg_demux_handle_page: processing ogg page (serial 20a147e6, pageno 644, granulepos 14226752, bos 0) 0:00:00.959992249 3581 0xa03ba28 WARN oggdemux gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown ogg pad for serial 20a147e6 detected WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 20a147e6 detected 0:00:01.472230434 3581 0xa03ba28 DEBUG oggdemux gstoggdemux.c:2864:gst_ogg_demux_chain: chain 0:00:01.472315918 3581 0xa03ba28 DEBUG oggdemux gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4084 bytes 0:00:01.472407410 3581 0xa03ba28 LOG oggdemux gstoggdemux.c:2769:gst_ogg_demux_handle_page: processing ogg page (serial 20a147e6, pageno 645, granulepos 14249280, bos 0) 0:00:01.472462583 3581 0xa03ba28 WARN oggdemux gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown ogg pad for serial 20a147e6 detected WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 20a147e6 detected 0:00:01.982984361 3581 0xa03ba28 DEBUG oggdemux gstoggdemux.c:2864:gst_ogg_demux_chain: chain 0:00:01.983093591 3581 0xa03ba28 DEBUG oggdemux gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4105 bytes 0:00:01.983188155 3581 0xa03ba28 LOG oggdemux gstoggdemux.c:2769:gst_ogg_demux_handle_page: processing ogg page (serial 20a147e6, pageno 646, granulepos 14271808, bos 0) 0:00:01.983244795 3581 0xa03ba28 WARN oggdemux gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown ogg pad for serial 20a147e6 detected WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 20a147e6 detected ^CCaught interrupt -- 0:00:02.470417436 3581 0xa03ba28 DEBUG oggdemux gstoggdemux.c:2864:gst_ogg_demux_chain: chain 0:00:02.470498660 3581 0xa03ba28 DEBUG oggdemux gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4148 bytes 0:00:02.470591128 3581 0xa03ba28 LOG oggdemux gstoggdemux.c:2769:gst_ogg_demux_handle_page: processing ogg page (serial 20a147e6, pageno 647, granulepos 14294336, bos 0) 0:00:02.470646581 3581 0xa03ba28 WARN oggdemux gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown ogg pad for serial 20a147e6 detected WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 20a147e6 detected handling interrupt. Interrupt: Stopping pipeline ... Execution ended after 2497263738 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... > Message: 2 > Date: Fri, 10 Dec 2010 17:37:23 -0500 > From: Timothy Braun > Subject: Re: [gst-devel] [NEVER WORKS] - > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode > stream. > To: Discussion of the development of GStreamer > > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > I'm not an expert, but have you tried running it with GST_DEBUG=3 (or even > 4) to see if any more details get dropped to the console? > > Tim > > On Fri, Dec 10, 2010 at 5:24 PM, Shamun toha md >wrote: > > > Dear Experts, > > > > I was trying a lot to get this work. But still have same issue with > > OGGDEMUG. I also tried with simple codecs like SPEEX but its always same > > error as following. Please advise. > > > > My system info: > > ================== > > [root at example ~]# uname -a > > Linux example 2.6.32.12-114.fc12.i686 #1 SMP Tue Apr 27 21:29:58 UTC 2010 > > i686 i686 i386 GNU/Linux > > > > Gstreamer info: > > ============= > > [root at example ~]# yum install gstreamer > > Loaded plugins: presto, refresh-packagekit > > Setting up Install Process > > Package gstreamer-0.10.29-1.fc12.i686 already installed and latest > version > > Nothing to do > > [root at example ~]# > > > > > > > > CLIENT - ERROR/BUG > > ================== > > > > [root at example ~]# gst-launch -v udpsrc multicast-group=127.0.0.1 > port=4444 > > ! oggdemux ! vorbisdec ! audioconvert ! alsasink > > Setting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstSystemClock > > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > > not decode stream. > > Additional debug info: > > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > > unknown ogg pad for serial 421c8315 detected > > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > > not decode stream. > > Additional debug info: > > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > > unknown ogg pad for serial 421c8315 detected > > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > > not decode stream. > > Additional debug info: > > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > > unknown ogg pad for serial 421c8315 detected > > ^CCaught interrupt -- handling interrupt. > > Interrupt: Stopping pipeline ... > > Execution ended after 1746771949 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > Setting pipeline to NULL ... > > Freeing pipeline ... > > [root at example ~]# > > > > > > > > SERVER - OK > > ============= > > > > > > [sun at example tmp]$ gst-launch -v autoaudiosrc num-buffers=100 ! speexenc > ! > > oggmux ! udpsink host=127.0.0.1 auto-multicast=true port=4444 > > Setting pipeline to PAUSED ... > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > actual-buffer-time = 23777210 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > actual-latency-time = 115986 > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: > caps > > = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > > caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstAudioSrcClock > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0: last-message = "Warning: > > speex is optimized for 8, 16 and 32 KHz" > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:sink: caps = > > audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > width=(int)16, > > depth=(int)16, rate=(int)44100, channels=(int)2 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0: > > caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:src: caps = > > audio/x-speex, rate=(int)44100, channels=(int)2, streamheader=(buffer)< > > > 5370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1996515720: caps = > > audio/x-speex, rate=(int)44100, channels=(int)2, streamheader=(buffer)< > > > 5370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = > > application/ogg, streamheader=(buffer)< > > > 4f676753000200000000000000008869007700000000aa8326b101505370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > 4f6767530000000000000000000088690077010000000c96225e01281f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = > > application/ogg, streamheader=(buffer)< > > > 4f676753000200000000000000008869007700000000aa8326b101505370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > 4f6767530000000000000000000088690077010000000c96225e01281f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > WARNING: from element /GstPipeline:pipeline0/GstUDPSink:udpsink0: > Internal > > data flow problem. > > Additional debug info: > > gstbasesink.c(3343): gst_base_sink_chain_unlocked (): > > /GstPipeline:pipeline0/GstUDPSink:udpsink0: > > Received buffer without a new-segment. Assuming timestamps start from 0. > > ^CCaught interrupt -- handling interrupt. > > Interrupt: Stopping pipeline ... > > Execution ended after 1126778907597 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = NULL > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1996515720: caps = > > NULL > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = NULL > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:src: caps = NULL > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:sink: caps = NULL > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: > caps > > = NULL > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > > caps = NULL > > Setting pipeline to NULL ... > > Freeing pipeline ... > > [sun at example tmp]$ clear > > > > [sun at example tmp]$ gst-launch -v autoaudiosrc ! audioconvert ! vorbisenc > ! > > oggmux ! udpsink host=127.0.0.1 auto-multicast=true port=4444 > > Setting pipeline to PAUSED ... > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > actual-buffer-time = 23777210 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > actual-latency-time = 115986 > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: > caps > > = audio/x-raw-float, endianness=(int)1234, width=(int)32, > rate=(int)44100, > > channels=(int)1 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > > caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, > > rate=(int)44100, channels=(int)1 > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstAudioSrcClock > > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = > > audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, > > channels=(int)1 > > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = > > audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, > > channels=(int)1 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0: > > caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, > > rate=(int)44100, channels=(int)1 > > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0: last-message = "encoding > at > > quality level 0.30" > > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0.GstPad:sink: caps = > > audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, > > channels=(int)1 > > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0.GstPad:src: caps = > > audio/x-vorbis, streamheader=(buffer)< > > 01766f72626973000000000144ac0000000000008038010000000000b801, > > > 03766f726269731d000000586970682e4f7267206c6962566f7262697320492032303039303730390000000001, > > > 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> > > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1109164821: caps = > > audio/x-vorbis, streamheader=(buffer)< > > 01766f72626973000000000144ac0000000000008038010000000000b801, > > > 03766f726269731d000000586970682e4f7267206c6962566f7262697320492032303039303730390000000001, > > > 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> > > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = > > application/ogg, streamheader=(buffer)< > > > 4f6767530002000000000000000015831c42000000004904efb7011e01766f72626973000000000144ac0000000000008038010000000000b801, > > > 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> > > > > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = > > application/ogg, streamheader=(buffer)< > > > 4f6767530002000000000000000015831c42000000004904efb7011e01766f72626973000000000144ac0000000000008038010000000000b801, > > > 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> > > > > WARNING: from element /GstPipeline:pipeline0/GstUDPSink:udpsink0: > Internal > > data flow problem. > > Additional debug info: > > gstbasesink.c(3343): gst_base_sink_chain_unlocked (): > > /GstPipeline:pipeline0/GstUDPSink:udpsink0: > > Received buffer without a new-segment. Assuming timestamps start from 0. > > > > > > > > > > Thanks & Regards > > Shamun > > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native support for > > PL/SQL, > > new data types, scalar functions, improved concurrency, built-in > packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > > ------------------------------ > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > > ------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > End of gstreamer-devel Digest, Vol 55, Issue 42 > *********************************************** > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shamun.toha at gmail.com Sat Dec 11 10:32:54 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sat, 11 Dec 2010 10:32:54 +0100 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 42 In-Reply-To: References: Message-ID: Hello, Here is the debug 4 info: Additional debug info: gstoggdemux.c(2838): gst_ogg_demux_handle_page (): /GstPipeline:pipeline0/GstOggDemux:oggdemux0: unknown ogg pad for serial 20a147e6 detected 0:00:45.313947600 3640 0x8361050 DEBUG GST_BUS gstbus.c:856:gst_bus_add_watch_full_unlocked: New source 0x84b5cf0 with id 118 0:00:45.313998164 3640 0x8361050 DEBUG GST_BUS gstbus.c:1056:gst_bus_poll: running mainloop 0x84883a8 0:00:45.314047541 3640 0x8361050 DEBUG GST_BUS gstbus.c:258:gst_bus_set_main_context: setting main context to 0x84bf240, GLib default context: 0x84bf240 0:00:45.750865868 3640 0x84b5db8 DEBUG GST_CLOCK gstclock.c:846:gst_clock_get_internal_time: internal time 0:55:58.447392596 0:00:45.750957708 3640 0x84b5db8 DEBUG GST_CLOCK gstclock.c:891:gst_clock_get_time: adjusted time 0:55:58.447392596 0:00:45.751007434 3640 0x84b5db8 DEBUG basesrc gstbasesrc.c:1988:gst_base_src_do_sync: no sync needed 0:00:45.751049478 3640 0x84b5db8 DEBUG basesrc gstbasesrc.c:2155:gst_base_src_get_range: buffer ok 0:00:45.751096969 3640 0x84b5db8 DEBUG oggdemux gstoggdemux.c:2864:gst_ogg_demux_chain: chain 0:00:45.751139641 3640 0x84b5db8 DEBUG oggdemux gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4159 bytes 0:00:45.751232808 3640 0x84b5db8 WARN oggdemux gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown ogg pad for serial 20a147e6 detected 0:00:45.751278343 3640 0x84b5db8 DEBUG GST_MESSAGE gstelement.c:1715:gst_element_message_full: start 0:00:45.751375700 3640 0x84b5db8 INFO GST_ERROR_SYSTEM gstelement.c:1746:gst_element_message_full: posting message: Could not decode stream. 0:00:45.751450150 3640 0x84b5db8 DEBUG GST_BUS gstbus.c:307:gst_bus_post: [msg 0x84b7200] posting on bus, type warning, GstMessageWarning, gerror=(GstGError)NULL, debug=(string)"gstoggdemux.c\(2838\):\ gst_ogg_demux_handle_page\ \(\):\ /GstPipeline:pipeline0/GstOggDemux:oggdemux0:\012unknown\ ogg\ pad\ for\ serial\ 20a147e6\ detected"; from source 0:00:45.751608687 3640 0x84b5db8 DEBUG bin gstbin.c:3002:gst_bin_handle_message_func: [msg 0x84b7200] handling child oggdemux0 message of type warning 0:00:45.751655061 3640 0x84b5db8 DEBUG bin gstbin.c:3276:gst_bin_handle_message_func: posting message upward 0:00:45.751698222 3640 0x84b5db8 DEBUG GST_BUS gstbus.c:307:gst_bus_post: [msg 0x84b7200] posting on bus, type warning, GstMessageWarning, gerror=(GstGError)NULL, debug=(string)"gstoggdemux.c\(2838\):\ gst_ogg_demux_handle_page\ \(\):\ /GstPipeline:pipeline0/GstOggDemux:oggdemux0:\012unknown\ ogg\ pad\ for\ serial\ 20a147e6\ detected"; from source 0:00:45.751789643 3640 0x84b5db8 DEBUG GST_BUS gstbus.c:337:gst_bus_post: [msg 0x84b7200] pushing on async queue 0:00:45.751835249 3640 0x84b5db8 DEBUG GST_BUS gstbus.c:342:gst_bus_post: [msg 0x84b7200] pushed on async queue 0:00:45.751897825 3640 0x84b5db8 DEBUG GST_BUS gstbus.c:333:gst_bus_post: [msg 0x84b7200] dropped 0:00:45.751944758 3640 0x84b5db8 INFO GST_ERROR_SYSTEM gstelement.c:1769:gst_element_message_full: posted warning message: Could not decode stream. 0:00:45.751949297 3640 0x8361050 DEBUG GST_BUS gstbus.c:498:gst_bus_timed_pop_filtered: got message 0x84b7200, warning, type mask is 4294967295 0:00:45.751998953 3640 0x84b5db8 DEBUG basesrc gstbasesrc.c:2114:gst_base_src_get_range: calling create offset 18446744073709551615 length 4096, time 0 0:00:45.752016344 3640 0x8361050 DEBUG GST_BUS gstbus.c:758:gst_bus_source_dispatch: source 0x84b5cf0 calling dispatch with 0x84b7200 0:00:45.752054546 3640 0x84b5db8 DEBUG GST_POLL gstpoll.c:1141:gst_poll_wait: timeout :99:99:99.999999999 0:00:45.752082831 3640 0x8361050 DEBUG GST_BUS gstbus.c:948:poll_func: mainloop 0x84883a8 quit 0:00:45.752142126 3640 0x8361050 DEBUG GST_BUS gstbus.c:763:gst_bus_source_dispatch: source 0x84b5cf0 handler returns 1 0:00:45.752190594 3640 0x8361050 DEBUG GST_BUS gstbus.c:1058:gst_bus_poll: mainloop stopped 0x84883a8 0:00:45.752232638 3640 0x8361050 DEBUG GST_BUS gstbus.c:1294:gst_bus_remove_signal_watch: removing signal watch 118 0:00:45.752282713 3640 0x8361050 DEBUG GST_BUS gstbus.c:784:gst_bus_source_finalize: finalize source 0x84b5cf0 0:00:45.752327900 3640 0x8361050 DEBUG GST_BUS gstbus.c:258:gst_bus_set_main_context: setting main context to (nil), GLib default context: 0x84bf240 0:00:45.752382934 3640 0x8361050 DEBUG GST_BUS gstbus.c:1071:gst_bus_poll: finished poll with message 0x84b7200 WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode stream. Additional debug info: Thanks & Regards Shamun > Message: 2 > Date: Fri, 10 Dec 2010 17:37:23 -0500 > From: Timothy Braun > Subject: Re: [gst-devel] [NEVER WORKS] - > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode > stream. > To: Discussion of the development of GStreamer > > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > I'm not an expert, but have you tried running it with GST_DEBUG=3 (or even > 4) to see if any more details get dropped to the console? > > Tim > > On Fri, Dec 10, 2010 at 5:24 PM, Shamun toha md >wrote: > > > Dear Experts, > > > > I was trying a lot to get this work. But still have same issue with > > OGGDEMUG. I also tried with simple codecs like SPEEX but its always same > > error as following. Please advise. > > > > My system info: > > ================== > > [root at example ~]# uname -a > > Linux example 2.6.32.12-114.fc12.i686 #1 SMP Tue Apr 27 21:29:58 UTC 2010 > > i686 i686 i386 GNU/Linux > > > > Gstreamer info: > > ============= > > [root at example ~]# yum install gstreamer > > Loaded plugins: presto, refresh-packagekit > > Setting up Install Process > > Package gstreamer-0.10.29-1.fc12.i686 already installed and latest > version > > Nothing to do > > [root at example ~]# > > > > > > > > CLIENT - ERROR/BUG > > ================== > > > > [root at example ~]# gst-launch -v udpsrc multicast-group=127.0.0.1 > port=4444 > > ! oggdemux ! vorbisdec ! audioconvert ! alsasink > > Setting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstSystemClock > > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > > not decode stream. > > Additional debug info: > > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > > unknown ogg pad for serial 421c8315 detected > > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > > not decode stream. > > Additional debug info: > > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > > unknown ogg pad for serial 421c8315 detected > > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > > not decode stream. > > Additional debug info: > > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > > unknown ogg pad for serial 421c8315 detected > > ^CCaught interrupt -- handling interrupt. > > Interrupt: Stopping pipeline ... > > Execution ended after 1746771949 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > Setting pipeline to NULL ... > > Freeing pipeline ... > > [root at example ~]# > > > > > > > > SERVER - OK > > ============= > > > > > > [sun at example tmp]$ gst-launch -v autoaudiosrc num-buffers=100 ! speexenc > ! > > oggmux ! udpsink host=127.0.0.1 auto-multicast=true port=4444 > > Setting pipeline to PAUSED ... > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > actual-buffer-time = 23777210 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > actual-latency-time = 115986 > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: > caps > > = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > > caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstAudioSrcClock > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0: last-message = "Warning: > > speex is optimized for 8, 16 and 32 KHz" > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:sink: caps = > > audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > width=(int)16, > > depth=(int)16, rate=(int)44100, channels=(int)2 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0: > > caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:src: caps = > > audio/x-speex, rate=(int)44100, channels=(int)2, streamheader=(buffer)< > > > 5370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1996515720: caps = > > audio/x-speex, rate=(int)44100, channels=(int)2, streamheader=(buffer)< > > > 5370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = > > application/ogg, streamheader=(buffer)< > > > 4f676753000200000000000000008869007700000000aa8326b101505370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > 4f6767530000000000000000000088690077010000000c96225e01281f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = > > application/ogg, streamheader=(buffer)< > > > 4f676753000200000000000000008869007700000000aa8326b101505370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > 4f6767530000000000000000000088690077010000000c96225e01281f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > WARNING: from element /GstPipeline:pipeline0/GstUDPSink:udpsink0: > Internal > > data flow problem. > > Additional debug info: > > gstbasesink.c(3343): gst_base_sink_chain_unlocked (): > > /GstPipeline:pipeline0/GstUDPSink:udpsink0: > > Received buffer without a new-segment. Assuming timestamps start from 0. > > ^CCaught interrupt -- handling interrupt. > > Interrupt: Stopping pipeline ... > > Execution ended after 1126778907597 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = NULL > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1996515720: caps = > > NULL > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = NULL > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:src: caps = NULL > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:sink: caps = NULL > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: > caps > > = NULL > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > > caps = NULL > > Setting pipeline to NULL ... > > Freeing pipeline ... > > [sun at example tmp]$ clear > > > > [sun at example tmp]$ gst-launch -v autoaudiosrc ! audioconvert ! vorbisenc > ! > > oggmux ! udpsink host=127.0.0.1 auto-multicast=true port=4444 > > Setting pipeline to PAUSED ... > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > actual-buffer-time = 23777210 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > actual-latency-time = 115986 > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: > caps > > = audio/x-raw-float, endianness=(int)1234, width=(int)32, > rate=(int)44100, > > channels=(int)1 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > > caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, > > rate=(int)44100, channels=(int)1 > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstAudioSrcClock > > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = > > audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, > > channels=(int)1 > > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = > > audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, > > channels=(int)1 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0: > > caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, > > rate=(int)44100, channels=(int)1 > > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0: last-message = "encoding > at > > quality level 0.30" > > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0.GstPad:sink: caps = > > audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)44100, > > channels=(int)1 > > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0.GstPad:src: caps = > > audio/x-vorbis, streamheader=(buffer)< > > 01766f72626973000000000144ac0000000000008038010000000000b801, > > > 03766f726269731d000000586970682e4f7267206c6962566f7262697320492032303039303730390000000001, > > > 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> > > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1109164821: caps = > > audio/x-vorbis, streamheader=(buffer)< > > 01766f72626973000000000144ac0000000000008038010000000000b801, > > > 03766f726269731d000000586970682e4f7267206c6962566f7262697320492032303039303730390000000001, > > > 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> > > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = > > application/ogg, streamheader=(buffer)< > > > 4f6767530002000000000000000015831c42000000004904efb7011e01766f72626973000000000144ac0000000000008038010000000000b801, > > > 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> > > > > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = > > application/ogg, streamheader=(buffer)< > > > 4f6767530002000000000000000015831c42000000004904efb7011e01766f72626973000000000144ac0000000000008038010000000000b801, > > > 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> > > > > WARNING: from element /GstPipeline:pipeline0/GstUDPSink:udpsink0: > Internal > > data flow problem. > > Additional debug info: > > gstbasesink.c(3343): gst_base_sink_chain_unlocked (): > > /GstPipeline:pipeline0/GstUDPSink:udpsink0: > > Received buffer without a new-segment. Assuming timestamps start from 0. > > > > > > > > > > Thanks & Regards > > Shamun > > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native support for > > PL/SQL, > > new data types, scalar functions, improved concurrency, built-in > packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > > ------------------------------ > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > > ------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > End of gstreamer-devel Digest, Vol 55, Issue 42 > *********************************************** > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shamun.toha at gmail.com Sat Dec 11 10:49:46 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sat, 11 Dec 2010 10:49:46 +0100 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 44 In-Reply-To: References: Message-ID: But why the following then works its in UDP, its doing multicast, exactly the same what i am trying with vorbis/ogg ? - When its transparent and no encode/decoder used it works, its the OGGDEMUX causing this. WHY THEN THE FOLLOWING CODE WORKS ??? 1, Sending my mic audio no encode/decode: ============================== [sun at example ~]$ gst-launch -v autoaudiosrc ! udpsink host=127.0.0.1 auto-multicast=true port=4444 Setting pipeline to PAUSED ... /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: actual-buffer-time = 23771768 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: actual-latency-time = 15986 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstAudioSrcClock /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 ^CCaught interrupt -- handling interrupt. Interrupt: Stopping pipeline ... Execution ended after 89601184492 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: caps = NULL /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: caps = NULL Setting pipeline to NULL ... Freeing pipeline ... 2, I can hear in my sound card my exact audio/ no drop nothing it works: ======================================================= [sun at example ~]$ gst-launch -v udpsrc multicast-group=127.0.0.1 port=4444 ! audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 ! autoaudiosink bash: syntax error near unexpected token `(' [sun at example ~]$ gst-launch -v udpsrc multicast-group=127.0.0.1 port=4444 ! audio/x-raw-int, endianness=1234, signed=true, width=16, depth=16, rate=44100, channels=2 ! autoaudiosinkSetting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock /GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0.GstGhostPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0.GstGhostPad:sink.GstProxyPad:proxypad0: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: volume = 0.010334 /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: mute = FALSE /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: volume = 0.031805 /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: mute = FALSE /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: volume = 0.038150 /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: mute = FALSE /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: volume = 1.000000 /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: mute = FALSE ^CCaught interrupt -- handling interrupt. Interrupt: Stopping pipeline ... Execution ended after 14477191074 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0.GstGhostPad:sink: caps = NULL /GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = NULL Setting pipeline to NULL ... Freeing pipeline ... [sun at example ~]$ Thanks & Regards Shamun On Sat, Dec 11, 2010 at 10:09 AM, < gstreamer-devel-request at lists.sourceforge.net> wrote: > Send gstreamer-devel mailing list submissions to > gstreamer-devel at lists.sourceforge.net > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > or, via email, send a message with subject or body 'help' to > gstreamer-devel-request at lists.sourceforge.net > > You can reach the person managing the list at > gstreamer-devel-owner at lists.sourceforge.net > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of gstreamer-devel digest..." > > > Today's Topics: > > 1. Re: [NEVER WORKS] - > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode > stream. (Edward Hervey) > 2. Re: How I test the FLV demux plugin (michael) > 3. Re: gstreamer-devel Digest, Vol 55, Issue 42 (Shamun toha md) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 11 Dec 2010 09:45:40 +0100 > From: Edward Hervey > Subject: Re: [gst-devel] [NEVER WORKS] - > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode > stream. > To: Discussion of the development of GStreamer > > Message-ID: <1292057140.2568.5.camel at deumeu> > Content-Type: text/plain; charset="us-ascii" > > On Fri, 2010-12-10 at 23:24 +0100, Shamun toha md wrote: > > Dear Experts, > > > > I was trying a lot to get this work. But still have same issue with > > OGGDEMUG. I also tried with simple codecs like SPEEX but its always > > same error as following. Please advise. > > > > > [root at example ~]# gst-launch -v udpsrc multicast-group=127.0.0.1 > > port=4444 ! oggdemux ! vorbisdec ! audioconvert ! alsasink > > > > > [sun at example tmp]$ gst-launch -v autoaudiosrc num-buffers=100 ! > > speexenc ! oggmux ! udpsink host=127.0.0.1 auto-multicast=true > > port=4444 > > > > [sun at example tmp]$ gst-launch -v autoaudiosrc ! audioconvert ! > > vorbisenc ! oggmux ! udpsink host=127.0.0.1 auto-multicast=true > > port=4444 > > > > You're sending data over UDP which, unlike TCP, does not guarantee: > * order of receiving packets > * packets arriving at all > > So basically your client pipeline is receiving random junk, I wouldn't > expect that to work. > > Switch to using either : > * TCP elements (Which won't do multicast) > * or RTP (de)payloaders > * or just use gst-rtsp-server on your server and playbin2 on your > client > > Edward > > > > > Thanks & Regards > > Shamun > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > > new data types, scalar functions, improved concurrency, built-in > packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------ > > Message: 2 > Date: Sat, 11 Dec 2010 16:49:37 +0800 > From: michael > Subject: Re: [gst-devel] How I test the FLV demux plugin > To: Discussion of the development of GStreamer > > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > Hi sudarshan: > I run it with the -v option, and the result like this: > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = video/x-h264, > pixel-aspect-ratio=(fraction)1/1, > > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal data > stream error. > Additional debug info: > gstflvdemux.c(2283): gst_flv_demux_loop (): > /GstPipeline:pipeline0/GstFlvDemux:demux: > stream stopped, reason not-linked > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL > Freeing pipeline ... > > I have found that the in function: gst_flv_demux_parse_tag_video() , the > call gst_pad_push(demux->video_pad, outbuf) returned the -1; > > On Sat, Dec 11, 2010 at 4:33 PM, sudarshan bisht > wrote: > > > Could you run the same pipeline with -v option and paste the log here ? > > > > > > On Sat, Dec 11, 2010 at 11:47 AM, michael wrote: > > > >> Hi all: > >> Now I am learning some about the flv plugin--flvdemux, I try to > demux > >> my flv test file like this: > >> gst-launch filesrc location=test.flv ! flvdemux name=demux > demux.video_00 > >> ! queue ! filesink location=test.ves demux.audio_00 ! queue ! filesink > >> location=test.aes > >> But when I run it , I just get the information like this: > >> Setting pipeline to PAUSED ... > >> Pipeline is PREROLLING ... > >> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal > >> data stream error. > >> Additional debug info: > >> gstflvdemux.c(2283): gst_flv_demux_loop (): > >> /GstPipeline:pipeline0/GstFlvDemux:demux: > >> stream stopped, reason not-linked > >> ERROR: pipeline doesn't want to preroll. > >> Setting pipeline to NULL ... > >> Freeing pipeline ... > >> > >> I just want test the flvdemux plugin , So How I fixed it? tks:) > >> > >> > >> > ------------------------------------------------------------------------------ > >> Oracle to DB2 Conversion Guide: Learn learn about native support for > >> PL/SQL, > >> new data types, scalar functions, improved concurrency, built-in > packages, > >> OCI, SQL*Plus, data movement tools, best practices and more. > >> http://p.sf.net/sfu/oracle-sfdev2dev > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >> > >> > > > > > > -- > > Regards, > > > > Sudarshan Bisht > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native support for > > PL/SQL, > > new data types, scalar functions, improved concurrency, built-in > packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > > ------------------------------ > > Message: 3 > Date: Sat, 11 Dec 2010 10:09:23 +0100 > From: Shamun toha md > Subject: Re: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 42 > To: gstreamer-devel at lists.sourceforge.net > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > YES, now i tried debug 4/5 for specific oggdemux getting as following. And > more details log 5 here: https://gist.github.com/737273 > > Debug 4: > > [sun at example ~]$ gst-launch -v --gst-debug=oggdemux:4 udpsrc > multicast-group=127.0.0.1 port=4444 ! oggdemux ! vorbisdec ! audioconvert ! > alsasink > > 0:00:00.029828546 3578 0x9035050 DEBUG oggdemux > gstoggdemux.c:3310:gst_ogg_demux_plugin_init: binding text domain > gst-plugins-base-0.10 to locale dir /usr/share/locale > Setting pipeline to PAUSED ... > 0:00:00.049287241 3578 0x9035050 DEBUG oggdemux > gstoggdemux.c:3213:gst_ogg_demux_sink_activate: activating > push > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > 0:00:00.502757325 3578 0x9192d38 DEBUG oggdemux > gstoggdemux.c:1275:gst_ogg_demux_sink_event: got a new segment > event > 0:00:00.502912789 3578 0x9192d38 DEBUG oggdemux > gstoggdemux.c:2864:gst_ogg_demux_chain: chain > 0:00:00.502958464 3578 0x9192d38 DEBUG oggdemux > gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 3996 > bytes > 0:00:00.503076704 3578 0x9192d38 WARN oggdemux > gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown > ogg pad for serial 20a147e6 detected > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > not decode stream. > Additional debug info: > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > unknown ogg pad for serial 20a147e6 detected > 0:00:01.022867801 3578 0x9192d38 DEBUG oggdemux > gstoggdemux.c:2864:gst_ogg_demux_chain: chain > 0:00:01.022950352 3578 0x9192d38 DEBUG oggdemux > gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4107 > bytes > 0:00:01.023052948 3578 0x9192d38 WARN oggdemux > gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown > ogg pad for serial 20a147e6 detected > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > not decode stream. > Additional debug info: > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > unknown ogg pad for serial 20a147e6 detected > 0:00:01.508798084 3578 0x9192d38 DEBUG oggdemux > gstoggdemux.c:2864:gst_ogg_demux_chain: chain > 0:00:01.508856191 3578 0x9192d38 DEBUG oggdemux > gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4283 > bytes > 0:00:01.508906336 3578 0x9192d38 WARN oggdemux > gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown > ogg pad for serial 20a147e6 detected > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > not decode stream. > Additional debug info: > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > unknown ogg pad for serial 20a147e6 detected > ^CCaught interrupt -- handling interrupt. > Interrupt: Stopping pipeline ... > Execution ended after 1749126474 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > Freeing pipeline ... > [sun at example ~]$ > > > Debug 5: > > [sun at example ~]$ gst-launch -v --gst-debug=oggdemux:5 udpsrc > multicast-group=127.0.0.1 port=4444 ! oggdemux ! vorbisdec ! audioconvert ! > alsasink > > 0:00:00.029476132 3581 0x9eee050 DEBUG oggdemux > gstoggdemux.c:3310:gst_ogg_demux_plugin_init: binding text domain > gst-plugins-base-0.10 to locale dir /usr/share/locale > Setting pipeline to PAUSED ... > 0:00:00.063592227 3581 0x9eee050 DEBUG oggdemux > gstoggdemux.c:3213:gst_ogg_demux_sink_activate: activating > push > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > 0:00:00.448037917 3581 0xa03ba28 DEBUG oggdemux > gstoggdemux.c:1275:gst_ogg_demux_sink_event: got a new segment > event > 0:00:00.448196035 3581 0xa03ba28 DEBUG oggdemux > gstoggdemux.c:2864:gst_ogg_demux_chain: chain > 0:00:00.448242129 3581 0xa03ba28 DEBUG oggdemux > gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4304 > bytes > 0:00:00.448371823 3581 0xa03ba28 LOG oggdemux > gstoggdemux.c:2769:gst_ogg_demux_handle_page: processing ogg > page > (serial 20a147e6, pageno 643, granulepos 14205248, bos 0) > 0:00:00.448430070 3581 0xa03ba28 WARN oggdemux > gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown > ogg pad for serial 20a147e6 detected > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > not decode stream. > Additional debug info: > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > unknown ogg pad for serial 20a147e6 detected > 0:00:00.959745085 3581 0xa03ba28 DEBUG oggdemux > gstoggdemux.c:2864:gst_ogg_demux_chain: chain > 0:00:00.959842373 3581 0xa03ba28 DEBUG oggdemux > gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 3985 > bytes > 0:00:00.959935260 3581 0xa03ba28 LOG oggdemux > gstoggdemux.c:2769:gst_ogg_demux_handle_page: processing ogg > page > (serial 20a147e6, pageno 644, granulepos 14226752, bos 0) > 0:00:00.959992249 3581 0xa03ba28 WARN oggdemux > gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown > ogg pad for serial 20a147e6 detected > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > not decode stream. > Additional debug info: > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > unknown ogg pad for serial 20a147e6 detected > 0:00:01.472230434 3581 0xa03ba28 DEBUG oggdemux > gstoggdemux.c:2864:gst_ogg_demux_chain: chain > 0:00:01.472315918 3581 0xa03ba28 DEBUG oggdemux > gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4084 > bytes > 0:00:01.472407410 3581 0xa03ba28 LOG oggdemux > gstoggdemux.c:2769:gst_ogg_demux_handle_page: processing ogg > page > (serial 20a147e6, pageno 645, granulepos 14249280, bos 0) > 0:00:01.472462583 3581 0xa03ba28 WARN oggdemux > gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown > ogg pad for serial 20a147e6 detected > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > not decode stream. > Additional debug info: > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > unknown ogg pad for serial 20a147e6 detected > 0:00:01.982984361 3581 0xa03ba28 DEBUG oggdemux > gstoggdemux.c:2864:gst_ogg_demux_chain: chain > 0:00:01.983093591 3581 0xa03ba28 DEBUG oggdemux > gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4105 > bytes > 0:00:01.983188155 3581 0xa03ba28 LOG oggdemux > gstoggdemux.c:2769:gst_ogg_demux_handle_page: processing ogg > page > (serial 20a147e6, pageno 646, granulepos 14271808, bos 0) > 0:00:01.983244795 3581 0xa03ba28 WARN oggdemux > gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown > ogg pad for serial 20a147e6 detected > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > not decode stream. > Additional debug info: > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > unknown ogg pad for serial 20a147e6 detected > ^CCaught interrupt -- 0:00:02.470417436 3581 0xa03ba28 DEBUG > oggdemux gstoggdemux.c:2864:gst_ogg_demux_chain: chain > 0:00:02.470498660 3581 0xa03ba28 DEBUG oggdemux > gstoggdemux.c:1314:gst_ogg_demux_submit_buffer: submitting 4148 > bytes > 0:00:02.470591128 3581 0xa03ba28 LOG oggdemux > gstoggdemux.c:2769:gst_ogg_demux_handle_page: processing ogg > page > (serial 20a147e6, pageno 647, granulepos 14294336, bos 0) > 0:00:02.470646581 3581 0xa03ba28 WARN oggdemux > gstoggdemux.c:2838:gst_ogg_demux_handle_page: warning: unknown > ogg pad for serial 20a147e6 detected > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could > not decode stream. > Additional debug info: > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > unknown ogg pad for serial 20a147e6 detected > handling interrupt. > Interrupt: Stopping pipeline ... > Execution ended after 2497263738 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > Freeing pipeline ... > > > > > > > > Message: 2 > > Date: Fri, 10 Dec 2010 17:37:23 -0500 > > From: Timothy Braun > > Subject: Re: [gst-devel] [NEVER WORKS] - > > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: Could not decode > > stream. > > To: Discussion of the development of GStreamer > > > > Message-ID: > > > > Content-Type: text/plain; charset="iso-8859-1" > > > > I'm not an expert, but have you tried running it with GST_DEBUG=3 (or > even > > 4) to see if any more details get dropped to the console? > > > > Tim > > > > On Fri, Dec 10, 2010 at 5:24 PM, Shamun toha md > >wrote: > > > > > Dear Experts, > > > > > > I was trying a lot to get this work. But still have same issue with > > > OGGDEMUG. I also tried with simple codecs like SPEEX but its always > same > > > error as following. Please advise. > > > > > > My system info: > > > ================== > > > [root at example ~]# uname -a > > > Linux example 2.6.32.12-114.fc12.i686 #1 SMP Tue Apr 27 21:29:58 UTC > 2010 > > > i686 i686 i386 GNU/Linux > > > > > > Gstreamer info: > > > ============= > > > [root at example ~]# yum install gstreamer > > > Loaded plugins: presto, refresh-packagekit > > > Setting up Install Process > > > Package gstreamer-0.10.29-1.fc12.i686 already installed and latest > > version > > > Nothing to do > > > [root at example ~]# > > > > > > > > > > > > CLIENT - ERROR/BUG > > > ================== > > > > > > [root at example ~]# gst-launch -v udpsrc multicast-group=127.0.0.1 > > port=4444 > > > ! oggdemux ! vorbisdec ! audioconvert ! alsasink > > > Setting pipeline to PAUSED ... > > > Pipeline is live and does not need PREROLL ... > > > Setting pipeline to PLAYING ... > > > New clock: GstSystemClock > > > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > Could > > > not decode stream. > > > Additional debug info: > > > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > > > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > > > unknown ogg pad for serial 421c8315 detected > > > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > Could > > > not decode stream. > > > Additional debug info: > > > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > > > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > > > unknown ogg pad for serial 421c8315 detected > > > WARNING: from element /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > Could > > > not decode stream. > > > Additional debug info: > > > gstoggdemux.c(2838): gst_ogg_demux_handle_page (): > > > /GstPipeline:pipeline0/GstOggDemux:oggdemux0: > > > unknown ogg pad for serial 421c8315 detected > > > ^CCaught interrupt -- handling interrupt. > > > Interrupt: Stopping pipeline ... > > > Execution ended after 1746771949 ns. > > > Setting pipeline to PAUSED ... > > > Setting pipeline to READY ... > > > Setting pipeline to NULL ... > > > Freeing pipeline ... > > > [root at example ~]# > > > > > > > > > > > > SERVER - OK > > > ============= > > > > > > > > > [sun at example tmp]$ gst-launch -v autoaudiosrc num-buffers=100 ! > speexenc > > ! > > > oggmux ! udpsink host=127.0.0.1 auto-multicast=true port=4444 > > > Setting pipeline to PAUSED ... > > > > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > > actual-buffer-time = 23777210 > > > > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > > actual-latency-time = 115986 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: > > caps > > > = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > > > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > > > > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > > > caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > > > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > > > Pipeline is live and does not need PREROLL ... > > > Setting pipeline to PLAYING ... > > > New clock: GstAudioSrcClock > > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0: last-message = "Warning: > > > speex is optimized for 8, 16 and 32 KHz" > > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:sink: caps = > > > audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > > width=(int)16, > > > depth=(int)16, rate=(int)44100, channels=(int)2 > > > > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0: > > > caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > > > width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 > > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:src: caps = > > > audio/x-speex, rate=(int)44100, channels=(int)2, streamheader=(buffer)< > > > > > > 5370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > > > > 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1996515720: caps = > > > audio/x-speex, rate=(int)44100, channels=(int)2, streamheader=(buffer)< > > > > > > 5370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > > > > 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = > > > application/ogg, streamheader=(buffer)< > > > > > > 4f676753000200000000000000008869007700000000aa8326b101505370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > > > > 4f6767530000000000000000000088690077010000000c96225e01281f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > > > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = > > > application/ogg, streamheader=(buffer)< > > > > > > 4f676753000200000000000000008869007700000000aa8326b101505370656578202020312e327263310000000000000000000000000000010000005000000044ac0000020000000400000002000000ffffffff800200000000000001000000000000000000000000000000, > > > > > > 4f6767530000000000000000000088690077010000000c96225e01281f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > > > > > > > WARNING: from element /GstPipeline:pipeline0/GstUDPSink:udpsink0: > > Internal > > > data flow problem. > > > Additional debug info: > > > gstbasesink.c(3343): gst_base_sink_chain_unlocked (): > > > /GstPipeline:pipeline0/GstUDPSink:udpsink0: > > > Received buffer without a new-segment. Assuming timestamps start from > 0. > > > ^CCaught interrupt -- handling interrupt. > > > Interrupt: Stopping pipeline ... > > > Execution ended after 1126778907597 ns. > > > Setting pipeline to PAUSED ... > > > Setting pipeline to READY ... > > > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = NULL > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1996515720: caps = > > > NULL > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:src: caps = NULL > > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:src: caps = NULL > > > /GstPipeline:pipeline0/GstSpeexEnc:speexenc0.GstPad:sink: caps = NULL > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: > > caps > > > = NULL > > > > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > > > caps = NULL > > > Setting pipeline to NULL ... > > > Freeing pipeline ... > > > [sun at example tmp]$ clear > > > > > > [sun at example tmp]$ gst-launch -v autoaudiosrc ! audioconvert ! > vorbisenc > > ! > > > oggmux ! udpsink host=127.0.0.1 auto-multicast=true port=4444 > > > Setting pipeline to PAUSED ... > > > > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > > actual-buffer-time = 23777210 > > > > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls: > > > actual-latency-time = 115986 > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src: > > caps > > > = audio/x-raw-float, endianness=(int)1234, width=(int)32, > > rate=(int)44100, > > > channels=(int)1 > > > > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src: > > > caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, > > > rate=(int)44100, channels=(int)1 > > > Pipeline is live and does not need PREROLL ... > > > Setting pipeline to PLAYING ... > > > New clock: GstAudioSrcClock > > > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = > > > audio/x-raw-float, endianness=(int)1234, width=(int)32, > rate=(int)44100, > > > channels=(int)1 > > > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps > = > > > audio/x-raw-float, endianness=(int)1234, width=(int)32, > rate=(int)44100, > > > channels=(int)1 > > > > > > /GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0: > > > caps = audio/x-raw-float, endianness=(int)1234, width=(int)32, > > > rate=(int)44100, channels=(int)1 > > > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0: last-message = > "encoding > > at > > > quality level 0.30" > > > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0.GstPad:sink: caps = > > > audio/x-raw-float, endianness=(int)1234, width=(int)32, > rate=(int)44100, > > > channels=(int)1 > > > /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0.GstPad:src: caps = > > > audio/x-vorbis, streamheader=(buffer)< > > > 01766f72626973000000000144ac0000000000008038010000000000b801, > > > > > > 03766f726269731d000000586970682e4f7267206c6962566f7262697320492032303039303730390000000001, > > > > > > 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> > > > > > > /GstPipeline:pipeline0/GstOggMux:oggmux0.GstPad:sink_1109164821: caps = > > > audio/x-vorbis, streamheader=(buffer)< > > > 01766f72626973000000000144ac0000000000008038010000000000b801, > > > > > > 03766f726269731d000000586970682e4f7267206c6962566f7262697320492032303039303730390000000001, > > > > > > 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> > > > > > > /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = > > > application/ogg, streamheader=(buffer)< > > > > > > 4f6767530002000000000000000015831c42000000004904efb7011e01766f72626973000000000144ac0000000000008038010000000000b801, > > > > > > 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> > > > > > > WARNING: from element /GstPipeline:pipeline0/GstUDPSink:udpsink0: > > Internal > > > data flow problem. > > > Additional debug info: > > > gstbasesink.c(3343): gst_base_sink_chain_unlocked (): > > > /GstPipeline:pipeline0/GstUDPSink:udpsink0: > > > Received buffer without a new-segment. Assuming timestamps start from > 0. > > > > > > > > > > > > > > > Thanks & Regards > > > Shamun > > > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 Conversion Guide: Learn learn about native support for > > > PL/SQL, > > > new data types, scalar functions, improved concurrency, built-in > > packages, > > > OCI, SQL*Plus, data movement tools, best practices and more. > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > > > ------------------------------ > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native support for > > PL/SQL, > > new data types, scalar functions, improved concurrency, built-in > packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > ------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > End of gstreamer-devel Digest, Vol 55, Issue 42 > > *********************************************** > > > -------------- next part -------------- > An HTML attachment was scrubbed... > > ------------------------------ > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > > ------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > End of gstreamer-devel Digest, Vol 55, Issue 44 > *********************************************** > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Sat Dec 11 11:05:06 2010 From: bilboed at gmail.com (Edward Hervey) Date: Sat, 11 Dec 2010 11:05:06 +0100 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 44 In-Reply-To: References: Message-ID: <1292061906.2568.12.camel@deumeu> On Sat, 2010-12-11 at 10:49 +0100, Shamun toha md wrote: > But why the following then works its in UDP, its doing multicast, > exactly the same what i am trying with vorbis/ogg ? Because if a raw audio packet gets dropped or reordered (which is insanely small in terms of samples [1]) you will not really noticed it and they are self-sufficient (they don't require the previous/next packest to be played back). But for compressed data one single packet dropped will cause corruption of the whole stream. > - When its transparent and no encode/decoder used it works, its the > OGGDEMUX causing this. WHY THEN THE FOLLOWING CODE WORKS ??? YOU SEEM TO HAVE A PROBLEM WITH YOUR CAPSLOCK KEY ! CAN YOU PLEASE FIX THAT ? THANKYOU Edward [1] a standard udp packet is 1500bytes, at 44100 samples per second x 2 (channels) x 2 (bytes per sample), that gives you around 8ms of audio per packet. 8ms of audio being dropped isn't really noticeable. From bisht.sudarshan at gmail.com Sat Dec 11 12:32:38 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Sat, 11 Dec 2010 13:32:38 +0200 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: References: Message-ID: Ok, now can you try following, gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 ! queue ! "video/x-h264, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb " ! filesink location=test.ves demux.audio_00 ! queue ! filesink location=test.aes On Sat, Dec 11, 2010 at 10:49 AM, michael wrote: > Hi sudarshan: > I run it with the -v option, and the result like this: > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = video/x-h264, > pixel-aspect-ratio=(fraction)1/1, > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal data > stream error. > Additional debug info: > gstflvdemux.c(2283): gst_flv_demux_loop (): > /GstPipeline:pipeline0/GstFlvDemux:demux: > stream stopped, reason not-linked > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL > Freeing pipeline ... > > I have found that the in function: gst_flv_demux_parse_tag_video() , the > call gst_pad_push(demux->video_pad, outbuf) returned the -1; > > On Sat, Dec 11, 2010 at 4:33 PM, sudarshan bisht < > bisht.sudarshan at gmail.com> wrote: > >> Could you run the same pipeline with -v option and paste the log here ? >> >> >> On Sat, Dec 11, 2010 at 11:47 AM, michael wrote: >> >>> Hi all: >>> Now I am learning some about the flv plugin--flvdemux, I try to >>> demux my flv test file like this: >>> gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 >>> ! queue ! filesink location=test.ves demux.audio_00 ! queue ! filesink >>> location=test.aes >>> But when I run it , I just get the information like this: >>> Setting pipeline to PAUSED ... >>> Pipeline is PREROLLING ... >>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal >>> data stream error. >>> Additional debug info: >>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>> stream stopped, reason not-linked >>> ERROR: pipeline doesn't want to preroll. >>> Setting pipeline to NULL ... >>> Freeing pipeline ... >>> >>> I just want test the flvdemux plugin , So How I fixed it? tks:) >>> >>> >>> ------------------------------------------------------------------------------ >>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>> PL/SQL, >>> new data types, scalar functions, improved concurrency, built-in >>> packages, >>> OCI, SQL*Plus, data movement tools, best practices and more. >>> http://p.sf.net/sfu/oracle-sfdev2dev >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> -- >> Regards, >> >> Sudarshan Bisht >> >> >> ------------------------------------------------------------------------------ >> Oracle to DB2 Conversion Guide: Learn learn about native support for >> PL/SQL, >> new data types, scalar functions, improved concurrency, built-in packages, >> OCI, SQL*Plus, data movement tools, best practices and more. >> http://p.sf.net/sfu/oracle-sfdev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Sat Dec 11 14:46:04 2010 From: mcygogo at gmail.com (michael) Date: Sat, 11 Dec 2010 21:46:04 +0800 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: References: Message-ID: Hi sudarshan: I followed your command, I got information : WARNING: erroneous pipeline: could not parse caps "video/x-h264,\ pixel-aspect-ratio=(fraction)1/1,\ codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" On Sat, Dec 11, 2010 at 7:32 PM, sudarshan bisht wrote: > Ok, now can you try following, > > gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 ! > queue ! "video/x-h264, pixel-aspect-ratio=(fraction)1/1, > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > " ! filesink location=test.ves demux.audio_00 ! queue ! filesink > location=test.aes > > > On Sat, Dec 11, 2010 at 10:49 AM, michael wrote: > >> Hi sudarshan: >> I run it with the -v option, and the result like this: >> Setting pipeline to PAUSED ... >> Pipeline is PREROLLING ... >> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = >> video/x-h264, pixel-aspect-ratio=(fraction)1/1, >> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal >> data stream error. >> Additional debug info: >> gstflvdemux.c(2283): gst_flv_demux_loop (): >> /GstPipeline:pipeline0/GstFlvDemux:demux: >> stream stopped, reason not-linked >> ERROR: pipeline doesn't want to preroll. >> Setting pipeline to NULL ... >> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL >> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL >> Freeing pipeline ... >> >> I have found that the in function: gst_flv_demux_parse_tag_video() , the >> call gst_pad_push(demux->video_pad, outbuf) returned the -1; >> >> On Sat, Dec 11, 2010 at 4:33 PM, sudarshan bisht < >> bisht.sudarshan at gmail.com> wrote: >> >>> Could you run the same pipeline with -v option and paste the log here ? >>> >>> >>> On Sat, Dec 11, 2010 at 11:47 AM, michael wrote: >>> >>>> Hi all: >>>> Now I am learning some about the flv plugin--flvdemux, I try to >>>> demux my flv test file like this: >>>> gst-launch filesrc location=test.flv ! flvdemux name=demux >>>> demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! queue ! >>>> filesink location=test.aes >>>> But when I run it , I just get the information like this: >>>> Setting pipeline to PAUSED ... >>>> Pipeline is PREROLLING ... >>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal >>>> data stream error. >>>> Additional debug info: >>>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>>> stream stopped, reason not-linked >>>> ERROR: pipeline doesn't want to preroll. >>>> Setting pipeline to NULL ... >>>> Freeing pipeline ... >>>> >>>> I just want test the flvdemux plugin , So How I fixed it? tks:) >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>> PL/SQL, >>>> new data types, scalar functions, improved concurrency, built-in >>>> packages, >>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> >>> -- >>> Regards, >>> >>> Sudarshan Bisht >>> >>> >>> ------------------------------------------------------------------------------ >>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>> PL/SQL, >>> new data types, scalar functions, improved concurrency, built-in >>> packages, >>> OCI, SQL*Plus, data movement tools, best practices and more. >>> http://p.sf.net/sfu/oracle-sfdev2dev >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> ------------------------------------------------------------------------------ >> Oracle to DB2 Conversion Guide: Learn learn about native support for >> PL/SQL, >> new data types, scalar functions, improved concurrency, built-in packages, >> OCI, SQL*Plus, data movement tools, best practices and more. >> http://p.sf.net/sfu/oracle-sfdev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > -- > Regards, > > Sudarshan Bisht > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Sat Dec 11 14:46:24 2010 From: mcygogo at gmail.com (michael) Date: Sat, 11 Dec 2010 21:46:24 +0800 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: References: Message-ID: and what the string means? On Sat, Dec 11, 2010 at 9:46 PM, michael wrote: > Hi sudarshan: > I followed your command, I got information : > WARNING: erroneous pipeline: could not parse caps "video/x-h264,\ > pixel-aspect-ratio=(fraction)1/1,\ > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" > > > > On Sat, Dec 11, 2010 at 7:32 PM, sudarshan bisht < > bisht.sudarshan at gmail.com> wrote: > >> Ok, now can you try following, >> >> gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 >> ! queue ! "video/x-h264, pixel-aspect-ratio=(fraction)1/1, >> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >> " ! filesink location=test.ves demux.audio_00 ! queue ! filesink >> location=test.aes >> >> >> On Sat, Dec 11, 2010 at 10:49 AM, michael wrote: >> >>> Hi sudarshan: >>> I run it with the -v option, and the result like this: >>> Setting pipeline to PAUSED ... >>> Pipeline is PREROLLING ... >>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = >>> video/x-h264, pixel-aspect-ratio=(fraction)1/1, >>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal >>> data stream error. >>> Additional debug info: >>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>> stream stopped, reason not-linked >>> ERROR: pipeline doesn't want to preroll. >>> Setting pipeline to NULL ... >>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL >>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL >>> Freeing pipeline ... >>> >>> I have found that the in function: gst_flv_demux_parse_tag_video() , the >>> call gst_pad_push(demux->video_pad, outbuf) returned the -1; >>> >>> On Sat, Dec 11, 2010 at 4:33 PM, sudarshan bisht < >>> bisht.sudarshan at gmail.com> wrote: >>> >>>> Could you run the same pipeline with -v option and paste the log here ? >>>> >>>> >>>> On Sat, Dec 11, 2010 at 11:47 AM, michael wrote: >>>> >>>>> Hi all: >>>>> Now I am learning some about the flv plugin--flvdemux, I try to >>>>> demux my flv test file like this: >>>>> gst-launch filesrc location=test.flv ! flvdemux name=demux >>>>> demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! queue ! >>>>> filesink location=test.aes >>>>> But when I run it , I just get the information like this: >>>>> Setting pipeline to PAUSED ... >>>>> Pipeline is PREROLLING ... >>>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal >>>>> data stream error. >>>>> Additional debug info: >>>>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>>>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>>>> stream stopped, reason not-linked >>>>> ERROR: pipeline doesn't want to preroll. >>>>> Setting pipeline to NULL ... >>>>> Freeing pipeline ... >>>>> >>>>> I just want test the flvdemux plugin , So How I fixed it? tks:) >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>>> PL/SQL, >>>>> new data types, scalar functions, improved concurrency, built-in >>>>> packages, >>>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> gstreamer-devel at lists.sourceforge.net >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> >>>> Sudarshan Bisht >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>> PL/SQL, >>>> new data types, scalar functions, improved concurrency, built-in >>>> packages, >>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>> PL/SQL, >>> new data types, scalar functions, improved concurrency, built-in >>> packages, >>> OCI, SQL*Plus, data movement tools, best practices and more. >>> http://p.sf.net/sfu/oracle-sfdev2dev >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> -- >> Regards, >> >> Sudarshan Bisht >> >> >> ------------------------------------------------------------------------------ >> Oracle to DB2 Conversion Guide: Learn learn about native support for >> PL/SQL, >> new data types, scalar functions, improved concurrency, built-in packages, >> OCI, SQL*Plus, data movement tools, best practices and more. >> http://p.sf.net/sfu/oracle-sfdev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shamun.toha at gmail.com Sat Dec 11 16:01:32 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sat, 11 Dec 2010 16:01:32 +0100 Subject: [gst-devel] CentOS 64 bit, how to get latest version 0.10.31 ? Message-ID: Hello, I am not sure what is that mean, but i thought i should ask it in mailing to make sure. I used an example in fedora works, but tried in CentOS 5.4 it does not work. Fedora gstreamer.i686 0.10.29-1.fc12 (working my test code) CentOS 5.4 gstreamer-0.10.20-3.el5.x86_64.rpm (does not work) CentOS 6.0 gstreamer-0.10.26-1.el6.x86_64.rpm (i am confused) How can i then test in CentOS, the latest Gstreamer releases ? Is there any guides or how tos to achieve that ? Or its impossible because CentOS 6.0 will sitll use the old version ? I am little bit confused now, will it be possible to have the latest and greatest ? Thanks & Regards Shamun -- ************************************************************************ The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination, or other use of this information, or taking of any action in reliance upon it by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. -------------- next part -------------- An HTML attachment was scrubbed... URL: From c.r.n.a at wanadoo.fr Sat Dec 11 16:31:33 2010 From: c.r.n.a at wanadoo.fr (Nicolas) Date: Sat, 11 Dec 2010 16:31:33 +0100 Subject: [gst-devel] Fakesink and GstNavigation In-Reply-To: <4CFFD7A0.9050201@hora-obscura.de> References: <4CFF7435.9020000@wanadoo.fr> <4CFFD7A0.9050201@hora-obscura.de> Message-ID: <4D039955.3010305@wanadoo.fr> Hi, Thanks for your help. I finally implement the navigation successfully ! Nicolas. From braunsquared at gmail.com Sat Dec 11 18:25:01 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Sat, 11 Dec 2010 12:25:01 -0500 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: References: Message-ID: <6721B70C-FFDF-444D-91A6-4E6BF7297EFE@gmail.com> Your best bet is to use gst-inspect and look at what the various templates of the pads are. It sounds like you need to do some conversion somewhere. Sent from my iPhone On Dec 11, 2010, at 8:46 AM, michael wrote: > Hi sudarshan: > I followed your command, I got information : > WARNING: erroneous pipeline: could not parse caps "video/x-h264,\ pixel-aspect-ratio=(fraction)1/1,\ codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" > > > > On Sat, Dec 11, 2010 at 7:32 PM, sudarshan bisht wrote: > Ok, now can you try following, > > gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 ! queue ! "video/x-h264, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb " ! filesink location=test.ves demux.audio_00 ! queue ! filesink location=test.aes > > > On Sat, Dec 11, 2010 at 10:49 AM, michael wrote: > Hi sudarshan: > I run it with the -v option, and the result like this: > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = video/x-h264, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal data stream error. > Additional debug info: > gstflvdemux.c(2283): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstFlvDemux:demux: > stream stopped, reason not-linked > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL > Freeing pipeline ... > > I have found that the in function: gst_flv_demux_parse_tag_video() , the call gst_pad_push(demux->video_pad, outbuf) returned the -1; > > On Sat, Dec 11, 2010 at 4:33 PM, sudarshan bisht wrote: > Could you run the same pipeline with -v option and paste the log here ? > > > On Sat, Dec 11, 2010 at 11:47 AM, michael wrote: > Hi all: > Now I am learning some about the flv plugin--flvdemux, I try to demux my flv test file like this: > gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! queue ! filesink location=test.aes > But when I run it , I just get the information like this: > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal data stream error. > Additional debug info: > gstflvdemux.c(2283): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstFlvDemux:demux: > stream stopped, reason not-linked > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > Freeing pipeline ... > > I just want test the flvdemux plugin , So How I fixed it? tks:) > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > Regards, > > Sudarshan Bisht > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > Regards, > > Sudarshan Bisht > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From smcnam at gmail.com Sat Dec 11 18:26:21 2010 From: smcnam at gmail.com (Sean McNamara) Date: Sat, 11 Dec 2010 12:26:21 -0500 Subject: [gst-devel] CentOS 64 bit, how to get latest version 0.10.31 ? In-Reply-To: References: Message-ID: Hi, On Sat, Dec 11, 2010 at 10:01 AM, Shamun toha md wrote: > Hello, > > I am not sure what is that mean, but i thought i should ask it in mailing to > make sure. I used an example in fedora works, but tried in CentOS 5.4 it > does not work. If I understand you correctly, you're saying that a certain pipeline example or gstreamer application works in Fedora 12, but not in CentOS 5.4. That is a valid reason for wanting to upgrade your gstreamer version, though you might simply be missing a certain plugin, or hitting an easily fixable bug. Without specific details about what "does not work" in particular, we can't help you more than that. BTW, CentOS 6.0 does not exist yet, only RHEL 6.0... > > Fedora???????? gstreamer.i686? 0.10.29-1.fc12 (working my test code) > > CentOS 5.4? gstreamer-0.10.20-3.el5.x86_64.rpm? (does not work) > CentOS 6.0? gstreamer-0.10.26-1.el6.x86_64.rpm? (i am confused) I'm confused, too! You aren't explaining yourself very well. You seem to be grabbing random RPMs and trying them? For your info, it's probably better to use a package manager, such as yum. Getting RPMs off of pbone or other RPM finding services does not guarantee that the built binaries you install will be compatible with the rest of your stack (i.e. that package's dependencies). > > How can i then test in CentOS, the latest Gstreamer releases ? Is there any > guides or how tos to achieve that ? Or its impossible because CentOS 6.0 > will sitll use the old version ? > > I am little bit confused now, will it be possible to have the latest and > greatest ? With free software, it is always possible :) The real question is whether you are willing to work for the desired result. Since Gstreamer is in userspace, and its dependencies are pretty flexible, it should be "easy" to upgrade the version by compiling from source. Another possibility is that you could find a repository online that has updated binary packages, though I don't know where such a repository would be, and such a thing wouldn't be supported by the Gstreamer project maintainers. Nor would it be supported by CentOS project or Red Hat. Compiling from vanilla sources might lend you some unofficial advice/support here among interested gstreamer users, though. Gstreamer project itself used to distribute binaries for RHEL/CentOS, but it seems they haven't been updated since 2008; for example, see: http://gstreamer.freedesktop.org/pkg/rhel/5/0.10/i386/gst/ (but DON'T install these; they are very old!) The short story is that yes, you can of course install the latest gstreamer and plugins from source, provided your system has the required dependencies (and the minimum versions), or provided that you compile the required dependencies yourself. This can take a lot of time, especially if you've never done it before, and special considerations are required if you plan to do this on many different computers, e.g. in an IT environment. In that case you may want to build your own RPM packages of it on a development box, and simply install them on each of the client computers. Your email does not really provide enough information for me to offer any further advice, but I hope this gets you started as to your options.... Sean > > Thanks & Regards > Shamun > > -- > > ************************************************************************ > The information transmitted is intended only for the person or entity to > which it is addressed and may contain confidential and/or privileged > material. > Any review, retransmission, dissemination, or other use of this information, > or taking of any action in reliance upon it by persons or entities > other than the intended recipient is prohibited. If you received this in > error, please contact the sender and delete the material from any computer. > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From josel.segura at gmx.es Sat Dec 11 19:40:19 2010 From: josel.segura at gmx.es (=?ISO-8859-1?B?Sm9z6Q==?= Luis Segura Lucas) Date: Sat, 11 Dec 2010 19:40:19 +0100 Subject: [gst-devel] Problem with xvimagesink and GTkWindow Message-ID: <20101211194019.042719e1@peter> Hello! I am using Gstreamer and GTK Python bindings to develop a program. In brief, my program create a GTK Window and put the xvimagesink on it using set_xwindow_id. When I close the GTK Window I need to stop the Gstreamer pipeline. Under certain circumstances, the program needs to re-play the pipeline and re-open the closed GTK Window. My problem is that, if I use the GTK Window "show" function after closing the window, this window doesn't emit signals like "show", "realize" and, the most important, "hide" (this callback is needed to stop the pipeline). I know that is more a GTK problem than a Gstreamer one, but I suppose I'm not the first on the list that have a similar problem. Regards, thanks in advance and, if you think it is off-topic, sorry o:-) From bisht.sudarshan at gmail.com Sun Dec 12 08:53:11 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Sun, 12 Dec 2010 13:23:11 +0530 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: <6721B70C-FFDF-444D-91A6-4E6BF7297EFE@gmail.com> References: <6721B70C-FFDF-444D-91A6-4E6BF7297EFE@gmail.com> Message-ID: I suggest you to use capsfilter, as you said in your earlier mails that gst_pad_push returns -1 , that means flvdemux is not able to push buffer to the next element and it may happen because of lack of caps negotiations. By the way how did you create this test.flv file ? and are you able to play this file with other players ? and give a try to following pipeline; gst-launch -v playbin uri=file:///path/test.flv On Sat, Dec 11, 2010 at 10:55 PM, Timothy Braun wrote: > Your best bet is to use gst-inspect and look at what the various templates > of the pads are. It sounds like you need to do some conversion somewhere. > > Sent from my iPhone > > On Dec 11, 2010, at 8:46 AM, michael wrote: > > Hi sudarshan: > I followed your command, I got information : > WARNING: erroneous pipeline: could not parse caps "video/x-h264,\ > pixel-aspect-ratio=(fraction)1/1,\ > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" > > > > On Sat, Dec 11, 2010 at 7:32 PM, sudarshan bisht < > bisht.sudarshan at gmail.com> wrote: > >> Ok, now can you try following, >> >> gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 >> ! queue ! "video/x-h264, pixel-aspect-ratio=(fraction)1/1, >> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >> " ! filesink location=test.ves demux.audio_00 ! queue ! filesink >> location=test.aes >> >> >> On Sat, Dec 11, 2010 at 10:49 AM, michael < >> mcygogo at gmail.com> wrote: >> >>> Hi sudarshan: >>> I run it with the -v option, and the result like this: >>> Setting pipeline to PAUSED ... >>> Pipeline is PREROLLING ... >>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = >>> video/x-h264, pixel-aspect-ratio=(fraction)1/1, >>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal >>> data stream error. >>> Additional debug info: >>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>> stream stopped, reason not-linked >>> ERROR: pipeline doesn't want to preroll. >>> Setting pipeline to NULL ... >>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL >>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL >>> Freeing pipeline ... >>> >>> I have found that the in function: gst_flv_demux_parse_tag_video() , the >>> call gst_pad_push(demux->video_pad, outbuf) returned the -1; >>> >>> On Sat, Dec 11, 2010 at 4:33 PM, sudarshan bisht < >>> bisht.sudarshan at gmail.com> wrote: >>> >>>> Could you run the same pipeline with -v option and paste the log here ? >>>> >>>> >>>> On Sat, Dec 11, 2010 at 11:47 AM, michael < >>>> mcygogo at gmail.com> wrote: >>>> >>>>> Hi all: >>>>> Now I am learning some about the flv plugin--flvdemux, I try to >>>>> demux my flv test file like this: >>>>> gst-launch filesrc location=test.flv ! flvdemux name=demux >>>>> demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! queue ! >>>>> filesink location=test.aes >>>>> But when I run it , I just get the information like this: >>>>> Setting pipeline to PAUSED ... >>>>> Pipeline is PREROLLING ... >>>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal >>>>> data stream error. >>>>> Additional debug info: >>>>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>>>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>>>> stream stopped, reason not-linked >>>>> ERROR: pipeline doesn't want to preroll. >>>>> Setting pipeline to NULL ... >>>>> Freeing pipeline ... >>>>> >>>>> I just want test the flvdemux plugin , So How I fixed it? tks:) >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>>> PL/SQL, >>>>> new data types, scalar functions, improved concurrency, built-in >>>>> packages, >>>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>>> >>>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> >>>>> gstreamer-devel at lists.sourceforge.net >>>>> >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> >>>> Sudarshan Bisht >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>> PL/SQL, >>>> new data types, scalar functions, improved concurrency, built-in >>>> packages, >>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>> >>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> >>>> gstreamer-devel at lists.sourceforge.net >>>> >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>> PL/SQL, >>> new data types, scalar functions, improved concurrency, built-in >>> packages, >>> OCI, SQL*Plus, data movement tools, best practices and more. >>> >>> http://p.sf.net/sfu/oracle-sfdev2dev >>> _______________________________________________ >>> gstreamer-devel mailing list >>> >>> gstreamer-devel at lists.sourceforge.net >>> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> -- >> Regards, >> >> Sudarshan Bisht >> >> >> ------------------------------------------------------------------------------ >> Oracle to DB2 Conversion Guide: Learn learn about native support for >> PL/SQL, >> new data types, scalar functions, improved concurrency, built-in packages, >> OCI, SQL*Plus, data movement tools, best practices and more. >> >> http://p.sf.net/sfu/oracle-sfdev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Sun Dec 12 10:16:14 2010 From: mcygogo at gmail.com (michael) Date: Sun, 12 Dec 2010 17:16:14 +0800 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: References: <6721B70C-FFDF-444D-91A6-4E6BF7297EFE@gmail.com> Message-ID: Hi sadarshan: I can play the file by your command: gst-launch -v playbin uri=file: ///path/test.flv On Sun, Dec 12, 2010 at 3:53 PM, sudarshan bisht wrote: > I suggest you to use capsfilter, as you said in your earlier mails > that gst_pad_push returns -1 , that means flvdemux is not able to push > buffer to the next element and it may happen because of lack of > caps negotiations. > > By the way how did you create this test.flv file ? and are you able to play > this file with other players ? > > and give a try to following pipeline; > gst-launch -v playbin uri=file:///path/test.flv > > > > On Sat, Dec 11, 2010 at 10:55 PM, Timothy Braun wrote: > >> Your best bet is to use gst-inspect and look at what the various templates >> of the pads are. It sounds like you need to do some conversion somewhere. >> >> Sent from my iPhone >> >> On Dec 11, 2010, at 8:46 AM, michael wrote: >> >> Hi sudarshan: >> I followed your command, I got information : >> WARNING: erroneous pipeline: could not parse caps "video/x-h264,\ >> pixel-aspect-ratio=(fraction)1/1,\ >> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" >> >> >> >> On Sat, Dec 11, 2010 at 7:32 PM, sudarshan bisht < >> bisht.sudarshan at gmail.com> wrote: >> >>> Ok, now can you try following, >>> >>> gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 >>> ! queue ! "video/x-h264, pixel-aspect-ratio=(fraction)1/1, >>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >>> " ! filesink location=test.ves demux.audio_00 ! queue ! filesink >>> location=test.aes >>> >>> >>> On Sat, Dec 11, 2010 at 10:49 AM, michael < >>> mcygogo at gmail.com> wrote: >>> >>>> Hi sudarshan: >>>> I run it with the -v option, and the result like this: >>>> Setting pipeline to PAUSED ... >>>> Pipeline is PREROLLING ... >>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = >>>> video/x-h264, pixel-aspect-ratio=(fraction)1/1, >>>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal >>>> data stream error. >>>> Additional debug info: >>>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>>> stream stopped, reason not-linked >>>> ERROR: pipeline doesn't want to preroll. >>>> Setting pipeline to NULL ... >>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL >>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL >>>> Freeing pipeline ... >>>> >>>> I have found that the in function: gst_flv_demux_parse_tag_video() , the >>>> call gst_pad_push(demux->video_pad, outbuf) returned the -1; >>>> >>>> On Sat, Dec 11, 2010 at 4:33 PM, sudarshan bisht < >>>> bisht.sudarshan at gmail.com> wrote: >>>> >>>>> Could you run the same pipeline with -v option and paste the log here >>>>> ? >>>>> >>>>> >>>>> On Sat, Dec 11, 2010 at 11:47 AM, michael < >>>>> mcygogo at gmail.com> wrote: >>>>> >>>>>> Hi all: >>>>>> Now I am learning some about the flv plugin--flvdemux, I try to >>>>>> demux my flv test file like this: >>>>>> gst-launch filesrc location=test.flv ! flvdemux name=demux >>>>>> demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! queue ! >>>>>> filesink location=test.aes >>>>>> But when I run it , I just get the information like this: >>>>>> Setting pipeline to PAUSED ... >>>>>> Pipeline is PREROLLING ... >>>>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal >>>>>> data stream error. >>>>>> Additional debug info: >>>>>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>>>>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>>>>> stream stopped, reason not-linked >>>>>> ERROR: pipeline doesn't want to preroll. >>>>>> Setting pipeline to NULL ... >>>>>> Freeing pipeline ... >>>>>> >>>>>> I just want test the flvdemux plugin , So How I fixed it? tks:) >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>>>> PL/SQL, >>>>>> new data types, scalar functions, improved concurrency, built-in >>>>>> packages, >>>>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>>>> >>>>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> >>>>> Sudarshan Bisht >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>>> PL/SQL, >>>>> new data types, scalar functions, improved concurrency, built-in >>>>> packages, >>>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>>> >>>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> >>>>> gstreamer-devel at lists.sourceforge.net >>>>> >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>> PL/SQL, >>>> new data types, scalar functions, improved concurrency, built-in >>>> packages, >>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>> >>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> >>>> gstreamer-devel at lists.sourceforge.net >>>> >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> >>> -- >>> Regards, >>> >>> Sudarshan Bisht >>> >>> >>> ------------------------------------------------------------------------------ >>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>> PL/SQL, >>> new data types, scalar functions, improved concurrency, built-in >>> packages, >>> OCI, SQL*Plus, data movement tools, best practices and more. >>> >>> http://p.sf.net/sfu/oracle-sfdev2dev >>> _______________________________________________ >>> gstreamer-devel mailing list >>> >>> gstreamer-devel at lists.sourceforge.net >>> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> ------------------------------------------------------------------------------ >> Oracle to DB2 Conversion Guide: Learn learn about native support for >> PL/SQL, >> new data types, scalar functions, improved concurrency, built-in packages, >> >> OCI, SQL*Plus, data movement tools, best practices and more. >> http://p.sf.net/sfu/oracle-sfdev2dev >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Oracle to DB2 Conversion Guide: Learn learn about native support for >> PL/SQL, >> new data types, scalar functions, improved concurrency, built-in packages, >> OCI, SQL*Plus, data movement tools, best practices and more. >> http://p.sf.net/sfu/oracle-sfdev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > -- > Regards, > > Sudarshan Bisht > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From p-t-r at users.sourceforge.net Sun Dec 12 10:12:22 2010 From: p-t-r at users.sourceforge.net (p-t-r) Date: Sun, 12 Dec 2010 10:12:22 +0100 Subject: [gst-devel] Pipeline gets stuck at PAUSED state In-Reply-To: References: Message-ID: <4D0491F6.1060403@users.sourceforge.net> Hi, I installed GStreamer 0.10 on an embedded system. When I try to run gst-launch like this gst-launch -v fakesrc num-buffers=5 ! fakesink it gets stuck at this line: Setting pipeline to PAUSED ... When I set GST_DEBUG to 5, the last output is as follows [...] 0:00:11.588028710 907 0x7160 LOG GST_MESSAGE gstmessage.c:184:gst_message_init: new message 0x21c18 0:00:11.589971165 907 0x7160 LOG GST_MESSAGE gstmessage.c:270:gst_message_new_custom: source src: creating new message 0x21c18 stream-status 0:00:11.592028220 907 0x7160 DEBUG GST_PADS gstpad.c:5216:do_stream_status: posting stream-status 0 0:00:11.594203823 907 0x7160 DEBUG GST_BUS gstbus.c:307:gst_bus_post: (null) 0:00:11.596117391 907 0x7160 DEBUG bin gstbin.c:3017:gst_bin_handle_message_func: [msg 0x21c18] handling child src message of type stream-status 0:00:11.598052272 907 0x7160 DEBUG bin gstbin.c:3302:gst_bin_handle_message_func: posting message upward 0:00:11.599900160 907 0x7160 DEBUG GST_BUS gstbus.c:307:gst_bus_post: (null) 0:00:11.601710630 907 0x7160 DEBUG GST_BUS gstbus.c:337:gst_bus_post: [msg 0x21c18] pushing on async queue 0:00:11.603738799 907 0x7160 DEBUG GST_BUS gstbus.c:342:gst_bus_post: [msg 0x21c18] pushed on async queue 0:00:11.605654865 907 0x7160 DEBUG GST_BUS gstbus.c:333:gst_bus_post: [msg 0x21c18] dropped 0:00:11.607649017 907 0x7160 DEBUG task gsttask.c:641:gst_task_set_state: Changing task 0x31020 to state 0 What could be the problem that prevents the pipeline to go into PLAYING state? Regards, Peter From shamun.toha at gmail.com Sun Dec 12 11:19:26 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sun, 12 Dec 2010 11:19:26 +0100 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 49 In-Reply-To: References: Message-ID: Hello, Thanks for the info. udpsrc cause ERROR: pipeline could not be constructed: no element "udpsrc". But this works in Gstreamer 0.10.29 but not in 0.10.20. Thanks & Regards Shamun > Message: 1 > Date: Sat, 11 Dec 2010 12:26:21 -0500 > From: Sean McNamara > Subject: Re: [gst-devel] CentOS 64 bit, how to get latest version > 0.10.31 ? > To: Discussion of the development of GStreamer > > Message-ID: > > Content-Type: text/plain; charset=ISO-8859-1 > > Hi, > > On Sat, Dec 11, 2010 at 10:01 AM, Shamun toha md > wrote: > > Hello, > > > > I am not sure what is that mean, but i thought i should ask it in mailing > to > > make sure. I used an example in fedora works, but tried in CentOS 5.4 it > > does not work. > > If I understand you correctly, you're saying that a certain pipeline > example or gstreamer application works in Fedora 12, but not in CentOS > 5.4. That is a valid reason for wanting to upgrade your gstreamer > version, though you might simply be missing a certain plugin, or > hitting an easily fixable bug. Without specific details about what > "does not work" in particular, we can't help you more than that. BTW, > CentOS 6.0 does not exist yet, only RHEL 6.0... > > > > > Fedora???????? gstreamer.i686? 0.10.29-1.fc12 (working my test code) > > > > CentOS 5.4? gstreamer-0.10.20-3.el5.x86_64.rpm? (does not work) > > CentOS 6.0? gstreamer-0.10.26-1.el6.x86_64.rpm? (i am confused) > > I'm confused, too! You aren't explaining yourself very well. You seem > to be grabbing random RPMs and trying them? For your info, it's > probably better to use a package manager, such as yum. Getting RPMs > off of pbone or other RPM finding services does not guarantee that the > built binaries you install will be compatible with the rest of your > stack (i.e. that package's dependencies). > > > > > How can i then test in CentOS, the latest Gstreamer releases ? Is there > any > > guides or how tos to achieve that ? Or its impossible because CentOS 6.0 > > will sitll use the old version ? > > > > I am little bit confused now, will it be possible to have the latest and > > greatest ? > > With free software, it is always possible :) The real question is > whether you are willing to work for the desired result. Since > Gstreamer is in userspace, and its dependencies are pretty flexible, > it should be "easy" to upgrade the version by compiling from source. > Another possibility is that you could find a repository online that > has updated binary packages, though I don't know where such a > repository would be, and such a thing wouldn't be supported by the > Gstreamer project maintainers. Nor would it be supported by CentOS > project or Red Hat. Compiling from vanilla sources might lend you some > unofficial advice/support here among interested gstreamer users, > though. > > Gstreamer project itself used to distribute binaries for RHEL/CentOS, > but it seems they haven't been updated since 2008; for example, see: > http://gstreamer.freedesktop.org/pkg/rhel/5/0.10/i386/gst/ (but DON'T > install these; they are very old!) > > The short story is that yes, you can of course install the latest > gstreamer and plugins from source, provided your system has the > required dependencies (and the minimum versions), or provided that you > compile the required dependencies yourself. This can take a lot of > time, especially if you've never done it before, and special > considerations are required if you plan to do this on many different > computers, e.g. in an IT environment. In that case you may want to > build your own RPM packages of it on a development box, and simply > install them on each of the client computers. > > Your email does not really provide enough information for me to offer > any further advice, but I hope this gets you started as to your > options.... > > > Sean > > > > > Thanks & Regards > > Shamun > > > > -- > > > > ************************************************************************ > > The information transmitted is intended only for the person or entity to > > which it is addressed and may contain confidential and/or privileged > > material. > > Any review, retransmission, dissemination, or other use of this > information, > > or taking of any action in reliance upon it by persons or entities > > other than the intended recipient is prohibited. If you received this in > > error, please contact the sender and delete the material from any > computer. > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > > new data types, scalar functions, improved concurrency, built-in > packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Sun Dec 12 11:41:52 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Sun, 12 Dec 2010 16:11:52 +0530 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: References: <6721B70C-FFDF-444D-91A6-4E6BF7297EFE@gmail.com> Message-ID: Ok, it seems playbin takes care of caps negotiations correctly when "codec_data" is a part of caps. So does this solve your problem ? Or still you want to use your old pipeline ? On Sun, Dec 12, 2010 at 2:46 PM, michael wrote: > Hi sadarshan: > I can play the file by your command: gst-launch -v playbin uri=file: > ///path/test.flv > > > On Sun, Dec 12, 2010 at 3:53 PM, sudarshan bisht < > bisht.sudarshan at gmail.com> wrote: > >> I suggest you to use capsfilter, as you said in your earlier mails >> that gst_pad_push returns -1 , that means flvdemux is not able to push >> buffer to the next element and it may happen because of lack of >> caps negotiations. >> >> By the way how did you create this test.flv file ? and are you able to >> play this file with other players ? >> >> and give a try to following pipeline; >> gst-launch -v playbin uri=file:///path/test.flv >> >> >> >> On Sat, Dec 11, 2010 at 10:55 PM, Timothy Braun wrote: >> >>> Your best bet is to use gst-inspect and look at what the various >>> templates of the pads are. It sounds like you need to do some conversion >>> somewhere. >>> >>> Sent from my iPhone >>> >>> On Dec 11, 2010, at 8:46 AM, michael wrote: >>> >>> Hi sudarshan: >>> I followed your command, I got information : >>> WARNING: erroneous pipeline: could not parse caps "video/x-h264,\ >>> pixel-aspect-ratio=(fraction)1/1,\ >>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" >>> >>> >>> >>> On Sat, Dec 11, 2010 at 7:32 PM, sudarshan bisht < >>> bisht.sudarshan at gmail.com> wrote: >>> >>>> Ok, now can you try following, >>>> >>>> gst-launch filesrc location=test.flv ! flvdemux name=demux >>>> demux.video_00 ! queue ! "video/x-h264, >>>> pixel-aspect-ratio=(fraction)1/1, >>>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >>>> " ! filesink location=test.ves demux.audio_00 ! queue ! filesink >>>> location=test.aes >>>> >>>> >>>> On Sat, Dec 11, 2010 at 10:49 AM, michael < >>>> mcygogo at gmail.com> wrote: >>>> >>>>> Hi sudarshan: >>>>> I run it with the -v option, and the result like this: >>>>> Setting pipeline to PAUSED ... >>>>> Pipeline is PREROLLING ... >>>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = >>>>> video/x-h264, pixel-aspect-ratio=(fraction)1/1, >>>>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >>>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: >>>>> Internal data stream error. >>>>> Additional debug info: >>>>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>>>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>>>> stream stopped, reason not-linked >>>>> ERROR: pipeline doesn't want to preroll. >>>>> Setting pipeline to NULL ... >>>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL >>>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL >>>>> Freeing pipeline ... >>>>> >>>>> I have found that the in function: gst_flv_demux_parse_tag_video() , >>>>> the call gst_pad_push(demux->video_pad, outbuf) returned the -1; >>>>> >>>>> On Sat, Dec 11, 2010 at 4:33 PM, sudarshan bisht < >>>>> bisht.sudarshan at gmail.com> wrote: >>>>> >>>>>> Could you run the same pipeline with -v option and paste the log here >>>>>> ? >>>>>> >>>>>> >>>>>> On Sat, Dec 11, 2010 at 11:47 AM, michael < >>>>>> mcygogo at gmail.com> wrote: >>>>>> >>>>>>> Hi all: >>>>>>> Now I am learning some about the flv plugin--flvdemux, I try to >>>>>>> demux my flv test file like this: >>>>>>> gst-launch filesrc location=test.flv ! flvdemux name=demux >>>>>>> demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! queue ! >>>>>>> filesink location=test.aes >>>>>>> But when I run it , I just get the information like this: >>>>>>> Setting pipeline to PAUSED ... >>>>>>> Pipeline is PREROLLING ... >>>>>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: >>>>>>> Internal data stream error. >>>>>>> Additional debug info: >>>>>>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>>>>>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>>>>>> stream stopped, reason not-linked >>>>>>> ERROR: pipeline doesn't want to preroll. >>>>>>> Setting pipeline to NULL ... >>>>>>> Freeing pipeline ... >>>>>>> >>>>>>> I just want test the flvdemux plugin , So How I fixed it? tks:) >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------------ >>>>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>>>>> PL/SQL, >>>>>>> new data types, scalar functions, improved concurrency, built-in >>>>>>> packages, >>>>>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>>>>> >>>>>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> >>>>>> Sudarshan Bisht >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>>>> PL/SQL, >>>>>> new data types, scalar functions, improved concurrency, built-in >>>>>> packages, >>>>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>>>> >>>>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>>> PL/SQL, >>>>> new data types, scalar functions, improved concurrency, built-in >>>>> packages, >>>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>>> >>>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> >>>>> gstreamer-devel at lists.sourceforge.net >>>>> >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> >>>> Sudarshan Bisht >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>> PL/SQL, >>>> new data types, scalar functions, improved concurrency, built-in >>>> packages, >>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>> >>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> >>>> gstreamer-devel at lists.sourceforge.net >>>> >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> ------------------------------------------------------------------------------ >>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>> PL/SQL, >>> new data types, scalar functions, improved concurrency, built-in >>> packages, >>> OCI, SQL*Plus, data movement tools, best practices and more. >>> http://p.sf.net/sfu/oracle-sfdev2dev >>> >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>> PL/SQL, >>> new data types, scalar functions, improved concurrency, built-in >>> packages, >>> OCI, SQL*Plus, data movement tools, best practices and more. >>> http://p.sf.net/sfu/oracle-sfdev2dev >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> -- >> Regards, >> >> Sudarshan Bisht >> >> >> ------------------------------------------------------------------------------ >> Oracle to DB2 Conversion Guide: Learn learn about native support for >> PL/SQL, >> new data types, scalar functions, improved concurrency, built-in packages, >> OCI, SQL*Plus, data movement tools, best practices and more. >> http://p.sf.net/sfu/oracle-sfdev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Sun Dec 12 12:34:19 2010 From: mcygogo at gmail.com (michael) Date: Sun, 12 Dec 2010 19:34:19 +0800 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: References: <6721B70C-FFDF-444D-91A6-4E6BF7297EFE@gmail.com> Message-ID: I do not just want to play the FLV file, I want to demux the FLV file trough the gstflvdemux plugin. So maybe I should go on to look for the reason why the gstflvdemux cannot work without the playbin. I can demux the MP4 file following this : gst-launch filesrc location=test.mp4 ! qtdemux name=demux demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! queue ! filesink location=test.aes But why this can NOT work with the flvdemux ? On Sun, Dec 12, 2010 at 6:41 PM, sudarshan bisht wrote: > > Ok, it seems playbin takes care of caps negotiations correctly when > "codec_data" is a part of caps. > > So does this solve your problem ? Or still you want to use your old > pipeline ? > > > > > On Sun, Dec 12, 2010 at 2:46 PM, michael wrote: > >> Hi sadarshan: >> I can play the file by your command: gst-launch -v playbin uri=file: >> ///path/test.flv >> >> >> On Sun, Dec 12, 2010 at 3:53 PM, sudarshan bisht < >> bisht.sudarshan at gmail.com> wrote: >> >>> I suggest you to use capsfilter, as you said in your earlier mails >>> that gst_pad_push returns -1 , that means flvdemux is not able to push >>> buffer to the next element and it may happen because of lack of >>> caps negotiations. >>> >>> By the way how did you create this test.flv file ? and are you able to >>> play this file with other players ? >>> >>> and give a try to following pipeline; >>> gst-launch -v playbin uri=file:///path/test.flv >>> >>> >>> >>> On Sat, Dec 11, 2010 at 10:55 PM, Timothy Braun wrote: >>> >>>> Your best bet is to use gst-inspect and look at what the various >>>> templates of the pads are. It sounds like you need to do some conversion >>>> somewhere. >>>> >>>> Sent from my iPhone >>>> >>>> On Dec 11, 2010, at 8:46 AM, michael wrote: >>>> >>>> Hi sudarshan: >>>> I followed your command, I got information : >>>> WARNING: erroneous pipeline: could not parse caps "video/x-h264,\ >>>> pixel-aspect-ratio=(fraction)1/1,\ >>>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" >>>> >>>> >>>> >>>> On Sat, Dec 11, 2010 at 7:32 PM, sudarshan bisht < >>>> bisht.sudarshan at gmail.com> wrote: >>>> >>>>> Ok, now can you try following, >>>>> >>>>> gst-launch filesrc location=test.flv ! flvdemux name=demux >>>>> demux.video_00 ! queue ! "video/x-h264, >>>>> pixel-aspect-ratio=(fraction)1/1, >>>>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >>>>> " ! filesink location=test.ves demux.audio_00 ! queue ! filesink >>>>> location=test.aes >>>>> >>>>> >>>>> On Sat, Dec 11, 2010 at 10:49 AM, michael < >>>>> mcygogo at gmail.com> wrote: >>>>> >>>>>> Hi sudarshan: >>>>>> I run it with the -v option, and the result like this: >>>>>> Setting pipeline to PAUSED ... >>>>>> Pipeline is PREROLLING ... >>>>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = >>>>>> video/x-h264, pixel-aspect-ratio=(fraction)1/1, >>>>>> codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb >>>>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: >>>>>> Internal data stream error. >>>>>> Additional debug info: >>>>>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>>>>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>>>>> stream stopped, reason not-linked >>>>>> ERROR: pipeline doesn't want to preroll. >>>>>> Setting pipeline to NULL ... >>>>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL >>>>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL >>>>>> Freeing pipeline ... >>>>>> >>>>>> I have found that the in function: gst_flv_demux_parse_tag_video() , >>>>>> the call gst_pad_push(demux->video_pad, outbuf) returned the -1; >>>>>> >>>>>> On Sat, Dec 11, 2010 at 4:33 PM, sudarshan bisht < >>>>>> bisht.sudarshan at gmail.com> wrote: >>>>>> >>>>>>> Could you run the same pipeline with -v option and paste the log here >>>>>>> ? >>>>>>> >>>>>>> >>>>>>> On Sat, Dec 11, 2010 at 11:47 AM, michael < >>>>>>> mcygogo at gmail.com> wrote: >>>>>>> >>>>>>>> Hi all: >>>>>>>> Now I am learning some about the flv plugin--flvdemux, I try to >>>>>>>> demux my flv test file like this: >>>>>>>> gst-launch filesrc location=test.flv ! flvdemux name=demux >>>>>>>> demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! queue ! >>>>>>>> filesink location=test.aes >>>>>>>> But when I run it , I just get the information like this: >>>>>>>> Setting pipeline to PAUSED ... >>>>>>>> Pipeline is PREROLLING ... >>>>>>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: >>>>>>>> Internal data stream error. >>>>>>>> Additional debug info: >>>>>>>> gstflvdemux.c(2283): gst_flv_demux_loop (): >>>>>>>> /GstPipeline:pipeline0/GstFlvDemux:demux: >>>>>>>> stream stopped, reason not-linked >>>>>>>> ERROR: pipeline doesn't want to preroll. >>>>>>>> Setting pipeline to NULL ... >>>>>>>> Freeing pipeline ... >>>>>>>> >>>>>>>> I just want test the flvdemux plugin , So How I fixed it? tks:) >>>>>>>> >>>>>>>> >>>>>>>> ------------------------------------------------------------------------------ >>>>>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>>>>>> PL/SQL, >>>>>>>> new data types, scalar functions, improved concurrency, built-in >>>>>>>> packages, >>>>>>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>>>>>> >>>>>>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>>>>>> _______________________________________________ >>>>>>>> gstreamer-devel mailing list >>>>>>>> >>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>> >>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> >>>>>>> Sudarshan Bisht >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------------ >>>>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>>>>> PL/SQL, >>>>>>> new data types, scalar functions, improved concurrency, built-in >>>>>>> packages, >>>>>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>>>>> >>>>>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>>>> PL/SQL, >>>>>> new data types, scalar functions, improved concurrency, built-in >>>>>> packages, >>>>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>>>> >>>>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> >>>>> Sudarshan Bisht >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>>> PL/SQL, >>>>> new data types, scalar functions, improved concurrency, built-in >>>>> packages, >>>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>>> >>>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> >>>>> gstreamer-devel at lists.sourceforge.net >>>>> >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>> PL/SQL, >>>> new data types, scalar functions, improved concurrency, built-in >>>> packages, >>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>> >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>>> PL/SQL, >>>> new data types, scalar functions, improved concurrency, built-in >>>> packages, >>>> OCI, SQL*Plus, data movement tools, best practices and more. >>>> http://p.sf.net/sfu/oracle-sfdev2dev >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> >>> -- >>> Regards, >>> >>> Sudarshan Bisht >>> >>> >>> ------------------------------------------------------------------------------ >>> Oracle to DB2 Conversion Guide: Learn learn about native support for >>> PL/SQL, >>> new data types, scalar functions, improved concurrency, built-in >>> packages, >>> OCI, SQL*Plus, data movement tools, best practices and more. >>> http://p.sf.net/sfu/oracle-sfdev2dev >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> ------------------------------------------------------------------------------ >> Oracle to DB2 Conversion Guide: Learn learn about native support for >> PL/SQL, >> new data types, scalar functions, improved concurrency, built-in packages, >> OCI, SQL*Plus, data movement tools, best practices and more. >> http://p.sf.net/sfu/oracle-sfdev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > -- > Regards, > > Sudarshan Bisht > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Sun Dec 12 12:42:34 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Sun, 12 Dec 2010 12:42:34 +0100 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: References: <6721B70C-FFDF-444D-91A6-4E6BF7297EFE@gmail.com> Message-ID: <1292154154.22115.5.camel@meany> On Sun, 2010-12-12 at 19:34 +0800, michael wrote: > I do not just want to play the FLV file, I want to demux the FLV file > trough the gstflvdemux plugin. > So maybe I should go on to look for the reason why the gstflvdemux > cannot work without the playbin. > I can demux the MP4 file following this : > gst-launch filesrc location=test.mp4 ! qtdemux name=demux > demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! > queue ! filesink location=test.aes > But why this can NOT work with the flvdemux ? Because you use the wrong pad names. use gst-inspect to see the possible srcpad names. Wim > > > > On Sun, Dec 12, 2010 at 6:41 PM, sudarshan bisht > wrote: > > > Ok, it seems playbin takes care of caps negotiations correctly > when "codec_data" is a part of caps. > > > So does this solve your problem ? Or still you want to use > your old pipeline ? > > > > > > > On Sun, Dec 12, 2010 at 2:46 PM, michael > wrote: > Hi sadarshan: > I can play the file by your command: gst-launch > -v playbin uri=file:///path/test.flv > > > > On Sun, Dec 12, 2010 at 3:53 PM, sudarshan bisht > wrote: > I suggest you to use capsfilter, as you said > in your earlier mails that gst_pad_push > returns -1 , that means flvdemux is not able > to push buffer to the next element and it may > happen because of lack of caps negotiations. > > > By the way how did you create this test.flv > file ? and are you able to play this file with > other players ? > > > and give a try to following pipeline; > gst-launch -v playbin > uri=file:///path/test.flv > > > > > > On Sat, Dec 11, 2010 at 10:55 PM, Timothy > Braun wrote: > Your best bet is to use gst-inspect > and look at what the various templates > of the pads are. It sounds like you > need to do some conversion somewhere. > > Sent from my iPhone > > > On Dec 11, 2010, at 8:46 AM, michael > wrote: > > > > > Hi sudarshan: > > I followed your command, I got > > information : > > WARNING: erroneous pipeline: could > > not parse caps "video/x-h264,\ > > pixel-aspect-ratio=(fraction)1/1,\ > > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" > > > > > > > > > > > > On Sat, Dec 11, 2010 at 7:32 PM, > > sudarshan bisht > > wrote: > > Ok, now can you try > > following, > > > > gst-launch filesrc > > location=test.flv ! flvdemux > > name=demux demux.video_00 ! > > queue ! "video/x-h264, > > pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb " ! filesink location=test.ves demux.audio_00 ! queue ! filesink location=test.aes > > > > > > > > On Sat, Dec 11, 2010 at > > 10:49 AM, michael > > wrote: > > Hi sudarshan: > > I run it with > > the -v option, and > > the result like > > this: > > > > Setting pipeline to > > PAUSED ... > > Pipeline is > > PREROLLING ... > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = video/x-h264, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > > ERROR: from > > element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal data stream error. > > Additional debug > > info: > > gstflvdemux.c(2283): > > gst_flv_demux_loop > > (): /GstPipeline:pipeline0/GstFlvDemux:demux: > > stream stopped, > > reason not-linked > > ERROR: pipeline > > doesn't want to > > preroll. > > Setting pipeline to > > NULL ... > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL > > Freeing pipeline ... > > > > > > I have found that > > the in function: > > gst_flv_demux_parse_tag_video() , the call gst_pad_push(demux->video_pad, outbuf) returned the -1; > > > > > > On Sat, Dec 11, 2010 > > at 4:33 PM, > > sudarshan bisht > > wrote: > > Could you > > run the same > > pipeline > > with -v > > option and > > paste the > > log here ? > > > > > > > > On Sat, Dec > > 11, 2010 at > > 11:47 AM, > > michael > > wrote: > > > > > > Hi > > all: > > > > Now > > I am > > learning some about the flv plugin--flvdemux, I try to demux my flv test file like this: > > gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! queue ! filesink location=test.aes > > > > But > > when > > I > > run > > it , > > I > > just > > get > > the > > information like this: > > Setting pipeline to PAUSED ... > > Pipeline is PREROLLING ... > > ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal data stream error. > > Additional debug info: > > gstflvdemux.c(2283): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstFlvDemux:demux: > > stream stopped, reason not-linked > > ERROR: pipeline doesn't want to preroll. > > Setting pipeline to NULL ... > > Freeing pipeline ... > > > > > > I > > just > > want > > test > > the > > flvdemux plugin , So How I fixed it? tks:) > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > > new > > data > > types, scalar functions, improved concurrency, built-in packages, > > OCI, > > SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > Regards, > > > > Sudarshan > > Bisht > > > > > > ------------------------------------------------------------------------------ > > Oracle to > > DB2 > > Conversion > > Guide: Learn > > learn about > > native > > support for > > PL/SQL, > > new data > > types, > > scalar > > functions, > > improved > > concurrency, > > built-in > > packages, > > OCI, > > SQL*Plus, > > data > > movement > > tools, best > > practices > > and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 > > Conversion Guide: > > Learn learn about > > native support for > > PL/SQL, > > new data types, > > scalar functions, > > improved > > concurrency, > > built-in packages, > > OCI, SQL*Plus, data > > movement tools, best > > practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel > > mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > Regards, > > > > Sudarshan Bisht > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion > > Guide: Learn learn about > > native support for PL/SQL, > > new data types, scalar > > functions, improved > > concurrency, built-in > > packages, > > OCI, SQL*Plus, data movement > > tools, best practices and > > more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: > > Learn learn about native support for > > PL/SQL, > > new data types, scalar functions, > > improved concurrency, built-in > > packages, > > OCI, SQL*Plus, data movement tools, > > best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn > learn about native support for PL/SQL, > new data types, scalar functions, > improved concurrency, built-in > packages, > OCI, SQL*Plus, data movement tools, > best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > Regards, > > Sudarshan Bisht > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn > about native support for PL/SQL, > new data types, scalar functions, improved > concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best > practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about > native support for PL/SQL, > new data types, scalar functions, improved > concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and > more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > Regards, > > Sudarshan Bisht > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native > support for PL/SQL, > new data types, scalar functions, improved concurrency, > built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From mcygogo at gmail.com Sun Dec 12 13:03:25 2010 From: mcygogo at gmail.com (michael) Date: Sun, 12 Dec 2010 20:03:25 +0800 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: <1292154154.22115.5.camel@meany> References: <6721B70C-FFDF-444D-91A6-4E6BF7297EFE@gmail.com> <1292154154.22115.5.camel@meany> Message-ID: Hi Wim: Do you means that I just used the wrong srcpad for flvdemux ? On Sun, Dec 12, 2010 at 7:42 PM, Wim Taymans wrote: > On Sun, 2010-12-12 at 19:34 +0800, michael wrote: > > I do not just want to play the FLV file, I want to demux the FLV file > > trough the gstflvdemux plugin. > > So maybe I should go on to look for the reason why the gstflvdemux > > cannot work without the playbin. > > I can demux the MP4 file following this : > > gst-launch filesrc location=test.mp4 ! qtdemux name=demux > > demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 ! > > queue ! filesink location=test.aes > > But why this can NOT work with the flvdemux ? > > Because you use the wrong pad names. use gst-inspect to see the possible > srcpad names. > > Wim > > > > > > > > On Sun, Dec 12, 2010 at 6:41 PM, sudarshan bisht > > wrote: > > > > > > Ok, it seems playbin takes care of caps negotiations correctly > > when "codec_data" is a part of caps. > > > > > > So does this solve your problem ? Or still you want to use > > your old pipeline ? > > > > > > > > > > > > > > On Sun, Dec 12, 2010 at 2:46 PM, michael > > wrote: > > Hi sadarshan: > > I can play the file by your command: gst-launch > > -v playbin uri=file:///path/test.flv > > > > > > > > On Sun, Dec 12, 2010 at 3:53 PM, sudarshan bisht > > wrote: > > I suggest you to use capsfilter, as you said > > in your earlier mails that gst_pad_push > > returns -1 , that means flvdemux is not able > > to push buffer to the next element and it may > > happen because of lack of caps negotiations. > > > > > > By the way how did you create this test.flv > > file ? and are you able to play this file with > > other players ? > > > > > > and give a try to following pipeline; > > gst-launch -v playbin > > uri=file:///path/test.flv > > > > > > > > > > > > On Sat, Dec 11, 2010 at 10:55 PM, Timothy > > Braun wrote: > > Your best bet is to use gst-inspect > > and look at what the various templates > > of the pads are. It sounds like you > > need to do some conversion somewhere. > > > > Sent from my iPhone > > > > > > On Dec 11, 2010, at 8:46 AM, michael > > wrote: > > > > > > > > > Hi sudarshan: > > > I followed your command, I got > > > information : > > > WARNING: erroneous pipeline: could > > > not parse caps "video/x-h264,\ > > > pixel-aspect-ratio=(fraction)1/1,\ > > > > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" > > > > > > > > > > > > > > > > > > On Sat, Dec 11, 2010 at 7:32 PM, > > > sudarshan bisht > > > wrote: > > > Ok, now can you try > > > following, > > > > > > gst-launch filesrc > > > location=test.flv ! flvdemux > > > name=demux demux.video_00 ! > > > queue ! "video/x-h264, > > > > pixel-aspect-ratio=(fraction)1/1, > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > " ! filesink location=test.ves demux.audio_00 ! queue ! filesink > location=test.aes > > > > > > > > > > > > On Sat, Dec 11, 2010 at > > > 10:49 AM, michael > > > wrote: > > > Hi sudarshan: > > > I run it with > > > the -v option, and > > > the result like > > > this: > > > > > > Setting pipeline to > > > PAUSED ... > > > Pipeline is > > > PREROLLING ... > > > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = video/x-h264, > pixel-aspect-ratio=(fraction)1/1, > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > > > ERROR: from > > > element > /GstPipeline:pipeline0/GstFlvDemux:demux: Internal data stream error. > > > Additional debug > > > info: > > > gstflvdemux.c(2283): > > > gst_flv_demux_loop > > > (): > /GstPipeline:pipeline0/GstFlvDemux:demux: > > > stream stopped, > > > reason not-linked > > > ERROR: pipeline > > > doesn't want to > > > preroll. > > > Setting pipeline to > > > NULL ... > > > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL > > > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL > > > Freeing pipeline ... > > > > > > > > > I have found that > > > the in function: > > > > gst_flv_demux_parse_tag_video() , the call gst_pad_push(demux->video_pad, > outbuf) returned the -1; > > > > > > > > > On Sat, Dec 11, 2010 > > > at 4:33 PM, > > > sudarshan bisht > > > < > bisht.sudarshan at gmail.com> wrote: > > > Could you > > > run the same > > > pipeline > > > with -v > > > option and > > > paste the > > > log here ? > > > > > > > > > > > > On Sat, Dec > > > 11, 2010 at > > > 11:47 AM, > > > michael > > > < > mcygogo at gmail.com> wrote: > > > > > > > > > Hi > > > all: > > > > > > Now > > > I am > > > > learning some about the flv plugin--flvdemux, I try to demux my flv test > file like this: > > > > gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video_00 ! > queue ! filesink location=test.ves demux.audio_00 ! queue ! filesink > location=test.aes > > > > > > But > > > when > > > I > > > run > > > it , > > > I > > > just > > > get > > > the > > > > information like this: > > > Setting > pipeline to PAUSED ... > > > > Pipeline is PREROLLING ... > > > ERROR: > from element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal data stream > error. > > > > Additional debug info: > > > > gstflvdemux.c(2283): gst_flv_demux_loop (): > /GstPipeline:pipeline0/GstFlvDemux:demux: > > > stream > stopped, reason not-linked > > > ERROR: > pipeline doesn't want to preroll. > > > Setting > pipeline to NULL ... > > > Freeing > pipeline ... > > > > > > > > > I > > > just > > > want > > > test > > > the > > > > flvdemux plugin , So How I fixed it? tks:) > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle > to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > > > new > > > data > > > types, > scalar functions, improved concurrency, built-in packages, > > > OCI, > > > > SQL*Plus, data movement tools, best practices and more. > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > _______________________________________________ > > > > gstreamer-devel mailing list > > > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > -- > > > Regards, > > > > > > Sudarshan > > > Bisht > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to > > > DB2 > > > Conversion > > > Guide: Learn > > > learn about > > > native > > > support for > > > PL/SQL, > > > new data > > > types, > > > scalar > > > functions, > > > improved > > > concurrency, > > > built-in > > > packages, > > > OCI, > > > SQL*Plus, > > > data > > > movement > > > tools, best > > > practices > > > and more. > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > _______________________________________________ > > > gstreamer-devel > mailing list > > > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 > > > Conversion Guide: > > > Learn learn about > > > native support for > > > PL/SQL, > > > new data types, > > > scalar functions, > > > improved > > > concurrency, > > > built-in packages, > > > OCI, SQL*Plus, data > > > movement tools, best > > > practices and more. > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > _______________________________________________ > > > gstreamer-devel > > > mailing list > > > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > -- > > > Regards, > > > > > > Sudarshan Bisht > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 Conversion > > > Guide: Learn learn about > > > native support for PL/SQL, > > > new data types, scalar > > > functions, improved > > > concurrency, built-in > > > packages, > > > OCI, SQL*Plus, data movement > > > tools, best practices and > > > more. > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > _______________________________________________ > > > gstreamer-devel mailing list > > > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 Conversion Guide: > > > Learn learn about native support for > > > PL/SQL, > > > new data types, scalar functions, > > > improved concurrency, built-in > > > packages, > > > OCI, SQL*Plus, data movement tools, > > > best practices and more. > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn > > learn about native support for PL/SQL, > > new data types, scalar functions, > > improved concurrency, built-in > > packages, > > OCI, SQL*Plus, data movement tools, > > best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > Regards, > > > > Sudarshan Bisht > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn > > about native support for PL/SQL, > > new data types, scalar functions, improved > > concurrency, built-in packages, > > OCI, SQL*Plus, data movement tools, best > > practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about > > native support for PL/SQL, > > new data types, scalar functions, improved > > concurrency, built-in packages, > > OCI, SQL*Plus, data movement tools, best practices and > > more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > Regards, > > > > Sudarshan Bisht > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native > > support for PL/SQL, > > new data types, scalar functions, improved concurrency, > > built-in packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > > new data types, scalar functions, improved concurrency, built-in > packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From 4ernov at gmail.com Sun Dec 12 13:07:23 2010 From: 4ernov at gmail.com (4ernov) Date: Sun, 12 Dec 2010 15:07:23 +0300 Subject: [gst-devel] How to make patch been noticed In-Reply-To: <201012110215.36152.4ernov@gmail.com> References: <201012110215.36152.4ernov@gmail.com> Message-ID: Hello, I filed a bug about gdppay/gdpdepay elements concerning their work with tee with dynamic connections and disconnections here: https://bugzilla.gnome.org/show_bug.cgi?id=635226 I worked on the problem and found a small bug in gdpdepay code. I fixed it and tested for my case and then proposed a small patch for this bug. But unfortunately nobody answers on the bug even after I fixed it by myself. Could you please help me with any information on how I can get some attention to this bug and patch which could perhaps be merged to the main tree? Thanks in advance, Alexey From wim.taymans at gmail.com Sun Dec 12 13:20:26 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Sun, 12 Dec 2010 13:20:26 +0100 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: References: <6721B70C-FFDF-444D-91A6-4E6BF7297EFE@gmail.com> <1292154154.22115.5.camel@meany> Message-ID: <1292156426.22115.6.camel@meany> On Sun, 2010-12-12 at 20:03 +0800, michael wrote: > Hi Wim: > Do you means that I just used the wrong srcpad for flvdemux ? Well yes, video_00 is not a pad that flvdemux will ever make according to gst-inspect. Wim > > On Sun, Dec 12, 2010 at 7:42 PM, Wim Taymans > wrote: > On Sun, 2010-12-12 at 19:34 +0800, michael wrote: > > I do not just want to play the FLV file, I want to demux the > FLV file > > trough the gstflvdemux plugin. > > So maybe I should go on to look for the reason why the > gstflvdemux > > cannot work without the playbin. > > I can demux the MP4 file following this : > > gst-launch filesrc location=test.mp4 ! qtdemux name=demux > > demux.video_00 ! queue ! filesink location=test.ves > demux.audio_00 ! > > queue ! filesink location=test.aes > > But why this can NOT work with the flvdemux ? > > > Because you use the wrong pad names. use gst-inspect to see > the possible > srcpad names. > > Wim > > > > > > > > > On Sun, Dec 12, 2010 at 6:41 PM, sudarshan bisht > > wrote: > > > > > > Ok, it seems playbin takes care of caps negotiations > correctly > > when "codec_data" is a part of caps. > > > > > > So does this solve your problem ? Or still you want > to use > > your old pipeline ? > > > > > > > > > > > > > > On Sun, Dec 12, 2010 at 2:46 PM, michael > > > wrote: > > Hi sadarshan: > > I can play the file by your command: > gst-launch > > -v playbin uri=file:///path/test.flv > > > > > > > > On Sun, Dec 12, 2010 at 3:53 PM, sudarshan > bisht > > wrote: > > I suggest you to use capsfilter, as > you said > > in your earlier mails that > gst_pad_push > > returns -1 , that means flvdemux is > not able > > to push buffer to the next element > and it may > > happen because of lack of caps > negotiations. > > > > > > By the way how did you create this > test.flv > > file ? and are you able to play this > file with > > other players ? > > > > > > and give a try to following > pipeline; > > gst-launch -v playbin > > uri=file:///path/test.flv > > > > > > > > > > > > On Sat, Dec 11, 2010 at 10:55 PM, > Timothy > > Braun > wrote: > > Your best bet is to use > gst-inspect > > and look at what the various > templates > > of the pads are. It sounds > like you > > need to do some conversion > somewhere. > > > > Sent from my iPhone > > > > > > On Dec 11, 2010, at 8:46 AM, > michael > > wrote: > > > > > > > > > Hi sudarshan: > > > I followed your > command, I got > > > information : > > > WARNING: erroneous > pipeline: could > > > not parse caps > "video/x-h264,\ > > > > pixel-aspect-ratio=(fraction)1/1,\ > > > > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" > > > > > > > > > > > > > > > > > > On Sat, Dec 11, 2010 at > 7:32 PM, > > > sudarshan bisht > > > > wrote: > > > Ok, now can you > try > > > following, > > > > > > gst-launch filesrc > > > > location=test.flv ! flvdemux > > > name=demux > demux.video_00 ! > > > queue ! > "video/x-h264, > > > > pixel-aspect-ratio=(fraction)1/1, > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb " ! filesink location=test.ves demux.audio_00 ! queue ! filesink location=test.aes > > > > > > > > > > > > On Sat, Dec 11, > 2010 at > > > 10:49 AM, michael > > > > wrote: > > > Hi > sudarshan: > > > I run > it with > > > the -v > option, and > > > the > result like > > > this: > > > > > > Setting > pipeline to > > > PAUSED ... > > > Pipeline > is > > > > PREROLLING ... > > > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps > = video/x-h264, pixel-aspect-ratio=(fraction)1/1, > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > > > ERROR: > from > > > > element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal > data stream error. > > > Additional > debug > > > info: > > > > gstflvdemux.c(2283): > > > > gst_flv_demux_loop > > > > (): /GstPipeline:pipeline0/GstFlvDemux:demux: > > > stream > stopped, > > > reason > not-linked > > > ERROR: > pipeline > > > doesn't > want to > > > preroll. > > > Setting > pipeline to > > > NULL ... > > > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps > = NULL > > > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps > = NULL > > > Freeing > pipeline ... > > > > > > > > > I have > found that > > > the in > function: > > > > gst_flv_demux_parse_tag_video() , the call > gst_pad_push(demux->video_pad, outbuf) returned the -1; > > > > > > > > > On Sat, > Dec 11, 2010 > > > at 4:33 > PM, > > > sudarshan > bisht > > > > wrote: > > > > Could you > > > > run the same > > > > pipeline > > > > with -v > > > > option and > > > > paste the > > > > log here ? > > > > > > > > > > > > On > Sat, Dec > > > > 11, 2010 at > > > > 11:47 AM, > > > > michael > > > > wrote: > > > > > > > > > > Hi > > > > all: > > > > > > > Now > > > > I am > > > > learning some about the flv plugin--flvdemux, I try to demux > my flv test file like this: > > > > gst-launch filesrc location=test.flv ! flvdemux name=demux > demux.video_00 ! queue ! filesink location=test.ves > demux.audio_00 ! queue ! filesink location=test.aes > > > > > > > But > > > > when > > > > I > > > > run > > > > it , > > > > I > > > > just > > > > get > > > > the > > > > information like this: > > > > Setting pipeline to PAUSED ... > > > > Pipeline is PREROLLING ... > > > > ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: > Internal data stream error. > > > > Additional debug info: > > > > gstflvdemux.c(2283): gst_flv_demux_loop > (): /GstPipeline:pipeline0/GstFlvDemux:demux: > > > > stream stopped, reason not-linked > > > > ERROR: pipeline doesn't want to preroll. > > > > Setting pipeline to NULL ... > > > > Freeing pipeline ... > > > > > > > > > > I > > > > just > > > > want > > > > test > > > > the > > > > flvdemux plugin , So How I fixed it? tks:) > > > > > > > > > > ------------------------------------------------------------------------------ > > > > Oracle to DB2 Conversion Guide: Learn learn about native > support for PL/SQL, > > > > new > > > > data > > > > types, scalar functions, improved concurrency, built-in > packages, > > > > OCI, > > > > SQL*Plus, data movement tools, best practices and more. > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > _______________________________________________ > > > > gstreamer-devel mailing list > > > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > -- > > > > Regards, > > > > > > > Sudarshan > > > > Bisht > > > > > > > > > > ------------------------------------------------------------------------------ > > > > Oracle to > > > > DB2 > > > > Conversion > > > > Guide: Learn > > > > learn about > > > > native > > > > support for > > > > PL/SQL, > > > > new data > > > > types, > > > > scalar > > > > functions, > > > > improved > > > > concurrency, > > > > built-in > > > > packages, > > > > OCI, > > > > SQL*Plus, > > > > data > > > > movement > > > > tools, best > > > > practices > > > > and more. > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > _______________________________________________ > > > > gstreamer-devel mailing list > > > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to > DB2 > > > Conversion > Guide: > > > Learn > learn about > > > native > support for > > > PL/SQL, > > > new data > types, > > > scalar > functions, > > > improved > > > > concurrency, > > > built-in > packages, > > > OCI, > SQL*Plus, data > > > movement > tools, best > > > practices > and more. > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > _______________________________________________ > > > > gstreamer-devel > > > mailing > list > > > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > -- > > > Regards, > > > > > > Sudarshan Bisht > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 > Conversion > > > Guide: Learn learn > about > > > native support for > PL/SQL, > > > new data types, > scalar > > > functions, > improved > > > concurrency, > built-in > > > packages, > > > OCI, SQL*Plus, > data movement > > > tools, best > practices and > > > more. > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > _______________________________________________ > > > gstreamer-devel > mailing list > > > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 Conversion > Guide: > > > Learn learn about native > support for > > > PL/SQL, > > > new data types, scalar > functions, > > > improved concurrency, > built-in > > > packages, > > > OCI, SQL*Plus, data > movement tools, > > > best practices and more. > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > _______________________________________________ > > > gstreamer-devel mailing > list > > > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion > Guide: Learn > > learn about native support > for PL/SQL, > > new data types, scalar > functions, > > improved concurrency, > built-in > > packages, > > OCI, SQL*Plus, data movement > tools, > > best practices and more. > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > _______________________________________________ > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > Regards, > > > > Sudarshan Bisht > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: > Learn learn > > about native support for PL/SQL, > > new data types, scalar functions, > improved > > concurrency, built-in packages, > > OCI, SQL*Plus, data movement tools, > best > > practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > > _______________________________________________ > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn > about > > native support for PL/SQL, > > new data types, scalar functions, improved > > concurrency, built-in packages, > > OCI, SQL*Plus, data movement tools, best > practices and > > more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > Regards, > > > > Sudarshan Bisht > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about > native > > support for PL/SQL, > > new data types, scalar functions, improved > concurrency, > > built-in packages, > > OCI, SQL*Plus, data movement tools, best practices > and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native > support for PL/SQL, > > new data types, scalar functions, improved concurrency, > built-in packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native > support for PL/SQL, > new data types, scalar functions, improved concurrency, > built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From mcygogo at gmail.com Sun Dec 12 13:32:03 2010 From: mcygogo at gmail.com (michael) Date: Sun, 12 Dec 2010 20:32:03 +0800 Subject: [gst-devel] How I test the FLV demux plugin In-Reply-To: <1292156426.22115.6.camel@meany> References: <6721B70C-FFDF-444D-91A6-4E6BF7297EFE@gmail.com> <1292154154.22115.5.camel@meany> <1292156426.22115.6.camel@meany> Message-ID: Hi Wim: thank you so much , I got it work like this : gst-launch filesrc location=test.flv ! flvdemux name=demux demux.video ! queue ! filesink location=test.ves demux.audio ! queue ! filesink location=test.aes On Sun, Dec 12, 2010 at 8:20 PM, Wim Taymans wrote: > On Sun, 2010-12-12 at 20:03 +0800, michael wrote: > > Hi Wim: > > Do you means that I just used the wrong srcpad for flvdemux ? > > Well yes, video_00 is not a pad that flvdemux will ever make according > to gst-inspect. > > Wim > > > > On Sun, Dec 12, 2010 at 7:42 PM, Wim Taymans > > wrote: > > On Sun, 2010-12-12 at 19:34 +0800, michael wrote: > > > I do not just want to play the FLV file, I want to demux the > > FLV file > > > trough the gstflvdemux plugin. > > > So maybe I should go on to look for the reason why the > > gstflvdemux > > > cannot work without the playbin. > > > I can demux the MP4 file following this : > > > gst-launch filesrc location=test.mp4 ! qtdemux name=demux > > > demux.video_00 ! queue ! filesink location=test.ves > > demux.audio_00 ! > > > queue ! filesink location=test.aes > > > But why this can NOT work with the flvdemux ? > > > > > > Because you use the wrong pad names. use gst-inspect to see > > the possible > > srcpad names. > > > > Wim > > > > > > > > > > > > > > On Sun, Dec 12, 2010 at 6:41 PM, sudarshan bisht > > > wrote: > > > > > > > > > Ok, it seems playbin takes care of caps negotiations > > correctly > > > when "codec_data" is a part of caps. > > > > > > > > > So does this solve your problem ? Or still you want > > to use > > > your old pipeline ? > > > > > > > > > > > > > > > > > > > > > On Sun, Dec 12, 2010 at 2:46 PM, michael > > > > > wrote: > > > Hi sadarshan: > > > I can play the file by your command: > > gst-launch > > > -v playbin uri=file:///path/test.flv > > > > > > > > > > > > On Sun, Dec 12, 2010 at 3:53 PM, sudarshan > > bisht > > > wrote: > > > I suggest you to use capsfilter, as > > you said > > > in your earlier mails that > > gst_pad_push > > > returns -1 , that means flvdemux is > > not able > > > to push buffer to the next element > > and it may > > > happen because of lack of caps > > negotiations. > > > > > > > > > By the way how did you create this > > test.flv > > > file ? and are you able to play this > > file with > > > other players ? > > > > > > > > > and give a try to following > > pipeline; > > > gst-launch -v playbin > > > uri=file:///path/test.flv > > > > > > > > > > > > > > > > > > On Sat, Dec 11, 2010 at 10:55 PM, > > Timothy > > > Braun > > wrote: > > > Your best bet is to use > > gst-inspect > > > and look at what the various > > templates > > > of the pads are. It sounds > > like you > > > need to do some conversion > > somewhere. > > > > > > Sent from my iPhone > > > > > > > > > On Dec 11, 2010, at 8:46 AM, > > michael > > > wrote: > > > > > > > > > > > > > Hi sudarshan: > > > > I followed your > > command, I got > > > > information : > > > > WARNING: erroneous > > pipeline: could > > > > not parse caps > > "video/x-h264,\ > > > > > > pixel-aspect-ratio=(fraction)1/1,\ > > > > > > > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" > > > > > > > > > > > > > > > > > > > > > > > > On Sat, Dec 11, 2010 at > > 7:32 PM, > > > > sudarshan bisht > > > > > > wrote: > > > > Ok, now can you > > try > > > > following, > > > > > > > > gst-launch filesrc > > > > > > location=test.flv ! flvdemux > > > > name=demux > > demux.video_00 ! > > > > queue ! > > "video/x-h264, > > > > > > pixel-aspect-ratio=(fraction)1/1, > > > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > " ! filesink location=test.ves demux.audio_00 ! queue ! filesink > location=test.aes > > > > > > > > > > > > > > > > On Sat, Dec 11, > > 2010 at > > > > 10:49 AM, michael > > > > > > wrote: > > > > Hi > > sudarshan: > > > > I run > > it with > > > > the -v > > option, and > > > > the > > result like > > > > this: > > > > > > > > Setting > > pipeline to > > > > PAUSED ... > > > > Pipeline > > is > > > > > > PREROLLING ... > > > > > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps > > = video/x-h264, pixel-aspect-ratio=(fraction)1/1, > > > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > > > > ERROR: > > from > > > > > > element /GstPipeline:pipeline0/GstFlvDemux:demux: Internal > > data stream error. > > > > Additional > > debug > > > > info: > > > > > > gstflvdemux.c(2283): > > > > > > gst_flv_demux_loop > > > > > > (): /GstPipeline:pipeline0/GstFlvDemux:demux: > > > > stream > > stopped, > > > > reason > > not-linked > > > > ERROR: > > pipeline > > > > doesn't > > want to > > > > preroll. > > > > Setting > > pipeline to > > > > NULL ... > > > > > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps > > = NULL > > > > > > /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps > > = NULL > > > > Freeing > > pipeline ... > > > > > > > > > > > > I have > > found that > > > > the in > > function: > > > > > > gst_flv_demux_parse_tag_video() , the call > > gst_pad_push(demux->video_pad, outbuf) returned the -1; > > > > > > > > > > > > On Sat, > > Dec 11, 2010 > > > > at 4:33 > > PM, > > > > sudarshan > > bisht > > > > > > wrote: > > > > > > Could you > > > > > > run the same > > > > > > pipeline > > > > > > with -v > > > > > > option and > > > > > > paste the > > > > > > log here ? > > > > > > > > > > > > > > > > On > > Sat, Dec > > > > > > 11, 2010 at > > > > > > 11:47 AM, > > > > > > michael > > > > > > wrote: > > > > > > > > > > > > > > Hi > > > > > > all: > > > > > > > > > > Now > > > > > > I am > > > > > > learning some about the flv plugin--flvdemux, I try to demux > > my flv test file like this: > > > > > > gst-launch filesrc location=test.flv ! flvdemux name=demux > > demux.video_00 ! queue ! filesink location=test.ves > > demux.audio_00 ! queue ! filesink location=test.aes > > > > > > > > > > But > > > > > > when > > > > > > I > > > > > > run > > > > > > it , > > > > > > I > > > > > > just > > > > > > get > > > > > > the > > > > > > information like this: > > > > > > Setting pipeline to PAUSED ... > > > > > > Pipeline is PREROLLING ... > > > > > > ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: > > Internal data stream error. > > > > > > Additional debug info: > > > > > > gstflvdemux.c(2283): gst_flv_demux_loop > > (): /GstPipeline:pipeline0/GstFlvDemux:demux: > > > > > > stream stopped, reason not-linked > > > > > > ERROR: pipeline doesn't want to preroll. > > > > > > Setting pipeline to NULL ... > > > > > > Freeing pipeline ... > > > > > > > > > > > > > > I > > > > > > just > > > > > > want > > > > > > test > > > > > > the > > > > > > flvdemux plugin , So How I fixed it? tks:) > > > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > > > > Oracle to DB2 Conversion Guide: Learn learn about native > > support for PL/SQL, > > > > > > new > > > > > > data > > > > > > types, scalar functions, improved concurrency, built-in > > packages, > > > > > > OCI, > > > > > > SQL*Plus, data movement tools, best practices and more. > > > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > > > _______________________________________________ > > > > > > gstreamer-devel mailing list > > > > > > gstreamer-devel at lists.sourceforge.net > > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > > > > > > -- > > > > > > Regards, > > > > > > > > > > Sudarshan > > > > > > Bisht > > > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > > > > Oracle to > > > > > > DB2 > > > > > > Conversion > > > > > > Guide: Learn > > > > > > learn about > > > > > > native > > > > > > support for > > > > > > PL/SQL, > > > > > > new data > > > > > > types, > > > > > > scalar > > > > > > functions, > > > > > > improved > > > > > > concurrency, > > > > > > built-in > > > > > > packages, > > > > > > OCI, > > > > > > SQL*Plus, > > > > > > data > > > > > > movement > > > > > > tools, best > > > > > > practices > > > > > > and more. > > > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > > > _______________________________________________ > > > > > > gstreamer-devel mailing list > > > > > > gstreamer-devel at lists.sourceforge.net > > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > > Oracle to > > DB2 > > > > Conversion > > Guide: > > > > Learn > > learn about > > > > native > > support for > > > > PL/SQL, > > > > new data > > types, > > > > scalar > > functions, > > > > improved > > > > > > concurrency, > > > > built-in > > packages, > > > > OCI, > > SQL*Plus, data > > > > movement > > tools, best > > > > practices > > and more. > > > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > > > _______________________________________________ > > > > > > gstreamer-devel > > > > mailing > > list > > > > > > gstreamer-devel at lists.sourceforge.net > > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > > > > > > -- > > > > Regards, > > > > > > > > Sudarshan Bisht > > > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > > Oracle to DB2 > > Conversion > > > > Guide: Learn learn > > about > > > > native support for > > PL/SQL, > > > > new data types, > > scalar > > > > functions, > > improved > > > > concurrency, > > built-in > > > > packages, > > > > OCI, SQL*Plus, > > data movement > > > > tools, best > > practices and > > > > more. > > > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > > > _______________________________________________ > > > > gstreamer-devel > > mailing list > > > > > > gstreamer-devel at lists.sourceforge.net > > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > > Oracle to DB2 Conversion > > Guide: > > > > Learn learn about native > > support for > > > > PL/SQL, > > > > new data types, scalar > > functions, > > > > improved concurrency, > > built-in > > > > packages, > > > > OCI, SQL*Plus, data > > movement tools, > > > > best practices and more. > > > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > > > _______________________________________________ > > > > gstreamer-devel mailing > > list > > > > > > gstreamer-devel at lists.sourceforge.net > > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 Conversion > > Guide: Learn > > > learn about native support > > for PL/SQL, > > > new data types, scalar > > functions, > > > improved concurrency, > > built-in > > > packages, > > > OCI, SQL*Plus, data movement > > tools, > > > best practices and more. > > > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > > _______________________________________________ > > > gstreamer-devel mailing list > > > > > gstreamer-devel at lists.sourceforge.net > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > -- > > > Regards, > > > > > > Sudarshan Bisht > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 Conversion Guide: > > Learn learn > > > about native support for PL/SQL, > > > new data types, scalar functions, > > improved > > > concurrency, built-in packages, > > > OCI, SQL*Plus, data movement tools, > > best > > > practices and more. > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > > _______________________________________________ > > > gstreamer-devel mailing list > > > > > gstreamer-devel at lists.sourceforge.net > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 Conversion Guide: Learn learn > > about > > > native support for PL/SQL, > > > new data types, scalar functions, improved > > > concurrency, built-in packages, > > > OCI, SQL*Plus, data movement tools, best > > practices and > > > more. > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > -- > > > Regards, > > > > > > Sudarshan Bisht > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 Conversion Guide: Learn learn about > > native > > > support for PL/SQL, > > > new data types, scalar functions, improved > > concurrency, > > > built-in packages, > > > OCI, SQL*Plus, data movement tools, best practices > > and more. > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 Conversion Guide: Learn learn about native > > support for PL/SQL, > > > new data types, scalar functions, improved concurrency, > > built-in packages, > > > OCI, SQL*Plus, data movement tools, best practices and more. > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native > > support for PL/SQL, > > new data types, scalar functions, improved concurrency, > > built-in packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > > new data types, scalar functions, improved concurrency, built-in > packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From yongnian.le at intel.com Mon Dec 13 03:55:14 2010 From: yongnian.le at intel.com (Le, Yongnian) Date: Mon, 13 Dec 2010 10:55:14 +0800 Subject: [gst-devel] Reduce GstXvImageSink rendering when its window is invisible Message-ID: Hi, I am working on multimedia application development using gstreamer framework. There might be many cases that some window is invisible (for example other window is in full screen), its related XvImageSink is still busy on calling gst_xvimagesink_show_frame for video rendering. I am wondering that we could have optimization to detect additional "Obscured" xevent and reduce the rendering frequency to very low value, like 1 fps, if such window is obscured. In this way, we can save precious resources for most applications, like CPU and power, at the same time, having minimal impact to end user. Does that approach make sense? What do you guys think of it? Thanks a lot for your feedback and comments. Best Regards Yongnian -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.henningsson at canonical.com Mon Dec 13 05:01:03 2010 From: david.henningsson at canonical.com (David Henningsson) Date: Mon, 13 Dec 2010 05:01:03 +0100 Subject: [gst-devel] [pulseaudio-discuss] [PATCH 0/3] Fighting rewinds In-Reply-To: References: <4D00DFAF.3000807@canonical.com> <4D0231DF.7040008@canonical.com> Message-ID: <4D059A7F.1000007@canonical.com> On 2010-12-10 16:37, pl bossart wrote: >>>> However, the problem is quite complex and there does not seem to be one >>>> perfect fix, it's more of an optimisation problem. GStreamer in >>>> particular >>>> sends out many small data packages, and PulseAudio does not handle that >>>> very >>>> well. >>> >>> That's the default behavior, but you can cut the traffic by using the >>> latency-time property in pulsesink. This makes sure you send bigger >>> buffers, up to the 64k limit that PulseAudio has internally. >> >> Thanks, that's good to know. Now, I'm just playing a simple audio file with >> totem, so obviously we want high latency and large packet sizes, but I'm not >> a gstreamer developer - do you have an idea of what can be at fault here? >> Should totem specify a big packet size (regardless of sink?), or can we just >> change the default packet size to something big for people not requesting >> anything in particular? > > The default pulsesink settings for latency/buffering are inherited > from a base class. I would agree with you that they should be large by > default and decreased explicitly when the application wants > low-latency (gaming, speech), but the opposite is done... > It's my understanding that higher-level components in the Meego/Qt > stack will specify bigger buffers/higher latencies, but for people who > use plain vanilla gstreamer optimizing for power requires a bit of > knowledge and manual configurations. This is unfortunate since every > single player will need to repeat this configuration. > -Pierre > Hmm, I enabled gstreamer debugging and spotted this (this is from an ogg file playback in totem): pulsesink.c:718:gst_pulseringbuffer_acquire: tlength: 70560 pulsesink.c:719:gst_pulseringbuffer_acquire: maxlength: -1 pulsesink.c:720:gst_pulseringbuffer_acquire: prebuf: 0 pulsesink.c:721:gst_pulseringbuffer_acquire: minreq: 3528 So tlength is actually reasonable. This means we have a segsize of 3528 and a segtotal of 20, then look at this code in gst_pulseringbuffer_commit: /* Only ever write segsize bytes at once. This will * also limit the PA shm buffer to segsize */ if (towrite > buf->spec.segsize) towrite = buf->spec.segsize; ...I'm not sure whether it is the segtotal=20 that's unreasonable, or if these lines are the offending ones, but the combination is actually causing gstreamer to send 20 small packets instead of one big packet => unnecessary overhead (and PA rewinds if we're not enough ahead). I tried changing "buf->spec.segsize" into "bufsize" (i e segsize * segtotal) and it started writing 8192 bytes at a time instead of 3528, which is slightly better, but still way far from tlength (or tlength/2, tlength/4, or whatever is an optimal number of segments). -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic From ds at entropywave.com Mon Dec 13 05:20:30 2010 From: ds at entropywave.com (David Schleef) Date: Sun, 12 Dec 2010 23:20:30 -0500 Subject: [gst-devel] Reduce GstXvImageSink rendering when its window is invisible In-Reply-To: References: Message-ID: <20101213042030.GA20961@cooker.entropywave.com> On Mon, Dec 13, 2010 at 10:55:14AM +0800, Le, Yongnian wrote: > Hi, > > I am working on multimedia application development using gstreamer > framework. There might be many cases that some window is invisible > (for example other window is in full screen), its related XvImageSink > is still busy on calling gst_xvimagesink_show_frame for video > rendering. I am wondering that we could have optimization to detect > additional "Obscured" xevent and reduce the rendering frequency to > very low value, like 1 fps, if such window is obscured. In this way, > we can save precious resources for most applications, like CPU and > power, at the same time, having minimal impact to end user. Does that > approach make sense? What do you guys think of it? Thanks a lot for > your feedback and comments. Right now, the UI toolkit (Gtk+, Qt) already keeps track of windows being mapped or unmapped, and in some cases, also the obscured or partially-obscured information. So it would be straightforward to extend the GstVideoSink base class to have a property that encourages degraded rendering. It is tempting to put the obscurity detection directly in the video sink, however, leaving this to the application allows for the possibility that the app is obscuring the video on purpose, yet doesn't want the quality degraded. Perhaps both methods are useful. In general, however, rendering is a tiny fraction of the cost of playing video, so it would be good to also push this information upstream, either as part of a QoS event (e.g., proportion=0.0 for video that will be obscured), or (better) an extension of the QoS event. That way, video decoders can avoid decoding the video. And, for bonus points, as there is a trend in the industry toward stream switching for video over the internet, the QoS information could be used to select a stream with low video bit rate, or audio-only stream. David From antoni.silvestre at gmail.com Mon Dec 13 11:59:25 2010 From: antoni.silvestre at gmail.com (=?ISO-8859-1?Q?Antoni_Silvestre_Padr=F3s?=) Date: Mon, 13 Dec 2010 11:59:25 +0100 Subject: [gst-devel] Getting remote endpoint from UDP traffic in a udpsrc element Message-ID: Hello, I'm trying to get the remote endpoint info from the last received UDP packet in a udpsrc GStreamer element. The only way I can think of doing this is obtaining the socket of the udpsrc and then use the use the recvfrom call, however this I guess would force me to somehow reinject the traffic I captured with that call back to the udpsrc element so it doesn't get lost. Is there any other way of doing this? I don't have very much experience programming with sockets so I might be missing something. Any help would be greatly appreciated. Toni Silvestre -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Mon Dec 13 12:40:34 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 13 Dec 2010 12:40:34 +0100 Subject: [gst-devel] Getting remote endpoint from UDP traffic in a udpsrc element In-Reply-To: References: Message-ID: <1292240434.22115.10.camel@meany> On Mon, 2010-12-13 at 11:59 +0100, Antoni Silvestre Padr?s wrote: > Hello, > > > I'm trying to get the remote endpoint info from the last received UDP > packet in a udpsrc GStreamer element. > The only way I can think of doing this is obtaining the socket of the > udpsrc and then use the use the recvfrom call, however this I guess > would force me to somehow reinject the traffic I captured with that > call back to the udpsrc element so it doesn't get lost. Is there any > other way of doing this? > > > I don't have very much experience programming with sockets so I might > be missing something. Udpsrc generates GstNetBuffers (see -base/gst-libs/get/netbuffer/gstnetbuffer.h) that contains the sender address. look in gstrtpbin for how you can use this address in your plugin then. Wim > > > Any help would be greatly appreciated. > > > Toni Silvestre > > > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From lichao19860508 at 126.com Mon Dec 13 12:58:51 2010 From: lichao19860508 at 126.com (bluesnail) Date: Mon, 13 Dec 2010 03:58:51 -0800 (PST) Subject: [gst-devel] How to enable gstreamer-vaapi? Message-ID: <1292241531109-3085318.post@n4.nabble.com> Hi,all I am trying to use gstreamer-vaapi-0.2.5 plugins for video playback using the hardware accelerated. I had successfully built vaapi/non-vaapi mplayer on this platform and they can work well. But when we try to enable gstreamer-vaapi, we encounter some problem. When I run this command:gst-launch-0.10 -v filesrc location=/path/to/video.mp4 ! qtdemux ! vaapidecode ! vaapisink fullscreen=true, X will not response or crash. The following are some vital messages. --vainfo: libva: libva version 0.31.1 libva: va_getDriverName() returns 0 libva: Trying to open /usr/lib/dri/pvr_drv_video.so libva: va_openDriver() returns 0 vainfo: VA API version: 0.31 vainfo: Driver version: Intel GMA500-MRST-5.3.0.32L.0029 (0X0000003C) vainfo: Supported profile and entrypoints VAProfileMPEG2Main : VAEntrypointVLD VAProfileMPEG2Main : VAEntrypointMoComp VAProfileMPEG4Simple : VAEntrypointVLD VAProfileMPEG4Simple : VAEntrypointEncSlice VAProfileMPEG4AdvancedSimple : VAEntrypointVLD VAProfileMPEG4AdvancedSimple : VAEntrypointEncSlice VAProfileH264Baseline : VAEntrypointVLD VAProfileH264Baseline : VAEntrypointEncSlice VAProfileH264Main : VAEntrypointVLD VAProfileH264Main : VAEntrypointEncSlice VAProfileH264High : VAEntrypointVLD VAProfileVC1Simple : VAEntrypointVLD VAProfileVC1Main : VAEntrypointVLD VAProfileVC1Advanced : VAEntrypointVLD VAProfileH263Baseline : VAEntrypointEncSlice I have install the following package: gstreamer-0.10.30 gst-plugins-base-0.10.30 gst-plugins-good-0.10.24 gst-plugins-bad-0.10.19 gst-plugins-ugly-0.10.15 gst-ffmpeg-0.10.10 ffmpeg When I configure gstreamer-vaapi use command ./configure --enable-vaapi-glx --enable-glx --enable-vaapi-glx, the log info shows like this: gstreamer-vaapi configuration summary: VA-API version ................... : 0.31.1 GLX support ...................... : yes VA/GLX support ................... : yes VaapiSink/GL ..................... : yes These are details: checking for X11... yes checking for GL/gl.h... yes checking for GL/glext.h... yes checking for GL/glx.h... yes checking for glXCreateContext in -lGL... yes checking for LIBVA... yes checking for old VA-API 0.29... no checking for LIBVA_X11... yes checking for LIBVA_GLX... yes checking for LIBAVCODEC... yes checking libavcodec/avcodec.h usability... yes checking libavcodec/avcodec.h presence... yes checking for libavcodec/avcodec.h... yes checking ffmpeg/avcodec.h usability... no checking ffmpeg/avcodec.h presence... no checking for ffmpeg/avcodec.h... no checking libavcodec/vaapi.h usability... yes checking libavcodec/vaapi.h presence... yes checking for libavcodec/vaapi.h... yes I also see that software requirements of gstreamer-vaapi need libva-dev >= 0.31.0-1+sds9 (VA/GLX),what is sds9? I download libva by git form git clone git://anongit.freedesktop.org/git/libva. It contains sds? Can anybody give me any suggestions? thanks a lot. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-enable-gstreamer-vaapi-tp3085318p3085318.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From cristianurban86 at gmail.com Mon Dec 13 14:40:32 2010 From: cristianurban86 at gmail.com (cristiurban) Date: Mon, 13 Dec 2010 05:40:32 -0800 (PST) Subject: [gst-devel] mpegaudioparse warnings Message-ID: <1292247632658-3085421.post@n4.nabble.com> Hello, I am using Phonon backend with gstreamer plugins to create an media player. Adding some code to my phonon backend where I enabled debugging on warning level I was able to capture the warnings thrown by "mpegaudioparse" element and capture it in my phonon backend in GST_MESSAGE_WARNINGS on the D-bus. The function in the element is static gboolean head_check (GstMPEGAudioParse * mp3parse, unsigned long head) where it does a header check of the audio file and when it finds a bad header it throws warnings on the D-bus . My problem is that there are to many warnings over 1 million messages. If I pause the application I still get this messages from the first detection. Nothing wrong is happining with my pipeline but my bus message is working hard to receive this warnings. I tried to modify the mpegaudioparse to give me a limited warnings to reduce the time consumption of the d-bus. I was not able to succed ... any help will be welcome. Thanks, Cristian Urban -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/mpegaudioparse-warnings-tp3085421p3085421.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Mon Dec 13 15:00:07 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 13 Dec 2010 14:00:07 +0000 Subject: [gst-devel] mpegaudioparse warnings In-Reply-To: <1292247632658-3085421.post@n4.nabble.com> References: <1292247632658-3085421.post@n4.nabble.com> Message-ID: <1292248807.16211.9.camel@zingle> On Mon, 2010-12-13 at 05:40 -0800, cristiurban wrote: > I am using Phonon backend with gstreamer plugins to create an media player. > Adding some code to my phonon backend where I enabled debugging on warning > level I was able to capture the warnings thrown by "mpegaudioparse" element > and capture it in my phonon backend in GST_MESSAGE_WARNINGS on the D-bus. > The function in the element is static gboolean > head_check (GstMPEGAudioParse * mp3parse, unsigned long head) where it does > a header check of the audio file and when it finds a bad header it throws > warnings on the D-bus . My problem is that there are to many warnings over 1 > million messages. If I pause the application I still get this messages from > the first detection. Nothing wrong is happining with my pipeline but my bus > message is working hard to receive this warnings. I tried to modify the > mpegaudioparse to give me a limited warnings to reduce the time consumption > of the d-bus. I was not able to succed ... any help will be welcome. Well, you probably shouldn't be posting GStreamer debug log lines onto any message busses, be it a GstBus or D-Bus. When you pause the pipeline, only the sinks stop rendering data (if you're playing back a local file). The rest of the pipeline will continue to process data until the queues fill up (and block upstream). Cheers -Tim From wmiller at sdr.com Mon Dec 13 17:49:09 2010 From: wmiller at sdr.com (Wes Miller) Date: Mon, 13 Dec 2010 08:49:09 -0800 (PST) Subject: [gst-devel] How do u set caps for udpsrc In-Reply-To: <1292244042611-3085367.post@n4.nabble.com> References: <1292244042611-3085367.post@n4.nabble.com> Message-ID: <1292258949698-3085734.post@n4.nabble.com> You can't just specify the caps string as you do with gst-launch. But it's almost as simple. See the help for Gstcaps and this function: GstCaps * gst_caps_from_string( const gchar *string ); Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-do-u-set-caps-for-udpsrc-tp3085367p3085734.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From msmith at xiph.org Mon Dec 13 19:23:17 2010 From: msmith at xiph.org (Michael Smith) Date: Mon, 13 Dec 2010 10:23:17 -0800 Subject: [gst-devel] Pipeline gets stuck at PAUSED state In-Reply-To: <4D0491F6.1060403@users.sourceforge.net> References: <4D0491F6.1060403@users.sourceforge.net> Message-ID: On Sun, Dec 12, 2010 at 1:12 AM, p-t-r wrote: > Hi, > > I installed GStreamer 0.10 on an embedded system. > When I try to run gst-launch like this > ?gst-launch -v fakesrc num-buffers=5 ! fakesink > it gets stuck at this line: > ?Setting pipeline to PAUSED ... That's a very simple pipeline that certainly should work. You should give us much, much more detail about your system, how you've built gstreamer, and so on. Presumably, if this isn't working, gstreamer itself isn't working right _at all_ - you should ensure that the core unit tests are passing in your environment, and if they're not, the ones that are failing may well give you a strong hint about where the problem might be. Mike From msmith at xiph.org Mon Dec 13 19:25:15 2010 From: msmith at xiph.org (Michael Smith) Date: Mon, 13 Dec 2010 10:25:15 -0800 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 49 In-Reply-To: References: Message-ID: On Sun, Dec 12, 2010 at 2:19 AM, Shamun toha md wrote: > Hello, > Thanks for the info. udpsrc cause?ERROR: pipeline could not be constructed: > no element "udpsrc". > But this works in Gstreamer 0.10.29 but not in 0.10.20. > Thanks & Regards > Shamun You seem to be confusing "GStreamer version" with "plugins you have installed". That message is pretty clear - you don't have a loadable udpsrc plugin installed. Possibly you don't have it at all, or possibly you have a version that requires a newer GStreamer. If it's the latter, then the cause is you installing random sets of incompatible software - don't do that. Mike From abarajase at gmail.com Mon Dec 13 21:48:40 2010 From: abarajase at gmail.com (Hector Barajas) Date: Mon, 13 Dec 2010 12:48:40 -0800 (PST) Subject: [gst-devel] mmap support on gstalsasink In-Reply-To: References: <20101209010209.GA22402@cooker.entropywave.com> <1291888715.1995.9.camel@meany> Message-ID: <1292273320653-3086035.post@n4.nabble.com> Thank you all for your comments I think I could start a patch on this that does not mess with default functionality and follows the recommendations given here, and then share it to you guys to get feedback from the experts, maybe the patch can be base for having this feature on gst base again, please let me know if you have any other points/considerations for the implementation. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/mmap-support-on-gstalsasink-tp3075546p3086035.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From braunsquared at gmail.com Mon Dec 13 23:33:07 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Mon, 13 Dec 2010 17:33:07 -0500 Subject: [gst-devel] GnlComposition Signals Message-ID: Hey Everyone, So I have a gnlcomposition which, for the matter of depicting the situation, has 5 sources. 3 sources play for 30 seconds followed by the remaining 2 sources for another 30 seconds. This should produce a 1 minute video. The composition has a videomixer in a gnloperation to mix the sources for final output. I'm running into a situation where the pipeline gets stuck during the transition from 3 sources to 2 sources. It seems the video mixer is trying to send a seek event to all it's sink pads, which fails on the pad which no-longer has a source. During my tests, the gnlcomposition will happily create new pads, ie. changing the order so it's 2 sources followed by 3 sources works fine. But the opposite doesn't seem to be true, the gnlcomposition doesn't remove unused pads to it's operations. This brings me to the question of, is there a signal I'm not handling appropriately? If so, any pointers on which one I should be listening for? Thanks, Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: From gurpreet at tataelxsi.co.in Tue Dec 14 07:36:05 2010 From: gurpreet at tataelxsi.co.in (Gurpreet) Date: Tue, 14 Dec 2010 12:06:05 +0530 Subject: [gst-devel] Regarding Avimux Message-ID: <4D071055.7070806@tataelxsi.co.in> Hi All.. I am using avimux for muxing. But i have one doubt. how it is managing audio and video data ? Like how it is differentating between audio and video data ? How they are knowing that this is audio data and this is video data ? -- / Gurpreet Singh Tata Elxsi Ltd. Bangalore +91-8022984127 +91-9019219511 / Rabba Mehar Kari Asi Ud-de Aasre Tere -------------- next part -------------- An HTML attachment was scrubbed... URL: From gurpreet at tataelxsi.co.in Tue Dec 14 07:44:39 2010 From: gurpreet at tataelxsi.co.in (Gurpreet) Date: Tue, 14 Dec 2010 12:14:39 +0530 Subject: [gst-devel] Regarding Avimux Message-ID: <4D071257.1040300@tataelxsi.co.in> Hi All.. I am using avimux for muxing. But i have one doubt. how it is managing audio and video data ? Like how it is differentating between audio and video data ? How they are knowing that this is audio data and this is video data ? Thanks Gurpreet -- / / -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Tue Dec 14 07:45:35 2010 From: msmith at xiph.org (Michael Smith) Date: Mon, 13 Dec 2010 22:45:35 -0800 Subject: [gst-devel] Regarding Avimux In-Reply-To: <4D071055.7070806@tataelxsi.co.in> References: <4D071055.7070806@tataelxsi.co.in> Message-ID: The caps (GstCaps objects) associated with all GStreamer data streams describe the data being carried - in much more detail than merely audio vs. video, but certainly including this information. This should be pretty obvious if you've read any of the documentation. Mike On Mon, Dec 13, 2010 at 10:36 PM, Gurpreet wrote: > Hi All.. > > ??? I am using avimux for muxing. But i have one doubt. how it is managing > audio and video data ? Like how it is differentating between audio and video > data ? How they are knowing that this is audio data and this is video data ? > -- > > Gurpreet Singh > Tata Elxsi Ltd. > Bangalore > +91-8022984127 > +91-9019219511 From shamun.toha at gmail.com Tue Dec 14 09:21:28 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Tue, 14 Dec 2010 09:21:28 +0100 Subject: [gst-devel] Java gstreamer wrapper. Impossible ? Message-ID: Hello, Is there any Java gstreamer wrapper please ? Which can be used for cross platform ? Thanks & Regards Shamun -------------- next part -------------- An HTML attachment was scrubbed... URL: From braunsquared at gmail.com Tue Dec 14 09:38:37 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Tue, 14 Dec 2010 03:38:37 -0500 Subject: [gst-devel] Java gstreamer wrapper. Impossible ? In-Reply-To: References: Message-ID: http://code.google.com/p/gstreamer-java/ On Tue, Dec 14, 2010 at 3:21 AM, Shamun toha md wrote: > Hello, > > Is there any Java gstreamer wrapper please ? Which can be used for cross > platform ? > > Thanks & Regards > Shamun > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gang.a.hu at intel.com Tue Dec 14 11:08:30 2010 From: gang.a.hu at intel.com (Hu, Gang A) Date: Tue, 14 Dec 2010 18:08:30 +0800 Subject: [gst-devel] Looking for multi-pads source element Message-ID: <1A42CE6F5F474C41B63392A5F80372B231B41978@shsmsx501.ccr.corp.intel.com> Hi The basesrc only have one pad. I want to add another static pads to basesrc. Is there any source element which have more than one pads? I am looking for such element for reference. Thanks. Best Regards! Hu Gang -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Tue Dec 14 11:42:43 2010 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 14 Dec 2010 11:42:43 +0100 Subject: [gst-devel] GnlComposition Signals In-Reply-To: References: Message-ID: <1292323363.27782.2.camel@localhost> On Mon, 2010-12-13 at 17:33 -0500, Timothy Braun wrote: > Hey Everyone, > So I have a gnlcomposition which, for the matter of depicting the > situation, has 5 sources. 3 sources play for 30 seconds followed by > the remaining 2 sources for another 30 seconds. This should produce a > 1 minute video. The composition has a videomixer in a gnloperation to > mix the sources for final output. > > I'm running into a situation where the pipeline gets stuck during > the transition from 3 sources to 2 sources. It seems the video mixer > is trying to send a seek event to all it's sink pads, which fails on > the pad which no-longer has a source. During my tests, the > gnlcomposition will happily create new pads, ie. changing the order so > it's 2 sources followed by 3 sources works fine. But the opposite > doesn't seem to be true, the gnlcomposition doesn't remove unused pads > to it's operations. > > This brings me to the question of, is there a signal I'm not > handling appropriately? If so, any pointers on which one I should be > listening for? If the element you put in a gnloperation has dynamic sink pads, the gnloperation will call gst_element_request_pad() and gst_element_release_request_pad() when needed. videomixer alone should work fine in those cases. If you have put a custom element with ghostpads, make sure you check when they are added/removed and properly request/release the targets of those ghostpads. Edward > > Thanks, > Tim > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Tue Dec 14 11:50:01 2010 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 14 Dec 2010 11:50:01 +0100 Subject: [gst-devel] Java gstreamer wrapper. Impossible ? In-Reply-To: References: Message-ID: <1292323801.27782.4.camel@localhost> On Tue, 2010-12-14 at 09:21 +0100, Shamun toha md wrote: > Hello, > > > > Is there any Java gstreamer wrapper please ? Which can be used for > cross platform ? Hi, Could you recommend me a good web search engine ? One where preferably I don't have to type any obvious keywords to get the answer to what I want ? Especially not when the first answer to the query would be what I was looking for. Edward > > > Thanks & Regards > Shamun > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From katcipis at inf.ufsc.br Tue Dec 14 16:17:39 2010 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Tue, 14 Dec 2010 13:17:39 -0200 Subject: [gst-devel] Looking for multi-pads source element In-Reply-To: <1A42CE6F5F474C41B63392A5F80372B231B41978@shsmsx501.ccr.corp.intel.com> References: <1A42CE6F5F474C41B63392A5F80372B231B41978@shsmsx501.ccr.corp.intel.com> Message-ID: On Tue, Dec 14, 2010 at 8:08 AM, Hu, Gang A wrote: > Hi > The basesrc only have one pad. I want to add another static pads to > basesrc. > Is there any source element which have more than one pads? I am looking > for such element for reference. > Thanks. > A tee ( http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-tee.html) doesn't resolve your problem? i think sources usually have only one pad because it is pretty easy to copy the source data to multiple streams using the tee element. Don't know if your case is different, hope this helps you. best regards, Katcipis > > Best Regards! > *Hu Gang* > > > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- http://www.getgnulinux.org/windows -------------- next part -------------- An HTML attachment was scrubbed... URL: From wb666greene at gmail.com Tue Dec 14 16:52:18 2010 From: wb666greene at gmail.com (wally_bkg) Date: Tue, 14 Dec 2010 07:52:18 -0800 (PST) Subject: [gst-devel] v4l2src->appsink appsrc->xvimagesink fails to change video norm from PAL to NTSC In-Reply-To: <1291766179.20631.31.camel@zingle> References: <1291738733147-3076696.post@n4.nabble.com> <1291765183409-3077453.post@n4.nabble.com> <1291766179.20631.31.camel@zingle> Message-ID: <1292341938465-3087351.post@n4.nabble.com> Tim-Philipp M?ller-2 wrote: > > On Tue, 2010-12-07 at 15:39 -0800, wally_bkg wrote: > >> Looks like another undocumented or unintentional change to gstreamer, >> need to >> set pipeline to READY instead of PAUSED now before using the interface. >> >> Change: >> >> gst_element_set_state (pipeline, GST_STATE_PAUSED); >> >> To: >> >> gst_element_set_state (pipeline, GST_STATE_READY); >> >> And the PAL to NTSC switch happens correctly now. > > I believe this change (open device when going to READY, not PAUSED) was > intentional, to align v4l2src with other elements (opening the device in > the NULL->READY state change), but by itself shouldn't break anything. > It shouldn't make a difference at the v4l2src level, since the device > will go to READY state before going to PAUSED state. > I've ran into this problem again now that I've split my pipeline into two pipelines. v4l2src->appsink appsrc->xvimagesink I assumed this issue was closed from the above, but we had a power failure so my system was reset and of course defaults to PAL on power-up. I was shocked to find it fail when I ran my code. wally at wahine:~/gst-learn$ ./VideoPipeline The v4l2 device is '/dev/video0'. Previous Norm: PAL Current Norm: NTSC Signals will be emitted. max buffers: 0 drop buffers: FALSE Reading from appsrc to xvimagesink Capturing from v4l2src to appsink isource failed to go into PLAYING state Main Loop Running... Error: Could not negotiate format Returned, stopping playback Deleting pipelines but simply running the program again after it terminates works: wally at wahine:~/gst-learn$ ./VideoPipeline The v4l2 device is '/dev/video0'. Previous Norm: NTSC Current Norm: NTSC Signals will be emitted. max buffers: 0 drop buffers: FALSE Reading from appsrc to xvimagesink Capturing from v4l2src to appsink on_new_buffer_from_source Callback entered. Main Loop Running... I'm using the appsink-src.c as a guide and don't understand why the v4l2 NORM switch doesn't seem to happen until the program errors and exits now. My "hello word" video program still works which is just: v4l2src->xvimagesink I can recreate the problem by stating XawTV, setting PAL mode and exiting it, leaving the card back in PAL mode. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/v4l2src-fails-to-change-video-norm-from-PAL-to-NTSC-tp3076696p3087351.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From mjbraun at leydenjar.com Tue Dec 14 17:42:21 2010 From: mjbraun at leydenjar.com (Matthew Braun) Date: Tue, 14 Dec 2010 10:42:21 -0600 Subject: [gst-devel] Cannot set caps on udpsrc0 to application/x-rtp under OS X Message-ID: Greetings! I'm streaming from a webcam attached to a BeagleBoard using: gst-launch -v v4l2src ! video/x-raw-yuv, width=640, height=480 ! ffmpegcolorspace ! TIVidenc1 codecName=h264enc engineName=codecServer ! rtph264pay pt=96 ! udpsink host= port=5000 Which runs and outputs caps of: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, sprop-parameter-sets=(string)\"Z0KAHukBQHpCAAAH0AAB1MAIAA\\=\\=\\,aM48gAA\\ =\", payload=(int)96, ssrc=(uint)1421770711, clock-base=(uint)3676471020, seqnum-base=(uint)12939 But when I run gst-launch on my OS X destination with % CAPS="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, sprop-parameter-sets=(string)\"Z0KAHukBQHpCAAAH0AAB1MAIAA\\=\\=\\,aM48gAA\\ =\", payload=(int)96, ssrc=(uint)3775499150, clock-base=(uint)2462209911, seqnum-base=(uint)27054" % gst-launch -v udpsrc caps=$CAPS port=5000 ! rtph264depay ! ffdec_h264 ! Autovideosink I get WARNING: erroneous pipeline: could not set property "caps" in element "udpsrc0" to "application/x-rtp," Trying without caps gives: "Input buffers need to have RTP caps set on them. This is usually achieved by setting the 'caps' property of the upstream source element (often udpsrc or appsrc), or by putting a capsfilter element before the depayloader and setting the 'caps' property on that. Also see http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README" Any ideas as to what I'm missing? Thanks for any guidance and/or suggestions! (% gst-launch --version gst-launch-0.10 version 0.10.31 GStreamer 0.10.31 ) -- Matthew Braun mjbraun at leydenjar.com From braunsquared at gmail.com Tue Dec 14 19:34:28 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Tue, 14 Dec 2010 13:34:28 -0500 Subject: [gst-devel] GnlComposition Signals In-Reply-To: <1292323363.27782.2.camel@localhost> References: <1292323363.27782.2.camel@localhost> Message-ID: Thank you Edward. That makes sense. I have a custom bin with ghosted pads in a gnlsource. I'm currently ghosting the pad upon creation and then adding the final internal link once decodebin2 tells me it's ready. On occasion, I run into a data flow error, but not consistently. I'm thinking it may be related to the order in which I'm adding pads. Is it better to ghost the bins pad after decodebin is connected? or does this order not matter? Also, if I'm ghosting the pad, do I have to manually remove it once streaming of that source is complete? If so, what signal should I be listening for from gnlsource/gnlcomposition to know that its completed it's duties? Thanks again. Best, Tim On Tue, Dec 14, 2010 at 5:42 AM, Edward Hervey wrote: > On Mon, 2010-12-13 at 17:33 -0500, Timothy Braun wrote: > > Hey Everyone, > > So I have a gnlcomposition which, for the matter of depicting the > > situation, has 5 sources. 3 sources play for 30 seconds followed by > > the remaining 2 sources for another 30 seconds. This should produce a > > 1 minute video. The composition has a videomixer in a gnloperation to > > mix the sources for final output. > > > > I'm running into a situation where the pipeline gets stuck during > > the transition from 3 sources to 2 sources. It seems the video mixer > > is trying to send a seek event to all it's sink pads, which fails on > > the pad which no-longer has a source. During my tests, the > > gnlcomposition will happily create new pads, ie. changing the order so > > it's 2 sources followed by 3 sources works fine. But the opposite > > doesn't seem to be true, the gnlcomposition doesn't remove unused pads > > to it's operations. > > > > This brings me to the question of, is there a signal I'm not > > handling appropriately? If so, any pointers on which one I should be > > listening for? > > If the element you put in a gnloperation has dynamic sink pads, the > gnloperation will call gst_element_request_pad() and > gst_element_release_request_pad() when needed. > videomixer alone should work fine in those cases. If you have put a > custom element with ghostpads, make sure you check when they are > added/removed and properly request/release the targets of those > ghostpads. > > Edward > > > > > Thanks, > > Tim > > > ------------------------------------------------------------------------------ > > Lotusphere 2011 > > Register now for Lotusphere 2011 and learn how > > to connect the dots, take your collaborative environment > > to the next level, and enter the era of Social Business. > > http://p.sf.net/sfu/lotusphere-d2d > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Tue Dec 14 22:04:44 2010 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 14 Dec 2010 22:04:44 +0100 Subject: [gst-devel] GnlComposition Signals In-Reply-To: References: <1292323363.27782.2.camel@localhost> Message-ID: <1292360684.10079.13.camel@deumeu> On Tue, 2010-12-14 at 13:34 -0500, Timothy Braun wrote: > Thank you Edward. That makes sense. > > I have a custom bin with ghosted pads in a gnlsource. I'm currently > ghosting the pad upon creation and then adding the final internal link > once decodebin2 tells me it's ready. If you do that... gnloperation will think there's a static number of sink pads. gnloperation will dynamically request/release pads only if the element it's controlling has a request sink pad template... meaning it'll work with videomixer, but you'll have to do your own subclass of GstBin with the proper pad templates to make it work (I'm assuming you want to put some extra elements before videomixer). > On occasion, I run into a data flow error, but not consistently. > I'm thinking it may be related to the order in which I'm adding pads. > Is it better to ghost the bins pad after decodebin is connected? or > does this order not matter? For sources it doesn't matter. > > Also, if I'm ghosting the pad, do I have to manually remove it once > streaming of that source is complete? If so, what signal should I be > listening for from gnlsource/gnlcomposition to know that its completed > it's duties? I'm assuming you mean for operations (doesn't matter for sources). For operations with dynamic pads you want to make sure that: * Your element (the one you put in gnloperation) * Has a sink pad template of type GST_PAD_REQUEST * you have overriden the GstElementClass::request_new_pad vmethod In there you request a pad from videomixer, add your extra elements, ghost the first element's sink pad, add that pad to yourself, and return that pad * you have overriden the GstElementClass::release_request_pad vmethod In there you remove that pad from yourself, release the sink pad you requested from videomixer, and remove all elements for that path * You listen to the 'input-priority-changed' signal from gnloperation This will notify you of changes in priority of incoming stream for each of your ghost sink pad. When you receive that, you change the 'associated' videomixer sinkpad for that ghostpad of yours by setting the 'zorder' property of that videomixer sinkpad to the priority you receive g_object_set(thatstream->videomixersinkpad, "zorder", thenewpriority, NULL); This assumes you need to insert extra elements before/after your videomixer in that operation (else you can just stick videomixer in a gnloperation and all of that will be taken care of). Hope this helps, Edward > > Thanks again. > > Best, > Tim > > On Tue, Dec 14, 2010 at 5:42 AM, Edward Hervey > wrote: > > On Mon, 2010-12-13 at 17:33 -0500, Timothy Braun wrote: > > Hey Everyone, > > So I have a gnlcomposition which, for the matter of > depicting the > > situation, has 5 sources. 3 sources play for 30 seconds > followed by > > the remaining 2 sources for another 30 seconds. This should > produce a > > 1 minute video. The composition has a videomixer in a > gnloperation to > > mix the sources for final output. > > > > I'm running into a situation where the pipeline gets stuck > during > > the transition from 3 sources to 2 sources. It seems the > video mixer > > is trying to send a seek event to all it's sink pads, which > fails on > > the pad which no-longer has a source. During my tests, the > > gnlcomposition will happily create new pads, ie. changing > the order so > > it's 2 sources followed by 3 sources works fine. But the > opposite > > doesn't seem to be true, the gnlcomposition doesn't remove > unused pads > > to it's operations. > > > > This brings me to the question of, is there a signal I'm > not > > handling appropriately? If so, any pointers on which one I > should be > > listening for? > > > If the element you put in a gnloperation has dynamic sink > pads, the > gnloperation will call gst_element_request_pad() and > gst_element_release_request_pad() when needed. > videomixer alone should work fine in those cases. If you have > put a > custom element with ghostpads, make sure you check when they > are > added/removed and properly request/release the targets of > those > ghostpads. > > Edward > > > > > Thanks, > > Tim > > > ------------------------------------------------------------------------------ > > Lotusphere 2011 > > Register now for Lotusphere 2011 and learn how > > to connect the dots, take your collaborative environment > > to the next level, and enter the era of Social Business. > > http://p.sf.net/sfu/lotusphere-d2d > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From radivojejovanovic at gmail.com Tue Dec 14 22:22:13 2010 From: radivojejovanovic at gmail.com (Radivoje Jovanovic) Date: Tue, 14 Dec 2010 13:22:13 -0800 Subject: [gst-devel] GST_FLOW_WRONG_STATE In-Reply-To: References: <1291853143.3531.43.camel@zingle> Message-ID: Thank you for all of your input. Unfortunately I did not write the decoders at all, but I do have the source code. What is considered as decoder latency? Is it the time for decoder to set up or something else? Cheers Ogi On Thu, Dec 9, 2010 at 7:45 PM, ved kpl wrote: > On Fri, Dec 10, 2010 at 1:39 AM, Radivoje Jovanovic > wrote: > > It seems like we are on a good path. after I tried the pipeline: > > gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue > > max-size-time=0 max-size-bytes=0 ! mpeg4dec ! MY_VIDEO_SINK t. ! queue > > ! faad ! alsasink > > (so I only give time and bytes for video side) the video and audio > worked > > just fine. > > > > Unfortunately this trick does not work for the other decoder I have which > > decodes h264 videos. > > > > Even with the pipeline: > > gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue > > max-size-time=0 max-size-bytes=0 ! h264dec ! MY_VIDEO_SINK t. ! queue > > ! faad ! alsasink > > The pipeline is stuck. This all has nothing to do with > GST_FLOW_WRONG_STATE > > since this error pops up all the time after CTRL-C operation. Any idea > why > > would h264 get stuck? > > To start off, try setting max-size-time/bytes/buffers=0 on the both > the queues (no limits) > If it works, it indicates that your h264dec has a very high > latency..and you can > narrow to the problem from here. > > > Another question is if I am giving too much memory to gstreamer with > > max-size-time=0 max-size-bytes=0? > > Well, It would depend on your codec details (resilution, etc), > environment/platform. You could possibly track down the issue. > > > Cheers > > Ogi > > > > > > On Wed, Dec 8, 2010 at 8:27 PM, ved kpl wrote: > >> > >> Hi, > >> > >> One possible reason could be that one of the queues is getting full > >> while the other remains empty > >> because of > >> (a) continuous audio/video buffers for a considerably longer time, > >> longer than the what the queues can hold. > >> (b) high latency at the your video decoder (mpeg4dec). > >> Hence the streaming thread is blocked & one of the sinks is not able > >> to commit the state to PAUSED. > >> (because it has not received a buffer yet). > >> > >> Yo can also set async=FALSE on the sinks and see. (not recommended) > >> > >> You can try disabling the the max time & max bytes properties of the > >> queue. > >> Try the following pipeline > >> > >> gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue > >> max-size-time=0 max-size-bytes=0 ! mpeg4dec ! MY_VIDEO_SINK t. ! queue > >> max-size-time=0 max-size-bytes=0 ! faad ! alsasink -v > >> > >> Note that this will increase the mem usage.. and even more if u set > >> max-size-buffers=0 as well. (queue without any limits!) > >> > >> Vikram > >> > >> On Thu, Dec 9, 2010 at 9:44 AM, ved kpl wrote: > >> > On Thu, Dec 9, 2010 at 6:26 AM, Radivoje Jovanovic > >> > wrote: > >> >> So the pipeline I am using is: > >> >> gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue ! > >> >> mpeg4dec ! MY_VIDEO_SINK t. ! queue ! faad ! alsasink > >> >> This pipeline will hang and after CTRL-C the mpeg4dec will show the > >> >> message > >> >> that is the result of calling gst_push_pad. This gst_push_pad will > >> >> return > >> >> GST_FLOW_WRONG_STATE. > >> >> > >> >> If I run gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! > >> >> queue > >> >> ! mpeg4dec ! MY_VIDEO_SINK > >> >> the video will play just fine > >> >> > >> >> If I run: > >> >> gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue ! > >> >> faad ! > >> >> alsasink > >> >> the audio from the audio will play just fine. > >> >> > >> >> I am not sure which code would be helpful since I have a lot of it. > >> >> Thanks for the GST_DEBUG info. > >> >> > >> >> Ogi > >> >> > >> >> > >> >> On Wed, Dec 8, 2010 at 4:05 PM, Tim-Philipp M?ller > >> >> wrote: > >> >>> > >> >>> On Wed, 2010-12-08 at 12:53 -0800, Radivoje Jovanovic wrote: > >> >>> > >> >>> > I have a video sink and ALSA driver to be used to play videos on > the > >> >>> > platform. If I play video or audio by them self everything works > >> >>> > fine, > >> >>> > but if I specify the whole pipeline for video and audio, the video > >> >>> > decoder (in pipeline right before the sink) comes back with the > >> >>> > message: > >> >>> > "The decoded frame did not successfully push out to downstream > >> >>> > element" > >> >>> > The message comes because the gst_pad_push have returned with > >> >>> > GST_FLOW_WRONG_STATE instead GST_FLOW_OK. > >> >>> > > >> >>> > Any idea why this might happen? > >> >>> > >> >>> It usually helps if you post your exact pipeline and/or any code > >> >>> snippets that go with it. > >> >>> > >> >>> FLOW_WRONG_STATE is what you get when a pad is flushing, which may > be > >> >>> normal (happens during a flushing seek, to make the old streaming > >> >>> thread > >> >>> stop) or because you forgot to set an element into PAUSED/PLAYING > >> >>> state > >> >>> (e.g. because you added it from a pad-added or new-decoded-pad > >> >>> callback > >> >>> or so). > >> >>> > >> >>> The GST_DEBUG=*:5 log might give you more information (just grep for > >> >>> wrong-state and read the lines before that). > >> >>> > >> >>> Cheers > >> >>> -Tim > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > ------------------------------------------------------------------------------ > >> >>> This SF Dev2Dev email is sponsored by: > >> >>> > >> >>> WikiLeaks The End of the Free Internet > >> >>> http://p.sf.net/sfu/therealnews-com > >> >>> _______________________________________________ > >> >>> gstreamer-devel mailing list > >> >>> gstreamer-devel at lists.sourceforge.net > >> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >> >> > >> >> > >> >> > >> >> > ------------------------------------------------------------------------------ > >> >> This SF Dev2Dev email is sponsored by: > >> >> > >> >> WikiLeaks The End of the Free Internet > >> >> http://p.sf.net/sfu/therealnews-com > >> >> _______________________________________________ > >> >> gstreamer-devel mailing list > >> >> gstreamer-devel at lists.sourceforge.net > >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >> >> > >> >> > >> > > >> > >> > >> > ------------------------------------------------------------------------------ > >> This SF Dev2Dev email is sponsored by: > >> > >> WikiLeaks The End of the Free Internet > >> http://p.sf.net/sfu/therealnews-com > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > ------------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From braunsquared at gmail.com Tue Dec 14 22:31:55 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Tue, 14 Dec 2010 16:31:55 -0500 Subject: [gst-devel] GnlComposition Signals In-Reply-To: <1292360684.10079.13.camel@deumeu> References: <1292323363.27782.2.camel@localhost> <1292360684.10079.13.camel@deumeu> Message-ID: Thank You Edward. It was a great help. I will make an attempt at it in the coming days. Much appreciated. Tim On Tue, Dec 14, 2010 at 4:04 PM, Edward Hervey wrote: > On Tue, 2010-12-14 at 13:34 -0500, Timothy Braun wrote: > > Thank you Edward. That makes sense. > > > > I have a custom bin with ghosted pads in a gnlsource. I'm currently > > ghosting the pad upon creation and then adding the final internal link > > once decodebin2 tells me it's ready. > > If you do that... gnloperation will think there's a static number of > sink pads. > gnloperation will dynamically request/release pads only if the element > it's controlling has a request sink pad template... meaning it'll work > with videomixer, but you'll have to do your own subclass of GstBin with > the proper pad templates to make it work (I'm assuming you want to put > some extra elements before videomixer). > > > On occasion, I run into a data flow error, but not consistently. > > I'm thinking it may be related to the order in which I'm adding pads. > > Is it better to ghost the bins pad after decodebin is connected? or > > does this order not matter? > > For sources it doesn't matter. > > > > > Also, if I'm ghosting the pad, do I have to manually remove it once > > streaming of that source is complete? If so, what signal should I be > > listening for from gnlsource/gnlcomposition to know that its completed > > it's duties? > > I'm assuming you mean for operations (doesn't matter for sources). > > For operations with dynamic pads you want to make sure that: > > * Your element (the one you put in gnloperation) > > * Has a sink pad template of type GST_PAD_REQUEST > * you have overriden the GstElementClass::request_new_pad vmethod > In there you request a pad from videomixer, add your extra > elements, ghost the first element's sink pad, add that pad to yourself, > and return that pad > * you have overriden the GstElementClass::release_request_pad vmethod > In there you remove that pad from yourself, release the sink pad > you requested from videomixer, and remove all elements for that path > > * You listen to the 'input-priority-changed' signal from gnloperation > > This will notify you of changes in priority of incoming stream for > each of your ghost sink pad. > When you receive that, you change the 'associated' videomixer sinkpad > for that ghostpad of yours by setting the 'zorder' property of that > videomixer sinkpad to the priority you receive > > g_object_set(thatstream->videomixersinkpad, "zorder", thenewpriority, > NULL); > > > This assumes you need to insert extra elements before/after your > videomixer in that operation (else you can just stick videomixer in a > gnloperation and all of that will be taken care of). > > Hope this helps, > > Edward > > > > > Thanks again. > > > > Best, > > Tim > > > > On Tue, Dec 14, 2010 at 5:42 AM, Edward Hervey > > wrote: > > > > On Mon, 2010-12-13 at 17:33 -0500, Timothy Braun wrote: > > > Hey Everyone, > > > So I have a gnlcomposition which, for the matter of > > depicting the > > > situation, has 5 sources. 3 sources play for 30 seconds > > followed by > > > the remaining 2 sources for another 30 seconds. This should > > produce a > > > 1 minute video. The composition has a videomixer in a > > gnloperation to > > > mix the sources for final output. > > > > > > I'm running into a situation where the pipeline gets stuck > > during > > > the transition from 3 sources to 2 sources. It seems the > > video mixer > > > is trying to send a seek event to all it's sink pads, which > > fails on > > > the pad which no-longer has a source. During my tests, the > > > gnlcomposition will happily create new pads, ie. changing > > the order so > > > it's 2 sources followed by 3 sources works fine. But the > > opposite > > > doesn't seem to be true, the gnlcomposition doesn't remove > > unused pads > > > to it's operations. > > > > > > This brings me to the question of, is there a signal I'm > > not > > > handling appropriately? If so, any pointers on which one I > > should be > > > listening for? > > > > > > If the element you put in a gnloperation has dynamic sink > > pads, the > > gnloperation will call gst_element_request_pad() and > > gst_element_release_request_pad() when needed. > > videomixer alone should work fine in those cases. If you have > > put a > > custom element with ghostpads, make sure you check when they > > are > > added/removed and properly request/release the targets of > > those > > ghostpads. > > > > Edward > > > > > > > > Thanks, > > > Tim > > > > > > ------------------------------------------------------------------------------ > > > Lotusphere 2011 > > > Register now for Lotusphere 2011 and learn how > > > to connect the dots, take your collaborative environment > > > to the next level, and enter the era of Social Business. > > > http://p.sf.net/sfu/lotusphere-d2d > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > ------------------------------------------------------------------------------ > > Lotusphere 2011 > > Register now for Lotusphere 2011 and learn how > > to connect the dots, take your collaborative environment > > to the next level, and enter the era of Social Business. > > http://p.sf.net/sfu/lotusphere-d2d > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > > Lotusphere 2011 > > Register now for Lotusphere 2011 and learn how > > to connect the dots, take your collaborative environment > > to the next level, and enter the era of Social Business. > > http://p.sf.net/sfu/lotusphere-d2d > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From thiago.sousa.santos at collabora.co.uk Wed Dec 15 02:19:40 2010 From: thiago.sousa.santos at collabora.co.uk (Thiago Sousa Santos) Date: Tue, 14 Dec 2010 22:19:40 -0300 Subject: [gst-devel] Looking for multi-pads source element In-Reply-To: <1A42CE6F5F474C41B63392A5F80372B231B41978@shsmsx501.ccr.corp.intel.com> References: <1A42CE6F5F474C41B63392A5F80372B231B41978@shsmsx501.ccr.corp.intel.com> Message-ID: <1292375980.12069.9.camel@blacksheep> On Tue, 2010-12-14 at 18:08 +0800, Hu, Gang A wrote: > Hi > The basesrc only have one pad. I want to add another static pads to > basesrc. > Is there any source element which have more than one pads? I am > looking for such element for reference. > Thanks. I'm assuming you're interested in camerabin2 and its 3pad source. I've been drafting a base class for it lately. Maybe we could discuss and get this working together? I've been wondering if it would be good to inherit from GstBaseSrc or GstPushSrc and add more pads to it. But I prefer writing it from GstElement itself as we get more flexibility and won't be surprised by future changes for GstBaseSrc or GstPushSrc. The initial implementation is already on http://gitorious.org/gstreamer-camerabin2/gst-plugins-bad/commits/camerabin2 > > Best Regards! > Hu Gang > -- Thiago > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From gang.a.hu at intel.com Wed Dec 15 03:55:13 2010 From: gang.a.hu at intel.com (Hu, Gang A) Date: Wed, 15 Dec 2010 10:55:13 +0800 Subject: [gst-devel] Looking for multi-pads source element In-Reply-To: <1292375980.12069.9.camel@blacksheep> References: <1A42CE6F5F474C41B63392A5F80372B231B41978@shsmsx501.ccr.corp.intel.com> <1292375980.12069.9.camel@blacksheep> Message-ID: <1A42CE6F5F474C41B63392A5F80372B231B41C85@shsmsx501.ccr.corp.intel.com> En, I am writing the source element for camerabin2. We can work together in the future. My currently plan is writing a GstCamBaseSrc inherited from GstElement. But I think the most of the code should be same as the GstBaseSrc. I also need to refer other element to know how to manage the pads. The camera source element inherits from GstCamBaseSrc. I will send out the first design later. Thanks. -----Original Message----- From: Thiago Sousa Santos [mailto:thiago.sousa.santos at collabora.co.uk] Sent: Wednesday, December 15, 2010 9:20 AM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] Looking for multi-pads source element On Tue, 2010-12-14 at 18:08 +0800, Hu, Gang A wrote: > Hi > The basesrc only have one pad. I want to add another static pads to > basesrc. > Is there any source element which have more than one pads? I am > looking for such element for reference. > Thanks. I'm assuming you're interested in camerabin2 and its 3pad source. I've been drafting a base class for it lately. Maybe we could discuss and get this working together? I've been wondering if it would be good to inherit from GstBaseSrc or GstPushSrc and add more pads to it. But I prefer writing it from GstElement itself as we get more flexibility and won't be surprised by future changes for GstBaseSrc or GstPushSrc. The initial implementation is already on http://gitorious.org/gstreamer-camerabin2/gst-plugins-bad/commits/camerabin2 > > Best Regards! > Hu Gang > -- Thiago > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Lotusphere 2011 Register now for Lotusphere 2011 and learn how to connect the dots, take your collaborative environment to the next level, and enter the era of Social Business. http://p.sf.net/sfu/lotusphere-d2d _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ved.kpl at gmail.com Wed Dec 15 05:06:47 2010 From: ved.kpl at gmail.com (ved kpl) Date: Wed, 15 Dec 2010 09:36:47 +0530 Subject: [gst-devel] GST_FLOW_WRONG_STATE In-Reply-To: References: <1291853143.3531.43.camel@zingle> Message-ID: It is the time taken taken for a input 'X' at the decoder to appear as the decoded output, which possibly could involve the startup latency as well,. depends on your implementation. (in READY->PAUSED). On Wed, Dec 15, 2010 at 2:52 AM, Radivoje Jovanovic wrote: > Thank you for all of your input. Unfortunately I did not write the decoders > at all, but I do have the source code. What is considered as decoder > latency? Is it the time for decoder to set up or something else? > Cheers > Ogi > > On Thu, Dec 9, 2010 at 7:45 PM, ved kpl wrote: >> >> On Fri, Dec 10, 2010 at 1:39 AM, Radivoje Jovanovic >> wrote: >> > It seems like we are on a good path. after I tried the pipeline: >> > ?gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue >> > max-size-time=0 max-size-bytes=0 ! mpeg4dec ! MY_VIDEO_SINK t. ! queue >> > ? ! faad ! alsasink >> > ?? (so I only give time and bytes for video side) the video and audio >> > worked >> > just fine. >> > >> > Unfortunately this trick does not work for the other decoder I have >> > which >> > decodes h264 videos. >> > >> > Even with the pipeline: >> > ?gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue >> > max-size-time=0 max-size-bytes=0 ! h264dec ! MY_VIDEO_SINK t. ! queue >> > ? ! faad ! alsasink >> > The pipeline is stuck. This all has nothing to do with >> > GST_FLOW_WRONG_STATE >> > since this error pops up all the time after CTRL-C operation. Any idea >> > why >> > would h264 get stuck? >> >> ?To start off, try setting max-size-time/bytes/buffers=0 on the both >> the queues (no limits) >> ?If it works, it indicates that your h264dec has a very high >> latency..and you can >> ?narrow to the problem from here. >> >> > Another question is if I am giving too much memory to gstreamer with >> > max-size-time=0 max-size-bytes=0? >> >> ?Well, It would depend on your codec details (resilution, etc), >> ?environment/platform. You could possibly track down the issue. >> >> > Cheers >> > Ogi >> > >> > >> > On Wed, Dec 8, 2010 at 8:27 PM, ved kpl wrote: >> >> >> >> Hi, >> >> >> >> One possible reason could be that one of the queues is getting full >> >> while the other remains empty >> >> because of >> >> (a) continuous audio/video buffers for a considerably longer time, >> >> longer than the what the queues can hold. >> >> (b) high latency at the your video decoder (mpeg4dec). >> >> Hence the streaming thread is blocked & one of the sinks is not able >> >> to commit the state to PAUSED. >> >> (because it has not received a buffer yet). >> >> >> >> Yo can also set async=FALSE on the sinks and see. (not recommended) >> >> >> >> You can try disabling the the max time & max bytes properties of the >> >> queue. >> >> Try the following pipeline >> >> >> >> ?gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue >> >> max-size-time=0 max-size-bytes=0 ! mpeg4dec ! MY_VIDEO_SINK t. ! queue >> >> max-size-time=0 max-size-bytes=0 ! faad ! alsasink -v >> >> >> >> Note that this will increase the mem usage.. and even more if u set >> >> max-size-buffers=0 as well. (queue without any limits!) >> >> >> >> Vikram >> >> >> >> On Thu, Dec 9, 2010 at 9:44 AM, ved kpl wrote: >> >> > On Thu, Dec 9, 2010 at 6:26 AM, Radivoje Jovanovic >> >> > wrote: >> >> >> So the pipeline I am using is: >> >> >> gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue >> >> >> ! >> >> >> mpeg4dec ! MY_VIDEO_SINK t. ! queue ! faad ! alsasink >> >> >> This pipeline will hang and after CTRL-C the mpeg4dec will show the >> >> >> message >> >> >> that is the result of calling gst_push_pad. This gst_push_pad will >> >> >> return >> >> >> GST_FLOW_WRONG_STATE. >> >> >> >> >> >> If I run gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t >> >> >> ! >> >> >> queue >> >> >> ! mpeg4dec ! MY_VIDEO_SINK >> >> >> the video will play just fine >> >> >> >> >> >> If I run: >> >> >> gst-launch-0.10 filesrc location=MY_MOVIE ! qtdemux name=t ! queue ! >> >> >> faad ! >> >> >> alsasink >> >> >> the audio from the audio will play just fine. >> >> >> >> >> >> I am not sure which code would be helpful since I have a lot of it. >> >> >> Thanks for the GST_DEBUG info. >> >> >> >> >> >> Ogi >> >> >> >> >> >> >> >> >> On Wed, Dec 8, 2010 at 4:05 PM, Tim-Philipp M?ller >> >> >> wrote: >> >> >>> >> >> >>> On Wed, 2010-12-08 at 12:53 -0800, Radivoje Jovanovic wrote: >> >> >>> >> >> >>> > I have a video sink and ALSA driver to be used to play videos on >> >> >>> > the >> >> >>> > platform. If I play video or audio by them self everything works >> >> >>> > fine, >> >> >>> > but if I specify the whole pipeline for video and audio, the >> >> >>> > video >> >> >>> > decoder (in pipeline right before the sink) comes back with the >> >> >>> > message: >> >> >>> > "The decoded frame did not successfully push out to downstream >> >> >>> > element" >> >> >>> > The message comes because the gst_pad_push have returned with >> >> >>> > GST_FLOW_WRONG_STATE instead GST_FLOW_OK. >> >> >>> > >> >> >>> > Any idea why this might happen? >> >> >>> >> >> >>> It usually helps if you post your exact pipeline and/or any code >> >> >>> snippets that go with it. >> >> >>> >> >> >>> FLOW_WRONG_STATE is what you get when a pad is flushing, which may >> >> >>> be >> >> >>> normal (happens during a flushing seek, to make the old streaming >> >> >>> thread >> >> >>> stop) or because you forgot to set an element into PAUSED/PLAYING >> >> >>> state >> >> >>> (e.g. because you added it from a pad-added or new-decoded-pad >> >> >>> callback >> >> >>> or so). >> >> >>> >> >> >>> The GST_DEBUG=*:5 log might give you more information (just grep >> >> >>> for >> >> >>> wrong-state and read the lines before that). >> >> >>> >> >> >>> Cheers >> >> >>> ?-Tim >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> ------------------------------------------------------------------------------ >> >> >>> This SF Dev2Dev email is sponsored by: >> >> >>> >> >> >>> WikiLeaks The End of the Free Internet >> >> >>> http://p.sf.net/sfu/therealnews-com >> >> >>> _______________________________________________ >> >> >>> gstreamer-devel mailing list >> >> >>> gstreamer-devel at lists.sourceforge.net >> >> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> ------------------------------------------------------------------------------ >> >> >> This SF Dev2Dev email is sponsored by: >> >> >> >> >> >> WikiLeaks The End of the Free Internet >> >> >> http://p.sf.net/sfu/therealnews-com >> >> >> _______________________________________________ >> >> >> gstreamer-devel mailing list >> >> >> gstreamer-devel at lists.sourceforge.net >> >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> >> >> >> >> > >> >> >> >> >> >> >> >> ------------------------------------------------------------------------------ >> >> This SF Dev2Dev email is sponsored by: >> >> >> >> WikiLeaks The End of the Free Internet >> >> http://p.sf.net/sfu/therealnews-com >> >> _______________________________________________ >> >> gstreamer-devel mailing list >> >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> > >> > >> > ------------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> > >> >> >> ------------------------------------------------------------------------------ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From gibrovacco at gmail.com Wed Dec 15 08:12:48 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 15 Dec 2010 09:12:48 +0200 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 44 In-Reply-To: <1292061906.2568.12.camel@deumeu> References: <1292061906.2568.12.camel@deumeu> Message-ID: Hi, On Sat, Dec 11, 2010 at 12:05 PM, Edward Hervey wrote: > On Sat, 2010-12-11 at 10:49 +0100, Shamun toha md wrote: >> But why the following then works its in UDP, its doing multicast, >> exactly the same what i am trying with vorbis/ogg ? > > ?Because if a raw audio packet gets dropped or reordered (which is > insanely small in terms of samples [1]) you will not really noticed it > and they are self-sufficient (they don't require the previous/next > packest to be played back). > ?But for compressed data one single packet dropped will cause > corruption of the whole stream. to add my 2 ? cents here, if you really want a reliable communication (negligible loss probability in normal conditions) you should use TCP instead than UDP as your transport protocol. Check the documentation about tcpclient(server)sink(src) for more info. Regards > >> - When its transparent and no encode/decoder used it ?works, its the >> OGGDEMUX causing this. WHY THEN THE FOLLOWING CODE WORKS ??? > > ?YOU SEEM TO HAVE A PROBLEM WITH YOUR CAPSLOCK KEY ! CAN YOU PLEASE FIX > THAT ? THANKYOU > > ?Edward > > [1] a standard udp packet is 1500bytes, at 44100 samples per second x 2 > (channels) x 2 (bytes per sample), that gives you around 8ms of audio > per packet. 8ms of audio being dropped isn't really noticeable. > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From gibrovacco at gmail.com Wed Dec 15 08:30:09 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 15 Dec 2010 09:30:09 +0200 Subject: [gst-devel] Cannot set caps on udpsrc0 to application/x-rtp under OS X In-Reply-To: References: Message-ID: Hi, I've never used a MAC, but maybe a general comment will help you.. On Tue, Dec 14, 2010 at 6:42 PM, Matthew Braun wrote: > Greetings! > > I'm streaming from a webcam attached to a BeagleBoard using: > ?gst-launch -v v4l2src ! video/x-raw-yuv, width=640, height=480 ! > ffmpegcolorspace ! TIVidenc1 codecName=h264enc engineName=codecServer ! > rtph264pay pt=96 ! udpsink host= port=5000 > > Which runs and outputs caps of: > ?caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, > encoding-name=(string)H264, > sprop-parameter-sets=(string)\"Z0KAHukBQHpCAAAH0AAB1MAIAA\\=\\=\\,aM48gAA\\ > =\", payload=(int)96, ssrc=(uint)1421770711, clock-base=(uint)3676471020, > seqnum-base=(uint)12939 > it's unlikely the real cause of your issue, but sprop-paramerters-set may change from session to session, more or less like ssrc, clock base and seqnum-base. A pipeline with such caps won't work more than once unless you copy and paste those details each time. As a suggestion, it's better not to specify them and let the RTP stack doing its work. > > But when I run gst-launch on my OS X destination with > ?% CAPS="application/x-rtp, media=(string)video, clock-rate=(int)90000, > encoding-name=(string)H264, > sprop-parameter-sets=(string)\"Z0KAHukBQHpCAAAH0AAB1MAIAA\\=\\=\\,aM48gAA\\ > =\", payload=(int)96, ssrc=(uint)3775499150, clock-base=(uint)2462209911, > seqnum-base=(uint)27054" "application/x-rtp,media=(string)video,encoding-name=(string)H264,clock-rate=(int)90000" should be enough > ?% ?gst-launch -v udpsrc caps=$CAPS port=5000 ! rtph264depay ! ffdec_h264 > ! Autovideosink > > I get > > WARNING: erroneous pipeline: could not set property "caps" in element > "udpsrc0" to "application/x-rtp," It looks like your caps have been cut right after the comma (,). Maybe you've a CLI syntactical issue with the way you specify them. Regards > > Trying without caps gives: > ?"Input buffers need to have RTP caps set on them. This is usually > achieved by setting the 'caps' property of the upstream source element > (often udpsrc or appsrc), or by putting a capsfilter element before the > depayloader and setting the 'caps' property on that. Also see > http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README" > > Any ideas as to what I'm missing? Thanks for any guidance and/or > suggestions! > > > (% gst-launch --version > ?gst-launch-0.10 version 0.10.31 > ?GStreamer 0.10.31 > ) > > > -- > Matthew Braun > mjbraun at leydenjar.com > > > > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From puneeth.mv at globaledgesoft.com Wed Dec 15 11:24:40 2010 From: puneeth.mv at globaledgesoft.com (Puneeth) Date: Wed, 15 Dec 2010 02:24:40 -0800 (PST) Subject: [gst-devel] problem in rtpmpapay and rtpmpadepay Message-ID: <1292408680963-3088737.post@n4.nabble.com> Hi All, I m trying to capture audio and encode to mp3 format and send through rtpmpapay and decode and playback in the other system, but i m not able to listen voice clearly. But it is good when send captured pcm format and playback using protocol rtppcmapay and rtppcmadepay. Here is the command for the reference, thank u in advance and will be greatly appreciated for the solution. Sender : gst-launch gstrtpbin name=rtpbin alsasrc ! audioconvert ! audioresample ! lamemp3enc ! rtpmpapay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5002 host=172.16.10.32 rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=172.16.10.32 sync=false async=false udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0 Receiver : gst-launch gstrtpbin name=rtpbin udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)MPA" port=5002 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtpmpadepay ! mp3parse ! ffdec_mp3 ! audioconvert ! volume volume=3 ! audioresample quality=10 ! alsasink udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=172.16.10.32 sync=false async=false -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/problem-in-rtpmpapay-and-rtpmpadepay-tp3088737p3088737.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From puneeth.mv at globaledgesoft.com Wed Dec 15 11:29:01 2010 From: puneeth.mv at globaledgesoft.com (Puneeth) Date: Wed, 15 Dec 2010 02:29:01 -0800 (PST) Subject: [gst-devel] How do u set caps for udpsrc In-Reply-To: <1292258949698-3085734.post@n4.nabble.com> References: <1292244042611-3085367.post@n4.nabble.com> <1292258949698-3085734.post@n4.nabble.com> Message-ID: <1292408941708-3088742.post@n4.nabble.com> Thank u very much but i did this using gst_caps_new_simple it is working fine....when i m saving to file it is working fine but facing problem when i m listening live audio data,there is little disturbance in playback. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-do-u-set-caps-for-udpsrc-tp3085367p3088742.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wmiller at sdr.com Wed Dec 15 15:54:30 2010 From: wmiller at sdr.com (Wes Miller) Date: Wed, 15 Dec 2010 06:54:30 -0800 (PST) Subject: [gst-devel] How do u set caps for udpsrc In-Reply-To: <1292408941708-3088742.post@n4.nabble.com> References: <1292244042611-3085367.post@n4.nabble.com> <1292258949698-3085734.post@n4.nabble.com> <1292408941708-3088742.post@n4.nabble.com> Message-ID: <1292424870964-3089161.post@n4.nabble.com> Hi, I'm not sure what you mean here: "but facing problem when i m listening live audio data,there is little disturbance in playback. " Are you using a different live audio source as input to the sender or did you replace the filesink with something like alsasink or pulsesink? If you are using alsasink or pulsesink you may need to set their sync propeerties to FALSE. And by "little disturbance" do you mean there is noise in the playback? There may be a problem with the sync properties of the sink (alsa- or file-) as noted above or your receiver caps could be mismatched to your sender. Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-do-u-set-caps-for-udpsrc-tp3085367p3089161.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ensonic at hora-obscura.de Wed Dec 15 16:01:49 2010 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 15 Dec 2010 17:01:49 +0200 Subject: [gst-devel] Looking for multi-pads source element In-Reply-To: <1292375980.12069.9.camel@blacksheep> References: <1A42CE6F5F474C41B63392A5F80372B231B41978@shsmsx501.ccr.corp.intel.com> <1292375980.12069.9.camel@blacksheep> Message-ID: <4D08D85D.6070401@hora-obscura.de> Hi, On 15.12.2010 03:19, Thiago Sousa Santos wrote: > On Tue, 2010-12-14 at 18:08 +0800, Hu, Gang A wrote: >> Hi >> The basesrc only have one pad. I want to add another static pads to >> basesrc. >> Is there any source element which have more than one pads? I am >> looking for such element for reference. >> Thanks. > I'm assuming you're interested in camerabin2 and its 3pad source. I've > been drafting a base class for it lately. Maybe we could discuss and get > this working together? > > I've been wondering if it would be good to inherit from GstBaseSrc or > GstPushSrc and add more pads to it. But I prefer writing it from > GstElement itself as we get more flexibility and won't be surprised by > future changes for GstBaseSrc or GstPushSrc. There is more need for a multi-pad basesrc - ladspa and lv2. This is the main reason sources there don't really work. I once started copying code from basesrc, but it becomes quite a copy'n'paste session :/ At some point in one of my other lives I plan to look at this again. Stefan > The initial implementation is already on > http://gitorious.org/gstreamer-camerabin2/gst-plugins-bad/commits/camerabin2 > >> >> Best Regards! >> Hu Gang >> > -- > Thiago > >> >> >> ------------------------------------------------------------------------------ >> Lotusphere 2011 >> Register now for Lotusphere 2011 and learn how >> to connect the dots, take your collaborative environment >> to the next level, and enter the era of Social Business. >> http://p.sf.net/sfu/lotusphere-d2d >> _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From robert.swain at gmail.com Wed Dec 15 16:43:46 2010 From: robert.swain at gmail.com (Rob) Date: Wed, 15 Dec 2010 16:43:46 +0100 Subject: [gst-devel] GStreamer Conference 2010: videos of talks from the Darwin room Message-ID: Hello, It seems that all the talks from the main room at the GStreamer Conference 2010 are up at http://gstconf.ubicast.tv/categories/conferences/ courtesy of Ubicast. Kudos to them for doing this, for their interesting setup and for talking about it at the conference itself. I was just looking for David's talk about Orc to show to a friend and it seems that the videos of the talks from the Darwin room are nowhere to be found. There was definitely a video camera in the Darwin room but there is no mention of the recordings here: http://www.gstreamer.net/conference/ . What happened to those recordings? Regards, Rob From antoni.silvestre at gmail.com Wed Dec 15 17:13:08 2010 From: antoni.silvestre at gmail.com (=?ISO-8859-1?Q?Antoni_Silvestre_Padr=F3s?=) Date: Wed, 15 Dec 2010 17:13:08 +0100 Subject: [gst-devel] Getting remote endpoint from UDP traffic in a udpsrc element In-Reply-To: <1292240434.22115.10.camel@meany> References: <1292240434.22115.10.camel@meany> Message-ID: Thanks a lot for your answer. I found the code you were referring to in the gstrtpsession source, and it would be perfect when writing a plug-in, however I'm not writing a plugin but a standalone application so I don't have direct access to buffers generated by plugins (and in my case inserting a tee and an appsink to the pipeline would not be very convenient). In my project I'm using the gstrtpbin element for a videoconferencing application, when I receive RTCP reports I'd like to know it's remote endpoint to be able to answer them, as I don't know the address and port of the stream sender beforehand I should get it from the socket. So it would be great if the udpsrc would have a property with the remote endpoint info of the last packet or could emit a signal whenever the remote endpoint location changes. I could add these features myself, nonetheless it would be great for me if my application would be directly compatible with any new udpsrc versions and gstreamer releases so if you think this feature would be interesting for the udpsrc plugin I could submit a patch with my changes so it may eventually be merged with the main code. If so, please tell me which would be the most convenient way doing it (creating a new property, a signal, ...) and I'll try to do it. Anyways, maybe I did not understand well your answer and there's no need for these changes or there is another solution. Thanks, Toni Silvestre On Mon, Dec 13, 2010 at 12:40, Wim Taymans wrote: > GstNetBuffer -------------- next part -------------- An HTML attachment was scrubbed... URL: From jpl at xiphos.ca Wed Dec 15 19:28:45 2010 From: jpl at xiphos.ca (Joshua Lamorie) Date: Wed, 15 Dec 2010 13:28:45 -0500 Subject: [gst-devel] Segmentation fault in g_type_check_instance_cast Message-ID: <1292437725.16916.26.camel@haddock.xiphos.ca> Gidday there, I'm having some issues with some code that runs fine as a 'standalone' application, but when I put it into a thread spawned by another application it has a segfault. The following is from GDB: [Switching to Thread 0x7ffff106f710 (LWP 4780)] IA__g_type_check_instance_cast (type_instance=0x7fffe4008030 [GstAppSrc], iface_type=3825213056) at gtype.c:3973 3973 check = is_instantiatable && iface && type_node_conforms_to_U (node, iface, TRUE, FALSE); (gdb) bt #0 IA__g_type_check_instance_cast (type_instance=0x7fffe4008030 [GstAppSrc], iface_type=3825213056) at gtype.c:3973 #1 0x00007ffff7b3fb64 in ESMRTPThreadMain (pArg=0x7fffec0052d0) at RTPSourcePlugin.c:283 #2 0x000000346fe64e84 in g_thread_create_proxy (data=0x7fffec005290) at gthread.c:1893 #3 0x000000346de07761 in start_thread (arg=0x7ffff106f710) at pthread_create.c:301 #4 0x000000346d2e14fd in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:115 Here's some pseudo code of the overall system. 1) Main C++ application loads and initializes itself. 2) Opens a .so containing a few extern C functions/symbols. 3) Calls an Init() function from the .so (which calls gst_init()) 4) Calls pthread_create() and sits on an accept() call. 4.1) The accept() returns and a message is read by the C++ app/thread. 4.2) This then calls an Execute() function from the .so. 4.3) The Execute() method creates a thread (either pthread_create or g_thread_create). 4.3.1) The newly created thread is pretty much copy and pasted from a functional 'standalone' gstreamer application and initializes itself. 4.3.2) During a call (cast) to GST_APP_SRC(GstElement *) a segfault occurs somewhere within the inlined functions. I've put this in gdb and it appears that all of the input arguments are non-NULL, but deep within the inlined functions I saw some references to what appear to be global locking variables. I'm concerned that my initialization doesn't quite match the standalone case, but I can't quite figure out what is going on 'behind-the-curtain' in the standalone case that is different from my multi-threaded example. Standalone code: GMainLoop *loop; int main(int iArgc, char * ppcArgv[]) { GstElement * poAppSrc; GstElement * poPipeline; /* ... snip ... */ /* initialize gstreamer */ gst_init(NULL,NULL); loop = g_main_loop_new (NULL, TRUE); poPipeline = gst_pipeline_new("ESMPipeline"); poAppSrc = gst_element_factory_make("appsrc", NULL); gst_app_src_set_stream_type(GST_APP_SRC(poAppSrc), GST_APP_STREAM_TYPE_STREAM); /* ... snip ... */ } Thread code: void * ESMRTPThreadMain(void * pArg) { GstElement * poPipeline = NULL; GstElement * poAppSrc = NULL; /* ... snip ... */ poPipeline = gst_pipeline_new("ESMPipeline"); poAppSrc = gst_element_factory_make("appsrc",NULL); if (poAppSrc == NULL) { printf("Problem generating appsrc element\n"); return(NULL); } /* set stream type */ gst_app_src_set_stream_type(GST_APP_SRC(poAppSrc), GST_APP_STREAM_TYPE_STREAM); /* ... snip ... */ } So what am I doing wrong in the initialization to cause the segfault? Thanks in advance. -- Joshua Lamorie, P. Eng. Chief Engineer Xiphos Technologies Inc. Email: jpl at xiphos.ca Phone: +1 (514) 847-9474 ext. 227 Fax: +1 (514) 847-9474 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. If you have received this in error, please contact the sender and delete this communication and any copy immediately. Thank you. From ds at entropywave.com Wed Dec 15 20:54:07 2010 From: ds at entropywave.com (David Schleef) Date: Wed, 15 Dec 2010 14:54:07 -0500 Subject: [gst-devel] GStreamer Conference 2010: videos of talks from the Darwin room In-Reply-To: References: Message-ID: <20101215195407.GA31650@cooker.entropywave.com> On Wed, Dec 15, 2010 at 04:43:46PM +0100, Rob wrote: > I was just looking for David's talk about Orc to show to a friend and > it seems that the videos of the talks from the Darwin room are nowhere > to be found. There was definitely a video camera in the Darwin room > but there is no mention of the recordings here: > http://www.gstreamer.net/conference/ . What happened to those > recordings? They will show up here when they are completed: http://free-electrons.com/community/videos/conferences/ David From rafael.lmsousa at gmail.com Wed Dec 15 21:02:33 2010 From: rafael.lmsousa at gmail.com (Rafael Sousa) Date: Wed, 15 Dec 2010 16:02:33 -0400 Subject: [gst-devel] distorted video encoded with x264 Message-ID: Hi all, I'm using the versions of gstreamer installed from ubuntu 10.10 apt-get. I've installed the gstreamer, gst-plugins-bad, gst-plugins-good, gst-plugins-ugly, gst-ffmpeg and I've installed the x264. The problem is that my pipeline gives as result a blurring video. My pipeline works to read a YUV file at the sender, encodes it in H.264, and sends it to receiver using RTP in encoding time. At the receiver, it receives the stream and stores it in YUV format again for posterior process. The pipeline is as follows: Sender: "gstrtpbin name=rtpbin filesrc location=paris_cif.yuv blocksize=152064 ! video/x-raw-yuv, width=352, height=288, framerate=(fraction)15/1, format=(fourcc)I420 ! x264enc, key-int-max=30, key-int-min=15, intra-refresh=true ! rtph264pay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 rtpbin.send_rtcp_src_0 ! udpsink host=5003 port=5001 sync=false async=false udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 " Receiver: "gstrtpbin name=rtpbin udpsrc caps=\"application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" port=5000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! ffdec_h264 ! filesink location=example.yuv udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false" I don't know why the video is blurring since I've tested it localhost, without loss rates. The command that I'm using to watch the result is: mplayer example.yuv -demuxer rawvideo -rawvideo cif I took a screenshot of the video, it is in annex. Please, anyone can help? thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Captura_de_tela.png Type: image/png Size: 125708 bytes Desc: not available URL: From braunsquared at gmail.com Wed Dec 15 21:42:01 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Wed, 15 Dec 2010 15:42:01 -0500 Subject: [gst-devel] distorted video encoded with x264 In-Reply-To: References: Message-ID: <69C57452-40E4-4172-883F-DF374D355CD7@gmail.com> I'm no expert on this, but ran into similar issues when making a local file. It ultimately ended up being the parameters I was passing to the x264enc element. I would take a look at those with gst-inspect and see if you can tune the output. Most notably the pass and quantizer parameters. Tim Sent from my iPhone On Dec 15, 2010, at 3:02 PM, Rafael Sousa wrote: > Hi all, > > I'm using the versions of gstreamer installed from ubuntu 10.10 apt-get. I've installed the gstreamer, gst-plugins-bad, gst-plugins-good, gst-plugins-ugly, gst-ffmpeg and I've installed the x264. > > The problem is that my pipeline gives as result a blurring video. My pipeline works to read a YUV file at the sender, encodes it in H.264, and sends it to receiver using RTP in encoding time. At the receiver, it receives the stream and stores it in YUV format again for posterior process. The pipeline is as follows: > > Sender: > > "gstrtpbin name=rtpbin filesrc location=paris_cif.yuv blocksize=152064 ! video/x-raw-yuv, width=352, height=288, framerate=(fraction)15/1, format=(fourcc)I420 ! x264enc, key-int-max=30, key-int-min=15, intra-refresh=true ! rtph264pay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 rtpbin.send_rtcp_src_0 ! udpsink host=5003 port=5001 sync=false async=false udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 " > > Receiver: > > "gstrtpbin name=rtpbin udpsrc caps=\"application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" port=5000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! ffdec_h264 ! filesink location=example.yuv udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false" > > I don't know why the video is blurring since I've tested it localhost, without loss rates. The command that I'm using to watch the result is: > > mplayer example.yuv -demuxer rawvideo -rawvideo cif > > I took a screenshot of the video, it is in annex. Please, anyone can help? > > thanks in advance > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From robert.swain at gmail.com Wed Dec 15 22:31:39 2010 From: robert.swain at gmail.com (Rob) Date: Wed, 15 Dec 2010 22:31:39 +0100 Subject: [gst-devel] GStreamer Conference 2010: videos of talks from the Darwin room In-Reply-To: <20101215195407.GA31650@cooker.entropywave.com> References: <20101215195407.GA31650@cooker.entropywave.com> Message-ID: On 15 December 2010 20:54, David Schleef wrote: > On Wed, Dec 15, 2010 at 04:43:46PM +0100, Rob wrote: >> I was just looking for David's talk about Orc to show to a friend and >> it seems that the videos of the talks from the Darwin room are nowhere >> to be found. There was definitely a video camera in the Darwin room >> but there is no mention of the recordings here: >> http://www.gstreamer.net/conference/ . What happened to those >> recordings? > > They will show up here when they are completed: > > http://free-electrons.com/community/videos/conferences/ Do we have any idea of when they will go up? It's coming up to 2 months since the conference now. Regards, Rob From shiva.varma at gmail.com Thu Dec 16 00:27:45 2010 From: shiva.varma at gmail.com (shiva varma) Date: Wed, 15 Dec 2010 15:27:45 -0800 Subject: [gst-devel] Problems converting audio pcap using pcapparse, rtpmp4depay and qtmux plugins Message-ID: Hi. I have a pcap file which contains mpeg4-generic audio carried in RTP I use the following pipeline gst-launch filesrc location=audio.rtp.pcap ! pcapparse ! "application/x-rtp, payload=96, media=audio, clock-rate=30" ! rtpmp4adepay ! "audio/mpeg, mpegversion=4" ! qtmux ! fakesink Below is the error I get Setting pipeline to PAUSED ... Pipeline is PREROLLING ... Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock ERROR: from element /GstPipeline:pipeline0/GstQTMux:qtmux0: Internal GStreamer error: negotiation problem. Please file a bug at http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer. Additional debug info: gstqtmux.c(1815): gst_qt_mux_add_buffer (): /GstPipeline:pipeline0/GstQTMux:qtmux0: format wasn't negotiated before buffer flow on pad audio_00 Execution ended after 300000 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... Am I missing on some caps that needs to be added to the pipeline? Thanks Shiva From shamun.toha at gmail.com Thu Dec 16 02:22:45 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Thu, 16 Dec 2010 02:22:45 +0100 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 51 In-Reply-To: References: Message-ID: I tried in my laptop Fedora 12 with gstreamer 0.10.29 works. But i put now in server Fedora 12 and gstreamer 0.10.29 but does not work same error for Centos 5.4 with gstreamer 0.10.20. [root at ip-10-234-178-139 ~]# gst-launch -v udpsrc port=4444 ! fakesink ERROR: pipeline could not be constructed: no element "udpsrc". [root at ip-10-234-178-139 ~]# Please kindly someone guide. I already searched lot of info but didnt came to a answer yet. Thanks & Regards Shamun On Sun, Dec 12, 2010 at 11:42 AM, < gstreamer-devel-request at lists.sourceforge.net> wrote: > Send gstreamer-devel mailing list submissions to > gstreamer-devel at lists.sourceforge.net > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > or, via email, send a message with subject or body 'help' to > gstreamer-devel-request at lists.sourceforge.net > > You can reach the person managing the list at > gstreamer-devel-owner at lists.sourceforge.net > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of gstreamer-devel digest..." > > > Today's Topics: > > 1. Re: gstreamer-devel Digest, Vol 55, Issue 49 (Shamun toha md) > 2. Re: How I test the FLV demux plugin (sudarshan bisht) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 12 Dec 2010 11:19:26 +0100 > From: Shamun toha md > Subject: Re: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 49 > To: gstreamer-devel at lists.sourceforge.net > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > Hello, > > Thanks for the info. udpsrc cause ERROR: pipeline could not be constructed: > no element "udpsrc". > > But this works in Gstreamer 0.10.29 but not in 0.10.20. > > Thanks & Regards > Shamun > > > > Message: 1 > > Date: Sat, 11 Dec 2010 12:26:21 -0500 > > From: Sean McNamara > > Subject: Re: [gst-devel] CentOS 64 bit, how to get latest version > > 0.10.31 ? > > To: Discussion of the development of GStreamer > > > > Message-ID: > > > > Content-Type: text/plain; charset=ISO-8859-1 > > > > Hi, > > > > On Sat, Dec 11, 2010 at 10:01 AM, Shamun toha md > > wrote: > > > Hello, > > > > > > I am not sure what is that mean, but i thought i should ask it in > mailing > > to > > > make sure. I used an example in fedora works, but tried in CentOS 5.4 > it > > > does not work. > > > > If I understand you correctly, you're saying that a certain pipeline > > example or gstreamer application works in Fedora 12, but not in CentOS > > 5.4. That is a valid reason for wanting to upgrade your gstreamer > > version, though you might simply be missing a certain plugin, or > > hitting an easily fixable bug. Without specific details about what > > "does not work" in particular, we can't help you more than that. BTW, > > CentOS 6.0 does not exist yet, only RHEL 6.0... > > > > > > > > Fedora???????? gstreamer.i686? 0.10.29-1.fc12 (working my test code) > > > > > > CentOS 5.4? gstreamer-0.10.20-3.el5.x86_64.rpm? (does not work) > > > CentOS 6.0? gstreamer-0.10.26-1.el6.x86_64.rpm? (i am confused) > > > > I'm confused, too! You aren't explaining yourself very well. You seem > > to be grabbing random RPMs and trying them? For your info, it's > > probably better to use a package manager, such as yum. Getting RPMs > > off of pbone or other RPM finding services does not guarantee that the > > built binaries you install will be compatible with the rest of your > > stack (i.e. that package's dependencies). > > > > > > > > How can i then test in CentOS, the latest Gstreamer releases ? Is there > > any > > > guides or how tos to achieve that ? Or its impossible because CentOS > 6.0 > > > will sitll use the old version ? > > > > > > I am little bit confused now, will it be possible to have the latest > and > > > greatest ? > > > > With free software, it is always possible :) The real question is > > whether you are willing to work for the desired result. Since > > Gstreamer is in userspace, and its dependencies are pretty flexible, > > it should be "easy" to upgrade the version by compiling from source. > > Another possibility is that you could find a repository online that > > has updated binary packages, though I don't know where such a > > repository would be, and such a thing wouldn't be supported by the > > Gstreamer project maintainers. Nor would it be supported by CentOS > > project or Red Hat. Compiling from vanilla sources might lend you some > > unofficial advice/support here among interested gstreamer users, > > though. > > > > Gstreamer project itself used to distribute binaries for RHEL/CentOS, > > but it seems they haven't been updated since 2008; for example, see: > > http://gstreamer.freedesktop.org/pkg/rhel/5/0.10/i386/gst/ (but DON'T > > install these; they are very old!) > > > > The short story is that yes, you can of course install the latest > > gstreamer and plugins from source, provided your system has the > > required dependencies (and the minimum versions), or provided that you > > compile the required dependencies yourself. This can take a lot of > > time, especially if you've never done it before, and special > > considerations are required if you plan to do this on many different > > computers, e.g. in an IT environment. In that case you may want to > > build your own RPM packages of it on a development box, and simply > > install them on each of the client computers. > > > > Your email does not really provide enough information for me to offer > > any further advice, but I hope this gets you started as to your > > options.... > > > > > > Sean > > > > > > > > Thanks & Regards > > > Shamun > > > > > > -- > > > > > > > ************************************************************************ > > > The information transmitted is intended only for the person or entity > to > > > which it is addressed and may contain confidential and/or privileged > > > material. > > > Any review, retransmission, dissemination, or other use of this > > information, > > > or taking of any action in reliance upon it by persons or entities > > > other than the intended recipient is prohibited. If you received this > in > > > error, please contact the sender and delete the material from any > > computer. > > > > > > > > > > > > ------------------------------------------------------------------------------ > > > Oracle to DB2 Conversion Guide: Learn learn about native support for > > PL/SQL, > > > new data types, scalar functions, improved concurrency, built-in > > packages, > > > OCI, SQL*Plus, data movement tools, best practices and more. > > > http://p.sf.net/sfu/oracle-sfdev2dev > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > > ------------------------------ > > Message: 2 > Date: Sun, 12 Dec 2010 16:11:52 +0530 > From: sudarshan bisht > Subject: Re: [gst-devel] How I test the FLV demux plugin > To: Discussion of the development of GStreamer > > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > Ok, it seems playbin takes care of caps negotiations correctly when > "codec_data" is a part of caps. > > So does this solve your problem ? Or still you want to use your old > pipeline > ? > > > > On Sun, Dec 12, 2010 at 2:46 PM, michael wrote: > > > Hi sadarshan: > > I can play the file by your command: gst-launch -v playbin uri=file: > > ///path/test.flv > > > > > > On Sun, Dec 12, 2010 at 3:53 PM, sudarshan bisht < > > bisht.sudarshan at gmail.com> wrote: > > > >> I suggest you to use capsfilter, as you said in your earlier mails > >> that gst_pad_push returns -1 , that means flvdemux is not able to push > >> buffer to the next element and it may happen because of lack of > >> caps negotiations. > >> > >> By the way how did you create this test.flv file ? and are you able to > >> play this file with other players ? > >> > >> and give a try to following pipeline; > >> gst-launch -v playbin uri=file:///path/test.flv > >> > >> > >> > >> On Sat, Dec 11, 2010 at 10:55 PM, Timothy Braun >wrote: > >> > >>> Your best bet is to use gst-inspect and look at what the various > >>> templates of the pads are. It sounds like you need to do some > conversion > >>> somewhere. > >>> > >>> Sent from my iPhone > >>> > >>> On Dec 11, 2010, at 8:46 AM, michael wrote: > >>> > >>> Hi sudarshan: > >>> I followed your command, I got information : > >>> WARNING: erroneous pipeline: could not parse caps "video/x-h264,\ > >>> pixel-aspect-ratio=(fraction)1/1,\ > >>> > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb\" > >>> > >>> > >>> > >>> On Sat, Dec 11, 2010 at 7:32 PM, sudarshan bisht << > bisht.sudarshan at gmail.com> > >>> bisht.sudarshan at gmail.com> wrote: > >>> > >>>> Ok, now can you try following, > >>>> > >>>> gst-launch filesrc location=test.flv ! flvdemux name=demux > >>>> demux.video_00 ! queue ! "video/x-h264, > >>>> pixel-aspect-ratio=(fraction)1/1, > >>>> > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > >>>> " ! filesink location=test.ves demux.audio_00 ! queue ! filesink > >>>> location=test.aes > >>>> > >>>> > >>>> On Sat, Dec 11, 2010 at 10:49 AM, michael < > >>>> mcygogo at gmail.com> wrote: > >>>> > >>>>> Hi sudarshan: > >>>>> I run it with the -v option, and the result like this: > >>>>> Setting pipeline to PAUSED ... > >>>>> Pipeline is PREROLLING ... > >>>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = > >>>>> video/x-h264, pixel-aspect-ratio=(fraction)1/1, > >>>>> > codec_data=(buffer)01640015ffe1001e67640015ac34e602808effc028402804000003000400000300ca3c58b67801000468e9bbcb > >>>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: > >>>>> Internal data stream error. > >>>>> Additional debug info: > >>>>> gstflvdemux.c(2283): gst_flv_demux_loop (): > >>>>> /GstPipeline:pipeline0/GstFlvDemux:demux: > >>>>> stream stopped, reason not-linked > >>>>> ERROR: pipeline doesn't want to preroll. > >>>>> Setting pipeline to NULL ... > >>>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:audio: caps = NULL > >>>>> /GstPipeline:pipeline0/GstFlvDemux:demux.GstPad:video: caps = NULL > >>>>> Freeing pipeline ... > >>>>> > >>>>> I have found that the in function: gst_flv_demux_parse_tag_video() , > >>>>> the call gst_pad_push(demux->video_pad, outbuf) returned the -1; > >>>>> > >>>>> On Sat, Dec 11, 2010 at 4:33 PM, sudarshan bisht << > bisht.sudarshan at gmail.com> > >>>>> bisht.sudarshan at gmail.com> wrote: > >>>>> > >>>>>> Could you run the same pipeline with -v option and paste the log > here > >>>>>> ? > >>>>>> > >>>>>> > >>>>>> On Sat, Dec 11, 2010 at 11:47 AM, michael < > >>>>>> mcygogo at gmail.com> wrote: > >>>>>> > >>>>>>> Hi all: > >>>>>>> Now I am learning some about the flv plugin--flvdemux, I try > to > >>>>>>> demux my flv test file like this: > >>>>>>> gst-launch filesrc location=test.flv ! flvdemux name=demux > >>>>>>> demux.video_00 ! queue ! filesink location=test.ves demux.audio_00 > ! queue ! > >>>>>>> filesink location=test.aes > >>>>>>> But when I run it , I just get the information like this: > >>>>>>> Setting pipeline to PAUSED ... > >>>>>>> Pipeline is PREROLLING ... > >>>>>>> ERROR: from element /GstPipeline:pipeline0/GstFlvDemux:demux: > >>>>>>> Internal data stream error. > >>>>>>> Additional debug info: > >>>>>>> gstflvdemux.c(2283): gst_flv_demux_loop (): > >>>>>>> /GstPipeline:pipeline0/GstFlvDemux:demux: > >>>>>>> stream stopped, reason not-linked > >>>>>>> ERROR: pipeline doesn't want to preroll. > >>>>>>> Setting pipeline to NULL ... > >>>>>>> Freeing pipeline ... > >>>>>>> > >>>>>>> I just want test the flvdemux plugin , So How I fixed it? tks:) > >>>>>>> > >>>>>>> > >>>>>>> > ------------------------------------------------------------------------------ > >>>>>>> Oracle to DB2 Conversion Guide: Learn learn about native support > for > >>>>>>> PL/SQL, > >>>>>>> new data types, scalar functions, improved concurrency, built-in > >>>>>>> packages, > >>>>>>> OCI, SQL*Plus, data movement tools, best practices and more. > >>>>>>> > >>>>>>> http://p.sf.net/sfu/oracle-sfdev2dev > >>>>>>> _______________________________________________ > >>>>>>> gstreamer-devel mailing list > >>>>>>> > >>>>>>> gstreamer-devel at lists.sourceforge.net > >>>>>>> > >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >>>>>>> > >>>>>>> > >>>>>> > >>>>>> > >>>>>> -- > >>>>>> Regards, > >>>>>> > >>>>>> Sudarshan Bisht > >>>>>> > >>>>>> > >>>>>> > ------------------------------------------------------------------------------ > >>>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for > >>>>>> PL/SQL, > >>>>>> new data types, scalar functions, improved concurrency, built-in > >>>>>> packages, > >>>>>> OCI, SQL*Plus, data movement tools, best practices and more. > >>>>>> > >>>>>> http://p.sf.net/sfu/oracle-sfdev2dev > >>>>>> _______________________________________________ > >>>>>> gstreamer-devel mailing list > >>>>>> > >>>>>> gstreamer-devel at lists.sourceforge.net > >>>>>> > >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >>>>>> > >>>>>> > >>>>> > >>>>> > >>>>> > ------------------------------------------------------------------------------ > >>>>> Oracle to DB2 Conversion Guide: Learn learn about native support for > >>>>> PL/SQL, > >>>>> new data types, scalar functions, improved concurrency, built-in > >>>>> packages, > >>>>> OCI, SQL*Plus, data movement tools, best practices and more. > >>>>> > >>>>> http://p.sf.net/sfu/oracle-sfdev2dev > >>>>> _______________________________________________ > >>>>> gstreamer-devel mailing list > >>>>> > >>>>> gstreamer-devel at lists.sourceforge.net > >>>>> > >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >>>>> > >>>>> > >>>> > >>>> > >>>> -- > >>>> Regards, > >>>> > >>>> Sudarshan Bisht > >>>> > >>>> > >>>> > ------------------------------------------------------------------------------ > >>>> Oracle to DB2 Conversion Guide: Learn learn about native support for > >>>> PL/SQL, > >>>> new data types, scalar functions, improved concurrency, built-in > >>>> packages, > >>>> OCI, SQL*Plus, data movement tools, best practices and more. > >>>> > >>>> http://p.sf.net/sfu/oracle-sfdev2dev > >>>> _______________________________________________ > >>>> gstreamer-devel mailing list > >>>> > >>>> gstreamer-devel at lists.sourceforge.net > >>>> > >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >>>> > >>>> > >>> > >>> > ------------------------------------------------------------------------------ > >>> Oracle to DB2 Conversion Guide: Learn learn about native support for > >>> PL/SQL, > >>> new data types, scalar functions, improved concurrency, built-in > >>> packages, > >>> OCI, SQL*Plus, data movement tools, best practices and more. > >>> http://p.sf.net/sfu/oracle-sfdev2dev > >>> > >>> _______________________________________________ > >>> gstreamer-devel mailing list > >>> gstreamer-devel at lists.sourceforge.net > >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >>> > >>> > >>> > >>> > ------------------------------------------------------------------------------ > >>> Oracle to DB2 Conversion Guide: Learn learn about native support for > >>> PL/SQL, > >>> new data types, scalar functions, improved concurrency, built-in > >>> packages, > >>> OCI, SQL*Plus, data movement tools, best practices and more. > >>> http://p.sf.net/sfu/oracle-sfdev2dev > >>> _______________________________________________ > >>> gstreamer-devel mailing list > >>> gstreamer-devel at lists.sourceforge.net > >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >>> > >>> > >> > >> > >> -- > >> Regards, > >> > >> Sudarshan Bisht > >> > >> > >> > ------------------------------------------------------------------------------ > >> Oracle to DB2 Conversion Guide: Learn learn about native support for > >> PL/SQL, > >> new data types, scalar functions, improved concurrency, built-in > packages, > >> OCI, SQL*Plus, data movement tools, best practices and more. > >> http://p.sf.net/sfu/oracle-sfdev2dev > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >> > >> > > > > > > > ------------------------------------------------------------------------------ > > Oracle to DB2 Conversion Guide: Learn learn about native support for > > PL/SQL, > > new data types, scalar functions, improved concurrency, built-in > packages, > > OCI, SQL*Plus, data movement tools, best practices and more. > > http://p.sf.net/sfu/oracle-sfdev2dev > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > -- > Regards, > > Sudarshan Bisht > -------------- next part -------------- > An HTML attachment was scrubbed... > > ------------------------------ > > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for > PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > > ------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > End of gstreamer-devel Digest, Vol 55, Issue 51 > *********************************************** > -------------- next part 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URL: From msmith at xiph.org Thu Dec 16 02:38:27 2010 From: msmith at xiph.org (Michael Smith) Date: Wed, 15 Dec 2010 17:38:27 -0800 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 51 In-Reply-To: References: Message-ID: On Wed, Dec 15, 2010 at 5:22 PM, Shamun toha md wrote: > I tried in my laptop Fedora 12 with gstreamer 0.10.29 works. But i put now > in server Fedora 12 and gstreamer 0.10.29 but does not work same error for > Centos 5.4 with gstreamer 0.10.20. > > [root at ip-10-234-178-139 ~]# gst-launch -v udpsrc port=4444 ! fakesink > ERROR: pipeline could not be constructed: no element "udpsrc". > [root at ip-10-234-178-139 ~]# You haven't installed the plugins that you're trying to use. I'm sure CentOS has packages for the gstreamer plugins; you'll need to install some of those. Mike From stuart68 at gmail.com Thu Dec 16 02:58:00 2010 From: stuart68 at gmail.com (Stuart Gray) Date: Thu, 16 Dec 2010 09:58:00 +0800 Subject: [gst-devel] gapless playback Message-ID: Hi, I am seeking some guidance with respect to getting gapless playback using decodebin2 and queue2. Our system comprises the following: arm9 processor gst-plugins-bad - 0.10.20-r11.0.6 gst-plugins-base - 0.10.31-r11.0.6 gst-plugins-good - 0.10.25-r11.0.6 gst-plugins-ugly - 0.10.16-r11.1.6 running on openembedded linux with kernel version 2.6.33 We are decoding audio only streams. This is a concern as reference was made to gapless playback for the latest releases: "Release notes for GStreamer Base Plug-ins?0.10.31 "Dance Like It's 1982" Bugs fixed in this release * 628028 : [uridecodebin] Don't enable progressive downloading for live streams * 623846 : typefinding: add support for " enhanced ac3 " (eac3) * 602437 : [playbin2] [gapless] Completely broken when switching between files with audio/video only " The above refers directly to playbin2, but is there anything related when using decodebin2 and queue2 as I would assume playbin2 would use these also. At present we have definite breaks with clicks between tracks. Any help is greatly appreciated.... -------------- next part -------------- An HTML attachment was scrubbed... URL: From luciana at fujii.eti.br Thu Dec 16 03:45:36 2010 From: luciana at fujii.eti.br (Luciana Fujii Pontello) Date: Thu, 16 Dec 2010 00:45:36 -0200 Subject: [gst-devel] Simple ogg pipeline doesn't work In-Reply-To: <4CFE74D1.5090903@hi-iberia.es> References: <4CFE74D1.5090903@hi-iberia.es> Message-ID: <1292467536.19370.37.camel@abacate> On Tue, 2010-12-07 at 18:54 +0100, Andres Gonzalez wrote: > Hi, > > I have written a simple python program that is supossed to copy an ogg > file, using gstreamer. The original ogg file weights 662135 bytes, but > the program doesn't write more than 46365. Why does this program get > blocked, any idea? Sorry for the late reply. Just an idea. Have you checked how are your queues? I have seen locks while using a source with both audio and video and a mux joining them in the same pipeline. My guess is that one queue is full and the other empty and the mux won't get buffer from one queue until it gets buffers from the other queue. You can try using multiqueue, or just increasing the limit of your queues and see if it changes anything. Regards, Luciana Fujii From rafael.lmsousa at gmail.com Thu Dec 16 05:11:34 2010 From: rafael.lmsousa at gmail.com (Rafael Sousa) Date: Thu, 16 Dec 2010 01:11:34 -0300 Subject: [gst-devel] distorted video encoded with x264 Message-ID: When I use the x264 build 85 that issue doesn't occur, the video is perfect, only in superior build that I tested (98 and 110) that the problem occur. I don't have any idea of what can be happening, I've already removed and installed the packages several times, I even upgraded my ubuntu system from 10.04 to 10.10 but no change. Anyone have any idea? I've tried all that I know. I need to use the build 98 or superior. thanks for the reply Message: 1 > Date: Wed, 15 Dec 2010 15:42:01 -0500 > From: Timothy Braun > Subject: Re: [gst-devel] distorted video encoded with x264 > To: Discussion of the development of GStreamer > > Message-ID: <69C57452-40E4-4172-883F-DF374D355CD7 at gmail.com> > Content-Type: text/plain; charset="us-ascii" > > I'm no expert on this, but ran into similar issues when making a local > file. It ultimately ended up being the parameters I was passing to the > x264enc element. I would take a look at those with gst-inspect and see if > you can tune the output. Most notably the pass and quantizer parameters. > > Tim > > Sent from my iPhone > > On Dec 15, 2010, at 3:02 PM, Rafael Sousa > wrote: > > > Hi all, > > > > I'm using the versions of gstreamer installed from ubuntu 10.10 apt-get. > I've installed the gstreamer, gst-plugins-bad, gst-plugins-good, > gst-plugins-ugly, gst-ffmpeg and I've installed the x264. > > > > The problem is that my pipeline gives as result a blurring video. My > pipeline works to read a YUV file at the sender, encodes it in H.264, and > sends it to receiver using RTP in encoding time. At the receiver, it > receives the stream and stores it in YUV format again for posterior process. > The pipeline is as follows: > > > > Sender: > > > > "gstrtpbin name=rtpbin filesrc location=paris_cif.yuv blocksize=152064 ! > video/x-raw-yuv, width=352, height=288, framerate=(fraction)15/1, > format=(fourcc)I420 ! x264enc, key-int-max=30, key-int-min=15, > intra-refresh=true ! rtph264pay ! rtpbin.send_rtp_sink_0 > rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 > rtpbin.send_rtcp_src_0 ! udpsink host=5003 port=5001 sync=false async=false > udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 " > > > > Receiver: > > > > "gstrtpbin name=rtpbin udpsrc > caps=\"application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" > port=5000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! ffdec_h264 ! > filesink location=example.yuv udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 > rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false" > > > > I don't know why the video is blurring since I've tested it localhost, > without loss rates. The command that I'm using to watch the result is: > > > > mplayer example.yuv -demuxer rawvideo -rawvideo cif > > > > I took a screenshot of the video, it is in annex. Please, anyone can > help? > > > > thanks in advance > > > > > ------------------------------------------------------------------------------ > > Lotusphere 2011 > > Register now for Lotusphere 2011 and learn how > > to connect the dots, take your collaborative environment > > to the next level, and enter the era of Social Business. > > http://p.sf.net/sfu/lotusphere-d2d > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From matheist76 at westnet.com.au Thu Dec 16 09:59:00 2010 From: matheist76 at westnet.com.au (matheist76 at westnet.com.au) Date: Thu, 16 Dec 2010 16:59:00 +0800 (WST) Subject: [gst-devel] Brisbane, Australia - The linux.conf.au 2011 In-Reply-To: <106768631.2063771292433541890.JavaMail.root@zim-store01.web.westnet.com.au> Message-ID: <1310668069.2245581292489940050.JavaMail.root@zim-store01.web.westnet.com.au> Hello, There is only a short time to get your linux.conf.au 2011 registration. It's the pass to the best Open Source and Linux Conference in the Southern hemisphere. Our Keynotes include Vinton G Cerf one of the founding fathers of the internet and Eric Allman the original author of Sendmail. We have heaps of talk from leaders in the community. Andrew Tridgell (SAMBA) and Rusty Russell (Linux Kernel developer) just to name a couple. There is also a special Rocket miniconf for people who like a bit of a kick start to their day. If that's a bit too much why not make a Linux powered coffee roaster. We will have the ambulance ready for the burn victims at the Arduino miniconf and there is sure to be much more to enjoy away from the conference. While your in Brisbane you might as well meet a few of the locals at Australia Zoo or take a dip in the South Bank Streets Beach. During the conference there will be the Australia Day celebrations. A perfect time to just be an Aussie. Even if your not. For further information about lca 2011 check out our website at http://lca 2011 .linux.org.au/ For Prices: http://lca 2011 .linux.org.au/register/prices For Keynotes: http://lca 2011 .linux.org.au/programme/keynotes Thanks Matthew Franklin lca 2011 Organiser -------------- next part -------------- An HTML attachment was scrubbed... URL: From acandido at hi-iberia.es Thu Dec 16 10:00:59 2010 From: acandido at hi-iberia.es (Andres Gonzalez) Date: Thu, 16 Dec 2010 10:00:59 +0100 Subject: [gst-devel] Simple ogg pipeline doesn't work In-Reply-To: <1292467536.19370.37.camel@abacate> References: <4CFE74D1.5090903@hi-iberia.es> <1292467536.19370.37.camel@abacate> Message-ID: <4D09D54B.5010600@hi-iberia.es> On 16/12/10 03:45, Luciana Fujii Pontello wrote: > On Tue, 2010-12-07 at 18:54 +0100, Andres Gonzalez wrote: >> Hi, >> >> I have written a simple python program that is supossed to copy an ogg >> file, using gstreamer. The original ogg file weights 662135 bytes, but >> the program doesn't write more than 46365. Why does this program get >> blocked, any idea? > Sorry for the late reply. > > Just an idea. Have you checked how are your queues? I have seen locks > while using a source with both audio and video and a mux joining them in > the same pipeline. My guess is that one queue is full and the other > empty and the mux won't get buffer from one queue until it gets buffers > from the other queue. You can try using multiqueue, or just increasing > the limit of your queues and see if it changes anything. > > Regards, > > Luciana Fujii > Many thanks for the idea. I'll work on it and post if I make any progress. Cheers, Andr?s From t.i.m at zen.co.uk Thu Dec 16 10:23:12 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 16 Dec 2010 09:23:12 +0000 Subject: [gst-devel] distorted video encoded with x264 In-Reply-To: References: Message-ID: <1292491392.6213.11.camel@zingle> On Thu, 2010-12-16 at 01:11 -0300, Rafael Sousa wrote: > When I use the x264 build 85 that issue doesn't occur, the video is > perfect, only in superior build that I tested (98 and 110) that the > problem occur. I don't have any idea of what can be happening, I've > already removed and installed the packages several times, I even > upgraded my ubuntu system from 10.04 to 10.10 but no change. Anyone > have any idea? I've tried all that I know. I need to use the build 98 > or superior. Just as a note, the x264enc plugin code has some special handling here and there depending on the x264 version compiled against. It assumes that the x264 version used at runtime is actually the one it was compiled against as well at the moment. This can lead to weird things happening when you upgrade/downgrade libx264 to a different version than the plugin was compiled against. So when you upgrade/downgrade libx264, you definitely need to recompile gst-plugins-ugly. (There may still be a bug of course..) Cheers -Tim > thanks for the reply > > Message: 1 > Date: Wed, 15 Dec 2010 15:42:01 -0500 > From: Timothy Braun > Subject: Re: [gst-devel] distorted video encoded with x264 > To: Discussion of the development of GStreamer > > Message-ID: <69C57452-40E4-4172-883F-DF374D355CD7 at gmail.com> > Content-Type: text/plain; charset="us-ascii" > > I'm no expert on this, but ran into similar issues when making > a local file. It ultimately ended up being the parameters I > was passing to the x264enc element. I would take a look at > those with gst-inspect and see if you can tune the output. > Most notably the pass and quantizer parameters. > > Tim > > Sent from my iPhone > > On Dec 15, 2010, at 3:02 PM, Rafael Sousa > wrote: > > > Hi all, > > > > I'm using the versions of gstreamer installed from ubuntu > 10.10 apt-get. I've installed the gstreamer, gst-plugins-bad, > gst-plugins-good, gst-plugins-ugly, gst-ffmpeg and I've > installed the x264. > > > > The problem is that my pipeline gives as result a blurring > video. My pipeline works to read a YUV file at the sender, > encodes it in H.264, and sends it to receiver using RTP in > encoding time. At the receiver, it receives the stream and > stores it in YUV format again for posterior process. The > pipeline is as follows: > > > > Sender: > > > > "gstrtpbin name=rtpbin filesrc location=paris_cif.yuv > blocksize=152064 ! video/x-raw-yuv, width=352, height=288, > framerate=(fraction)15/1, format=(fourcc)I420 ! x264enc, > key-int-max=30, key-int-min=15, intra-refresh=true ! > rtph264pay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! > udpsink host=127.0.0.1 port=5000 rtpbin.send_rtcp_src_0 ! > udpsink host=5003 port=5001 sync=false async=false udpsrc > port=5005 ! rtpbin.recv_rtcp_sink_0 " > > > > Receiver: > > > > "gstrtpbin name=rtpbin udpsrc caps= > \"application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" port=5000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! ffdec_h264 ! filesink location=example.yuv udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false" > > > > I don't know why the video is blurring since I've tested it > localhost, without loss rates. The command that I'm using to > watch the result is: > > > > mplayer example.yuv -demuxer rawvideo -rawvideo cif > > > > I took a screenshot of the video, it is in annex. Please, > anyone can help? > > > > thanks in advance > > > > > ------------------------------------------------------------------------------ > > Lotusphere 2011 > > Register now for Lotusphere 2011 and learn how > > to connect the dots, take your collaborative environment > > to the next level, and enter the era of Social Business. > > http://p.sf.net/sfu/lotusphere-d2d > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From t.i.m at zen.co.uk Thu Dec 16 10:36:18 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 16 Dec 2010 09:36:18 +0000 Subject: [gst-devel] distorted video encoded with x264 In-Reply-To: <1292491392.6213.11.camel@zingle> References: <1292491392.6213.11.camel@zingle> Message-ID: <1292492178.6213.13.camel@zingle> On Thu, 2010-12-16 at 09:23 +0000, Tim-Philipp M?ller wrote: > It assumes that the x264 version used at runtime is actually the one it > was compiled against as well at the moment. Actually, ignore all this. Apparently the plugin will just fail to link/load if the versions are different... Cheers -Tim From robert.swain at gmail.com Thu Dec 16 10:40:27 2010 From: robert.swain at gmail.com (Rob) Date: Thu, 16 Dec 2010 10:40:27 +0100 Subject: [gst-devel] x264 build 110 incompatible with gstreamer? In-Reply-To: References: Message-ID: On 10 December 2010 16:05, Rafael Sousa wrote: > I'm trying to encode a yuv video to h.264, for this, I installed the lastest > builds of gstreamer, plugins-bad ugly and good, gst-ffmpeg and x264 (build > 110). But when I play the encoded video the image is blurred. I tried to > downgrade the build of x264 to build 85 and it worked fine. There is some > incompatibility with the gstreamer and x264's lastest build? I'm using > ubuntu 10.04. So, you compiled a newer version of x264 and then recompiled the same version of -ugly that you were using against this new version of libx264? There may have been issues with newer x264 that were fixed since in newer -ugly releases or git. Unfortunately I don't remember all the details. Regards, Rob From robert.swain at gmail.com Thu Dec 16 10:53:38 2010 From: robert.swain at gmail.com (Rob) Date: Thu, 16 Dec 2010 10:53:38 +0100 Subject: [gst-devel] distorted video encoded with x264 In-Reply-To: References: Message-ID: On 15 December 2010 21:02, Rafael Sousa wrote: > I'm using the versions of gstreamer installed from ubuntu 10.10 apt-get. > I've installed the gstreamer, gst-plugins-bad, gst-plugins-good, > gst-plugins-ugly, gst-ffmpeg and I've installed the x264. > The problem is that my pipeline gives as result a blurring video. My > pipeline works to read a YUV file at the sender, encodes it in H.264, and > sends it to receiver using RTP in encoding time. At the receiver, it > receives the stream and stores it in YUV format again for posterior process. Using a newer version of x264 without recompiling -ugly against that newer version will not make use of that newer version of x264. In fact, x264enc should fail to load unless you still have the x264 version against which it was linked. > The pipeline is as follows: > Sender: > "gstrtpbin name=rtpbin filesrc location=paris_cif.yuv blocksize=152064 > !?video/x-raw-yuv, width=352, height=288, framerate=(fraction)15/1, > format=(fourcc)I420 !?x264enc, key-int-max=30, key-int-min=15, > intra-refresh=true !?rtph264pay > !?rtpbin.send_rtp_sink_0?rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 > port=5000?rtpbin.send_rtcp_src_0 ! udpsink host=5003 port=5001 sync=false > async=false?udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 " Why don't you set a rate control method? The default is 1-pass ABR 2Mbps which should be overkill for 352x288 15fps. > Receiver: > "gstrtpbin name=rtpbin?udpsrc > caps=\"application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264"?port=5000 > ! rtpbin.recv_rtp_sink_0?rtpbin. ! rtph264depay ! ffdec_h264 ! filesink > location=example.yuv udpsrc port=5001 ! > rtpbin.recv_rtcp_sink_0?rtpbin.send_rtcp_src_0 ! udpsink port=5005 > sync=false async=false" > I don't know why the video is blurring since I've tested it localhost, > without loss rates. The command that I'm using to watch the result is: > mplayer example.yuv -demuxer rawvideo -rawvideo cif > I took a screenshot of the video, it is in annex. Please, anyone can help? The screenshot looks like heavy over-quantisation and subsequent very heavy in-loop deblocking. This normally means that, considering the bit rate should be overkill for this resolution and frame rate, x264enc is not configured correctly. I strongly recommend you actually set a rate control method, even if it means just choosing a bit rate appropriate for your content. Then I would recommend making use of the presets and tunings. For your purposes you probably want a faster preset (whatever your cpu can handle though default of medium may be OK, if not, you could reduce the preset until it consistently and reliably encodes in realtime on your machine) and the zerolatency tuning. If you're still having issues, you made need to build gst from git against x264 from git to see if the issue persists with current development code. If it does, please report back. Regards, Rob From wb666greene at gmail.com Thu Dec 16 15:40:22 2010 From: wb666greene at gmail.com (wally_bkg) Date: Thu, 16 Dec 2010 06:40:22 -0800 (PST) Subject: [gst-devel] Why does this file encoding pipeline not work? Message-ID: <1292510422444-3090988.post@n4.nabble.com> I've been playing around with the various encoders in gstreamer to get an idea of quality/size/cpu tradeoffs, and so far x264enc is looking like the best overall compromise, at least based on what is shipped with Ubuntu 10.04. I need a readily available target with a reasonable lifespan and update policy, Ubuntu LTS versions seem to me the best compromise at present, so for the time being I need to live with whatever limitations exist in the gstreamer-0.10.28 its ships with. I can create a test file with: gst-launch -e v4l2src ! video/x-raw-yuv, framerate=\(fraction\)30000/1001, width=640, height=480 ! x264enc ! avimux ! filesink location=test.avi This file plays back fine in Totem, although Totem will not play it if I change the name to test.mp4, although changing it to test.mpr or test.264 will work. What's up with this? Also, while the files I record with these various gstreamer pipelines all play back fine in Totem (using gstreamer, Doh!), most all of them will not play back on a Windows system that is my "gold standard" for playing obscure media formats people give me to convert for their Powerpoint presentations. However trying to get an idea of CPU usage in a more realistic analog of my application the following pipeline locks up displaying only a single frame and produces an output file of 0 bytes: gst-launch -e v4l2src ! video/x-raw-yuv, framerate=\(fraction\)30000/1001, width=640, height=480 ! tee name=t ! queue ! xvimagesink t. ! queue ! x264enc ! avimux ! filesink location=test.avi I get: Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock And the display window is locked on a single frame .... When I press control-C in the terminal window I get: Caught interrupt -- handling interrupt. Interrupt: Stopping pipeline ... EOS on shutdown enabled -- Forcing EOS on the pipeline Waiting for EOS... which hangs until I press control-C again. If I close the output window when its locked up instead of pressing control-C in the terminal window I get this when the window is closed: ERROR: from element /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0: Output window was closed Additional debug info: xvimagesink.c(1299): gst_xvimagesink_handle_xevents (): /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0 Execution ended after 7073325548 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... I'd like to point out, that "man gst-launch" appears to have no mention of the -e option, which I learned about on this list a while back. My C code that effectively does: vrl2src->videoscale->queue->appsink -\ \appsrc->queue->xvimagesink \appsrc->queue->x264enc->filesink (with the above caps set on the appsink, & appsrc elements) Seems to work ok, although I haven't got to the point of independently starting/stopping the recoding pipeline, initially all three start at once. I'm pretty happy with the ~72% CPU usages shown by top (quad core) and can run two instances of the program writing to different out directories, although there may be recording glitches introduced when doing this, I need further testing. The videoscale element does nothing for most of my capture cards as they can output 640x480 frames, but I have a USB Haupapauge HRV-950Q that can only do 720x480 on its analog inputs. Video scale is very expensive in this case, increasing CPU usage by ~25% -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Why-does-this-file-encoding-pipeline-not-work-tp3090988p3090988.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From rafael.lmsousa at gmail.com Thu Dec 16 17:34:57 2010 From: rafael.lmsousa at gmail.com (Rafael Sousa) Date: Thu, 16 Dec 2010 12:34:57 -0400 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 66 In-Reply-To: References: Message-ID: Hi Tim, I've recompiled the gst-plugins-ugly and all others plugins related, like plugins-ugly, plugins-bad, plugins-base, plugins-good , gst-ffmpeg, ffmpeg and gstreamer. I'm certain that the integration is complete by inserting some printfs in the code and making sure that it is print on the stdout. I still don't know what to do =(. > > Message: 2 > Date: Thu, 16 Dec 2010 09:23:12 +0000 > From: Tim-Philipp M?ller > Subject: Re: [gst-devel] distorted video encoded with x264 > To: gstreamer-devel at lists.sourceforge.net > Message-ID: <1292491392.6213.11.camel at zingle> > Content-Type: text/plain; charset="UTF-8" > > On Thu, 2010-12-16 at 01:11 -0300, Rafael Sousa wrote: > > > When I use the x264 build 85 that issue doesn't occur, the video is > > perfect, only in superior build that I tested (98 and 110) that the > > problem occur. I don't have any idea of what can be happening, I've > > already removed and installed the packages several times, I even > > upgraded my ubuntu system from 10.04 to 10.10 but no change. Anyone > > have any idea? I've tried all that I know. I need to use the build 98 > > or superior. > > Just as a note, the x264enc plugin code has some special handling here > and there depending on the x264 version compiled against. It assumes > that the x264 version used at runtime is actually the one it was > compiled against as well at the moment. This can lead to weird things > happening when you upgrade/downgrade libx264 to a different version than > the plugin was compiled against. > > So when you upgrade/downgrade libx264, you definitely need to recompile > gst-plugins-ugly. > > (There may still be a bug of course..) > > Cheers > -Tim > > > > thanks for the reply > > > > Message: 1 > > Date: Wed, 15 Dec 2010 15:42:01 -0500 > > From: Timothy Braun > > Subject: Re: [gst-devel] distorted video encoded with x264 > > To: Discussion of the development of GStreamer > > > > Message-ID: <69C57452-40E4-4172-883F-DF374D355CD7 at gmail.com> > > Content-Type: text/plain; charset="us-ascii" > > > > I'm no expert on this, but ran into similar issues when making > > a local file. It ultimately ended up being the parameters I > > was passing to the x264enc element. I would take a look at > > those with gst-inspect and see if you can tune the output. > > Most notably the pass and quantizer parameters. > > > > Tim > > > > Sent from my iPhone > > > > On Dec 15, 2010, at 3:02 PM, Rafael Sousa > > wrote: > > > > > Hi all, > > > > > > I'm using the versions of gstreamer installed from ubuntu > > 10.10 apt-get. I've installed the gstreamer, gst-plugins-bad, > > gst-plugins-good, gst-plugins-ugly, gst-ffmpeg and I've > > installed the x264. > > > > > > The problem is that my pipeline gives as result a blurring > > video. My pipeline works to read a YUV file at the sender, > > encodes it in H.264, and sends it to receiver using RTP in > > encoding time. At the receiver, it receives the stream and > > stores it in YUV format again for posterior process. The > > pipeline is as follows: > > > > > > Sender: > > > > > > "gstrtpbin name=rtpbin filesrc location=paris_cif.yuv > > blocksize=152064 ! video/x-raw-yuv, width=352, height=288, > > framerate=(fraction)15/1, format=(fourcc)I420 ! x264enc, > > key-int-max=30, key-int-min=15, intra-refresh=true ! > > rtph264pay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! > > udpsink host=127.0.0.1 port=5000 rtpbin.send_rtcp_src_0 ! > > udpsink host=5003 port=5001 sync=false async=false udpsrc > > port=5005 ! rtpbin.recv_rtcp_sink_0 " > > > > > > Receiver: > > > > > > "gstrtpbin name=rtpbin udpsrc caps= > > > \"application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" > port=5000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! ffdec_h264 ! > filesink location=example.yuv udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 > rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false" > > > > > > I don't know why the video is blurring since I've tested it > > localhost, without loss rates. The command that I'm using to > > watch the result is: > > > > > > mplayer example.yuv -demuxer rawvideo -rawvideo cif > > > > > > I took a screenshot of the video, it is in annex. Please, > > anyone can help? > > > > > > thanks in advance > > > > > > > > > ------------------------------------------------------------------------------ > > > Lotusphere 2011 > > > Register now for Lotusphere 2011 and learn how > > > to connect the dots, take your collaborative environment > > > to the next level, and enter the era of Social Business. > > > http://p.sf.net/sfu/lotusphere-d2d > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > > Lotusphere 2011 > > Register now for Lotusphere 2011 and learn how > > to connect the dots, take your collaborative environment > > to the next level, and enter the era of Social Business. > > http://p.sf.net/sfu/lotusphere-d2d > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jpl at xiphos.ca Thu Dec 16 18:45:32 2010 From: jpl at xiphos.ca (Joshua Lamorie) Date: Thu, 16 Dec 2010 12:45:32 -0500 Subject: [gst-devel] Segmentation fault in g_type_check_instance_cast In-Reply-To: <1292437725.16916.26.camel@haddock.xiphos.ca> References: <1292437725.16916.26.camel@haddock.xiphos.ca> Message-ID: <1292521532.21031.301.camel@haddock.xiphos.ca> Well, I have isolated the problem to be something in the way that GST_TYPE_APP_SRC returns a reference to the g_type. For some reason, the way that I've built my application (on a 64-bit system) is mixing in a few 32-bit references, so instead of returning 0x7fff12345678 it returns 0x12345678 and a segfault occurs. I've rebuilt my code on a 'pure' 32-bit system, and everything appears to be okay. Thanks Joshua On Wed, 2010-12-15 at 13:28 -0500, Joshua Lamorie wrote: > Gidday there, > > I'm having some issues with some code that runs fine as a 'standalone' > application, but when I put it into a thread spawned by another > application it has a segfault. > > The following is from GDB: > > [Switching to Thread 0x7ffff106f710 (LWP 4780)] > IA__g_type_check_instance_cast (type_instance=0x7fffe4008030 > [GstAppSrc], iface_type=3825213056) at gtype.c:3973 > 3973 check = is_instantiatable && iface && type_node_conforms_to_U > (node, iface, TRUE, FALSE); > > (gdb) bt > #0 IA__g_type_check_instance_cast (type_instance=0x7fffe4008030 > [GstAppSrc], iface_type=3825213056) at gtype.c:3973 > #1 0x00007ffff7b3fb64 in ESMRTPThreadMain (pArg=0x7fffec0052d0) at > RTPSourcePlugin.c:283 > #2 0x000000346fe64e84 in g_thread_create_proxy (data=0x7fffec005290) at > gthread.c:1893 > #3 0x000000346de07761 in start_thread (arg=0x7ffff106f710) at > pthread_create.c:301 > #4 0x000000346d2e14fd in clone () > at ../sysdeps/unix/sysv/linux/x86_64/clone.S:115 > > > Here's some pseudo code of the overall system. > > 1) Main C++ application loads and initializes itself. > 2) Opens a .so containing a few extern C functions/symbols. > 3) Calls an Init() function from the .so (which calls gst_init()) > 4) Calls pthread_create() and sits on an accept() call. > 4.1) The accept() returns and a message is read by the C++ app/thread. > 4.2) This then calls an Execute() function from the .so. > 4.3) The Execute() method creates a thread (either pthread_create or > g_thread_create). > 4.3.1) The newly created thread is pretty much copy and pasted from a > functional 'standalone' gstreamer application and initializes itself. > 4.3.2) During a call (cast) to GST_APP_SRC(GstElement *) a segfault > occurs somewhere within the inlined functions. > > I've put this in gdb and it appears that all of the input arguments are > non-NULL, but deep within the inlined functions I saw some references to > what appear to be global locking variables. I'm concerned that my > initialization doesn't quite match the standalone case, but I can't > quite figure out what is going on 'behind-the-curtain' in the standalone > case that is different from my multi-threaded example. > > Standalone code: > > GMainLoop *loop; > int main(int iArgc, char * ppcArgv[]) { > GstElement * poAppSrc; > GstElement * poPipeline; > /* ... snip ... */ > /* initialize gstreamer */ > gst_init(NULL,NULL); > loop = g_main_loop_new (NULL, TRUE); > poPipeline = gst_pipeline_new("ESMPipeline"); > poAppSrc = gst_element_factory_make("appsrc", NULL); > gst_app_src_set_stream_type(GST_APP_SRC(poAppSrc), > GST_APP_STREAM_TYPE_STREAM); > /* ... snip ... */ > } > > Thread code: > > void * ESMRTPThreadMain(void * pArg) { > GstElement * poPipeline = NULL; > GstElement * poAppSrc = NULL; > /* ... snip ... */ > poPipeline = gst_pipeline_new("ESMPipeline"); > poAppSrc = gst_element_factory_make("appsrc",NULL); > if (poAppSrc == NULL) { > printf("Problem generating appsrc element\n"); > return(NULL); > } > /* set stream type */ > gst_app_src_set_stream_type(GST_APP_SRC(poAppSrc), > GST_APP_STREAM_TYPE_STREAM); > /* ... snip ... */ > } > > So what am I doing wrong in the initialization to cause the segfault? > > Thanks in advance. > -- Joshua Lamorie, P. Eng. Chief Engineer Xiphos Technologies Inc. Email: jpl at xiphos.ca Phone: +1 (514) 847-9474 ext. 227 Fax: +1 (514) 847-9474 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. If you have received this in error, please contact the sender and delete this communication and any copy immediately. Thank you. From shiva.varma at gmail.com Fri Dec 17 01:13:53 2010 From: shiva.varma at gmail.com (shiva varma) Date: Thu, 16 Dec 2010 16:13:53 -0800 (PST) Subject: [gst-devel] Problems converting audio pcap using pcapparse, rtpmp4depay and qtmux plugins In-Reply-To: References: Message-ID: <1292544833133-3091917.post@n4.nabble.com> Update I have tried the following pipeline and I get a similar error gst-launch filesrc location=audio.pcap ! pcapparse ! "application/x-rtp, payload=96, clock-rate=44100, encoding-name=MP4A-LATM, cpresent=0, config=NULL" ! gstrtpjitterbuffer ! rtpmp4adepay ! "audio/mpeg, mpegversion=4, stream-format=raw, channels=2, rate=44100" ! qtmux ! fakesink -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Problems-converting-audio-pcap-using-pcapparse-rtpmp4depay-and-qtmux-plugins-tp3090099p3091917.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bertd at tplogic.com Fri Dec 17 08:04:53 2010 From: bertd at tplogic.com (Bert Douglas) Date: Thu, 16 Dec 2010 23:04:53 -0800 Subject: [gst-devel] video for linux test source -- vivi.ko module -- missing in ubuntu 10.10 Message-ID: Hello helpful gstreamer folks, I converted to ubuntu 10.10 and it seems there is no kernel module for vivi, the virtual video device. It was there in ubuntu 10.04. Anybody else encountered this problem? I have never built a custom kernel, and was hoping to remain blissfully ignorant. Thanks, Bert Douglas -------------- next part -------------- An HTML attachment was scrubbed... URL: From bard_kuo at compalcomm.com Fri Dec 17 09:53:42 2010 From: bard_kuo at compalcomm.com (bard) Date: Fri, 17 Dec 2010 00:53:42 -0800 (PST) Subject: [gst-devel] xvimagesink Could not initialise Xv output and No port available Message-ID: <1292576022802-3092246.post@n4.nabble.com> Hi All, My xvimagesink does not work but ximagesink does. I use xvimagesink as below /home/user # gst-launch videotestsrc ! xvimagesink Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0: Could not initialise Xv output Additional debug info: xvimagesink.c(1446): gst_xvimagesink_get_xv_support (): /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0: No port available Setting pipeline to NULL ... Freeing pipeline ... /home/user # What cause "could not initialise Xv output" and "No port available" ? Is there something wrong ? How do I check ? Thanks, Bard -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/xvimagesink-Could-not-initialise-Xv-output-and-No-port-available-tp3092246p3092246.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Fri Dec 17 10:10:57 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 17 Dec 2010 09:10:57 +0000 Subject: [gst-devel] xvimagesink Could not initialise Xv output and No port available In-Reply-To: <1292576022802-3092246.post@n4.nabble.com> References: <1292576022802-3092246.post@n4.nabble.com> Message-ID: <1292577057.23642.1.camel@zingle> On Fri, 2010-12-17 at 00:53 -0800, bard wrote: > ERROR: from element /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0: > Could not initialise Xv output > Additional debug info: > xvimagesink.c(1446): gst_xvimagesink_get_xv_support (): > /GstPipeline:pipeline0/GstXvImageSink:xvimagesink0: > No port available > > What cause "could not initialise Xv output" and "No port available" ? > > Is there something wrong ? How do I check ? What does the 'xvinfo' command show? What's the output of $ GST_DEBUG_NO_COLOR=1 GST_DEBUG=xv*:5 gst-launch-0.10 videotestsrc ! xvimagesink ? Cheers -Tim From bard_kuo at compalcomm.com Fri Dec 17 10:39:16 2010 From: bard_kuo at compalcomm.com (bard) Date: Fri, 17 Dec 2010 01:39:16 -0800 (PST) Subject: [gst-devel] xvimagesink Could not initialise Xv output and No port available In-Reply-To: <1292577057.23642.1.camel@zingle> References: <1292576022802-3092246.post@n4.nabble.com> <1292577057.23642.1.camel@zingle> Message-ID: <1292578756801-3092288.post@n4.nabble.com> Thanks for reply, 1.xvinfo: X-Video Extension version 2.2 screen #0 no adaptors present 2.GST_DEBUG_NO_COLOR=1 GST_DEBUG=xv*:5 gst-launch-0.10 videotestsrc ! xvimagesink <--- I will try to turn on the debug and get back to you . Thanks, Bard -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/xvimagesink-Could-not-initialise-Xv-output-and-No-port-available-tp3092246p3092288.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Fri Dec 17 10:50:32 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 17 Dec 2010 09:50:32 +0000 Subject: [gst-devel] xvimagesink Could not initialise Xv output and No port available In-Reply-To: <1292578756801-3092288.post@n4.nabble.com> References: <1292576022802-3092246.post@n4.nabble.com> <1292577057.23642.1.camel@zingle> <1292578756801-3092288.post@n4.nabble.com> Message-ID: <1292579432.23642.15.camel@zingle> On Fri, 2010-12-17 at 01:39 -0800, bard wrote: Hi, > 1.xvinfo: > X-Video Extension version 2.2 > screen #0 > no adaptors present > > 2.GST_DEBUG_NO_COLOR=1 GST_DEBUG=xv*:5 gst-launch-0.10 videotestsrc ! > xvimagesink <--- I will try to turn on the debug and get back to you . Well, if xvinfo says there are no adapters, that's the problem (xvimagesink reports the same after all). It's more of an X11/driver issue though than a GStreamer issue. Cheers -Tim From bard_kuo at compalcomm.com Fri Dec 17 11:06:47 2010 From: bard_kuo at compalcomm.com (bard) Date: Fri, 17 Dec 2010 02:06:47 -0800 (PST) Subject: [gst-devel] xvimagesink Could not initialise Xv output and No port available In-Reply-To: <1292579432.23642.15.camel@zingle> References: <1292576022802-3092246.post@n4.nabble.com> <1292577057.23642.1.camel@zingle> <1292578756801-3092288.post@n4.nabble.com> <1292579432.23642.15.camel@zingle> Message-ID: <1292580407505-3092313.post@n4.nabble.com> Thank you Tim! If I want to dig more , how do I do to make sure it is not gstreamer's issue ? Thanks, Bard -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/xvimagesink-Could-not-initialise-Xv-output-and-No-port-available-tp3092246p3092313.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Fri Dec 17 11:25:33 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 17 Dec 2010 10:25:33 +0000 Subject: [gst-devel] xvimagesink Could not initialise Xv output and No port available In-Reply-To: <1292580407505-3092313.post@n4.nabble.com> References: <1292576022802-3092246.post@n4.nabble.com> <1292577057.23642.1.camel@zingle> <1292578756801-3092288.post@n4.nabble.com> <1292579432.23642.15.camel@zingle> <1292580407505-3092313.post@n4.nabble.com> Message-ID: <1292581533.23642.17.camel@zingle> On Fri, 2010-12-17 at 02:06 -0800, bard wrote: > If I want to dig more , how do I do to make sure it is not gstreamer's issue? The output of xvinfo indicates that it's not a GStreamer issue. From nico at inattendu.org Fri Dec 17 11:32:20 2010 From: nico at inattendu.org (Nicolas Bertrand) Date: Fri, 17 Dec 2010 14:32:20 +0400 Subject: [gst-devel] xvimagesink Could not initialise Xv output and No port available In-Reply-To: <1292580407505-3092313.post@n4.nabble.com> References: <1292576022802-3092246.post@n4.nabble.com> <1292577057.23642.1.camel@zingle> <1292578756801-3092288.post@n4.nabble.com> <1292579432.23642.15.camel@zingle> <1292580407505-3092313.post@n4.nabble.com> Message-ID: <4D0B3C34.10203@inattendu.org> Le 17/12/2010 14:06, bard a ?crit : > Thank you Tim! > > If I want to dig more , how do I do to make sure it is not gstreamer's issue > ? > > Thanks, > Bard > It's not a gstreamer error because the error is reported by your X11 driver. It can come for many reasons, for example I have this problem on a Nvidia based card, because I didn't have installed the proprietary divers Try to use autovideosink instead of ximagesink/xvimagesink to pass through this error. From bard_kuo at compalcomm.com Fri Dec 17 13:03:22 2010 From: bard_kuo at compalcomm.com (bard) Date: Fri, 17 Dec 2010 04:03:22 -0800 (PST) Subject: [gst-devel] xvimagesink Could not initialise Xv output and No port available In-Reply-To: <4D0B3C34.10203@inattendu.org> References: <1292576022802-3092246.post@n4.nabble.com> <1292577057.23642.1.camel@zingle> <1292578756801-3092288.post@n4.nabble.com> <1292579432.23642.15.camel@zingle> <1292580407505-3092313.post@n4.nabble.com> <4D0B3C34.10203@inattendu.org> Message-ID: <1292587402740-3092433.post@n4.nabble.com> Thank you ! It helps me a lot. Now I know there is no problem in target for xvimagesink. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/xvimagesink-Could-not-initialise-Xv-output-and-No-port-available-tp3092246p3092433.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From shiva.varma at gmail.com Fri Dec 17 20:53:29 2010 From: shiva.varma at gmail.com (shiva varma) Date: Fri, 17 Dec 2010 11:53:29 -0800 (PST) Subject: [gst-devel] Converting MPEG4-GENERIC pcap audio using rtpmp4depay Message-ID: <1292615609799-3093035.post@n4.nabble.com> When I use rtpmp4depay to convert MPEG4-GENERIC audio in pcap, I get the "reason not-negotiated" error gst-launch filesrc location=audio.pcap ! pcapparse src-ip=10.22.156.148 ! "application/x-rtp, media=audio, clock-rate=48000, encoding-name=MPEG4-GENERIC" ! rtpmp4gdepay ! fakesink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstFileSrc:filesrc0: Internal data flow error. Additional debug info: gstbasesrc.c(2550): gst_base_src_loop (): /GstPipeline:pipeline0/GstFileSrc:filesrc0: streaming task paused, reason not-negotiated (-4) ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... Can you point me to the possible cause of the error Thanks -Shiva -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Converting-MPEG4-GENERIC-pcap-audio-using-rtpmp4depay-tp3093035p3093035.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wim.taymans at gmail.com Fri Dec 17 21:01:10 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Fri, 17 Dec 2010 21:01:10 +0100 Subject: [gst-devel] Converting MPEG4-GENERIC pcap audio using rtpmp4depay In-Reply-To: <1292615609799-3093035.post@n4.nabble.com> References: <1292615609799-3093035.post@n4.nabble.com> Message-ID: <1292616070.17279.44.camel@meany> On Fri, 2010-12-17 at 11:53 -0800, shiva varma wrote: > When I use rtpmp4depay to convert MPEG4-GENERIC audio in pcap, I get the > "reason not-negotiated" error > > gst-launch filesrc location=audio.pcap ! pcapparse src-ip=10.22.156.148 ! > "application/x-rtp, media=audio, clock-rate=48000, > encoding-name=MPEG4-GENERIC" ! rtpmp4gdepay ! fakesink > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > ERROR: from element /GstPipeline:pipeline0/GstFileSrc:filesrc0: Internal > data flow error. > Additional debug info: > gstbasesrc.c(2550): gst_base_src_loop (): > /GstPipeline:pipeline0/GstFileSrc:filesrc0: > streaming task paused, reason not-negotiated (-4) > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > Freeing pipeline ... > > Can you point me to the possible cause of the error Your caps are incomplete, gst-inspect rtpmp4gdepay will show that 'streamtype' and 'mode' should be specified on the caps. Wim > > Thanks > -Shiva > From shiva.varma at gmail.com Fri Dec 17 21:18:06 2010 From: shiva.varma at gmail.com (shiva varma) Date: Fri, 17 Dec 2010 12:18:06 -0800 (PST) Subject: [gst-devel] Converting MPEG4-GENERIC pcap audio using rtpmp4depay In-Reply-To: <1292616070.17279.44.camel@meany> References: <1292615609799-3093035.post@n4.nabble.com> <1292616070.17279.44.camel@meany> Message-ID: <1292617086168-3093068.post@n4.nabble.com> Hi Wim Thanks for your response I tried adding mode and streamtype to the caps... gst-launch filesrc location=cts.mpeg4.PT96.pcap ! pcapparse ! "application/x-rtp, payload=96, media=audio, clock-rate=48000, encoding-name=MPEG4-GENERIC, mode=generic, streamtype=4" ! rtpmp4gdepay ! fakesink gst-launch filesrc location=cts.mpeg4.PT96.pcap ! pcapparse ! "application/x-rtp, payload=96, media=audio, clock-rate=48000, encoding-name=MPEG4-GENERIC, mode=generic, streamtype=5" ! rtpmp4gdepay ! fakesink Also tried with mode= AAC-lbr and AAC-hbr, I am still running into the same error.... Thanks Shiva -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Converting-MPEG4-GENERIC-pcap-audio-using-rtpmp4depay-tp3093035p3093068.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From amit.sharma1986 at gmail.com Sat Dec 18 10:13:24 2010 From: amit.sharma1986 at gmail.com (amit sharma) Date: Sat, 18 Dec 2010 14:43:24 +0530 Subject: [gst-devel] Error : Linking exe with gstreamer shared lib Message-ID: Hi, I have successfully created lib for gstreamer and supported plugin by using NDK, now i want to test the working of so file, for that I create a application and create a exe by using NDK . Next step is to run the exe on android-emulator, for that first I load the exe into directory /data/temp by using adb push command, then push all the shared library which is required to execute the exe i.e. libglib-2.0.so libgmodule-2.0.so libgobject-2.0.so libgthread-2.0.so libgstcoreelements-gstplugin.so libgstreamer-0.10-libs.so libgstcoreindexers-gstplugin.so libgstreamer.so libgst-plugins-base-libs-0.10.so libgstapp.so So everything is now in the same folder. Next step is to link the library at run time so for that I set the path by using LD_LIBRARY_PATH So up to now everything seem to be OK. Final Step: To execute the exe. Error: The strange thing that i observe that exe is able to execute some of the basic api of gstreamer like as, gst_init (&argc, &argv); bin = gst_pipeline_new ("pipeline"); But while executing following command, exe not able to create appsrc element. appsrc = gst_element_factory_make ("appsrc", NULL); --------> Not able to create this element I have crossed check and I found I have all the .so file that is required to execute the code. GLib: Cannot convert message: Could not open converter from 'UTF-8' to 'ASCII' ----> I am getting this message also while executing the code. So, If any body face such type of error or want to share something on this issue then please share your valueable point. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From rohitratri at gmail.com Sat Dec 18 16:42:35 2010 From: rohitratri at gmail.com (Rohit Atri) Date: Sat, 18 Dec 2010 21:12:35 +0530 Subject: [gst-devel] Selective decoding of video packets Message-ID: Hi, I m writing an application which reads mp4 packets from a file (bin1: filesrc->qtdemux->appsink), do some analysis on a set(say 100 - depending on the duration of the video) of packets and choose one single packet and send it for decoding (bin2: appsrc->ffdec_mpeg4->appsink) and then save this image... repeat the process for all packets in the video. I've a few questions - 1. Are the elements I'm using apt? and do I've to do some thread management since I've to pipelines? 2. a. I'm able to receive packets from bin1, do some analysis and send a single packet(iFrame) for decoding.. but bin2 doesn't return the packet back(the application hangs!).. b. When I turn on the logs with level-4, I do get the packet back!!! - so it could be timing issue, I suspect. c. The bin1 returns say 200 or so packets and the app receives an EOS message from it! - why is that? 3. Does appsink return packets as and when it receives them or does it wait to preroll? Please help me understand the situation better. I've hit a roadblock! Please let me know if you need some logs... Thanks Rohit -------------- next part -------------- An HTML attachment was scrubbed... URL: From gibrovacco at gmail.com Sat Dec 18 16:55:19 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sat, 18 Dec 2010 17:55:19 +0200 Subject: [gst-devel] Problems converting audio pcap using pcapparse, rtpmp4depay and qtmux plugins In-Reply-To: <1292544833133-3091917.post@n4.nabble.com> References: <1292544833133-3091917.post@n4.nabble.com> Message-ID: Hi, try getting the proper playload type and port to set in the pcapparse element. You achieve it easily by opening your dump with Wireshark. Besides, which sw did you use to collect the network dump? And which command? Regards On Fri, Dec 17, 2010 at 2:13 AM, shiva varma wrote: > > Update > I have tried the following pipeline and I get a similar error > > gst-launch filesrc location=audio.pcap ! pcapparse ! "application/x-rtp, > payload=96, clock-rate=44100, encoding-name=MP4A-LATM, cpresent=0, > config=NULL" ! gstrtpjitterbuffer ! rtpmp4adepay ! "audio/mpeg, > mpegversion=4, stream-format=raw, channels=2, rate=44100" ! qtmux ! fakesink > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Problems-converting-audio-pcap-using-pcapparse-rtpmp4depay-and-qtmux-plugins-tp3090099p3091917.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From gibrovacco at gmail.com Sun Dec 19 09:53:33 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sun, 19 Dec 2010 10:53:33 +0200 Subject: [gst-devel] gapless playback In-Reply-To: References: Message-ID: Hi, On Thu, Dec 16, 2010 at 3:58 AM, Stuart Gray wrote: > Hi, > > I am seeking some guidance with respect to getting gapless playback using > decodebin2 and queue2. > > Our system comprises the following: > > arm9 processor > gst-plugins-bad - 0.10.20-r11.0.6 > gst-plugins-base - 0.10.31-r11.0.6 > gst-plugins-good - 0.10.25-r11.0.6 > gst-plugins-ugly - 0.10.16-r11.1.6 > running on openembedded linux with kernel version 2.6.33 > > We are decoding audio only streams. This is a concern as reference was made > to gapless playback for the latest releases: > > "Release notes for GStreamer Base Plug-ins?0.10.31 "Dance Like It's 1982" > > Bugs fixed in this release > > ? ? ?* 628028 : [uridecodebin] Don't enable progressive downloading for live > streams > ? ? ?* 623846 : typefinding: add support for " enhanced ac3 " (eac3) > ? ? ?* 602437 : [playbin2] [gapless] Completely broken when switching > between files with audio/video only > " > > The above refers directly to playbin2, but is there anything related when > using decodebin2 and queue2 as I would assume playbin2 would use these also. > > At present we have definite breaks with clicks between tracks. > > Any help is greatly appreciated.... afaiu the gapless playback concept in playbin2 is based on the "about-to-finish" signal: http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-playbin2.html#GstPlayBin2-about-to-finish which appears to be playbin2 specific, but let's not give up with hopes now. As the signal is emitted from the handler of the "drained" signal, in turn emitted from uridecodebin/decodebin2, you can use it to prepare the next track in your app. Still you'll have to implement by yourself the handling logic as in the following functions (I'm sorry not to put a link to the gitweb source but freedesktop is awfully unstable today): - drained_cb: the handler of the "drained" signal. - activate_group: (the most noteworthy) re-sets the pipeline and puts in it in the appropriate state. Essentially, you'll need to set all the pipeline properties in the signal handler, as implemented in activate_group. These (intermediate) functions may be interesting to understand how playbin2 internally handles the transition: - gst_play_bin_set_suburi: to store the next URI to be played. - setup_next_source: called by the drained_cb callback after the new URI has been set with gst_play_bin_set_uri. It essentially prepares the playbin2 internals for the next uri. all the functions are in [gst-plugins-good]/gst/playback/gstplaybin2.c This is not much, but hopefully it's at least a good starting point.. Regards > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From sebastian.droege at collabora.co.uk Sun Dec 19 10:00:47 2010 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Sun, 19 Dec 2010 10:00:47 +0100 Subject: [gst-devel] gapless playback In-Reply-To: References: Message-ID: <1292749247.15976.233.camel@odin.lan> On Thu, 2010-12-16 at 09:58 +0800, Stuart Gray wrote: > Hi, > > I am seeking some guidance with respect to getting gapless playback > using decodebin2 and queue2. > > Our system comprises the following: > > arm9 processor > gst-plugins-bad - 0.10.20-r11.0.6 > gst-plugins-base - 0.10.31-r11.0.6 > gst-plugins-good - 0.10.25-r11.0.6 > gst-plugins-ugly - 0.10.16-r11.1.6 > running on openembedded linux with kernel version 2.6.33 > > We are decoding audio only streams. This is a concern as reference was > made to gapless playback for the latest releases: > > "Release notes for GStreamer Base Plug-ins?0.10.31 "Dance Like It's > 1982" > > Bugs fixed in this release > > * 628028 : [uridecodebin] Don't enable progressive downloading > for live streams > * 623846 : typefinding: add support for " enhanced ac3 " (eac3) > * 602437 : [playbin2] [gapless] Completely broken when switching > between files with audio/video only > " > > The above refers directly to playbin2, but is there anything related > when using decodebin2 and queue2 as I would assume playbin2 would use > these also. > > At present we have definite breaks with clicks between tracks. The gapless playback is handled inside playbin2/playsink only. You can't do it with only decodebin2 unless you're building the same logic around it. Note however, that short before 0.10.31 release the main fix for bug #602437 was disabled again because it caused problems in other, standard scenarios that worked before. As such gapless playback is still broken in almost all cases (unless you're playing audio-only streams only). As you're decoding audio-only streams you might want to take a look at rhythmbox's gapless playback too. For audio-only streams there's a much easier solution for gapless playback: you would simply write an element that converts the different audio streams that it gets into one, continous stream and then passes it to the sink. Just make sure that you *never* change the sinks (changing the complete pipeline state will do that) state to something below PLAYING, otherwise you'll hear gaps again. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: From gibrovacco at gmail.com Sun Dec 19 10:07:05 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sun, 19 Dec 2010 11:07:05 +0200 Subject: [gst-devel] problem in rtpmpapay and rtpmpadepay In-Reply-To: <1292408680963-3088737.post@n4.nabble.com> References: <1292408680963-3088737.post@n4.nabble.com> Message-ID: Hi, On Wed, Dec 15, 2010 at 12:24 PM, Puneeth wrote: > > Hi All, > > ? ? I m trying to capture audio and encode to mp3 format and send through > rtpmpapay and decode and playback in the other system, but i m not able to > listen voice clearly. But it is good when send captured pcm format and > playback using protocol rtppcmapay and rtppcmadepay. > You'd need some basic sanity check on your system while running the pipeline. Encoding and decoding mp3 requires more computational power than raw pcm: check the CPU usage with "top". If it's anywhere close to 100% you've found a culprit. Possible solutions involve increasing the lamemp3enc output bitrate, enforcing mono encoding, setting "encoding-engine-quality" to "fast", getting a better performing encoder or a system with an higher computational power. If the problem is not CPU load, I suggest you to get some debug with GST_DEBUG=2. Regards > ?Here is the command for the reference, thank u in advance and will be > greatly appreciated for the solution. > > Sender : gst-launch gstrtpbin name=rtpbin alsasrc ! audioconvert ! > audioresample ! lamemp3enc ! rtpmpapay ! rtpbin.send_rtp_sink_0 > rtpbin.send_rtp_src_0 ! udpsink port=5002 host=172.16.10.32 > rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=172.16.10.32 sync=false > async=false udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0 > > Receiver : gst-launch gstrtpbin name=rtpbin udpsrc > caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)MPA" > port=5002 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtpmpadepay ! mp3parse ! > ffdec_mp3 ! audioconvert ! volume volume=3 ! audioresample quality=10 ! > alsasink udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! > udpsink port=5007 host=172.16.10.32 sync=false async=false > > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/problem-in-rtpmpapay-and-rtpmpadepay-tp3088737p3088737.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From gibrovacco at gmail.com Sun Dec 19 16:01:40 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sun, 19 Dec 2010 17:01:40 +0200 Subject: [gst-devel] ways to pause the pipeline, something i am confused. In-Reply-To: References: Message-ID: Hi, late reply and short, but maybe useful.. 2010/12/9 mingqian Han : > Hi all, > ??? Say, I am going to pause the pipeline, in order to do some realtime > change of some elements. I can set the pipeline to PAUSED state to achieve > it, and I can also block some pads to pause the pipeline from playing.? What > is the difference? Or limitations? setting a pipelined to PAUSED state is the official way, blocking pads isn't so, by using the second approach, you're going through a lot of troubles where you may have separate threads (e.g. a tee with a queue). Regards > > Thanks and regards, > > Rafael > > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From rohitratri at gmail.com Sun Dec 19 16:51:51 2010 From: rohitratri at gmail.com (Rohit Atri) Date: Sun, 19 Dec 2010 21:21:51 +0530 Subject: [gst-devel] Selective decoding of video packets In-Reply-To: References: Message-ID: Come on, guys! Someone's got to know where I'm stuck. All you GST gurus out there, please help me out. Thanks On Sat, Dec 18, 2010 at 9:12 PM, Rohit Atri wrote: > Hi, > > I m writing an application which reads mp4 packets from a file (bin1: > filesrc->qtdemux->appsink), do some analysis on a set(say 100 - depending on > the duration of the video) of packets and choose one single packet and send > it for decoding (bin2: appsrc->ffdec_mpeg4->appsink) and then save this > image... repeat the process for all packets in the video. > I've a few questions - > 1. Are the elements I'm using apt? and do I've to do some thread > management since I've to pipelines? > 2. a. I'm able to receive packets from bin1, do some analysis and send a > single packet(iFrame) for decoding.. but bin2 doesn't return the packet > back(the application hangs!).. > b. When I turn on the logs with level-4, I do get the packet back!!! - so > it could be timing issue, I suspect. > c. The bin1 returns say 200 or so packets and the app receives an EOS > message from it! - why is that? > 3. Does appsink return packets as and when it receives them or does it wait > to preroll? > > Please help me understand the situation better. I've hit a roadblock! > Please let me know if you need some logs... > > Thanks > Rohit > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gibrovacco at gmail.com Sun Dec 19 17:56:33 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sun, 19 Dec 2010 18:56:33 +0200 Subject: [gst-devel] [gst-embedded] rtsp client with rtpdec problem In-Reply-To: References: <54DFF4F1A9254246A7328EDAEAD9CDD70F65EE2A97@MSX6.ITRI.DS> Message-ID: Ok, looks like for the n-th time I just replied to the sender instead of the mailing list.. 2010/12/19 Marco Ballesio : > Hi, > > On Sat, Dec 4, 2010 at 10:59 AM, wrote: >> Dear, >> >> I design a rtsp client player using gst-plugin-good, >> >> but I using "rtpdec" replace "rtpbin", >> >> my client could receive "sender report" from server, >> >> but could not send "receive report" to the server. >> >> I try using "rtpbin" , and it could do both receive/send actions. >> >> How could I send "receive report" to the server when I using "rtpdec" ? >> > > similarly to what happens in gstrtpbin, you must use the same value > for when connecting the pads rtcp_src_ and > recv_rtp_src___. This way the gstrtpsession element > within the rtpdec will send RTCP packets (likely one each 5 seconds) > to the udpsink where you connected rtcp_src_. > > Regards > >> regards, >> >> Albert >> >> ???????????????????????????????????????????? >> This email may contain confidential information. Please do not use or >> disclose it in any way and delete it if you are not the intended recipient. >> ------------------------------------------------------------------------------ >> >> _______________________________________________ >> Gstreamer-embedded mailing list >> Gstreamer-embedded at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded >> >> > From gibrovacco at gmail.com Sun Dec 19 18:03:44 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sun, 19 Dec 2010 19:03:44 +0200 Subject: [gst-devel] Distortion when using directsoundsrc In-Reply-To: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1F03@SRV-VS06.TELEVIC.COM> References: <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1CB2@SRV-VS06.TELEVIC.COM> <5D1AAB0BD4000A4982D8E556DD882D45ACEF8F1F03@SRV-VS06.TELEVIC.COM> Message-ID: Hi, On Thu, Dec 9, 2010 at 10:41 AM, Callewaert Sven wrote: > On my Vista computer I hear some distortions on the audio that is captured > from my microphone. The distortion is a periodical noisy crack. Playing > around with the buffer-time parameter changes the number of cracks, but it > doesn?t seem possible to make them go away. I use the ossbuild of gstreamer > version 0.10.7. On my Windows 7 computer there is no problem. This is the > pipeline: > > > > gst-launch directsoundsrc ! directsoundsink > > > > Is there any way to get rid of the distortion? > I've never used GStreamer under Microsoft stuff, btw maybe I could get more help by attaching here the log you get with, let's say, GST_DEBUG=3 (no copy-paste in the email please). Does it get better if you add a queue element bw the source and the sink? Regards > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From braunsquared at gmail.com Sun Dec 19 18:14:20 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Sun, 19 Dec 2010 12:14:20 -0500 Subject: [gst-devel] Selective decoding of video packets In-Reply-To: References: Message-ID: I'm not guru with GST by any means. But some things you may want to look into below: On Sun, Dec 19, 2010 at 10:51 AM, Rohit Atri wrote: > Come on, guys! Someone's got to know where I'm stuck. > All you GST gurus out there, please help me out. > > Thanks > > On Sat, Dec 18, 2010 at 9:12 PM, Rohit Atri wrote: > >> Hi, >> >> I m writing an application which reads mp4 packets from a file (bin1: >> filesrc->qtdemux->appsink), do some analysis on a set(say 100 - depending on >> the duration of the video) of packets and choose one single packet and send >> it for decoding (bin2: appsrc->ffdec_mpeg4->appsink) and then save this >> image... repeat the process for all packets in the video. >> I've a few questions - >> 1. Are the elements I'm using apt? and do I've to do some thread >> management since I've to pipelines? >> > GST does quite a bit of thread management on it's own. It builds the logic into the pipeline using various layers of queues and multiqueues. > 2. a. I'm able to receive packets from bin1, do some analysis and send a >> single packet(iFrame) for decoding.. but bin2 doesn't return the packet >> back(the application hangs!).. >> > Is it possible that the single packet isn't enough to fulfill the segment? Take a look at the debug output of ffdec, is it spitting off buffer underflow warnings? > b. When I turn on the logs with level-4, I do get the packet back!!! - so >> it could be timing issue, I suspect. >> > Not sure where this is coming from, but you may want to throw a queue between your appsrc and ffdec so it would be appsrc->queue->ffdec > c. The bin1 returns say 200 or so packets and the app receives an EOS >> message from it! - why is that? >> > Not sure where the EOS is coming from, but you may want to look into blocking pads to control pipeline flow. Also take a look at the 'eos' property on the appsink element, this may be what's causing your issue, but I'm just guessing here. > 3. Does appsink return packets as and when it receives them or does it wait >> to preroll? >> >> No idea here, sorry. > Please help me understand the situation better. I've hit a roadblock! >> Please let me know if you need some logs... >> >> Thanks >> Rohit >> > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gibrovacco at gmail.com Sun Dec 19 18:14:55 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sun, 19 Dec 2010 19:14:55 +0200 Subject: [gst-devel] examples for demux In-Reply-To: References: Message-ID: Hi, 2010/12/9 mingqian Han : > Hi all, > ??? I have read? the plugin writer's guide, but I still cannot figure out > how to write a demux plugin. Is there some examples to get started? > Sure, there are plenty of them. qtdemux is, for instance, an excellent one: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/qtdemux Regards > Thanks & regards, > > Rafael > > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From gibrovacco at gmail.com Sun Dec 19 18:39:41 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sun, 19 Dec 2010 19:39:41 +0200 Subject: [gst-devel] Splitting Video in a Recording Pipeline In-Reply-To: References: Message-ID: Hi, as all my emails of today, I'm pretty in late.. I hope this helps anyway. On Thu, Dec 9, 2010 at 4:15 PM, LIJIN SYAM K wrote: > Hi > > ? I am,trying to implement a recorder that will split a larger media file or > a live source in to smaller chunks. > First i created a pipeline to record the steam from filesrc and rtspsrc . > For Implementing the file change i put a callback in the encoder element > which keep on checking the size of the file created by the pipeline. The encoder element would not consider the overhead generated by the muxer and by other possible streams (for instance, if you're doing this with the video encoder you would not get the quantity of audio data written). When debugging, I usually install a data probe on the filesrc sink pad in order to get the proper size of the file while writing it, but maybe there are more elegant ways to handle the thing. Be careful about the execution context! http://www.gstreamer.net/data/doc/gstreamer/head/manual/html/chapter-dataaccess.html > When > the filesize is reached the predefined size then i will put the pipeline in > to GST_STATE_PAUSE.?? Then i move the recorded file to the destination > .,after that that the state of the pipeline is set to GST_STATE_NULL. > unref the main pipeline. it's actually possible to re-use the same pipeline (after reconfiguring its properties). > Here the first video is created and after that the > same pipeline is created and the video is recorded . I can Implement file > change in this way..But is this the right way to Implement my idea..? > > Pipeline Created --->> GST_STATE_READY---->> GST_STATE_PAUSE > ---->>GST_STATE_PLAYING-----{ > after a predefined file size splitting file } -->> GST_STATE_PAUSE--->> > GST_STATE_READY-->>GST_STATE_NULL-->>unref the pipeline I wonder if segment events may be of any help here.. Regards > > After this the same method is repeated to create the second chunk. Is there > any better way to Implement idea...? > > > Regards > > LijiN > ------------------------------------------------------------------------------ > This SF Dev2Dev email is sponsored by: > > WikiLeaks The End of the Free Internet > http://p.sf.net/sfu/therealnews-com > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From ensonic at hora-obscura.de Sun Dec 19 19:43:10 2010 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sun, 19 Dec 2010 20:43:10 +0200 Subject: [gst-devel] Selective decoding of video packets In-Reply-To: References: Message-ID: <4D0E523E.8020505@hora-obscura.de> Am 18.12.2010 17:42, schrieb Rohit Atri: > Hi, > > I m writing an application which reads mp4 packets from a file (bin1: > filesrc->qtdemux->appsink), do some analysis on a set(say 100 - depending on the > duration of the video) of packets and choose one single packet and send it for > decoding (bin2: appsrc->ffdec_mpeg4->appsink) and then save this image... repeat > the process for all packets in the video. The approach sounds a bit flawed. You should send the complete stream to the decoder. Videodecoding is rarely stateless (mjpeg would be). If you just want certain frames to be saved in the end. You could e.g. send some custom synchronized downstream event from teh demuxer side. Then based on the event you can handled the decoded frames after the decoder. Stefan > I've a few questions - > 1. Are the elements I'm using apt? and do I've to do some thread management > since I've to pipelines? > 2. a. I'm able to receive packets from bin1, do some analysis and send a single > packet(iFrame) for decoding.. but bin2 doesn't return the packet back(the > application hangs!).. > b. When I turn on the logs with level-4, I do get the packet back!!! - so it > could be timing issue, I suspect. > c. The bin1 returns say 200 or so packets and the app receives an EOS message > from it! - why is that? > 3. Does appsink return packets as and when it receives them or does it wait to > preroll? > > Please help me understand the situation better. I've hit a roadblock! Please let > me know if you need some logs... > > Thanks > Rohit > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > > > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From t.i.m at zen.co.uk Mon Dec 20 10:39:44 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 20 Dec 2010 09:39:44 +0000 Subject: [gst-devel] [gst-cvs] gstreamer: micro-optim: if (x) is cheaper than if (x > 0) for unsigned integers In-Reply-To: <20101203191706.GA15328@cooker.entropywave.com> References: <20101203110404.52B5910057@kemper.freedesktop.org> <1291376839.22392.20.camel@zingle> <1291377573.4291.19441.camel@localhost> <20101203191706.GA15328@cooker.entropywave.com> Message-ID: <1292837984.3331.10.camel@zingle> On Fri, 2010-12-03 at 14:17 -0500, David Schleef wrote: > > I saw gcc asm where it was doing an expensive compare as opposed to a > > fast != 0. So yes, needed. Worth it ? Well I did title it "micro-optim". > > What is an expensive compare? cmp $0, %eax and test %eax, %eax are > both one cycle. It is somewhat amusing that gcc-4.5 generates > different code for these, when it should know they are identical. > gcc-4.4 doesn't, so I consider that a regression, but certainly not > an important one. Edward, in light of David's comments (and your own), can we revert this please? (At the very least I'd like to see the not-quite-right parts like the refcount checks where refcount is a signed int reverted and where it used to warn on refcount <= 0, but now only warns on refcount == 0) Cheers -Tim From osmoma at gmail.com Mon Dec 20 11:43:16 2010 From: osmoma at gmail.com (Osmo Antero Maatta) Date: Mon, 20 Dec 2010 11:43:16 +0100 Subject: [gst-devel] Capturing EOS message from the bus. Message-ID: <4D0F3344.6090706@gmail.com> Hello, I have created an audio-recorder application using the GStreamer framework. My question is: How to catch and react to the EOS (end of stream, end of recording) message? When starting a recording the state change from NULL -> PLAYING works fine. My program now updates the GUI (buttons, icons in the GUI) when the recording changes state from NULL -> READY -> PLAYING. This works OK. Also state change from PLAYING -> PAUSED works fine. My program captures the PAUSED message and updates the GUI. This is OK. But how to catch the STOP message (when the recording stops/ends) ? Obviously Gstreamer has no STOP/STREAM END message. Are there event-chain for PLAYING -> PAUSED -> STOP/NULL? *I want to update the GUI solely via the Gstreamer's state-change messages/events. * Why? Because the recording can be controlled from various places; directly from the GUI using buttons, it can be controlled from DBus by Media Players (like Amarok/RhythmBox/Banshee and there are threads involved etc.). So I want to update the GUI solely via (event's) state changes. Here is how I setup the callback functions to receive events/messages from the bus. See: rec_state_changed_cb(...) function in http://www.futuredesktop.com/tmp/gst-recorder.c The message/event handlers are set up like this (in the rec_create_pipeline(...) function) : // Add a message handlers GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline)); gst_bus_add_signal_watch(bus); // Detect state changes g_signal_connect(bus, "message::state-changed", G_CALLBACK(rec_state_changed_cb), NULL); // Monitor sound level/amplitude g_signal_connect(bus, "message::element", G_CALLBACK(rec_level_message_cb), NULL); // Catch error messages g_signal_connect(bus, "message::error", G_CALLBACK(rec_pipeline_error_cb), NULL); // EOS g_signal_connect(bus, "message::eos", G_CALLBACK(rec_eos_msg_cb), NULL); gst_object_unref(bus); ---- Thanks, Osmo Antero (Moma) http://www.futuredesktop.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From cristianurban86 at gmail.com Mon Dec 20 13:40:13 2010 From: cristianurban86 at gmail.com (cristiurban) Date: Mon, 20 Dec 2010 04:40:13 -0800 (PST) Subject: [gst-devel] Example of Video player In-Reply-To: <1292810894762-3095045.post@n4.nabble.com> References: <1292810894762-3095045.post@n4.nabble.com> Message-ID: <1292848813764-3095442.post@n4.nabble.com> I can use the command below to play video file: gst-launch playbin uri=file:////home/jinkui/Downloads/sample.m4v Try using gst-launch -v playbin uri=file:////home/jinkui/Downloads/sample.m4v to see what elements playbin takes to created the pipeline or try gst-launch filesrc location=sample.m4v ! ffdec_mpeg4 ! ffmpegcolorspace ! ximagesink --gst-debug-level= but the command below would be error: gst-launch filesrc location=sample.m4v ! ffdec_mpeg4 ! ffmpegcolorspace ! ximagesink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/ffdec_mpeg4:ffdec_mpeg40: Internal GStreamer error: negotiation problem. Please file a bug at http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer. Additional debug info: gstffmpegdec.c(2729): gst_ffmpegdec_chain (): /GstPipeline:pipeline0/ffdec_mpeg4:ffdec_mpeg40: ffdec_mpeg4: input format was not set before data start ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... can someone give me a example of video player about m4v files using ffdec_mpeg4? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Example-of-Video-player-tp3095045p3095442.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From florian.iragne at viewsurf.com Mon Dec 20 14:53:40 2010 From: florian.iragne at viewsurf.com (Florian Iragne) Date: Mon, 20 Dec 2010 14:53:40 +0100 Subject: [gst-devel] muxing into mp4 files Message-ID: <4D0F5FE4.5010902@viewsurf.com> Hi, i'm currently using ffmux_ipod to mux a h264 stream into an 'ipod/iphone' compliant mp4 file. It works. However, from the begining, i've noticed some video quality problems (the video seems to "freeze" on one frame every 5 seconds. Since it match the keyframe interval i've defined on the coder, i thought it was an artifact due to the coder. Today, i've tried to mp4mux/qtmux to achieve the same goal (mp4 file ipod/iphone compliant). I'm not able to play these files on iOS device, however the strange video quality problem has disappeared. So few questions : * is there a way to force mp4mux to mux into an ios compliant mp4 file? I've tried lots of parameters tunning without success * since the quality problem is the same using ffmux_mp4 and ffmux_ipod, i think it is related somewhat to ffmpeg. Does anybody have a idea about the problem? I haven't found anything in the bug tracker. thanks for any help Florian From gwunder at eoir.com Mon Dec 20 17:03:56 2010 From: gwunder at eoir.com (Greg Wunder) Date: Mon, 20 Dec 2010 11:03:56 -0500 Subject: [gst-devel] GStreamer Database Integration Message-ID: <451B4D021F581D4AB9B270C067B7062C0AD9EF81@NEOEVS.eoir.com> Are there any examples of GStreamer use with databases of any kind? I'd like to pull data from the GstBus and push it to a database if possible. -------------- next part -------------- An HTML attachment was scrubbed... URL: From nathanael at gnat.ca Mon Dec 20 17:58:43 2010 From: nathanael at gnat.ca (Nathanael D. Noblet) Date: Mon, 20 Dec 2010 09:58:43 -0700 Subject: [gst-devel] muxing into mp4 files In-Reply-To: <4D0F5FE4.5010902@viewsurf.com> References: <4D0F5FE4.5010902@viewsurf.com> Message-ID: <4D0F8B43.9050909@gnat.ca> On 12/20/2010 06:53 AM, Florian Iragne wrote: > Hi, > > i'm currently using ffmux_ipod to mux a h264 stream into an > 'ipod/iphone' compliant mp4 file. > > It works. However, from the begining, i've noticed some video quality > problems (the video seems to "freeze" on one frame every 5 seconds. > Since it match the keyframe interval i've defined on the coder, i > thought it was an artifact due to the coder. > > Today, i've tried to mp4mux/qtmux to achieve the same goal (mp4 file > ipod/iphone compliant). I'm not able to play these files on iOS device, > however the strange video quality problem has disappeared. > > So few questions : > * is there a way to force mp4mux to mux into an ios compliant mp4 > file? I've tried lots of parameters tunning without success > * since the quality problem is the same using ffmux_mp4 and > ffmux_ipod, i think it is related somewhat to ffmpeg. Does anybody have > a idea about the problem? I haven't found anything in the bug tracker. > > thanks for any help I can't really help you directly with the problem, however I would suggest looking at the winFF program. I think it uses ffmpeg and has profiles for these types of conversions. You can either use that to create the ipod converted file, or perhaps seeing the options it is using find where gstreamer / your pipeline have issues. -- Nathanael d. Noblet From osmoma at gmail.com Mon Dec 20 19:21:49 2010 From: osmoma at gmail.com (Osmo Antero Maatta) Date: Mon, 20 Dec 2010 19:21:49 +0100 Subject: [gst-devel] Capturing EOS message from the bus. [SOLVED] In-Reply-To: <4D0F3344.6090706@gmail.com> References: <4D0F3344.6090706@gmail.com> Message-ID: <4D0F9EBD.503@gmail.com> Re-hi, Ok, I found a solution by studying the source of gnome-media-2.30.0/grecord. My code now calls rec_set_state_to_null() function that sets the pipeline to GST_STATE_NULL state. Rec_state_changed_cb(...) function receives now messages so the state goes from PLAYING -> PAUSED -> READY/NULL. And the GUI is updated appropriately. Very good. I can now bake the code to my recorder. void rec_set_state_to_null() { // Copyright notice: // Copied from gnome-media-2.30.0/grecord. // Set state of pipeline to GST_STATE_NULL. GstMessage *msg; GstState cur_state, pending; GstBus *bus; if (!GST_IS_PIPELINE(g_rec.pipeline)) return; gst_element_get_state(g_rec.pipeline, &cur_state, &pending, 0); if (cur_state == GST_STATE_NULL && pending == GST_STATE_VOID_PENDING) return; if (cur_state == GST_STATE_NULL && pending != GST_STATE_VOID_PENDING) { gst_element_set_state (g_rec.pipeline, GST_STATE_NULL); return; } gst_element_set_state(g_rec.pipeline, GST_STATE_READY); gst_element_get_state(g_rec.pipeline, NULL, NULL, -1); bus = gst_element_get_bus(g_rec.pipeline); if (GST_IS_BUS(bus)) { while ((msg = gst_bus_pop(bus))) { gst_bus_async_signal_func(bus, msg, NULL); gst_message_unref(msg); } gst_object_unref(bus); } gst_element_set_state(g_rec.pipeline, GST_STATE_NULL); } Thanks. Osmo Antero (Moma) http://www.futuredesktop.org On 12/20/2010 11:43 AM, Osmo Antero Maatta wrote: > Hello, > I have created an audio-recorder application using the GStreamer > framework. My question is: > How to catch and react to the EOS (end of stream, end of recording) > message? > > When starting a recording the state change from NULL -> PLAYING works > fine. > My program now updates the GUI (buttons, icons in the GUI) when the > recording changes state from NULL -> READY -> PLAYING. This works OK. > > Also state change from PLAYING -> PAUSED works fine. My program > captures the PAUSED message and updates the GUI. This is OK. > > But how to catch the STOP message (when the recording stops/ends) ? > Obviously Gstreamer has no STOP/STREAM END message. Are there > event-chain for PLAYING -> PAUSED -> STOP/NULL? > > *I want to update the GUI solely via the Gstreamer's state-change > messages/events. * > Why? Because the recording can be controlled from various places; > directly from the GUI using buttons, it can be controlled from DBus by > Media Players (like Amarok/RhythmBox/Banshee and there are threads > involved etc.). So I want to update the GUI solely via (event's) state > changes. > > Here is how I setup the callback functions to receive events/messages > from the bus. > See: rec_state_changed_cb(...) function in > http://www.futuredesktop.com/tmp/gst-recorder.c > > The message/event handlers are set up like this (in the > rec_create_pipeline(...) function) : > > // Add a message handlers > GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline)); > gst_bus_add_signal_watch(bus); > > // Detect state changes > g_signal_connect(bus, "message::state-changed", > G_CALLBACK(rec_state_changed_cb), NULL); > > // Monitor sound level/amplitude > g_signal_connect(bus, "message::element", > G_CALLBACK(rec_level_message_cb), NULL); > > // Catch error messages > g_signal_connect(bus, "message::error", > G_CALLBACK(rec_pipeline_error_cb), NULL); > > // EOS > g_signal_connect(bus, "message::eos", G_CALLBACK(rec_eos_msg_cb), NULL); > > gst_object_unref(bus); > ---- > > Thanks, > Osmo Antero (Moma) > http://www.futuredesktop.org > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Mon Dec 20 20:40:00 2010 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 20 Dec 2010 21:40:00 +0200 Subject: [gst-devel] Example of Video player In-Reply-To: <1292848813764-3095442.post@n4.nabble.com> References: <1292810894762-3095045.post@n4.nabble.com> <1292848813764-3095442.post@n4.nabble.com> Message-ID: <4D0FB110.3060501@hora-obscura.de> Am 20.12.2010 14:40, schrieb cristiurban: > > I can use the command below to play video file: > gst-launch playbin uri=file:////home/jinkui/Downloads/sample.m4v > Try using gst-launch -v playbin > uri=file:////home/jinkui/Downloads/sample.m4v > to see what elements playbin takes to created the pipeline > or try gst-launch filesrc location=sample.m4v ! ffdec_mpeg4 ! > ffmpegcolorspace ! ximagesink --gst-debug-level= debugging> > > > > but the command below would be error: > gst-launch filesrc location=sample.m4v ! ffdec_mpeg4 ! ffmpegcolorspace ! > ximagesink Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > ERROR: from element /GstPipeline:pipeline0/ffdec_mpeg4:ffdec_mpeg40: > Internal GStreamer error: negotiation problem. Please file a bug at > http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer. > Additional debug info: > gstffmpegdec.c(2729): gst_ffmpegdec_chain (): > /GstPipeline:pipeline0/ffdec_mpeg4:ffdec_mpeg40: > ffdec_mpeg4: input format was not set before data start > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > Freeing pipeline ... gst-launch filesrc location=sample.m4v ! mpeg4videoparse ! ffdec_mpeg4 ! ffmpegcolorspace ! ximagesink you need to use a container (demuxer) or a parser. Stefan > > > can someone give me a example of video player about m4v files using > ffdec_mpeg4? From radivojejovanovic at gmail.com Mon Dec 20 22:32:00 2010 From: radivojejovanovic at gmail.com (Radivoje Jovanovic) Date: Mon, 20 Dec 2010 13:32:00 -0800 Subject: [gst-devel] problem with audio over gstreamer? Message-ID: Hello, I have ALSA driver that is working just fine on the system. If I use aplay WAV_FILE it works just fine. If I play the same file through gstreamer like: gst-launch filesrc location=WAV_FILE ! wavparse ! alsasink it works just fine, but if I play movie on my board the sound is glitching. So I try using just audio part of the pipeline: gst-launch filesrc location=MY_MPEG4_MOVIE ! qtdemux name=t ! queue ! faad ! alsasink the audio is glitchy. Any idea why? If not, is there any other AAC decoder I might try? Cheers Ogi -------------- next part -------------- An HTML attachment was scrubbed... URL: From shamun.toha at gmail.com Mon Dec 20 22:46:53 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Mon, 20 Dec 2010 22:46:53 +0100 Subject: [gst-devel] Gstreamer is impossible to use with web browser. Message-ID: I dont know how many of you tried like this (non stop 1 week testing and testing till midnight). But i did really a big try till now 20, December 2010, and i gave up, because i want to have a answer either a big yes or no, finally came to a answer which is "NO". GStreamer using java applet is impossible (do not ever try and spend your time): - I wrote very simple source src and sink application in java applet and i used gstreamer wrapper - I converted it as java applet - I signed it with my details (completely signed), which works in my Linux box having gstreamer pre installed by simply doing "java -jar myapplet.jar" - I played it in Linux all browsers (including firefox/konquer/chrome/opera) result does not work, even i have signed applet + java all updates - I installed Virtual OS Windows XP/Vista/7 and downloaded same .jar files and gstreamer in Windows machine, gave try with (IE6/ IE7/ IE8/ IE9 and Firefox/ Chrome/ Opera/ Sea monkey/ Safari/ Flock/ Avant) DOES not work What i can do to solve this ? I must install gstreamer (i know then it works) ? But why i cant embed gstreamer with my applet web install/web start ..etc etc??? Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From florian.iragne at viewsurf.com Mon Dec 20 22:38:56 2010 From: florian.iragne at viewsurf.com (Florian Iragne) Date: Mon, 20 Dec 2010 22:38:56 +0100 Subject: [gst-devel] muxing into mp4 files In-Reply-To: <4D0F8B43.9050909@gnat.ca> References: <4D0F5FE4.5010902@viewsurf.com> <4D0F8B43.9050909@gnat.ca> Message-ID: <4D0FCCF0.8080404@viewsurf.com> Thanks for your answer, i know where are the issues : on one side, ffmux_ipod gives me a mp4 file iOS compliant, but visualy not perfect. On the other side, mp4mux gives an mp4 file visualy "almost" perfect, but not playable with iOS devices. so the problem in my gstreamer pipeline is clearly the muxer element. However, after having search through all the pipeline elements properties, i haven't found any that has an effect on either the quality of the output or its compliance to iOS limits (and yes, the files have to be playable under iOS, unfortunately). here are the elements of my pipeline : rtspsrc ! rtph264depay byte-stream=false ! capsfilter ! mp4mux (or ffmux_ipod) ! filesink thanks for youur help Florian From wmiller at sdr.com Mon Dec 20 22:57:59 2010 From: wmiller at sdr.com (Wes Miller) Date: Mon, 20 Dec 2010 13:57:59 -0800 (PST) Subject: [gst-devel] get GstElement from param Message-ID: <1292882279201-3097713.post@n4.nabble.com> Slightly out of context here, but... I know it's out there somewhere, but, among other problems, most of the examples on the web are "temporarly unavailable", and there doesn't seem to be a g_param_spec_get_object() function out there. After enumerating the properties of a gst element, I get down to the point where testing the params with G_IS_PARAM_SPEC_OBJECT( param ) is true. I'd like to know more about the object that is alluded to here. How do I get back a GstElement* for the element that is the "object of type GstElement"? Also, what is the correct forum for g_whatever() discussions? Thanks and Happy Holidays, Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/get-GstElement-from-param-tp3097713p3097713.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Mon Dec 20 23:32:57 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 20 Dec 2010 22:32:57 +0000 Subject: [gst-devel] get GstElement from param In-Reply-To: <1292882279201-3097713.post@n4.nabble.com> References: <1292882279201-3097713.post@n4.nabble.com> Message-ID: <1292884377.11020.1.camel@zingle> On Mon, 2010-12-20 at 13:57 -0800, Wes Miller wrote: > I know it's out there somewhere, but, among other problems, most of the > examples on the web are "temporarly unavailable", and there doesn't seem to > be a g_param_spec_get_object() function out there. > > After enumerating the properties of a gst element, I get down to the point > where testing the params with G_IS_PARAM_SPEC_OBJECT( param ) is true. I'd > like to know more about the object that is alluded to here. How do I get > back a GstElement* for the element that is the "object of type GstElement"? Use the source? http://cgit.freedesktop.org/gstreamer/gstreamer/tree/tools/gst-inspect.c#n574 > Also, what is the correct forum for g_whatever() discussions? http://mail.gnome.org/mailman/listinfo/gtk-app-devel-list Cheers -Tim From wmiller at sdr.com Mon Dec 20 23:48:06 2010 From: wmiller at sdr.com (Wes Miller) Date: Mon, 20 Dec 2010 14:48:06 -0800 (PST) Subject: [gst-devel] get GstElement from param In-Reply-To: <1292884377.11020.1.camel@zingle> References: <1292882279201-3097713.post@n4.nabble.com> <1292884377.11020.1.camel@zingle> Message-ID: <1292885286785-3104789.post@n4.nabble.com> Tim, Thanks for the quick reply. That is the code I was looking at. Rather than knowing the type_name, I want the GstElement* so I can run it back through this same function to get information about THAT element. Goal is to enumerate the properties of, say, playbin, and to find that property 'audio-sink" is a "reference" to an object of type GstElement then to get the audio-sink's GstElement pointer and explore it's properties to find out that the, say, pulsesink has it's sync property set to false. Alice is in wonderland, walking backwards speaking APL. Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/get-GstElement-from-param-tp3097713p3104789.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Tue Dec 21 00:04:46 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 20 Dec 2010 23:04:46 +0000 Subject: [gst-devel] get GstElement from param In-Reply-To: <1292885286785-3104789.post@n4.nabble.com> References: <1292882279201-3097713.post@n4.nabble.com> <1292884377.11020.1.camel@zingle> <1292885286785-3104789.post@n4.nabble.com> Message-ID: <1292886286.11020.5.camel@zingle> On Mon, 2010-12-20 at 14:48 -0800, Wes Miller wrote: > That is the code I was looking at. Rather than knowing the type_name, I > want the GstElement* so I can run it back through this same function to get > information about THAT element. > > Goal is to enumerate the properties of, say, playbin, and to find that > property 'audio-sink" is a "reference" to an object of type GstElement then > to get the audio-sink's GstElement pointer and explore it's properties to > find out that the, say, pulsesink has it's sync property set to false. > > Alice is in wonderland, walking backwards speaking APL. GstElement *element = NULL; g_object_get (playbin, param->name, &element, NULL); ... gst_object_unref (element); (or go via the GValue using g_object_get_property() and then g_value_get_object() as the gst-inspect code does) -Tim From fthiery at gmail.com Tue Dec 21 00:45:59 2010 From: fthiery at gmail.com (Florent) Date: Tue, 21 Dec 2010 00:45:59 +0100 Subject: [gst-devel] GStreamer Conference 2010: videos of talks from the Darwin room In-Reply-To: References: <20101215195407.GA31650@cooker.entropywave.com> Message-ID: Hi Do we have any idea of when they will go up? It's coming up to 2 > months since the conference now. > I'll ask again Florent -------------- next part -------------- An HTML attachment was scrubbed... URL: From marc.leeman at gmail.com Tue Dec 21 01:45:28 2010 From: marc.leeman at gmail.com (Marc Leeman) Date: Tue, 21 Dec 2010 01:45:28 +0100 Subject: [gst-devel] Gstreamer is impossible to use with web browser. In-Reply-To: References: Message-ID: <20101221004528.GO711@crichton.homelinux.org> > - I installed Virtual OS Windows XP/Vista/7 and downloaded same .jar files > and gstreamer in Windows machine, gave try with > (IE6/ IE7/ IE8/ IE9 and Firefox/ Chrome/ Opera/ Sea monkey/ Safari/ Flock/ > Avant) > > DOES not work Odd, we got an html 5/webkit based setup working quite easily on squeeze. We probably must have used something else without realising it if it does not work. -- greetz, marc Human beings were created by water to transport it uphill. crichton 2.6.26 #1 PREEMPT Tue Jul 29 21:17:59 CDT 2008 GNU/Linux -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Digital signature URL: From wmiller at sdr.com Tue Dec 21 02:18:39 2010 From: wmiller at sdr.com (Wes Miller) Date: Mon, 20 Dec 2010 17:18:39 -0800 (PST) Subject: [gst-devel] get GstElement from param In-Reply-To: <1292886286.11020.5.camel@zingle> References: <1292882279201-3097713.post@n4.nabble.com> <1292884377.11020.1.camel@zingle> <1292885286785-3104789.post@n4.nabble.com> <1292886286.11020.5.camel@zingle> Message-ID: <1292894319170-3113780.post@n4.nabble.com> Thanks, Tim. that did the trick. That and a bunch of relearning to use STL lists. Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/get-GstElement-from-param-tp3097713p3113780.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From shiva.varma at gmail.com Tue Dec 21 02:24:50 2010 From: shiva.varma at gmail.com (shiva varma) Date: Mon, 20 Dec 2010 17:24:50 -0800 (PST) Subject: [gst-devel] pcapparse seems to work, the output from pcapparse to rtpmp4gdepay has a problem Message-ID: <1292894690583-3114174.post@n4.nabble.com> Hi All. Would greatly appreciate any help regarding the problem I have been facing wirh rtpmp4gdepay I have a pcap file, which contains AAC-LD stream, with the following properties Bitrate: 64 kbps/channel RTP Payload: IETF RFC 3640, AAC-hbr mode Default Dynamic Payload Number: 96 When I pipe the pcap file to a filesink using pcapparse I can see that the file has some size indicating the rtp stream was written to it, following is the pipeline gst-launch filesrc location=audioCaptureFiltered.pcap ! pcapparse src-ip=10.22.156.148 ! "application/x-rtp, payload=96, media=audio, clock-rate=64000, encoding-name=MPEG4-GENERIC, mode=AAC-hbr, streamtype=(string)4" ! filesink location=abc.out However if I add rtpmp4gdepay to the pcapparse output, I observe that the size of the file is zero.... I do not get any errors, even when I set gst-debug-level to 5.. Here is the command output gst-launch filesrc location=audioCaptureFiltered.pcap ! pcapparse src-ip=10.22.156.148 src-port=16464 ! "application/x-rtp, payload=96, media=audio, clock-rate=64000, encoding-name=MPEG4-GENERIC, mode=AAC-hbr, streamtype=(string)4" ! rtpmp4gdepay ! filesink location=abc.out Setting pipeline to PAUSED ... Pipeline is PREROLLING ... Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock Got EOS from element "pipeline0". Execution ended after 151000 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... Is there something I am missing, please let me know Thanks Shiva -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/pcapparse-seems-to-work-the-output-from-pcapparse-to-rtpmp4gdepay-has-a-problem-tp3114174p3114174.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From dingchengus1 at yahoo.com Tue Dec 21 03:16:48 2010 From: dingchengus1 at yahoo.com (ding cheng) Date: Mon, 20 Dec 2010 18:16:48 -0800 (PST) Subject: [gst-devel] webcam rtsp server supports only one session Message-ID: <959999.34052.qm@web120012.mail.ne1.yahoo.com> I modify test-mp4.c?launch string to use v4l2src as stream source: ? "v4l2src device=/dev/video0 ! ffmpegcolorspace !? videoscale ! ffenc_mpeg4 ! rtpmp4vpay pt=96 name=pay0". ? It?can only support one session. But the origional test-mp4.c supports multiple sessions without problem. ? Can somebody give me a clue? Thanks,? ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Tue Dec 21 09:20:15 2010 From: mcygogo at gmail.com (michael) Date: Tue, 21 Dec 2010 16:20:15 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? Message-ID: Hi all: I want to record the video from my book's camera, and is there one pipeline can do this? tks :) -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Tue Dec 21 09:24:04 2010 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 21 Dec 2010 09:24:04 +0100 Subject: [gst-devel] Gstreamer is impossible to use with web browser. In-Reply-To: References: Message-ID: <1292919844.2596.11.camel@deumeu> On Mon, 2010-12-20 at 22:46 +0100, Shamun toha md wrote: > I dont know how many of you tried like this (non stop 1 week testing > and testing till midnight). But i did really a big try till now 20, > December 2010, and i gave up, because i want to have a answer either a > big yes or no, finally came to a answer which is "NO". > > What i can do to solve this ? I must install gstreamer (i know then it > works) ? But why i cant embed gstreamer with my applet web install/web > start ..etc etc??? You did embed both gstreamer AND the plugins you needed in that applet, right ? I don't know much about the applet side of java, but a few basic questions come to mind: * you'll need JNI bindings/wrappers. Can one distribute an applet with JNI bindings? * How do you build 'applets' that depend on shared libraries ? * If you need some plugins that depend on other libraries, have you also embedded those ? * GStreamer is compiled for one arch at a time, aren't applets meant to be arch-/platform-agnostic ? Have you tried a minimalistic (gst_init()) version ? What errors (like debugging statements or java console errors) do you see ? Does your applet find the gstreamer symbols alright ? Take a few steps back, try to answer some fundamental questions like those above and maybe the 'real' problem will show up (for which somebody might be able to provide more pointers, or better yet, *you* might be able to find answers on the internet). Edward > > > > > Thanks & Regards > > > > > > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From wim.taymans at gmail.com Tue Dec 21 09:24:17 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 21 Dec 2010 09:24:17 +0100 Subject: [gst-devel] webcam rtsp server supports only one session In-Reply-To: <959999.34052.qm@web120012.mail.ne1.yahoo.com> References: <959999.34052.qm@web120012.mail.ne1.yahoo.com> Message-ID: <1292919857.17279.57.camel@meany> Check out the 'shared' property on the media factory, it allows you to share 1 instance of a pipeline with multiple clients. Wim On Mon, 2010-12-20 at 18:16 -0800, ding cheng wrote: > I modify test-mp4.c launch string to use v4l2src as stream source: > > "v4l2src device=/dev/video0 ! ffmpegcolorspace ! videoscale ! > ffenc_mpeg4 ! rtpmp4vpay pt=96 name=pay0". > > It can only support one session. But the origional test-mp4.c supports > multiple sessions without problem. > Can somebody give me a clue? Thanks, > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From mcygogo at gmail.com Tue Dec 21 09:58:08 2010 From: mcygogo at gmail.com (michael) Date: Tue, 21 Dec 2010 16:58:08 +0800 Subject: [gst-devel] webcam rtsp server supports only one session In-Reply-To: <1292919857.17279.57.camel@meany> References: <959999.34052.qm@web120012.mail.ne1.yahoo.com> <1292919857.17279.57.camel@meany> Message-ID: where is your test-mp4.c ? On Tue, Dec 21, 2010 at 4:24 PM, Wim Taymans wrote: > Check out the 'shared' property on the media factory, it allows you to > share 1 instance of a pipeline with multiple clients. > > Wim > > On Mon, 2010-12-20 at 18:16 -0800, ding cheng wrote: > > I modify test-mp4.c launch string to use v4l2src as stream source: > > > > "v4l2src device=/dev/video0 ! ffmpegcolorspace ! videoscale ! > > ffenc_mpeg4 ! rtpmp4vpay pt=96 name=pay0". > > > > It can only support one session. But the origional test-mp4.c supports > > multiple sessions without problem. > > Can somebody give me a clue? Thanks, > > > > > > > > > ------------------------------------------------------------------------------ > > Lotusphere 2011 > > Register now for Lotusphere 2011 and learn how > > to connect the dots, take your collaborative environment > > to the next level, and enter the era of Social Business. > > http://p.sf.net/sfu/lotusphere-d2d > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From lazyt at gmx.net Tue Dec 21 10:45:34 2010 From: lazyt at gmx.net (Thomas Loewe) Date: Tue, 21 Dec 2010 01:45:34 -0800 (PST) Subject: [gst-devel] [souphttpsrc] pausing after gst_base_src_get_range() = unexpected Message-ID: <1292924734822-3138390.post@n4.nabble.com> Hi, for some mpeg2/h264 transportstreams i get the obove message and streaming stops after 2 seconds. When I set the multiqueue option "max-size-time=5000000000" it stops after 5 seconds and so on. I use this pipeline in my player: |-> >-[ multiqueue ]-> >-[ mpeg2dec/ffdec_h264 ]-> >-[ deinterlace ]-> >-[ vid:dvbsuboverlay ]-> >-[ clockoverlay ]-> >-[ textoverlay ]-> >-[ videobalance ]-> >-[ videoscale ]-> >-[ queue ]-> >-[ autovideosink ] | |-> >-[ multiqueue ]-> >-[ mad/a52dec ]-> >-[ audioconvert ]-> >-[ audioresample ]-> >-| |-> >-[ queue ]-> >-[ mpeg2tsdemux ]-> >-[ multiqueue ]-> >-[ mad/a52dec ]-> >-[ audioconvert ]-> >-[ audioresample ]-> >-[ input-selector ]-> >-[ volume ]-> >-[ equalizer ]-> >-[ queue ]-> >-[ autoaudiosink ] | |-> >-[ multiqueue ]-> >-[ mad/a52dec ]-> >-[ audioconvert ]-> >-[ audioresample ]-> >-| | | [ souphttpsrc ]-> >-[ identity ]-> >-[ tee ] |-> >-[ multiqueue ]-> >-[ appsink ] | | | |-> >-[ multiqueue ]-> >-[ sub:dvbsuboverlay ] | |-> >-[ queue ]-> >-[ valve ]-> >-[ filesink ] The pipeline is dynamicly build via pad-added signals. All added elements are set to GST_STATE_PAUSED and after the pads-done signal (emitted by me because the demuxer doesn't send it himself) the pipeline ist set to GST_STATE_PLAYING. This works fine for most of the streams, but for some it simply stucks after the buffer time and the tee element sends "gst_pad_event_default_dispatch: Sending event 08468B40 (eos) to all internally linked pads" right after the "pausing after gst_base_src_get_range() = unexpected" message. Any hint what could cause this problem? Thank you, Thomas -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/souphttpsrc-pausing-after-gst-base-src-get-range-unexpected-tp3138390p3138390.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From malloblenne at gmail.com Tue Dec 21 11:16:21 2010 From: malloblenne at gmail.com (Mauro Brenna) Date: Tue, 21 Dec 2010 11:16:21 +0100 Subject: [gst-devel] RTP+MJPEG - Sniffing packets with wireshark, How clients join the rtp server Message-ID: Hello, I am studying the RTP protocol in order to implement a server for RTP with MJPEG payload using RFC 2435. I read the RFC and I tried to stream and receive a video in rtp using gstreamer. I read about : http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-rtpjpegpay.html which is exactly the implementation of the RFC I read. I tested the streaming and reception of two camera outputs using a pipeline which I have found in the Internet and which seems to work: #Streaming rtp mjpeg gst-launch-0.10 -v gstrtpbin name=rtpbin v4l2src device=/dev/video0 ! video/x-raw-rgb,width=640,height=480,framerate=\(fraction\)30/1 ! ffmpegcolorspace ! jpegenc ! rtpjpegpay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! multiudpsink clients="127.0.0.1:9996" rtpbin.send_rtcp_src_0 ! multiudpsink clients="127.0.0.1:9997" sync=false async=false udpsrc port=10000 ! rtpbin.recv_rtcp_sink_0 v4l2src device=/dev/video1 ! video/x-raw-rgb,width=640,height=480,framerate=\(fraction\)30/1 ! ffmpegcolorspace ! jpegenc ! rtpjpegpay ! rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! multiudpsink clients="127.0.0.1:9998" rtpbin.send_rtcp_src_1 ! multiudpsink clients="127.0.0.1:9999" sync=false async=false udpsrc port=10001 ! rtpbin.recv_rtcp_sink_1 #Playing rtp mjpeg gst-launch-0.10 -v gstrtpbin name=rtpbin udpsrc caps="application/x-rtp, media=\(string\)video, clock-rate=\(int\)90000, encoding-name=\(string\)JPEG, ssrc=\(guint\)469657143, payload=\(int\)96, clock-base=\(guint\)2841649723, seqnum-base=\(guint\)39869" port=9996 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpjpegdepay ! jpegdec ! videorate ! video/x-raw-yuv ,framerate=30/1 ! ffmpegcolorspace ! xvimagesink udpsrc port=9997 ! rtpbin.recv_rtcp_sink_1 rtpbin.send_rtcp_src_1 ! multiudpsink clients="127.0.0.1:10001" sync=false async=false & gst-launch-0.10 -v gstrtpbin name=rtpbin udpsrc caps="application/x-rtp, media=\(string\)video, clock-rate=\(int\)90000, encoding-name=\(string\)JPEG, ssrc=\(guint\)469657143, payload=\(int\)96, clock-base=\(guint\)2841649723, seqnum-base=\(guint\)39869" port=9998 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpjpegdepay ! jpegdec ! videorate ! video/x-raw-yuv ,framerate=30/1 ! ffmpegcolorspace ! xvimagesink udpsrc port=9999 ! rtpbin.recv_rtcp_sink_1 rtpbin.send_rtcp_src_1 ! multiudpsink clients="127.0.0.1:10001" sync=false async=false Everything works but I would like to ask to this mailinglist a couple of questions about the implementation and RTP itself: 1) I tried to sniff the packets using wireshark. I can see the packets that are sent using UDP but not the RTP encapsulation (I see only the field data of the UDP packet). Is is a problem of wireshark for RTP (RFC 2435) or the gstreamer plug-in is not compliant to the standard? I tested the same thing in VLC that streams RTP packets using RFC 2250. In wireshark I can see the RTP packet with the correct payload type. 2) I do not understand very well how the client (player) join the RTP server (streamer). I read the general RFC 1889. In the section about RTP Control Protocol, it is not clear how the initialization works. I mean, when I start the streamer, does it start to send RTP+MJPEG packets to the unicast/multicast destination even if nobody is listening to it? Or does it start sending packets after the reception of a particular packet from the client ? I read about feedbacks packets from sender/receiver and about the BYE (Goodbye RTCP packet). However, I have not found a sort of JOIN packet. Thanks in advance for the replies, Mauro From shamun.toha at gmail.com Tue Dec 21 11:26:20 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Tue, 21 Dec 2010 11:26:20 +0100 Subject: [gst-devel] Linux to Windows Message-ID: Because Gstreamer features are more under Linux platform. But same package if required to port for Windows, seems a hell is just sitting out there. If we try with Winbuilds, it does not fit all the features that is available for linux distros. Question: 1) Is it not possible install Cygwin/Mingw in Windows XP/Vista/7 2) Put the GStreamer source and compile there 3) Use Cygwin way of GStreamer Impossible, anyone ever tried ? Can then JAVA wrapper will work ? Can JAVA Applet will work ? Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From wl2776 at gmail.com Tue Dec 21 13:11:27 2010 From: wl2776 at gmail.com (wl2776) Date: Tue, 21 Dec 2010 04:11:27 -0800 (PST) Subject: [gst-devel] Linux to Windows In-Reply-To: References: Message-ID: <1292933487616-3147100.post@n4.nabble.com> Everything is possible. Try OSSBuild. http://code.google.com/p/ossbuild/ Shamun toha md wrote: > > Because Gstreamer features are more under Linux platform. But same package > if required to port for Windows, seems a hell is just sitting out there. > If we try with Winbuilds, it does not fit all the features that is > available > for linux distros. > > Question: > 1) Is it not possible install Cygwin/Mingw in Windows XP/Vista/7 > > 2) Put the GStreamer source and compile there > > 3) Use Cygwin way of GStreamer > > Impossible, anyone ever tried ? Can then JAVA wrapper will work ? Can JAVA > Applet will work ? > > > Thanks & Regards > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Linux-to-Windows-tp3140849p3147100.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From cristianurban86 at gmail.com Tue Dec 21 13:47:13 2010 From: cristianurban86 at gmail.com (cristiurban) Date: Tue, 21 Dec 2010 04:47:13 -0800 (PST) Subject: [gst-devel] ffmpegcolorspace warning Message-ID: <1292935633394-3148890.post@n4.nabble.com> I have a application using phonon (gstreamer as engine) and when I am trying to play a video file I get this warning and no stream is played : ** (:1393): WARNING **: ffmpegcsp1: size 28 is not a multiple of unit size 460800 Something to de with ffmpegcolorspace -> http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-ffmpegcolorspace.html Does anyone have an ideea where to look, or give me a hint of what I am missing? Note: if I do a playbin pipeline it works just fine playing the video, so the problem is at phonon level. Thanks. Cristi. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/ffmpegcolorspace-warning-tp3148890p3148890.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From sebastian.droege at collabora.co.uk Tue Dec 21 14:45:54 2010 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Tue, 21 Dec 2010 14:45:54 +0100 Subject: [gst-devel] ffmpegcolorspace warning In-Reply-To: <1292935633394-3148890.post@n4.nabble.com> References: <1292935633394-3148890.post@n4.nabble.com> Message-ID: <1292939154.15976.242.camel@odin.lan> On Tue, 2010-12-21 at 04:47 -0800, cristiurban wrote: > I have a application using phonon (gstreamer as engine) and when I am trying > to play a video file I get this warning and no stream is played : > ** (:1393): WARNING **: ffmpegcsp1: size 28 is not a multiple of > unit size 460800 > Something to de with ffmpegcolorspace -> > http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-ffmpegcolorspace.html > Does anyone have an ideea where to look, or give me a hint of what I am > missing? > Note: if I do a playbin pipeline it works just fine playing the video, so > the problem is at phonon level. This is probably a problem with phonon then, something is passing buffers to ffmpegcolorspace that are not a complete frame. Please file a bugreport against phonon -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: From timothejahan at gmail.com Tue Dec 21 15:00:57 2010 From: timothejahan at gmail.com (timothe jahan) Date: Tue, 21 Dec 2010 15:00:57 +0100 Subject: [gst-devel] Synchronising live audio from 2 different capture card Message-ID: Dears, I am thinking of setting up a multichannel live streaming solution using gstreamer. Up to now my biggest unknown is the possibility to adjust gstreamer plugin parameters on-the-fly, and especially one to control the pitch of an audio resampler. I would need to change it either manually (myself during the encoding) or via smart algorythm. Does anyone woule have already done something like that ? To understand the background: I want to synchronise the audio from a multichannel audio card to the one of a video capture card to eventually get the equivalent of a very expensive video+multichannel audio card. The idea would be to compare a reference audio channel on both card and based on the result to speed up (a little bit !!) or slow down the channels from the audio only card. Many thanks, Timothe -------------- next part -------------- An HTML attachment was scrubbed... URL: From dingchengus1 at yahoo.com Tue Dec 21 15:48:27 2010 From: dingchengus1 at yahoo.com (ding cheng) Date: Tue, 21 Dec 2010 06:48:27 -0800 (PST) Subject: [gst-devel] webcam rtsp server supports only one session In-Reply-To: Message-ID: <427513.47540.qm@web120017.mail.ne1.yahoo.com> It is one of test programs in example directory. You can use test-mp4 to stream a mp4 file. --- On Tue, 12/21/10, michael wrote: From: michael Subject: Re: [gst-devel] webcam rtsp server supports only one session To: "Discussion of the development of GStreamer" Date: Tuesday, December 21, 2010, 12:58 AM where is your test-mp4.c ? On Tue, Dec 21, 2010 at 4:24 PM, Wim Taymans wrote: Check out the 'shared' property on the media factory, it allows you to share 1 instance of a pipeline with multiple clients. Wim On Mon, 2010-12-20 at 18:16 -0800, ding cheng wrote: > I modify test-mp4.c launch string to use v4l2src as stream source: > > "v4l2src device=/dev/video0 ! ffmpegcolorspace ! ?videoscale ! > ffenc_mpeg4 ! rtpmp4vpay pt=96 name=pay0". > > It can only support one session. But the origional test-mp4.c supports > multiple sessions without problem. > Can somebody give me a clue? Thanks, > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Lotusphere 2011 Register now for Lotusphere 2011 and learn how to connect the dots, take your collaborative environment to the next level, and enter the era of Social Business. http://p.sf.net/sfu/lotusphere-d2d _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -----Inline Attachment Follows----- ------------------------------------------------------------------------------ Lotusphere 2011 Register now for Lotusphere 2011 and learn how to connect the dots, take your collaborative environment to the next level, and enter the era of Social Business. http://p.sf.net/sfu/lotusphere-d2d -----Inline Attachment Follows----- _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From rohitratri at gmail.com Tue Dec 21 16:14:04 2010 From: rohitratri at gmail.com (Rohit Atri) Date: Tue, 21 Dec 2010 20:44:04 +0530 Subject: [gst-devel] YUV to RGB conversion of decoder output Message-ID: Hi, I've an application which selectively decodes iFrames in a video (mpeg4 only for now) and hence I use ffdec_mpeg4 in the pipeline. The output I get is in YUV format and I need to convert it to RGB so I can build a QPixmap (Qt image representation class) from it. How do I do that? I tried - setting the caps as "video/x-raw-rgb" for the src pad of decoder but that didn't work. Also I tried putting a capsfilter with caps = "video/x-raw-rgb" and followed by a ffmpegcolorspace element but that didn't quite work. I don't want to set other structures such as width height depth etc since that could vary from video to video. PS - I could just write my own yuv->rgb routine but want to check if this can be done in the pipeline itself. Thanks Rohit -------------- next part -------------- An HTML attachment was scrubbed... URL: From lijinsyam at gmail.com Tue Dec 21 16:37:57 2010 From: lijinsyam at gmail.com (liJin) Date: Tue, 21 Dec 2010 21:07:57 +0530 Subject: [gst-devel] webcam rtsp server supports only one session In-Reply-To: <427513.47540.qm@web120017.mail.ne1.yahoo.com> References: <427513.47540.qm@web120017.mail.ne1.yahoo.com> Message-ID: Hi.. For sharing media factory = gst_rtsp_media_factory_new (); * gst_rtsp_media_factory_set_shared(factory,TRUE);* gst_rtsp_media_factory_set_launch (factory, str); I think this will help.... On Tue, Dec 21, 2010 at 8:18 PM, ding cheng wrote: > It is one of test programs in example directory. You can use test-mp4 to > stream a mp4 file. > > --- On *Tue, 12/21/10, michael * wrote: > > > From: michael > Subject: Re: [gst-devel] webcam rtsp server supports only one session > To: "Discussion of the development of GStreamer" < > gstreamer-devel at lists.sourceforge.net> > Date: Tuesday, December 21, 2010, 12:58 AM > > > where is your test-mp4.c ? > > On Tue, Dec 21, 2010 at 4:24 PM, Wim Taymans > > wrote: > > Check out the 'shared' property on the media factory, it allows you to > share 1 instance of a pipeline with multiple clients. > > Wim > > On Mon, 2010-12-20 at 18:16 -0800, ding cheng wrote: > > I modify test-mp4.c launch string to use v4l2src as stream source: > > > > "v4l2src device=/dev/video0 ! ffmpegcolorspace ! videoscale ! > > ffenc_mpeg4 ! rtpmp4vpay pt=96 name=pay0". > > > > It can only support one session. But the origional test-mp4.c supports > > multiple sessions without problem. > > Can somebody give me a clue? Thanks, > > > > > > > > > ------------------------------------------------------------------------------ > > Lotusphere 2011 > > Register now for Lotusphere 2011 and learn how > > to connect the dots, take your collaborative environment > > to the next level, and enter the era of Social Business. > > http://p.sf.net/sfu/lotusphere-d2d > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > -----Inline Attachment Follows----- > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > > -----Inline Attachment Follows----- > > _______________________________________________ > > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Tue Dec 21 16:46:17 2010 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 21 Dec 2010 16:46:17 +0100 Subject: [gst-devel] YUV to RGB conversion of decoder output In-Reply-To: References: Message-ID: <1292946377.30176.12.camel@localhost> On Tue, 2010-12-21 at 20:44 +0530, Rohit Atri wrote: > Hi, > > I've an application which selectively decodes iFrames in a video > (mpeg4 only for now) and hence I use ffdec_mpeg4 in the pipeline. The > output I get is in YUV format and I need to convert it to RGB so I can > build a QPixmap (Qt image representation class) from it. > > > How do I do that? > > > I tried - setting the caps as "video/x-raw-rgb" for the src pad of > decoder but that didn't work. Also I tried putting a capsfilter with > caps = "video/x-raw-rgb" and followed by a ffmpegcolorspace element > but that didn't quite work. You need to put your capsfilter *after* ffmpegcolorspace (you want it to output RGB regardless of what it has in input). Edward > I don't want to set other structures such as width height depth etc > since that could vary from video to video. > > > PS - I could just write my own yuv->rgb routine but want to check if > this can be done in the pipeline itself. > > > Thanks > Rohit > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From rohitratri at gmail.com Tue Dec 21 16:53:01 2010 From: rohitratri at gmail.com (Rohit Atri) Date: Tue, 21 Dec 2010 21:23:01 +0530 Subject: [gst-devel] YUV to RGB conversion of decoder output In-Reply-To: <1292946377.30176.12.camel@localhost> References: <1292946377.30176.12.camel@localhost> Message-ID: Oh god! I kinda feel embarrassed now :) But the embarrassment was worth the lesson I learnt! Thanks Edward, that worked. On Tue, Dec 21, 2010 at 9:16 PM, Edward Hervey wrote: > On Tue, 2010-12-21 at 20:44 +0530, Rohit Atri wrote: > > Hi, > > > > I've an application which selectively decodes iFrames in a video > > (mpeg4 only for now) and hence I use ffdec_mpeg4 in the pipeline. The > > output I get is in YUV format and I need to convert it to RGB so I can > > build a QPixmap (Qt image representation class) from it. > > > > > > How do I do that? > > > > > > I tried - setting the caps as "video/x-raw-rgb" for the src pad of > > decoder but that didn't work. Also I tried putting a capsfilter with > > caps = "video/x-raw-rgb" and followed by a ffmpegcolorspace element > > but that didn't quite work. > > You need to put your capsfilter *after* ffmpegcolorspace (you want it > to output RGB regardless of what it has in input). > > Edward > > > I don't want to set other structures such as width height depth etc > > since that could vary from video to video. > > > > > > PS - I could just write my own yuv->rgb routine but want to check if > > this can be done in the pipeline itself. > > > > > > Thanks > > Rohit > > > ------------------------------------------------------------------------------ > > Lotusphere 2011 > > Register now for Lotusphere 2011 and learn how > > to connect the dots, take your collaborative environment > > to the next level, and enter the era of Social Business. > > http://p.sf.net/sfu/lotusphere-d2d > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dingchengus1 at yahoo.com Tue Dec 21 18:23:09 2010 From: dingchengus1 at yahoo.com (ding cheng) Date: Tue, 21 Dec 2010 09:23:09 -0800 (PST) Subject: [gst-devel] webcam rtsp server supports only one session In-Reply-To: Message-ID: <521397.62278.qm@web120001.mail.ne1.yahoo.com> To LiJim and Wim, many thanks. It works. --- On Tue, 12/21/10, liJin wrote: From: liJin Subject: Re: [gst-devel] webcam rtsp server supports only one session To: "Discussion of the development of GStreamer" Date: Tuesday, December 21, 2010, 7:37 AM Hi.. For sharing media ? ?? ? ? factory = gst_rtsp_media_factory_new (); ?? ? ? gst_rtsp_media_factory_set_shared(factory,TRUE); ?? ? ? gst_rtsp_media_factory_set_launch (factory, str); I think this will help.... ? On Tue, Dec 21, 2010 at 8:18 PM, ding cheng wrote: It is one of test programs in example directory. You can use test-mp4 to stream a mp4 file. --- On Tue, 12/21/10, michael wrote: From: michael Subject: Re: [gst-devel] webcam rtsp server supports only one session To: "Discussion of the development of GStreamer" Date: Tuesday, December 21, 2010, 12:58 AM where is your test-mp4.c ? On Tue, Dec 21, 2010 at 4:24 PM, Wim Taymans wrote: Check out the 'shared' property on the media factory, it allows you to share 1 instance of a pipeline with multiple clients. Wim On Mon, 2010-12-20 at 18:16 -0800, ding cheng wrote: > I modify test-mp4.c launch string to use v4l2src as stream source: > > "v4l2src device=/dev/video0 ! ffmpegcolorspace ! ?videoscale ! > ffenc_mpeg4 ! rtpmp4vpay pt=96 name=pay0". > > It can only support one session. But the origional test-mp4.c supports > multiple sessions without problem. > Can somebody give me a clue? Thanks, > > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Lotusphere 2011 Register now for Lotusphere 2011 and learn how to connect the dots, take your collaborative environment to the next level, and enter the era of Social Business. http://p.sf.net/sfu/lotusphere-d2d _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -----Inline Attachment Follows----- ------------------------------------------------------------------------------ Lotusphere 2011 Register now for Lotusphere 2011 and learn how to connect the dots, take your collaborative environment to the next level, and enter the era of Social Business. http://p.sf.net/sfu/lotusphere-d2d -----Inline Attachment Follows----- _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Lotusphere 2011 Register now for Lotusphere 2011 and learn how to connect the dots, take your collaborative environment to the next level, and enter the era of Social Business. http://p.sf.net/sfu/lotusphere-d2d _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -----Inline Attachment Follows----- ------------------------------------------------------------------------------ Lotusphere 2011 Register now for Lotusphere 2011 and learn how to connect the dots, take your collaborative environment to the next level, and enter the era of Social Business. http://p.sf.net/sfu/lotusphere-d2d -----Inline Attachment Follows----- _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Tue Dec 21 19:07:10 2010 From: msmith at xiph.org (Michael Smith) Date: Tue, 21 Dec 2010 10:07:10 -0800 Subject: [gst-devel] Linux to Windows In-Reply-To: References: Message-ID: On Tue, Dec 21, 2010 at 2:26 AM, Shamun toha md wrote: > Because Gstreamer features are more under Linux platform. But same package > if required to port for Windows, seems a hell is just sitting out there. > If we try with Winbuilds, it does not fit all the features that is available > for linux distros. > Question: > 1) Is it not possible install Cygwin/Mingw in Windows XP/Vista/7 That is certainly possible. > 2) Put the GStreamer source and compile there That is possible too, though it's complex and absurdly slow - using winbuilds is a much better idea. > 3) Use Cygwin way of GStreamer > Impossible, anyone ever tried ? Can then JAVA wrapper will work ? Can JAVA > Applet will work ? It should be possible (though it may require some work) to get the java wrapper to work on windows. Java applets running native code are pointless though - hard to develop, hard to deploy, not helpful to anyone. Don't do that. Mike From gwunder at eoir.com Tue Dec 21 19:38:27 2010 From: gwunder at eoir.com (Greg Wunder) Date: Tue, 21 Dec 2010 13:38:27 -0500 Subject: [gst-devel] g_object_notify in filesrc Message-ID: <451B4D021F581D4AB9B270C067B7062C0AD9EF92@NEOEVS.eoir.com> When the filesrc plugin emits the g_object_notify location parameter, how can another plug-in in the chain pick this up? -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Tue Dec 21 19:48:49 2010 From: msmith at xiph.org (Michael Smith) Date: Tue, 21 Dec 2010 10:48:49 -0800 Subject: [gst-devel] g_object_notify in filesrc In-Reply-To: <451B4D021F581D4AB9B270C067B7062C0AD9EF92@NEOEVS.eoir.com> References: <451B4D021F581D4AB9B270C067B7062C0AD9EF92@NEOEVS.eoir.com> Message-ID: On Tue, Dec 21, 2010 at 10:38 AM, Greg Wunder wrote: > When the filesrc plugin emits the g_object_notify location parameter, how > can another plug-in in the chain pick this up? Your application can connect to this notify signal to find out about this (though it seems to me that this would be a strange thing to want to do), but other plugins in the pipeline shouldn't ever do that - that's not how gstreamer works. If you describe what you're trying to accomplish we might be able to help point you in the right direction. Mike From gwunder at eoir.com Tue Dec 21 19:53:12 2010 From: gwunder at eoir.com (Greg Wunder) Date: Tue, 21 Dec 2010 13:53:12 -0500 Subject: [gst-devel] g_object_notify in filesrc References: <451B4D021F581D4AB9B270C067B7062C0AD9EF92@NEOEVS.eoir.com> Message-ID: <451B4D021F581D4AB9B270C067B7062C0AD9EF93@NEOEVS.eoir.com> I have a pipeline I run as such: gst-launch filesrc location=C:/Test0001.jpg ! decodebin ! freeze ! ffmpegcolorspace ! ppdet ! glimagesink I want ppdet to access the filesrc location parameter, but it doesn't appear that this is sent to the GstBus or as a GstEvent. How can I accomplish this? -----Original Message----- From: Michael Smith [mailto:msmith at xiph.org] Sent: Tue 12/21/2010 1:48 PM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] g_object_notify in filesrc On Tue, Dec 21, 2010 at 10:38 AM, Greg Wunder wrote: > When the filesrc plugin emits the g_object_notify location parameter, how > can another plug-in in the chain pick this up? Your application can connect to this notify signal to find out about this (though it seems to me that this would be a strange thing to want to do), but other plugins in the pipeline shouldn't ever do that - that's not how gstreamer works. If you describe what you're trying to accomplish we might be able to help point you in the right direction. Mike ------------------------------------------------------------------------------ Forrester recently released a report on the Return on Investment (ROI) of Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even within 7 months. Over 3 million businesses have gone Google with Google Apps: an online email calendar, and document program that's accessible from your browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Tue Dec 21 20:03:05 2010 From: msmith at xiph.org (Michael Smith) Date: Tue, 21 Dec 2010 11:03:05 -0800 Subject: [gst-devel] g_object_notify in filesrc In-Reply-To: <451B4D021F581D4AB9B270C067B7062C0AD9EF93@NEOEVS.eoir.com> References: <451B4D021F581D4AB9B270C067B7062C0AD9EF92@NEOEVS.eoir.com> <451B4D021F581D4AB9B270C067B7062C0AD9EF93@NEOEVS.eoir.com> Message-ID: On Tue, Dec 21, 2010 at 10:53 AM, Greg Wunder wrote: > I have a pipeline I run as such: > > gst-launch filesrc location=C:/Test0001.jpg ! decodebin ! freeze ! > ffmpegcolorspace ! ppdet ! glimagesink > > I want ppdet to access the filesrc location parameter, but it doesn't appear > that this is sent to the GstBus or as a GstEvent. > > How can I accomplish this? That's correct - the filename in use isn't sent. The way GStreamer works, elements downstream aren't supposed to know about _what_ is providing the data - they just receive the data and process it. You might equally be using a source that _doesn't_ read a file at all, so there's no filename to use - so your element would be buggy if it relied on this information, and that's why GStreamer doesn't provide it. I don't know what your 'ppdet' element does - why do you want the filename in use? You probably need to figure out a different way to implement whatever you're doing. Mike From yongnian.le at intel.com Wed Dec 22 03:00:30 2010 From: yongnian.le at intel.com (Le, Yongnian) Date: Wed, 22 Dec 2010 10:00:30 +0800 Subject: [gst-devel] Reduce GstXvImageSink rendering when its window is invisible In-Reply-To: <20101213042030.GA20961@cooker.entropywave.com> References: <20101213042030.GA20961@cooker.entropywave.com> Message-ID: Thanks for suggestions and I moved the obscurity detection to the application layer (Gtk/Qt) and let application to inform video sink through property when it encourages degraded rendering. Currently I am designing the event propagation mechanism. I feel that we need avoid subtype QoS event, otherwise all related handlers need to be updated for such special event (proportion=0.0) But I am not sure whether we need to extend QoS event or add a special new event called "OBSCURED" or else. If we extend QoS event, it looks a little bit not graceful/general for it only leads to degraded quality and only applies to video. What do you think? 1. event extension: GStreamer::Event::QOS::OBSCURED 2. add new event: GStreamer::Event::OBSCURED /*original event types*/ GStreamer::Event::FlushStart GStreamer::Event::FlushStop GStreamer::Event::EOS GStreamer::Event::NewSegment GStreamer::Event::Tag GStreamer::Event::BufferSize GStreamer::Event::QOS GStreamer::Event::Seek GStreamer::Event::Navigation GStreamer::Event::Custom::UP GStreamer::Event::Custom::DS GStreamer::Event::Custom::DS::OOB GStreamer::Event::Custom::Both GStreamer::Event::Custom::Both::OOB Best Regards Yongnian -----Original Message----- From: David Schleef [mailto:ds at entropywave.com] Sent: Monday, December 13, 2010 12:21 PM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] Reduce GstXvImageSink rendering when its window is invisible On Mon, Dec 13, 2010 at 10:55:14AM +0800, Le, Yongnian wrote: > Hi, > > I am working on multimedia application development using gstreamer > framework. There might be many cases that some window is invisible > (for example other window is in full screen), its related XvImageSink > is still busy on calling gst_xvimagesink_show_frame for video > rendering. I am wondering that we could have optimization to detect > additional "Obscured" xevent and reduce the rendering frequency to > very low value, like 1 fps, if such window is obscured. In this way, > we can save precious resources for most applications, like CPU and > power, at the same time, having minimal impact to end user. Does that > approach make sense? What do you guys think of it? Thanks a lot for > your feedback and comments. Right now, the UI toolkit (Gtk+, Qt) already keeps track of windows being mapped or unmapped, and in some cases, also the obscured or partially-obscured information. So it would be straightforward to extend the GstVideoSink base class to have a property that encourages degraded rendering. It is tempting to put the obscurity detection directly in the video sink, however, leaving this to the application allows for the possibility that the app is obscuring the video on purpose, yet doesn't want the quality degraded. Perhaps both methods are useful. In general, however, rendering is a tiny fraction of the cost of playing video, so it would be good to also push this information upstream, either as part of a QoS event (e.g., proportion=0.0 for video that will be obscured), or (better) an extension of the QoS event. That way, video decoders can avoid decoding the video. And, for bonus points, as there is a trend in the industry toward stream switching for video over the internet, the QoS information could be used to select a stream with low video bit rate, or audio-only stream. David ------------------------------------------------------------------------------ Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, new data types, scalar functions, improved concurrency, built-in packages, OCI, SQL*Plus, data movement tools, best practices and more. http://p.sf.net/sfu/oracle-sfdev2dev _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From mcygogo at gmail.com Wed Dec 22 05:40:14 2010 From: mcygogo at gmail.com (michael) Date: Wed, 22 Dec 2010 12:40:14 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: Message-ID: anyone help? On Tue, Dec 21, 2010 at 4:20 PM, michael wrote: > Hi all: > I want to record the video from my book's camera, and is there one > pipeline can do this? tks :) > -------------- next part -------------- An HTML attachment was scrubbed... URL: From lijinsyam at gmail.com Wed Dec 22 06:29:07 2010 From: lijinsyam at gmail.com (liJin) Date: Wed, 22 Dec 2010 10:59:07 +0530 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: Message-ID: Hi ...Michael I am not an expert ....but i think this will help gst-launch v4l2src device=/dev/video0 ! 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink On Wed, Dec 22, 2010 at 10:10 AM, michael wrote: > anyone help? > > > On Tue, Dec 21, 2010 at 4:20 PM, michael wrote: > >> Hi all: >> I want to record the video from my book's camera, and is there one >> pipeline can do this? tks :) >> > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From braunsquared at gmail.com Wed Dec 22 06:36:31 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Wed, 22 Dec 2010 00:36:31 -0500 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: Message-ID: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> Be careful with the capsfilter you have in there. If that isn't the actual caps of the stream it will never preroll. The camera will be the deciding factor for the caps of this stream. If the results you are looking for match the caps below, I would add a videorate, videoscale and ffmpeg_colorspace before the capsfilter. Best of luck, Tim Sent from my iPhone On Dec 22, 2010, at 12:29 AM, liJin wrote: > Hi ...Michael > > I am not an expert ....but i think this will help > > gst-launch v4l2src device=/dev/video0 ! 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink > > > > On Wed, Dec 22, 2010 at 10:10 AM, michael wrote: > anyone help? > > > On Tue, Dec 21, 2010 at 4:20 PM, michael wrote: > Hi all: > I want to record the video from my book's camera, and is there one pipeline can do this? tks :) > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Wed Dec 22 07:26:30 2010 From: mcygogo at gmail.com (michael) Date: Wed, 22 Dec 2010 14:26:30 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> Message-ID: tks, this is work: gst-launch v4l2src device=/dev/video0 ! 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink On Wed, Dec 22, 2010 at 1:36 PM, Timothy Braun wrote: > Be careful with the capsfilter you have in there. If that isn't the actual > caps of the stream it will never preroll. The camera will be the deciding > factor for the caps of this stream. If the results you are looking for match > the caps below, I would add a videorate, videoscale and ffmpeg_colorspace > before the capsfilter. > > Best of luck, > Tim > > Sent from my iPhone > > On Dec 22, 2010, at 12:29 AM, liJin wrote: > > Hi ...Michael > > I am not an expert ....but i think this will help > > gst-launch v4l2src device=/dev/video0 ! > 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink > > > > On Wed, Dec 22, 2010 at 10:10 AM, michael < > mcygogo at gmail.com> wrote: > >> anyone help? >> >> >> On Tue, Dec 21, 2010 at 4:20 PM, michael < >> mcygogo at gmail.com> wrote: >> >>> Hi all: >>> I want to record the video from my book's camera, and is there one >>> pipeline can do this? tks :) >>> >> >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google >> Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: >> >> http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: > http://p.sf.net/sfu/googleapps-sfnew > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Wed Dec 22 07:28:51 2010 From: mcygogo at gmail.com (michael) Date: Wed, 22 Dec 2010 14:28:51 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> Message-ID: And now , I want to record the video to a MP4 file with the h264 encoding format , How can I do ? On Wed, Dec 22, 2010 at 2:26 PM, michael wrote: > tks, this is work: gst-launch v4l2src device=/dev/video0 ! > 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink > > > On Wed, Dec 22, 2010 at 1:36 PM, Timothy Braun wrote: > >> Be careful with the capsfilter you have in there. If that isn't the actual >> caps of the stream it will never preroll. The camera will be the deciding >> factor for the caps of this stream. If the results you are looking for match >> the caps below, I would add a videorate, videoscale and ffmpeg_colorspace >> before the capsfilter. >> >> Best of luck, >> Tim >> >> Sent from my iPhone >> >> On Dec 22, 2010, at 12:29 AM, liJin wrote: >> >> Hi ...Michael >> >> I am not an expert ....but i think this will help >> >> gst-launch v4l2src device=/dev/video0 ! >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink >> >> >> >> On Wed, Dec 22, 2010 at 10:10 AM, michael < >> mcygogo at gmail.com> wrote: >> >>> anyone help? >>> >>> >>> On Tue, Dec 21, 2010 at 4:20 PM, michael < >>> mcygogo at gmail.com> wrote: >>> >>>> Hi all: >>>> I want to record the video from my book's camera, and is there one >>>> pipeline can do this? tks :) >>>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google >>> Apps: >>> an online email calendar, and document program that's accessible from >>> your >>> browser. Read the Forrester report: >>> >>> http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> >>> gstreamer-devel at lists.sourceforge.net >>> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google >> Apps: >> an online email calendar, and document program that's accessible from your >> >> browser. Read the Forrester report: >> >> http://p.sf.net/sfu/googleapps-sfnew >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google >> Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gibrovacco at gmail.com Wed Dec 22 08:31:18 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 22 Dec 2010 09:31:18 +0200 Subject: [gst-devel] pcapparse seems to work, the output from pcapparse to rtpmp4gdepay has a problem In-Reply-To: <1292894690583-3114174.post@n4.nabble.com> References: <1292894690583-3114174.post@n4.nabble.com> Message-ID: Hi, On Tue, Dec 21, 2010 at 3:24 AM, shiva varma wrote: > > Hi All. > ?Would greatly appreciate any help regarding the problem I have been facing > wirh rtpmp4gdepay > > I have a pcap file, which contains AAC-LD stream, with the following > properties > Bitrate: 64 kbps/channel > RTP Payload: IETF RFC 3640, AAC-hbr mode > Default Dynamic Payload Number: 96 > > When I pipe the pcap file to a filesink using pcapparse I can see that the > file has some size indicating the rtp stream was written to it, following is > the pipeline > > gst-launch ?filesrc location=audioCaptureFiltered.pcap ! pcapparse > src-ip=10.22.156.148 ! "application/x-rtp, payload=96, media=audio, > clock-rate=64000, encoding-name=MPEG4-GENERIC, mode=AAC-hbr, > streamtype=(string)4" ! ?filesink location=abc.out > > However if I add rtpmp4gdepay to the pcapparse output, I observe that the > size of the file is zero.... You probably need to set the proper src-port as well in the pcapparse element. > I do not get any errors, even when I set gst-debug-level to 5.. Probably all of the packets are filtered away, so it's not, strictly speaking, an error. > > Here is the command output > > gst-launch ?filesrc location=audioCaptureFiltered.pcap ! pcapparse > src-ip=10.22.156.148 src-port=16464 ! "application/x-rtp, payload=96, > media=audio, clock-rate=64000, encoding-name=MPEG4-GENERIC, mode=AAC-hbr, > streamtype=(string)4" ! ?rtpmp4gdepay ! filesink location=abc.out strange, here you've set the port and still doesn't it work? I note that in my tests I don't set the src-ip, maybe I got your same issue.. Regards > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > Pipeline is PREROLLED ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > Got EOS from element "pipeline0". > Execution ended after 151000 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > Freeing pipeline ... > > Is there something I am missing, please let me know > > Thanks > Shiva > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/pcapparse-seems-to-work-the-output-from-pcapparse-to-rtpmp4gdepay-has-a-problem-tp3114174p3114174.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From gibrovacco at gmail.com Wed Dec 22 09:03:40 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 22 Dec 2010 10:03:40 +0200 Subject: [gst-devel] RTP+MJPEG - Sniffing packets with wireshark, How clients join the rtp server In-Reply-To: References: Message-ID: Hi, these are really generic RTP/wireshark specific questions so I might not be the more appropriate guy on this planet to reply, btw a few answers below.. On Tue, Dec 21, 2010 at 12:16 PM, Mauro Brenna wrote: ..snip.. > > > Everything works but I would like to ask to this mailinglist a couple > of questions about the implementation and RTP itself: > > 1) I tried to sniff the packets using wireshark. I can see the packets > that are sent using UDP but not the RTP encapsulation (I see only the > field data of the UDP packet). Is is a problem of wireshark for RTP > (RFC 2435) or the gstreamer plug-in is not compliant to the standard? > I tested the same thing in VLC that streams RTP packets using RFC > 2250. In wireshark I can see the RTP packet with the correct payload > type. It's usually up to a sender/receiver negotiation to agree that an UDP communication is actually RTP and which are its parameters. Standard ways to perform such a negotiation are RTSP, SIP/SDP, XMPP, etc. etc. . Afaik, by no means this negotiation, aka "the session initiation" or "signaling phase", is described in RFC1889. As such a negotiation is not mandatory, in case the two ends of the communication "already know" the basic parameters of the stream (basically, destination IP and port) the latter can happily occur without any violation to the standards. Now, what has Wireshark to do with all of this? Basically, it is a very smart tool which, other than sniffing the network traffic, also processes it (maybe you noted it in the lower part of its UI ;) ). This happens also for the negotiation packets and so, when it discovers that an UDP stream matches with the parameters of, for instance, a previous RTSP session initiation (appearing in the network dump), it's "dissected" as RTP. This kind of dissection fails in a few cases: 1) No negotiation occurred, as in the pipelines you posted above. 2) The negotiation protocol in unknown to Wireshark. For instance, you designed and implemented it by yourself: in such a case you'll have to pay the fee for doing proprietary stuff :). 3) The capture Wireshark is analysing began AFTER a "dissectable" negotiation, so the tool does not have a clue about its contents. > > 2) I do not understand very well how the client (player) join the RTP > server (streamer). I read the general RFC 1889. In the section about > RTP Control Protocol, it is not clear how the initialization works. RTCP is not about session initiation, as RFC1889 says, it is "to allow monitoring of the data delivery". Read above for more details. > I > mean, when I start the streamer, does it start to send RTP+MJPEG > packets to the unicast/multicast destination even if nobody is > listening to it? Yes, it does. How could it happen without signaling otherwise? > Or does it start sending packets after the reception > of a particular packet from the client ? I read about feedbacks > packets from sender/receiver and about the BYE (Goodbye RTCP packet). > However, I have not found a sort of JOIN packet. > Because -unless you're using means other than RFC1889- there aren't any. Probably VLC performs a session initiation (I guess RTSP, you should see it in the network capture). If you need such a feature, on the GStreamer side you'll need to use the "rtspsrc" element. Regards > > Thanks in advance for the replies, > > Mauro > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From lijinsyam at gmail.com Wed Dec 22 09:06:37 2010 From: lijinsyam at gmail.com (liJin) Date: Wed, 22 Dec 2010 13:36:37 +0530 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> Message-ID: Try this..... gst-launch v4l2src ! video/x-raw-yuv,width=320,height=240 ! ffmpegcolorspace ! ffenc_mpeg4 ! avimux ! filesink location=1.mp4 Here i encoded in to mpeg4 format in mp4 container ..If you want h264 stream then ..replace ffenc_mpeg4... Reagrds LiJin On Wed, Dec 22, 2010 at 11:58 AM, michael wrote: > And now , I want to record the video to a MP4 file with the h264 encoding > format , How can I do ? > > > On Wed, Dec 22, 2010 at 2:26 PM, michael wrote: > >> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink >> >> >> On Wed, Dec 22, 2010 at 1:36 PM, Timothy Braun wrote: >> >>> Be careful with the capsfilter you have in there. If that isn't the >>> actual caps of the stream it will never preroll. The camera will be the >>> deciding factor for the caps of this stream. If the results you are looking >>> for match the caps below, I would add a videorate, videoscale and >>> ffmpeg_colorspace before the capsfilter. >>> >>> Best of luck, >>> Tim >>> >>> Sent from my iPhone >>> >>> On Dec 22, 2010, at 12:29 AM, liJin wrote: >>> >>> Hi ...Michael >>> >>> I am not an expert ....but i think this will help >>> >>> gst-launch v4l2src device=/dev/video0 ! >>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink >>> >>> >>> >>> On Wed, Dec 22, 2010 at 10:10 AM, michael < >>> mcygogo at gmail.com> wrote: >>> >>>> anyone help? >>>> >>>> >>>> On Tue, Dec 21, 2010 at 4:20 PM, michael < >>>> mcygogo at gmail.com> wrote: >>>> >>>>> Hi all: >>>>> I want to record the video from my book's camera, and is there one >>>>> pipeline can do this? tks :) >>>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Forrester recently released a report on the Return on Investment (ROI) >>>> of >>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>>> within 7 months. Over 3 million businesses have gone Google with Google >>>> Apps: >>>> an online email calendar, and document program that's accessible from >>>> your >>>> browser. Read the Forrester report: >>>> >>>> http://p.sf.net/sfu/googleapps-sfnew >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> >>>> gstreamer-devel at lists.sourceforge.net >>>> >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google >>> Apps: >>> an online email calendar, and document program that's accessible from >>> your >>> browser. Read the Forrester report: >>> >>> http://p.sf.net/sfu/googleapps-sfnew >>> >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google >>> Apps: >>> an online email calendar, and document program that's accessible from >>> your >>> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wl2776 at gmail.com Wed Dec 22 09:10:45 2010 From: wl2776 at gmail.com (wl2776) Date: Wed, 22 Dec 2010 00:10:45 -0800 (PST) Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> Message-ID: <1293005445306-3160261.post@n4.nabble.com> michael young wrote: > > And now , I want to record the video to a MP4 file with the h264 encoding > format , How can I do ? > > On Wed, Dec 22, 2010 at 2:26 PM, michael wrote: > >> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink > > Looks like you can't, because h264 encoder is unavailable. gst-inspect | grep h264 doesn't show it. However, general idea is to encode, then mux, and then save to a file. gst-launch v4l2src device=/dev/video0 ! 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffenc_mpeg4 ! mp4mux !filesink location=/home/user/file. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From gibrovacco at gmail.com Wed Dec 22 09:14:01 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 22 Dec 2010 10:14:01 +0200 Subject: [gst-devel] RTP+MJPEG - Sniffing packets with wireshark, How clients join the rtp server In-Reply-To: References: Message-ID: Hi, one note I forgot to write before (sorry for the spam).. On Wed, Dec 22, 2010 at 10:03 AM, Marco Ballesio wrote: ..snip.. >> 2) I do not understand very well how the client (player) join the RTP >> server (streamer). I read the general RFC 1889. In the section about >> RTP Control Protocol, it is not clear how the initialization works. > > RTCP is not about session initiation, as RFC1889 says, it is "to allow > monitoring of the data delivery". ?Read above for more details. And also, unless you're dealing with very old HW/SW, you should use RFC 3550 in place of RFC 1889 (not that changes are so drastic though). Regards > >> I >> mean, when I start the streamer, does it start to send RTP+MJPEG >> packets to the unicast/multicast destination even if nobody is >> listening to it? > > Yes, it does. How could it happen without signaling otherwise? > >> Or does it start sending packets after the reception >> of a particular packet from the client ? I read about feedbacks >> packets from sender/receiver and about the BYE (Goodbye RTCP packet). >> However, I have not found a sort of JOIN packet. >> > > Because -unless you're using means other than RFC1889- there aren't > any. Probably VLC performs a session initiation (I guess RTSP, you > should see it in the network capture). If you need such a feature, on > the GStreamer side you'll need to use the "rtspsrc" element. > > Regards > >> >> Thanks in advance for the replies, >> >> Mauro >> >> ------------------------------------------------------------------------------ >> Lotusphere 2011 >> Register now for Lotusphere 2011 and learn how >> to connect the dots, take your collaborative environment >> to the next level, and enter the era of Social Business. >> http://p.sf.net/sfu/lotusphere-d2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > From braunsquared at gmail.com Wed Dec 22 09:38:15 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Wed, 22 Dec 2010 03:38:15 -0500 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: <1293005445306-3160261.post@n4.nabble.com> References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> Message-ID: H264 is encoded with x264enc element. Sent from my iPhone On Dec 22, 2010, at 3:10 AM, wl2776 wrote: > > > michael young wrote: >> >> And now , I want to record the video to a MP4 file with the h264 encoding >> format , How can I do ? >> >> On Wed, Dec 22, 2010 at 2:26 PM, michael wrote: >> >>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink >> >> > > Looks like you can't, because h264 encoder is unavailable. > gst-inspect | grep h264 > doesn't show it. > However, general idea is to encode, then mux, and then save to a file. > gst-launch v4l2src device=/dev/video0 ! > 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffenc_mpeg4 ! > mp4mux !filesink location=/home/user/file. > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From antoni.silvestre at gmail.com Wed Dec 22 10:32:27 2010 From: antoni.silvestre at gmail.com (=?ISO-8859-1?Q?Antoni_Silvestre_Padr=F3s?=) Date: Wed, 22 Dec 2010 10:32:27 +0100 Subject: [gst-devel] Gst Locks and signals Message-ID: Hi, I am using GstRtpBin for my application and I have a problem with signals and locks. I need the program to hold back the emission of certain gstrtpbin signals while I am executing some functions. Stopping the execution of the callbacks of those signals when they have already been called doesn't work because when those callbacks have been called the gstrtpbin code that has emitted them has already acquired the lock for the gstrtpbin object and my concurrent part of the code uses functions like gst_element_get_request_pad that also need to acquire that same lock. One solution would be to acquire the lock of the gstrtpbin object myself every time I run that part of the code that is not compatible with the signals. That would hold back the emission of gstrtpbin signals, however as gst object locks are not recursive when I use functions like gst_element_get_request_pad that also acquire the gst lock, my code gets deadlocked by locking the same mutex two times by the same thread. One workaround would be releasing the gst object lock just before any call to such functions but then obviously I would have the risk of a signal emission occurring in between the lock release and the call to the gst function that requires a lock. Right now I can't think of any elegant solution to this problem, anyone has any ideas? Thanks, Toni Silvestre -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Wed Dec 22 11:18:00 2010 From: mcygogo at gmail.com (michael) Date: Wed, 22 Dec 2010 18:18:00 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> Message-ID: Now I can record the video to a AVI file with mpeg4 encoding by : gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! avimux ! filesink location=record.avi But I got two issues: 1: I want record as a MP4 file, I do it like this: gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=record.avi I can record the video , but when I playback it by mplayer, I got a error: Playing ./record.mp4. libavformat file format detected. [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found LAVF_header: av_open_input_stream() failed ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 2: I want encode the video to H264, so I do like this: gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, framerate=30/1' ! ffmpegcolorspace ! x264enc ! avimux ! filesink location=record.avi But when I run the pipeline I just get the error info like this: Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Internal data flow error. Additional debug info: gstbasesrc.c(2550): gst_base_src_loop (): /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: streaming task paused, reason not-negotiated (-4) Execution ended after 1727716724 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... Some one can help me fix the issues? Tks :) On 12/22/10, Timothy Braun wrote: > H264 is encoded with x264enc element. > > Sent from my iPhone > > On Dec 22, 2010, at 3:10 AM, wl2776 wrote: > >> >> >> michael young wrote: >>> >>> And now , I want to record the video to a MP4 file with the h264 encoding >>> format , How can I do ? >>> >>> On Wed, Dec 22, 2010 at 2:26 PM, michael wrote: >>> >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink >>> >>> >> >> Looks like you can't, because h264 encoder is unavailable. >> gst-inspect | grep h264 >> doesn't show it. >> However, general idea is to encode, then mux, and then save to a file. >> gst-launch v4l2src device=/dev/video0 ! >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffenc_mpeg4 ! >> mp4mux !filesink location=/home/user/file. >> >> -- >> View this message in context: >> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google >> Apps: >> an online email calendar, and document program that's accessible from your >> >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Wed Dec 22 11:21:03 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Wed, 22 Dec 2010 11:21:03 +0100 Subject: [gst-devel] Gst Locks and signals In-Reply-To: References: Message-ID: <1293013264.17279.111.camel@meany> On Wed, 2010-12-22 at 10:32 +0100, Antoni Silvestre Padr?s wrote: > Hi, I am using GstRtpBin for my application and I have a problem with > signals and locks. I need the program to hold back the emission of > certain gstrtpbin signals while I am executing some functions. > Stopping the execution of the callbacks of those signals when they > have already been called doesn't work because when those callbacks > have been called the gstrtpbin code that has emitted them has already > acquired the lock for the gstrtpbin object and my concurrent part of > the code uses functions like gst_element_get_request_pad that also > need to acquire that same lock. > > > One solution would be to acquire the lock of the gstrtpbin object > myself every time I run that part of the code that is not compatible > with the signals. That would hold back the emission of gstrtpbin > signals, however as gst object locks are not recursive when I use > functions like gst_element_get_request_pad that also acquire the gst > lock, my code gets deadlocked by locking the same mutex two times by > the same thread. One workaround would be releasing the gst object lock > just before any call to such functions but then obviously I would have > the risk of a signal emission occurring in between the lock release > and the call to the gst function that requires a lock. > > > Right now I can't think of any elegant solution to this problem, > anyone has any ideas? What are you trying to do? all signals are emitted without any locks, if not that's a bug. What signals are you talking about, what actions are you performing that causes deadlocks? Wim > > > Thanks, > Toni Silvestre > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From antoni.silvestre at gmail.com Wed Dec 22 11:29:59 2010 From: antoni.silvestre at gmail.com (=?ISO-8859-1?Q?Antoni_Silvestre_Padr=F3s?=) Date: Wed, 22 Dec 2010 11:29:59 +0100 Subject: [gst-devel] Gst Locks and signals In-Reply-To: <1293013264.17279.111.camel@meany> References: <1293013264.17279.111.camel@meany> Message-ID: If I haven't done anything wrong with my debugging, the signal request-pt-map for the GstRtpBin element is emitted with the element locked, and in my case my concurrent code was getting deadlocked in the function gst_element_get_request_pad. thanks, Toni Silvestre On Wed, Dec 22, 2010 at 11:21, Wim Taymans wrote: > On Wed, 2010-12-22 at 10:32 +0100, Antoni Silvestre Padr?s wrote: > > Hi, I am using GstRtpBin for my application and I have a problem with > > signals and locks. I need the program to hold back the emission of > > certain gstrtpbin signals while I am executing some functions. > > Stopping the execution of the callbacks of those signals when they > > have already been called doesn't work because when those callbacks > > have been called the gstrtpbin code that has emitted them has already > > acquired the lock for the gstrtpbin object and my concurrent part of > > the code uses functions like gst_element_get_request_pad that also > > need to acquire that same lock. > > > > > > One solution would be to acquire the lock of the gstrtpbin object > > myself every time I run that part of the code that is not compatible > > with the signals. That would hold back the emission of gstrtpbin > > signals, however as gst object locks are not recursive when I use > > functions like gst_element_get_request_pad that also acquire the gst > > lock, my code gets deadlocked by locking the same mutex two times by > > the same thread. One workaround would be releasing the gst object lock > > just before any call to such functions but then obviously I would have > > the risk of a signal emission occurring in between the lock release > > and the call to the gst function that requires a lock. > > > > > > Right now I can't think of any elegant solution to this problem, > > anyone has any ideas? > > What are you trying to do? all signals are emitted without any locks, if > not that's a bug. What signals are you talking about, what actions are > you performing that causes deadlocks? > > Wim > > > > > > > Thanks, > > Toni Silvestre > > > ------------------------------------------------------------------------------ > > Forrester recently released a report on the Return on Investment (ROI) of > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > > an online email calendar, and document program that's accessible from > your > > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Wed Dec 22 11:48:33 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Wed, 22 Dec 2010 11:48:33 +0100 Subject: [gst-devel] Gst Locks and signals In-Reply-To: References: <1293013264.17279.111.camel@meany> Message-ID: <1293014913.17279.114.camel@meany> On Wed, 2010-12-22 at 11:29 +0100, Antoni Silvestre Padr?s wrote: > If I haven't done anything wrong with my debugging, the > signal request-pt-map for the GstRtpBin element is emitted with the > element locked, and in my case my concurrent code was getting > deadlocked in the function gst_element_get_request_pad. > Ok, gstrtpsession was not releasing a lock, fixed in git with this commit: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2b53cbe923419a87df9cccd8fbcb911e61b46c9e Wim > > thanks, > Toni Silvestre > > On Wed, Dec 22, 2010 at 11:21, Wim Taymans > wrote: > > On Wed, 2010-12-22 at 10:32 +0100, Antoni Silvestre Padr?s > wrote: > > Hi, I am using GstRtpBin for my application and I have a > problem with > > signals and locks. I need the program to hold back the > emission of > > certain gstrtpbin signals while I am executing some > functions. > > Stopping the execution of the callbacks of those signals > when they > > have already been called doesn't work because when those > callbacks > > have been called the gstrtpbin code that has emitted them > has already > > acquired the lock for the gstrtpbin object and my concurrent > part of > > the code uses functions like gst_element_get_request_pad > that also > > need to acquire that same lock. > > > > > > One solution would be to acquire the lock of the gstrtpbin > object > > myself every time I run that part of the code that is not > compatible > > with the signals. That would hold back the emission of > gstrtpbin > > signals, however as gst object locks are not recursive when > I use > > functions like gst_element_get_request_pad that also acquire > the gst > > lock, my code gets deadlocked by locking the same mutex two > times by > > the same thread. One workaround would be releasing the gst > object lock > > just before any call to such functions but then obviously I > would have > > the risk of a signal emission occurring in between the lock > release > > and the call to the gst function that requires a lock. > > > > > > Right now I can't think of any elegant solution to this > problem, > > anyone has any ideas? > > > What are you trying to do? all signals are emitted without any > locks, if > not that's a bug. What signals are you talking about, what > actions are > you performing that causes deadlocks? > > Wim > > > > > > > Thanks, > > Toni Silvestre > > > ------------------------------------------------------------------------------ > > Forrester recently released a report on the Return on > Investment (ROI) of > > Google Apps. They found a 300% ROI, 38%-56% cost savings, > and break-even > > within 7 months. Over 3 million businesses have gone Google > with Google Apps: > > an online email calendar, and document program that's > accessible from your > > browser. Read the Forrester report: > http://p.sf.net/sfu/googleapps-sfnew > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on > Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and > break-even > within 7 months. Over 3 million businesses have gone Google > with Google Apps: > an online email calendar, and document program that's > accessible from your > browser. Read the Forrester report: > http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From antoni.silvestre at gmail.com Wed Dec 22 11:58:27 2010 From: antoni.silvestre at gmail.com (=?ISO-8859-1?Q?Antoni_Silvestre_Padr=F3s?=) Date: Wed, 22 Dec 2010 11:58:27 +0100 Subject: [gst-devel] Gst Locks and signals In-Reply-To: <1293014913.17279.114.camel@meany> References: <1293013264.17279.111.camel@meany> <1293014913.17279.114.camel@meany> Message-ID: Thanks a lot for such a fast fix! Toni Silvestre On Wed, Dec 22, 2010 at 11:48, Wim Taymans wrote: > On Wed, 2010-12-22 at 11:29 +0100, Antoni Silvestre Padr?s wrote: > > If I haven't done anything wrong with my debugging, the > > signal request-pt-map for the GstRtpBin element is emitted with the > > element locked, and in my case my concurrent code was getting > > deadlocked in the function gst_element_get_request_pad. > > > Ok, gstrtpsession was not releasing a lock, fixed in git with this > commit: > > http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2b53cbe923419a87df9cccd8fbcb911e61b46c9e > > Wim > > > > thanks, > > Toni Silvestre > > > > On Wed, Dec 22, 2010 at 11:21, Wim Taymans > > wrote: > > > > On Wed, 2010-12-22 at 10:32 +0100, Antoni Silvestre Padr?s > > wrote: > > > Hi, I am using GstRtpBin for my application and I have a > > problem with > > > signals and locks. I need the program to hold back the > > emission of > > > certain gstrtpbin signals while I am executing some > > functions. > > > Stopping the execution of the callbacks of those signals > > when they > > > have already been called doesn't work because when those > > callbacks > > > have been called the gstrtpbin code that has emitted them > > has already > > > acquired the lock for the gstrtpbin object and my concurrent > > part of > > > the code uses functions like gst_element_get_request_pad > > that also > > > need to acquire that same lock. > > > > > > > > > One solution would be to acquire the lock of the gstrtpbin > > object > > > myself every time I run that part of the code that is not > > compatible > > > with the signals. That would hold back the emission of > > gstrtpbin > > > signals, however as gst object locks are not recursive when > > I use > > > functions like gst_element_get_request_pad that also acquire > > the gst > > > lock, my code gets deadlocked by locking the same mutex two > > times by > > > the same thread. One workaround would be releasing the gst > > object lock > > > just before any call to such functions but then obviously I > > would have > > > the risk of a signal emission occurring in between the lock > > release > > > and the call to the gst function that requires a lock. > > > > > > > > > Right now I can't think of any elegant solution to this > > problem, > > > anyone has any ideas? > > > > > > What are you trying to do? all signals are emitted without any > > locks, if > > not that's a bug. What signals are you talking about, what > > actions are > > you performing that causes deadlocks? > > > > Wim > > > > > > > > > > > Thanks, > > > Toni Silvestre > > > > > > ------------------------------------------------------------------------------ > > > Forrester recently released a report on the Return on > > Investment (ROI) of > > > Google Apps. They found a 300% ROI, 38%-56% cost savings, > > and break-even > > > within 7 months. Over 3 million businesses have gone Google > > with Google Apps: > > > an online email calendar, and document program that's > > accessible from your > > > browser. Read the Forrester report: > > http://p.sf.net/sfu/googleapps-sfnew > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > ------------------------------------------------------------------------------ > > Forrester recently released a report on the Return on > > Investment (ROI) of > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and > > break-even > > within 7 months. Over 3 million businesses have gone Google > > with Google Apps: > > an online email calendar, and document program that's > > accessible from your > > browser. Read the Forrester report: > > http://p.sf.net/sfu/googleapps-sfnew > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > ------------------------------------------------------------------------------ > > Forrester recently released a report on the Return on Investment (ROI) of > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > > an online email calendar, and document program that's accessible from > your > > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From malloblenne at gmail.com Wed Dec 22 12:20:04 2010 From: malloblenne at gmail.com (Mauro Brenna) Date: Wed, 22 Dec 2010 12:20:04 +0100 Subject: [gst-devel] RTP+MJPEG - Sniffing packets with wireshark, How clients join the rtp server In-Reply-To: References: Message-ID: Hi, thank you for the replies. I looked for some more information about wireshark and it appears that wireshark can detect the RFC 2435 (RTP for MJPEG). http://www.wireshark.org/docs/dfref/j/jpeg.html So, I expected, like it is for VLC using RFC 2250, that every UDP packet containing an RTP packet as payload will be detected. During my tests, I see only standard UDP packets. I read something more about RTP and JPEG, I would like to know if I can use the RFC 2435 for images and another RFC for audio and synchronize audio and video using RTCP. In this case, I would like to know if I can do the same for N video sources in RFC 2435. In the negative case I believe I will be constraint to use RFC 2250 for streaming both video and audio. Moreover, can I stream directly JPEG frames using RFC2250? IMHO,Since it isn't written in the RFC I believe it isn't possible. I know that these questions are a bit OT because they are more general than gstreamer use, but I am trying to understand the real potentiality of the protocol for developing my realtime application. Thanks, Mauro From gibrovacco at gmail.com Wed Dec 22 13:27:18 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Wed, 22 Dec 2010 14:27:18 +0200 Subject: [gst-devel] RTP+MJPEG - Sniffing packets with wireshark, How clients join the rtp server In-Reply-To: References: Message-ID: Hi, On Wed, Dec 22, 2010 at 1:20 PM, Mauro Brenna wrote: > Hi, > > thank you for the replies. > > I looked for some more information about wireshark and it appears that > wireshark can detect the RFC 2435 (RTP for MJPEG). > > http://www.wireshark.org/docs/dfref/j/jpeg.html "detect" is the wrong term, at least in the context of the link you sent. Wireshark can "display" the content of these packets or, that is the same, it has a dissector for the protocol. In my (limited) experience with the tool, the automagic detection occurs only if you have a signalling phase Wireshark understands. You want to just "see" the contents and you don't care about automagic, right-click a packet containing RFC2435 -> "Decode as.." -> "RTP" in the scroll down list. For more questions about Wireshark, please refer to the appropriate contacts: http://www.wireshark.org/lists/ > > So, I expected, like it is for VLC using RFC 2250, that every UDP > packet containing an RTP packet as payload will be detected. During my > tests, I see only standard UDP packets. Please, check if you have a signalling phase in the VLC capture. It's very easy, you should see some "RTSP" packets going back and forth before the actual stream begins. > > I read something more about RTP and JPEG, I would like to know if I > can use the RFC 2435 for images and another RFC for audio git clone git://anongit.freedesktop.org/gstreamer/gst-plugins-good grep -r RFC gst-plugins-good/gst/rtp | grep audio gives me: gst-plugins-good/gst/rtp/gstrtpL16pay.c: "Payload-encode Raw audio into RTP packets (RFC 3551)", gst-plugins-good/gst/rtp/gstrtpac3depay.c: "Extracts AC3 audio from RTP packets (RFC 4184)", gst-plugins-good/gst/rtp/gstrtpamrdepay.c: "Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)", gst-plugins-good/gst/rtp/gstrtpamrpay.c: "Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267)", gst-plugins-good/gst/rtp/gstrtpbvdepay.c: "Extracts BroadcomVoice audio from RTP packets (RFC 4298)", gst-plugins-good/gst/rtp/gstrtpbvpay.c: "Packetize BroadcomVoice audio streams into RTP packets (RFC 4298)", gst-plugins-good/gst/rtp/gstrtpg722pay.c: "Payload-encode Raw audio into RTP packets (RFC 3551)", gst-plugins-good/gst/rtp/gstrtpg723depay.c: "Extracts G.723 audio from RTP packets (RFC 3551)", gst-plugins-good/gst/rtp/gstrtpg723pay.c: GST_STATIC_CAPS ("audio/G723, " /* according to RFC 3551 */ gst-plugins-good/gst/rtp/gstrtpg729depay.c: "Extracts G.729 audio from RTP packets (RFC 3551)", gst-plugins-good/gst/rtp/gstrtpg729pay.c: GST_STATIC_CAPS ("audio/G729, " /* according to RFC 3555 */ gst-plugins-good/gst/rtp/gstrtpilbcdepay.c: "Extracts iLBC audio from RTP packets (RFC 3952)", gst-plugins-good/gst/rtp/gstrtpmp4adepay.c: "Extracts MPEG4 audio from RTP packets (RFC 3016)", gst-plugins-good/gst/rtp/gstrtpmp4apay.c: "Payload MPEG4 audio as RTP packets (RFC 3016)", gst-plugins-good/gst/rtp/gstrtpmpadepay.c: "Extracts MPEG audio from RTP packets (RFC 2038)", gst-plugins-good/gst/rtp/gstrtpmpapay.c: "Payload MPEG audio as RTP packets (RFC 2038)", gst-plugins-good/gst/rtp/gstrtpmparobustdepay.c: "Extracts MPEG audio from RTP packets (RFC 5219)", gst-plugins-good/gst/rtp/gstrtpqcelpdepay.c: "Extracts QCELP (PureVoice) audio from RTP packets (RFC 2658)", gst-plugins-good/gst/rtp/gstrtpqdmdepay.c: "Extracts QDM2 audio from RTP packets (no RFC)", gst-plugins-good/gst/rtp/gstrtpvorbispay.c: "Payload-encode Vorbis audio into RTP packets (RFC 5215)", (Disclaimer: that is not a comprehensive list) > and > synchronize audio and video using RTCP. RTP is actually designed to grant synchronisation. Unless you've some peculiar requirements, all you have to use is a proper GStreamer pipeline with gstrtpbin. > In this case, I would like to > know if I can do the same for N video sources in RFC 2435. In theory, yes. In practice, I never tried to synchronise multiple video streams over RTP (and you mat run into issues like synchronising the clock of the various source devices). > In the > negative case I believe I will be constraint to use RFC 2250 for > streaming both video and audio. RFC 2250 says "RTP Payload Format for MPEG1/MPEG2 Video". Taking apart the fact it is only for video and quite obsolete, are you limited to this technology because of customer requirements? You could achieve better quality (and lower bandwidth) using, for instance, h264 and rfc3984. > Moreover, can I stream directly JPEG frames using RFC2250? IMHO,Since > it isn't written in the RFC I believe it isn't possible. on the same folder of before, grep -r RFC gst-plugins-good/gst/rtp | grep JPEG gives some answers ;). > > I know that these questions are a bit OT because they are more general > than gstreamer use, but I am trying to understand the real > potentiality of the protocol for developing my realtime application. Sure it depends on how "standard" are your requirements. I've seen nothing too complex from your descriptions so far. Regards > > > Thanks, > > Mauro > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. ?Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From Topi.Santakivi at digia.com Wed Dec 22 13:31:35 2010 From: Topi.Santakivi at digia.com (Santakivi Topi) Date: Wed, 22 Dec 2010 14:31:35 +0200 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> Message-ID: <4D11EFA7.8080300@digia.com> On 12/22/2010 12:18 PM, michael wrote: > Now I can record the video to a AVI file with mpeg4 encoding by : > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink > location=record.avi > > But I got two issues: > 1: I want record as a MP4 file, I do it like this: > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink > location=record.avi > I can record the video , but when I playback it by mplayer, I got a error: > Playing ./record.mp4. > libavformat file format detected. > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found > LAVF_header: av_open_input_stream() failed > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 When recording, with the above gst-launch line, do you stop the pipeline with Ctrl+C..? It could be that the file container doesn't get fully written in case of such an interrupt and that's why the clip doesn't play. For example, if you use this pipeline and stop it with Ctrl+C, the clip doesn't play: gst-launch-0.10 videotestsrc ! 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 But this pipeline produces a valid video file: gst-launch-0.10 videotestsrc num-buffers=200 ! 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 > > 2: I want encode the video to H264, so I do like this: > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink > location=record.avi How about adding the format to the caps, e.g . v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink location=record.avi ? BR, Topi > But when I run the pipeline I just get the error info like this: > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Internal > data flow error. > Additional debug info: > gstbasesrc.c(2550): gst_base_src_loop (): > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: > streaming task paused, reason not-negotiated (-4) > Execution ended after 1727716724 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > Freeing pipeline ... > > Some one can help me fix the issues? Tks :) > > > On 12/22/10, Timothy Braun > wrote: > > H264 is encoded with x264enc element. > > > > Sent from my iPhone > > > > On Dec 22, 2010, at 3:10 AM, wl2776 > wrote: > > > >> > >> > >> michael young wrote: > >>> > >>> And now , I want to record the video to a MP4 file with the h264 > encoding > >>> format , How can I do ? > >>> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael > wrote: > >>> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink > >>> > >>> > >> > >> Looks like you can't, because h264 encoder is unavailable. > >> gst-inspect | grep h264 > >> doesn't show it. > >> However, general idea is to encode, then mux, and then save to a file. > >> gst-launch v4l2src device=/dev/video0 ! > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffenc_mpeg4 ! > >> mp4mux !filesink location=/home/user/file. > >> > >> -- > >> View this message in context: > >> > http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html > >> Sent from the GStreamer-devel mailing list archive at Nabble.com. > >> > >> > ------------------------------------------------------------------------------ > >> Forrester recently released a report on the Return on Investment > (ROI) of > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > >> within 7 months. Over 3 million businesses have gone Google with Google > >> Apps: > >> an online email calendar, and document program that's accessible > from your > >> > >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > > Forrester recently released a report on the Return on Investment (ROI) of > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > > within 7 months. Over 3 million businesses have gone Google with Google > > Apps: > > an online email calendar, and document program that's accessible from > your > > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From mcygogo at gmail.com Wed Dec 22 14:17:42 2010 From: mcygogo at gmail.com (michael) Date: Wed, 22 Dec 2010 21:17:42 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: <4D11EFA7.8080300@digia.com> References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> Message-ID: Hi Santakivi: I have tried your method, 1: I run it like this: gst-launch videotestsrc num-buffers=200 ! video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 then, I can got a right MP4 file, and I can palyback it. But I record the video from my capture like this: gst-launch v4l2src num-buffers=200 ! video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 I also got a file , But it cannot playback , I just get info like this: Playing record.mp4. libavformat file format detected. [mpeg4 @ 0x9958c10]Error, header damaged or not MPEG4 header (f_code=0) [mpeg4 @ 0x9958c10]header damaged [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9957640]Could not find codec parameters (Video: mpeg4, yuv420p) LAVF_header: av_find_stream_info() failed ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 Quicktime/MOV file format detected. [mov] Video stream found, -vid 0 VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) open: No such file or directory [MGA] Couldn't open: /dev/mga_vid open: No such file or directory [MGA] Couldn't open: /dev/mga_vid [VO_TDFXFB] Can't open /dev/fb0: Permission denied. [VO_3DFX] Unable to open /dev/3dfx. ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) ========================================================================== Audio: no sound Starting playback... VDec: vo config request - 640 x 480 (preferred colorspace: Planar YV12) VDec: using Planar YV12 as output csp (no 0) Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. VO: [xv] 640x480 => 640x480 Planar YV12 V: 0.0 200/200 ??% ??% ??,?% 0 0 Exiting... (End of file) 2: I run this: gst-launch v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240,framerate=30/1' ! ffmpegcolorspace ! x264enc ! avimux ! filesink location=record.avi But I just got error like this: Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Could not negotiate format Additional debug info: gstbasesrc.c(2755): gst_base_src_start (): /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Check your filtered caps, if any Setting pipeline to NULL ... Freeing pipeline ... I don't know the fourcc means what, then I modified it to this: gst-launch v4l2src ! 'video/x-raw-yuv,format=I420,width=320,height=240,framerate=30/1' ! ffmpegcolorspace ! x264enc ! avimux ! filesink location=record.avi But I just got warning like this: WARNING: erroneous pipeline: could not link v4l2src0 to ffmpegcsp0 and cannot record the video. And I replaced the format I420 to others just like YVU9 YV12 and so on what I found by: gst-inspect v4l2src, but there is no one can work :( On Wed, Dec 22, 2010 at 8:31 PM, Santakivi Topi wrote: > On 12/22/2010 12:18 PM, michael wrote: > > Now I can record the video to a AVI file with mpeg4 encoding by : > > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink > > location=record.avi > > > > But I got two issues: > > 1: I want record as a MP4 file, I do it like this: > > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink > > location=record.avi > > I can record the video , but when I playback it by mplayer, I got a > error: > > Playing ./record.mp4. > > libavformat file format detected. > > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found > > LAVF_header: av_open_input_stream() failed > > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 > > When recording, with the above gst-launch line, do you stop > the pipeline with Ctrl+C..? > > It could be that the file container doesn't get fully written > in case of such an interrupt and that's why the clip doesn't play. > > For example, if you use this pipeline and > stop it with Ctrl+C, the clip doesn't play: > > gst-launch-0.10 videotestsrc ! > 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace > ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 > > But this pipeline produces a valid video file: > > gst-launch-0.10 videotestsrc num-buffers=200 ! > 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace > ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 > > > > > 2: I want encode the video to H264, so I do like this: > > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink > > location=record.avi > > How about adding the format to the caps, e.g . > v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, > framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink > location=record.avi ? > > BR, > Topi > > > But when I run the pipeline I just get the error info like this: > > Setting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstSystemClock > > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Internal > > data flow error. > > Additional debug info: > > gstbasesrc.c(2550): gst_base_src_loop (): > > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: > > streaming task paused, reason not-negotiated (-4) > > Execution ended after 1727716724 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > Setting pipeline to NULL ... > > Freeing pipeline ... > > > > Some one can help me fix the issues? Tks :) > > > > > > On 12/22/10, Timothy Braun > > wrote: > > > H264 is encoded with x264enc element. > > > > > > Sent from my iPhone > > > > > > On Dec 22, 2010, at 3:10 AM, wl2776 > > wrote: > > > > > >> > > >> > > >> michael young wrote: > > >>> > > >>> And now , I want to record the video to a MP4 file with the h264 > > encoding > > >>> format , How can I do ? > > >>> > > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael > > wrote: > > >>> > > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! > > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink > > >>> > > >>> > > >> > > >> Looks like you can't, because h264 encoder is unavailable. > > >> gst-inspect | grep h264 > > >> doesn't show it. > > >> However, general idea is to encode, then mux, and then save to a > file. > > >> gst-launch v4l2src device=/dev/video0 ! > > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffenc_mpeg4 > ! > > >> mp4mux !filesink location=/home/user/file. > > >> > > >> -- > > >> View this message in context: > > >> > > > http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html > > >> Sent from the GStreamer-devel mailing list archive at Nabble.com. > > >> > > >> > > > ------------------------------------------------------------------------------ > > >> Forrester recently released a report on the Return on Investment > > (ROI) of > > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and > break-even > > >> within 7 months. Over 3 million businesses have gone Google with > Google > > >> Apps: > > >> an online email calendar, and document program that's accessible > > from your > > >> > > >> browser. Read the Forrester report: > http://p.sf.net/sfu/googleapps-sfnew > > >> _______________________________________________ > > >> gstreamer-devel mailing list > > >> gstreamer-devel at lists.sourceforge.net > > > > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > ------------------------------------------------------------------------------ > > > Forrester recently released a report on the Return on Investment (ROI) > of > > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and > break-even > > > within 7 months. Over 3 million businesses have gone Google with > Google > > > Apps: > > > an online email calendar, and document program that's accessible from > > your > > > browser. Read the Forrester report: > http://p.sf.net/sfu/googleapps-sfnew > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From lijinsyam at gmail.com Wed Dec 22 14:17:39 2010 From: lijinsyam at gmail.com (liJin) Date: Wed, 22 Dec 2010 18:47:39 +0530 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: <4D11EFA7.8080300@digia.com> References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> Message-ID: I think thats the issue... gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=1.mp4 run with* -e *option On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi wrote: > On 12/22/2010 12:18 PM, michael wrote: > > Now I can record the video to a AVI file with mpeg4 encoding by : > > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink > > location=record.avi > > > > But I got two issues: > > 1: I want record as a MP4 file, I do it like this: > > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink > > location=record.avi > > I can record the video , but when I playback it by mplayer, I got a > error: > > Playing ./record.mp4. > > libavformat file format detected. > > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found > > LAVF_header: av_open_input_stream() failed > > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 > > When recording, with the above gst-launch line, do you stop > the pipeline with Ctrl+C..? > > It could be that the file container doesn't get fully written > in case of such an interrupt and that's why the clip doesn't play. > > For example, if you use this pipeline and > stop it with Ctrl+C, the clip doesn't play: > > gst-launch-0.10 videotestsrc ! > 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace > ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 > > But this pipeline produces a valid video file: > > gst-launch-0.10 videotestsrc num-buffers=200 ! > 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace > ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 > > > > > 2: I want encode the video to H264, so I do like this: > > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink > > location=record.avi > > How about adding the format to the caps, e.g . > v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, > framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink > location=record.avi ? > > BR, > Topi > > > But when I run the pipeline I just get the error info like this: > > Setting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstSystemClock > > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Internal > > data flow error. > > Additional debug info: > > gstbasesrc.c(2550): gst_base_src_loop (): > > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: > > streaming task paused, reason not-negotiated (-4) > > Execution ended after 1727716724 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > Setting pipeline to NULL ... > > Freeing pipeline ... > > > > Some one can help me fix the issues? Tks :) > > > > > > On 12/22/10, Timothy Braun > > wrote: > > > H264 is encoded with x264enc element. > > > > > > Sent from my iPhone > > > > > > On Dec 22, 2010, at 3:10 AM, wl2776 > > wrote: > > > > > >> > > >> > > >> michael young wrote: > > >>> > > >>> And now , I want to record the video to a MP4 file with the h264 > > encoding > > >>> format , How can I do ? > > >>> > > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael > > wrote: > > >>> > > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! > > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink > > >>> > > >>> > > >> > > >> Looks like you can't, because h264 encoder is unavailable. > > >> gst-inspect | grep h264 > > >> doesn't show it. > > >> However, general idea is to encode, then mux, and then save to a > file. > > >> gst-launch v4l2src device=/dev/video0 ! > > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffenc_mpeg4 > ! > > >> mp4mux !filesink location=/home/user/file. > > >> > > >> -- > > >> View this message in context: > > >> > > > http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html > > >> Sent from the GStreamer-devel mailing list archive at Nabble.com. > > >> > > >> > > > ------------------------------------------------------------------------------ > > >> Forrester recently released a report on the Return on Investment > > (ROI) of > > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and > break-even > > >> within 7 months. Over 3 million businesses have gone Google with > Google > > >> Apps: > > >> an online email calendar, and document program that's accessible > > from your > > >> > > >> browser. Read the Forrester report: > http://p.sf.net/sfu/googleapps-sfnew > > >> _______________________________________________ > > >> gstreamer-devel mailing list > > >> gstreamer-devel at lists.sourceforge.net > > > > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > ------------------------------------------------------------------------------ > > > Forrester recently released a report on the Return on Investment (ROI) > of > > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and > break-even > > > within 7 months. Over 3 million businesses have gone Google with > Google > > > Apps: > > > an online email calendar, and document program that's accessible from > > your > > > browser. Read the Forrester report: > http://p.sf.net/sfu/googleapps-sfnew > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Wed Dec 22 14:27:18 2010 From: mcygogo at gmail.com (michael) Date: Wed, 22 Dec 2010 21:27:18 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> Message-ID: I just ran it : gst-launch -e v4l2src num-buffers=200 ! video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 And got a mp4 file , but it still cannot be playbacked, and give the error info: Playing record.mp4. libavformat file format detected. [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to guess time_increment_bits [mpeg4 @ 0x9376c10]my guess is 5 bits ;) [mpeg4 @ 0x9376c10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag [mpeg4 @ 0x9376c10]picture size invalid (0x0) [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters (Video: mpeg4, yuv420p) LAVF_header: av_find_stream_info() failed ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 Quicktime/MOV file format detected. [mov] Video stream found, -vid 0 VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) open: No such file or directory [MGA] Couldn't open: /dev/mga_vid open: No such file or directory [MGA] Couldn't open: /dev/mga_vid [VO_TDFXFB] Can't open /dev/fb0: Permission denied. [VO_3DFX] Unable to open /dev/3dfx. ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) ========================================================================== Audio: no sound Starting playback... VDec: vo config request - 640 x 480 (preferred colorspace: Planar YV12) VDec: using Planar YV12 as output csp (no 0) Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. VO: [xv] 640x480 => 640x480 Planar YV12 V: 0.0 200/200 ??% ??% ??,?% 0 0 Exiting... (End of file) But the error info not like the old one! On Wed, Dec 22, 2010 at 9:17 PM, liJin wrote: > I think thats the issue... > > > gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink > location=1.mp4 > > run with* -e *option > > > > On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi wrote: > >> On 12/22/2010 12:18 PM, michael wrote: >> > Now I can record the video to a AVI file with mpeg4 encoding by : >> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink >> > location=record.avi >> > >> > But I got two issues: >> > 1: I want record as a MP4 file, I do it like this: >> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink >> > location=record.avi >> > I can record the video , but when I playback it by mplayer, I got a >> error: >> > Playing ./record.mp4. >> > libavformat file format detected. >> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >> > LAVF_header: av_open_input_stream() failed >> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >> >> When recording, with the above gst-launch line, do you stop >> the pipeline with Ctrl+C..? >> >> It could be that the file container doesn't get fully written >> in case of such an interrupt and that's why the clip doesn't play. >> >> For example, if you use this pipeline and >> stop it with Ctrl+C, the clip doesn't play: >> >> gst-launch-0.10 videotestsrc ! >> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace >> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >> >> But this pipeline produces a valid video file: >> >> gst-launch-0.10 videotestsrc num-buffers=200 ! >> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace >> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >> >> > >> > 2: I want encode the video to H264, so I do like this: >> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >> > location=record.avi >> >> How about adding the format to the caps, e.g . >> v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >> location=record.avi ? >> >> BR, >> Topi >> >> > But when I run the pipeline I just get the error info like this: >> > Setting pipeline to PAUSED ... >> > Pipeline is live and does not need PREROLL ... >> > Setting pipeline to PLAYING ... >> > New clock: GstSystemClock >> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Internal >> > data flow error. >> > Additional debug info: >> > gstbasesrc.c(2550): gst_base_src_loop (): >> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >> > streaming task paused, reason not-negotiated (-4) >> > Execution ended after 1727716724 ns. >> > Setting pipeline to PAUSED ... >> > Setting pipeline to READY ... >> > Setting pipeline to NULL ... >> > Freeing pipeline ... >> > >> > Some one can help me fix the issues? Tks :) >> > >> > >> > On 12/22/10, Timothy Braun > > > wrote: >> > > H264 is encoded with x264enc element. >> > > >> > > Sent from my iPhone >> > > >> > > On Dec 22, 2010, at 3:10 AM, wl2776 > > > wrote: >> > > >> > >> >> > >> >> > >> michael young wrote: >> > >>> >> > >>> And now , I want to record the video to a MP4 file with the h264 >> > encoding >> > >>> format , How can I do ? >> > >>> >> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael > > > wrote: >> > >>> >> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >> xvimagesink >> > >>> >> > >>> >> > >> >> > >> Looks like you can't, because h264 encoder is unavailable. >> > >> gst-inspect | grep h264 >> > >> doesn't show it. >> > >> However, general idea is to encode, then mux, and then save to a >> file. >> > >> gst-launch v4l2src device=/dev/video0 ! >> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffenc_mpeg4 >> ! >> > >> mp4mux !filesink location=/home/user/file. >> > >> >> > >> -- >> > >> View this message in context: >> > >> >> > >> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >> > >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> > >> >> > >> >> > >> ------------------------------------------------------------------------------ >> > >> Forrester recently released a report on the Return on Investment >> > (ROI) of >> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >> break-even >> > >> within 7 months. Over 3 million businesses have gone Google with >> Google >> > >> Apps: >> > >> an online email calendar, and document program that's accessible >> > from your >> > >> >> > >> browser. Read the Forrester report: >> http://p.sf.net/sfu/googleapps-sfnew >> > >> _______________________________________________ >> > >> gstreamer-devel mailing list >> > >> gstreamer-devel at lists.sourceforge.net >> > >> > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > >> > > >> > >> ------------------------------------------------------------------------------ >> > > Forrester recently released a report on the Return on Investment >> (ROI) of >> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and >> break-even >> > > within 7 months. Over 3 million businesses have gone Google with >> Google >> > > Apps: >> > > an online email calendar, and document program that's accessible from >> > your >> > > browser. Read the Forrester report: >> http://p.sf.net/sfu/googleapps-sfnew >> > > _______________________________________________ >> > > gstreamer-devel mailing list >> > > gstreamer-devel at lists.sourceforge.net >> > >> > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google >> Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From markus.t.vartiainen at nokia.com Wed Dec 22 14:42:45 2010 From: markus.t.vartiainen at nokia.com (Markus Vartiainen) Date: Wed, 22 Dec 2010 15:42:45 +0200 Subject: [gst-devel] Compilation error in "[gst-cvs] gst-plugins-bad: camerabin: fix for latest GTK+ API changes" ? In-Reply-To: <20101220193458.518EE10057@kemper.freedesktop.org> References: <20101220193458.518EE10057@kemper.freedesktop.org> Message-ID: <4D120055.9040303@nokia.com> Hi, I just skimmed through latest commits and could not help noticing that there is a typo in the variable "store" (it is written "sotre"): > +#if GTK_CHECK_VERSION (2, 91, 6) > + GtkListStore *sotre; > + GtkTreeIter iter; > +#endif The variable is later used as "store", like this: #if GTK_CHECK_VERSION (2, 91, 6) store = GTK_LIST_STORE (gtk_combo_box_get_model (ui_cbbox_resolution)); gtk_list_store_append (store, &iter); gtk_list_store_set (store, &iter, 0, item_str->str, -1); #else To me it seems like this would not even compile, if the #if statements resolve to true, since there is no other variable "store" defined in the scope. Regards, Markus Vartiainen ext Stefan Kost wrote: > Module: gst-plugins-bad > Branch: master > Commit: 516c977c76ba4cd86a7147aab7416a33d76f870f > URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-bad/commit/?id=516c977c76ba4cd86a7147aab7416a33d76f870f > > Author: Saleem Abdulrasool > Date: Mon Dec 20 07:19:04 2010 -0800 > > camerabin: fix for latest GTK+ API changes > > --- > > tests/examples/camerabin/gst-camera.c | 14 ++++++++++++++ > 1 files changed, 14 insertions(+), 0 deletions(-) > > diff --git a/tests/examples/camerabin/gst-camera.c b/tests/examples/camerabin/gst-camera.c > index c837ba5..e439bb8 100644 > --- a/tests/examples/camerabin/gst-camera.c > +++ b/tests/examples/camerabin/gst-camera.c > @@ -276,7 +276,11 @@ my_bus_sync_callback (GstBus * bus, GstMessage * message, gpointer data) > > /* FIXME: make sure to get XID in main thread */ > gst_x_overlay_set_window_handle (GST_X_OVERLAY (message->src), > +#if GTK_CHECK_VERSION (2, 91, 6) > + GDK_WINDOW_XID (gtk_widget_get_window (ui_drawing)); > +#else > GDK_WINDOW_XWINDOW (gtk_widget_get_window (ui_drawing))); > +#endif > > gst_message_unref (message); > return GST_BUS_DROP; > @@ -1157,6 +1161,10 @@ create_menu_items_from_structure (GstStructure * structure) > > for (j = 0; j < num_framerates; j++) { > GstCaps *video_caps; > +#if GTK_CHECK_VERSION (2, 91, 6) > + GtkListStore *sotre; > + GtkTreeIter iter; > +#endif > > if (framerate_list) { > const GValue *item = gst_value_list_get_value (framerate_list, j); > @@ -1167,7 +1175,13 @@ create_menu_items_from_structure (GstStructure * structure) > g_string_append_printf (item_str, " (%" GST_FOURCC_FORMAT ")", > GST_FOURCC_ARGS (fourcc)); > g_string_append_printf (item_str, ", %dx%d at %d/%d", w, h, n, d); > +#if GTK_CHECK_VERSION (2, 91, 6) > + store = GTK_LIST_STORE (gtk_combo_box_get_model (ui_cbbox_resolution)); > + gtk_list_store_append (store, &iter); > + gtk_list_store_set (store, &iter, 0, item_str->str, -1); > +#else > gtk_combo_box_append_text (ui_cbbox_resolution, item_str->str); > +#endif > > video_caps = > gst_caps_new_simple (structure_name, "format", GST_TYPE_FOURCC, > > > ------------------------------------------------------------------------------ > Lotusphere 2011 > Register now for Lotusphere 2011 and learn how > to connect the dots, take your collaborative environment > to the next level, and enter the era of Social Business. > http://p.sf.net/sfu/lotusphere-d2d > _______________________________________________ > gstreamer-cvs mailing list > gstreamer-cvs at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-cvs From rafael.lmsousa at gmail.com Wed Dec 22 15:11:02 2010 From: rafael.lmsousa at gmail.com (Rafael Sousa) Date: Wed, 22 Dec 2010 10:11:02 -0400 Subject: [gst-devel] distorted video encoded with x264 In-Reply-To: References: Message-ID: Hi all, I've fixed my problem by adding the following parameter to the x264enc: "pass=qual quantizer=20 tune=zerolatency" the video presented was perfect... thanks for the replies. On Thu, Dec 16, 2010 at 12:11 AM, Rafael Sousa wrote: > When I use the x264 build 85 that issue doesn't occur, the video is > perfect, only in superior build that I tested (98 and 110) that the problem > occur. I don't have any idea of what can be happening, I've already removed > and installed the packages several times, I even upgraded my ubuntu system > from 10.04 to 10.10 but no change. Anyone have any idea? I've tried all that > I know. I need to use the build 98 or superior. > > thanks for the reply > > Message: 1 > >> Date: Wed, 15 Dec 2010 15:42:01 -0500 >> From: Timothy Braun >> Subject: Re: [gst-devel] distorted video encoded with x264 >> To: Discussion of the development of GStreamer >> >> Message-ID: <69C57452-40E4-4172-883F-DF374D355CD7 at gmail.com> >> Content-Type: text/plain; charset="us-ascii" >> >> I'm no expert on this, but ran into similar issues when making a local >> file. It ultimately ended up being the parameters I was passing to the >> x264enc element. I would take a look at those with gst-inspect and see if >> you can tune the output. Most notably the pass and quantizer parameters. >> >> Tim >> >> Sent from my iPhone >> >> >> On Dec 15, 2010, at 3:02 PM, Rafael Sousa >> wrote: >> >> > Hi all, >> > >> > I'm using the versions of gstreamer installed from ubuntu 10.10 apt-get. >> I've installed the gstreamer, gst-plugins-bad, gst-plugins-good, >> gst-plugins-ugly, gst-ffmpeg and I've installed the x264. >> > >> > The problem is that my pipeline gives as result a blurring video. My >> pipeline works to read a YUV file at the sender, encodes it in H.264, and >> sends it to receiver using RTP in encoding time. At the receiver, it >> receives the stream and stores it in YUV format again for posterior process. >> The pipeline is as follows: >> > >> > Sender: >> > >> > "gstrtpbin name=rtpbin filesrc location=paris_cif.yuv blocksize=152064 ! >> video/x-raw-yuv, width=352, height=288, framerate=(fraction)15/1, >> format=(fourcc)I420 ! x264enc, key-int-max=30, key-int-min=15, >> intra-refresh=true ! rtph264pay ! rtpbin.send_rtp_sink_0 >> rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 >> rtpbin.send_rtcp_src_0 ! udpsink host=5003 port=5001 sync=false async=false >> udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 " >> > >> > Receiver: >> > >> > "gstrtpbin name=rtpbin udpsrc >> caps=\"application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" >> port=5000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! ffdec_h264 ! >> filesink location=example.yuv udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 >> rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false" >> > >> > I don't know why the video is blurring since I've tested it localhost, >> without loss rates. The command that I'm using to watch the result is: >> > >> > mplayer example.yuv -demuxer rawvideo -rawvideo cif >> > >> > I took a screenshot of the video, it is in annex. Please, anyone can >> help? >> > >> > thanks in advance >> > >> > >> ------------------------------------------------------------------------------ >> > Lotusphere 2011 >> > Register now for Lotusphere 2011 and learn how >> > to connect the dots, take your collaborative environment >> > to the next level, and enter the era of Social Business. >> > http://p.sf.net/sfu/lotusphere-d2d >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From braunsquared at gmail.com Wed Dec 22 15:58:01 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Wed, 22 Dec 2010 09:58:01 -0500 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> Message-ID: <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> I was never able to get the ffmpeg mpeg4 mixer to work quite right. I found that mp4mux, an extension of qtmux, works much better. If you inspect the ffmpeg mp4 muxer they actually suggest that you use a different one as it's unstable. Best, Tim Sent from my iPhone On Dec 22, 2010, at 8:27 AM, michael wrote: > I just ran it : > gst-launch -e v4l2src num-buffers=200 ! video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 > And got a mp4 file , but it still cannot be playbacked, and give the error info: > Playing record.mp4. > libavformat file format detected. > [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to guess time_increment_bits > [mpeg4 @ 0x9376c10]my guess is 5 bits ;) > [mpeg4 @ 0x9376c10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag > [mpeg4 @ 0x9376c10]picture size invalid (0x0) > [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters (Video: mpeg4, yuv420p) > LAVF_header: av_find_stream_info() failed > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 > Quicktime/MOV file format detected. > [mov] Video stream found, -vid 0 > VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) > open: No such file or directory > [MGA] Couldn't open: /dev/mga_vid > open: No such file or directory > [MGA] Couldn't open: /dev/mga_vid > [VO_TDFXFB] Can't open /dev/fb0: Permission denied. > [VO_3DFX] Unable to open /dev/3dfx. > ========================================================================== > Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family > Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) > ========================================================================== > Audio: no sound > Starting playback... > VDec: vo config request - 640 x 480 (preferred colorspace: Planar YV12) > VDec: using Planar YV12 as output csp (no 0) > Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. > VO: [xv] 640x480 => 640x480 Planar YV12 > V: 0.0 200/200 ??% ??% ??,?% 0 0 > > Exiting... (End of file) > > But the error info not like the old one! > > On Wed, Dec 22, 2010 at 9:17 PM, liJin wrote: > I think thats the issue... > > > gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=1.mp4 > > run with -e option > > > > On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi wrote: > On 12/22/2010 12:18 PM, michael wrote: > > Now I can record the video to a AVI file with mpeg4 encoding by : > > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink > > location=record.avi > > > > But I got two issues: > > 1: I want record as a MP4 file, I do it like this: > > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink > > location=record.avi > > I can record the video , but when I playback it by mplayer, I got a error: > > Playing ./record.mp4. > > libavformat file format detected. > > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found > > LAVF_header: av_open_input_stream() failed > > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 > > When recording, with the above gst-launch line, do you stop > the pipeline with Ctrl+C..? > > It could be that the file container doesn't get fully written > in case of such an interrupt and that's why the clip doesn't play. > > For example, if you use this pipeline and > stop it with Ctrl+C, the clip doesn't play: > > gst-launch-0.10 videotestsrc ! > 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace > ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 > > But this pipeline produces a valid video file: > > gst-launch-0.10 videotestsrc num-buffers=200 ! > 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace > ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 > > > > > 2: I want encode the video to H264, so I do like this: > > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, > > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink > > location=record.avi > > How about adding the format to the caps, e.g . > v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, > framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink > location=record.avi ? > > BR, > Topi > > > But when I run the pipeline I just get the error info like this: > > Setting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstSystemClock > > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Internal > > data flow error. > > Additional debug info: > > gstbasesrc.c(2550): gst_base_src_loop (): > > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: > > streaming task paused, reason not-negotiated (-4) > > Execution ended after 1727716724 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > Setting pipeline to NULL ... > > Freeing pipeline ... > > > > Some one can help me fix the issues? Tks :) > > > > > > On 12/22/10, Timothy Braun > > wrote: > > > H264 is encoded with x264enc element. > > > > > > Sent from my iPhone > > > > > > On Dec 22, 2010, at 3:10 AM, wl2776 > > wrote: > > > > > >> > > >> > > >> michael young wrote: > > >>> > > >>> And now , I want to record the video to a MP4 file with the h264 > > encoding > > >>> format , How can I do ? > > >>> > > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael > > wrote: > > >>> > > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! > > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink > > >>> > > >>> > > >> > > >> Looks like you can't, because h264 encoder is unavailable. > > >> gst-inspect | grep h264 > > >> doesn't show it. > > >> However, general idea is to encode, then mux, and then save to a file. > > >> gst-launch v4l2src device=/dev/video0 ! > > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffenc_mpeg4 ! > > >> mp4mux !filesink location=/home/user/file. > > >> > > >> -- > > >> View this message in context: > > >> > > http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html > > >> Sent from the GStreamer-devel mailing list archive at Nabble.com. > > >> > > >> > > ------------------------------------------------------------------------------ > > >> Forrester recently released a report on the Return on Investment > > (ROI) of > > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > > >> within 7 months. Over 3 million businesses have gone Google with Google > > >> Apps: > > >> an online email calendar, and document program that's accessible > > from your > > >> > > >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > > >> _______________________________________________ > > >> gstreamer-devel mailing list > > >> gstreamer-devel at lists.sourceforge.net > > > > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > ------------------------------------------------------------------------------ > > > Forrester recently released a report on the Return on Investment (ROI) of > > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > > > within 7 months. Over 3 million businesses have gone Google with Google > > > Apps: > > > an online email calendar, and document program that's accessible from > > your > > > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Wed Dec 22 16:10:32 2010 From: mcygogo at gmail.com (michael) Date: Wed, 22 Dec 2010 23:10:32 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> Message-ID: I have tested changing the muxer to qtmux, but it still not work On Wed, Dec 22, 2010 at 10:58 PM, Timothy Braun wrote: > I was never able to get the ffmpeg mpeg4 mixer to work quite right. I found > that mp4mux, an extension of qtmux, works much better. If you inspect the > ffmpeg mp4 muxer they actually suggest that you use a different one as it's > unstable. > > Best, > Tim > > Sent from my iPhone > > On Dec 22, 2010, at 8:27 AM, michael wrote: > > I just ran it : > gst-launch -e v4l2src num-buffers=200 ! > video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! > ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 > And got a mp4 file , but it still cannot be playbacked, and give the error > info: > Playing record.mp4. > libavformat file format detected. > [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to guess > time_increment_bits > [mpeg4 @ 0x9376c10]my guess is 5 bits ;) > [mpeg4 @ 0x9376c10]looks like this file was encoded with > (divx4/(old)xvid/opendivx) -> forcing low_delay flag > [mpeg4 @ 0x9376c10]picture size invalid (0x0) > [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters > (Video: mpeg4, yuv420p) > LAVF_header: av_find_stream_info() failed > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 > Quicktime/MOV file format detected. > [mov] Video stream found, -vid 0 > VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) > open: No such file or directory > [MGA] Couldn't open: /dev/mga_vid > open: No such file or directory > [MGA] Couldn't open: /dev/mga_vid > [VO_TDFXFB] Can't open /dev/fb0: Permission denied. > [VO_3DFX] Unable to open /dev/3dfx. > ========================================================================== > Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family > Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) > ========================================================================== > Audio: no sound > Starting playback... > VDec: vo config request - 640 x 480 (preferred colorspace: Planar YV12) > VDec: using Planar YV12 as output csp (no 0) > Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. > VO: [xv] 640x480 => 640x480 Planar YV12 > V: 0.0 200/200 ??% ??% ??,?% 0 0 > > Exiting... (End of file) > > But the error info not like the old one! > > On Wed, Dec 22, 2010 at 9:17 PM, liJin < > lijinsyam at gmail.com> wrote: > >> I think thats the issue... >> >> >> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >> framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink >> location=1.mp4 >> >> run with* -e *option >> >> >> >> On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi < >> Topi.Santakivi at digia.com> wrote: >> >>> On 12/22/2010 12:18 PM, michael wrote: >>> > Now I can record the video to a AVI file with mpeg4 encoding by : >>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink >>> > location=record.avi >>> > >>> > But I got two issues: >>> > 1: I want record as a MP4 file, I do it like this: >>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink >>> > location=record.avi >>> > I can record the video , but when I playback it by mplayer, I got a >>> error: >>> > Playing ./record.mp4. >>> > libavformat file format detected. >>> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >>> > LAVF_header: av_open_input_stream() failed >>> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>> >>> When recording, with the above gst-launch line, do you stop >>> the pipeline with Ctrl+C..? >>> >>> It could be that the file container doesn't get fully written >>> in case of such an interrupt and that's why the clip doesn't play. >>> >>> For example, if you use this pipeline and >>> stop it with Ctrl+C, the clip doesn't play: >>> >>> gst-launch-0.10 videotestsrc ! >>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace >>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>> >>> But this pipeline produces a valid video file: >>> >>> gst-launch-0.10 videotestsrc num-buffers=200 ! >>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace >>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>> >>> > >>> > 2: I want encode the video to H264, so I do like this: >>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >>> > location=record.avi >>> >>> How about adding the format to the caps, e.g . >>> v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >>> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >>> location=record.avi ? >>> >>> BR, >>> Topi >>> >>> > But when I run the pipeline I just get the error info like this: >>> > Setting pipeline to PAUSED ... >>> > Pipeline is live and does not need PREROLL ... >>> > Setting pipeline to PLAYING ... >>> > New clock: GstSystemClock >>> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>> Internal >>> > data flow error. >>> > Additional debug info: >>> > gstbasesrc.c(2550): gst_base_src_loop (): >>> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>> > streaming task paused, reason not-negotiated (-4) >>> > Execution ended after 1727716724 ns. >>> > Setting pipeline to PAUSED ... >>> > Setting pipeline to READY ... >>> > Setting pipeline to NULL ... >>> > Freeing pipeline ... >>> > >>> > Some one can help me fix the issues? Tks :) >>> > >>> > >>> > On 12/22/10, Timothy Braun < >>> braunsquared at gmail.com >>> > braunsquared at gmail.com>> wrote: >>> > > H264 is encoded with x264enc element. >>> > > >>> > > Sent from my iPhone >>> > > >>> > > On Dec 22, 2010, at 3:10 AM, wl2776 < >>> wl2776 at gmail.com >>> > wl2776 at gmail.com>> wrote: >>> > > >>> > >> >>> > >> >>> > >> michael young wrote: >>> > >>> >>> > >>> And now , I want to record the video to a MP4 file with the h264 >>> > encoding >>> > >>> format , How can I do ? >>> > >>> >>> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael < >>> mcygogo at gmail.com >>> > mcygogo at gmail.com>> wrote: >>> > >>> >>> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >>> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>> xvimagesink >>> > >>> >>> > >>> >>> > >> >>> > >> Looks like you can't, because h264 encoder is unavailable. >>> > >> gst-inspect | grep h264 >>> > >> doesn't show it. >>> > >> However, general idea is to encode, then mux, and then save to a >>> file. >>> > >> gst-launch v4l2src device=/dev/video0 ! >>> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>> ffenc_mpeg4 ! >>> > >> mp4mux !filesink location=/home/user/file. >>> > >> >>> > >> -- >>> > >> View this message in context: >>> > >> >>> > >>> >>> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >>> > >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >>> > >> >>> > >> >>> > >>> ------------------------------------------------------------------------------ >>> > >> Forrester recently released a report on the Return on Investment >>> > (ROI) of >>> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>> break-even >>> > >> within 7 months. Over 3 million businesses have gone Google with >>> Google >>> > >> Apps: >>> > >> an online email calendar, and document program that's accessible >>> > from your >>> > >> >>> > >> browser. Read the Forrester report: >>> >>> http://p.sf.net/sfu/googleapps-sfnew >>> > >> _______________________________________________ >>> > >> gstreamer-devel mailing list >>> > >> >>> gstreamer-devel at lists.sourceforge.net >>> > >>> gstreamer-devel at lists.sourceforge.net> >>> > >> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> > > >>> > > >>> > >>> ------------------------------------------------------------------------------ >>> > > Forrester recently released a report on the Return on Investment >>> (ROI) of >>> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>> break-even >>> > > within 7 months. Over 3 million businesses have gone Google with >>> Google >>> > > Apps: >>> > > an online email calendar, and document program that's accessible >>> from >>> > your >>> > > browser. Read the Forrester report: >>> >>> http://p.sf.net/sfu/googleapps-sfnew >>> > > _______________________________________________ >>> > > gstreamer-devel mailing list >>> > > >>> gstreamer-devel at lists.sourceforge.net >>> > >>> gstreamer-devel at lists.sourceforge.net> >>> > > >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> > > >>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google >>> Apps: >>> an online email calendar, and document program that's accessible from >>> your >>> browser. Read the Forrester report: >>> >>> http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> >>> gstreamer-devel at lists.sourceforge.net >>> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >> >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google >> Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: >> >> http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: > http://p.sf.net/sfu/googleapps-sfnew > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From braunsquared at gmail.com Wed Dec 22 16:24:52 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Wed, 22 Dec 2010 10:24:52 -0500 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> Message-ID: Not qtmux, use mp4mux. On Wed, Dec 22, 2010 at 10:10 AM, michael wrote: > I have tested changing the muxer to qtmux, but it still not work > > > On Wed, Dec 22, 2010 at 10:58 PM, Timothy Braun wrote: > >> I was never able to get the ffmpeg mpeg4 mixer to work quite right. I >> found that mp4mux, an extension of qtmux, works much better. If you inspect >> the ffmpeg mp4 muxer they actually suggest that you use a different one as >> it's unstable. >> >> Best, >> Tim >> >> Sent from my iPhone >> >> On Dec 22, 2010, at 8:27 AM, michael wrote: >> >> I just ran it : >> gst-launch -e v4l2src num-buffers=200 ! >> video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! >> ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 >> And got a mp4 file , but it still cannot be playbacked, and give the error >> info: >> Playing record.mp4. >> libavformat file format detected. >> [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to >> guess time_increment_bits >> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >> [mpeg4 @ 0x9376c10]looks like this file was encoded with >> (divx4/(old)xvid/opendivx) -> forcing low_delay flag >> [mpeg4 @ 0x9376c10]picture size invalid (0x0) >> [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) >> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters >> (Video: mpeg4, yuv420p) >> LAVF_header: av_find_stream_info() failed >> ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >> Quicktime/MOV file format detected. >> [mov] Video stream found, -vid 0 >> VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) >> open: No such file or directory >> [MGA] Couldn't open: /dev/mga_vid >> open: No such file or directory >> [MGA] Couldn't open: /dev/mga_vid >> [VO_TDFXFB] Can't open /dev/fb0: Permission denied. >> [VO_3DFX] Unable to open /dev/3dfx. >> ========================================================================== >> Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family >> Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) >> ========================================================================== >> Audio: no sound >> Starting playback... >> VDec: vo config request - 640 x 480 (preferred colorspace: Planar YV12) >> VDec: using Planar YV12 as output csp (no 0) >> Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. >> VO: [xv] 640x480 => 640x480 Planar YV12 >> V: 0.0 200/200 ??% ??% ??,?% 0 0 >> >> Exiting... (End of file) >> >> But the error info not like the old one! >> >> On Wed, Dec 22, 2010 at 9:17 PM, liJin < >> lijinsyam at gmail.com> wrote: >> >>> I think thats the issue... >>> >>> >>> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>> framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink >>> location=1.mp4 >>> >>> run with* -e *option >>> >>> >>> >>> On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi < >>> Topi.Santakivi at digia.com> wrote: >>> >>>> On 12/22/2010 12:18 PM, michael wrote: >>>> > Now I can record the video to a AVI file with mpeg4 encoding by : >>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink >>>> > location=record.avi >>>> > >>>> > But I got two issues: >>>> > 1: I want record as a MP4 file, I do it like this: >>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink >>>> > location=record.avi >>>> > I can record the video , but when I playback it by mplayer, I got a >>>> error: >>>> > Playing ./record.mp4. >>>> > libavformat file format detected. >>>> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >>>> > LAVF_header: av_open_input_stream() failed >>>> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>> >>>> When recording, with the above gst-launch line, do you stop >>>> the pipeline with Ctrl+C..? >>>> >>>> It could be that the file container doesn't get fully written >>>> in case of such an interrupt and that's why the clip doesn't play. >>>> >>>> For example, if you use this pipeline and >>>> stop it with Ctrl+C, the clip doesn't play: >>>> >>>> gst-launch-0.10 videotestsrc ! >>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace >>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>> >>>> But this pipeline produces a valid video file: >>>> >>>> gst-launch-0.10 videotestsrc num-buffers=200 ! >>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace >>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>> >>>> > >>>> > 2: I want encode the video to H264, so I do like this: >>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >>>> > location=record.avi >>>> >>>> How about adding the format to the caps, e.g . >>>> v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >>>> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >>>> location=record.avi ? >>>> >>>> BR, >>>> Topi >>>> >>>> > But when I run the pipeline I just get the error info like this: >>>> > Setting pipeline to PAUSED ... >>>> > Pipeline is live and does not need PREROLL ... >>>> > Setting pipeline to PLAYING ... >>>> > New clock: GstSystemClock >>>> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>> Internal >>>> > data flow error. >>>> > Additional debug info: >>>> > gstbasesrc.c(2550): gst_base_src_loop (): >>>> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>> > streaming task paused, reason not-negotiated (-4) >>>> > Execution ended after 1727716724 ns. >>>> > Setting pipeline to PAUSED ... >>>> > Setting pipeline to READY ... >>>> > Setting pipeline to NULL ... >>>> > Freeing pipeline ... >>>> > >>>> > Some one can help me fix the issues? Tks :) >>>> > >>>> > >>>> > On 12/22/10, Timothy Braun < >>>> braunsquared at gmail.com >>>> > braunsquared at gmail.com>> wrote: >>>> > > H264 is encoded with x264enc element. >>>> > > >>>> > > Sent from my iPhone >>>> > > >>>> > > On Dec 22, 2010, at 3:10 AM, wl2776 < >>>> wl2776 at gmail.com >>>> > wl2776 at gmail.com>> wrote: >>>> > > >>>> > >> >>>> > >> >>>> > >> michael young wrote: >>>> > >>> >>>> > >>> And now , I want to record the video to a MP4 file with the h264 >>>> > encoding >>>> > >>> format , How can I do ? >>>> > >>> >>>> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael < >>>> mcygogo at gmail.com >>>> > mcygogo at gmail.com>> wrote: >>>> > >>> >>>> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >>>> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>> xvimagesink >>>> > >>> >>>> > >>> >>>> > >> >>>> > >> Looks like you can't, because h264 encoder is unavailable. >>>> > >> gst-inspect | grep h264 >>>> > >> doesn't show it. >>>> > >> However, general idea is to encode, then mux, and then save to a >>>> file. >>>> > >> gst-launch v4l2src device=/dev/video0 ! >>>> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>> ffenc_mpeg4 ! >>>> > >> mp4mux !filesink location=/home/user/file. >>>> > >> >>>> > >> -- >>>> > >> View this message in context: >>>> > >> >>>> > >>>> >>>> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >>>> > >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >>>> > >> >>>> > >> >>>> > >>>> ------------------------------------------------------------------------------ >>>> > >> Forrester recently released a report on the Return on Investment >>>> > (ROI) of >>>> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>> break-even >>>> > >> within 7 months. Over 3 million businesses have gone Google with >>>> Google >>>> > >> Apps: >>>> > >> an online email calendar, and document program that's accessible >>>> > from your >>>> > >> >>>> > >> browser. Read the Forrester report: >>>> >>>> http://p.sf.net/sfu/googleapps-sfnew >>>> > >> _______________________________________________ >>>> > >> gstreamer-devel mailing list >>>> > >> >>>> gstreamer-devel at lists.sourceforge.net >>>> > >>>> gstreamer-devel at lists.sourceforge.net> >>>> > >> >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> > > >>>> > > >>>> > >>>> ------------------------------------------------------------------------------ >>>> > > Forrester recently released a report on the Return on Investment >>>> (ROI) of >>>> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>> break-even >>>> > > within 7 months. Over 3 million businesses have gone Google with >>>> Google >>>> > > Apps: >>>> > > an online email calendar, and document program that's accessible >>>> from >>>> > your >>>> > > browser. Read the Forrester report: >>>> >>>> http://p.sf.net/sfu/googleapps-sfnew >>>> > > _______________________________________________ >>>> > > gstreamer-devel mailing list >>>> > > >>>> gstreamer-devel at lists.sourceforge.net >>>> > >>>> gstreamer-devel at lists.sourceforge.net> >>>> > > >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> > > >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Forrester recently released a report on the Return on Investment (ROI) >>>> of >>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>>> within 7 months. Over 3 million businesses have gone Google with Google >>>> Apps: >>>> an online email calendar, and document program that's accessible from >>>> your >>>> browser. Read the Forrester report: >>>> >>>> http://p.sf.net/sfu/googleapps-sfnew >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> >>>> gstreamer-devel at lists.sourceforge.net >>>> >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google >>> Apps: >>> an online email calendar, and document program that's accessible from >>> your >>> browser. Read the Forrester report: >>> >>> http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> >>> gstreamer-devel at lists.sourceforge.net >>> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google >> Apps: >> an online email calendar, and document program that's accessible from your >> >> browser. Read the Forrester report: >> >> http://p.sf.net/sfu/googleapps-sfnew >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google >> Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From leon at tim-online.nl Wed Dec 22 23:25:32 2010 From: leon at tim-online.nl (Leon Bogaert) Date: Wed, 22 Dec 2010 22:25:32 +0000 Subject: [gst-devel] decodebin & gapless Message-ID: Hi all, Could anyone give me a hand with how to use decodebin and make gapless playback work? I've tried searching for some example code but I couldn't really find anything. Thanks in advance! Regards, Leon From mcygogo at gmail.com Thu Dec 23 02:46:31 2010 From: mcygogo at gmail.com (michael) Date: Thu, 23 Dec 2010 09:46:31 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> Message-ID: neither qtmux nor mp4mux is not work well On Wed, Dec 22, 2010 at 11:24 PM, Timothy Braun wrote: > Not qtmux, use mp4mux. > > > On Wed, Dec 22, 2010 at 10:10 AM, michael wrote: > >> I have tested changing the muxer to qtmux, but it still not work >> >> >> On Wed, Dec 22, 2010 at 10:58 PM, Timothy Braun wrote: >> >>> I was never able to get the ffmpeg mpeg4 mixer to work quite right. I >>> found that mp4mux, an extension of qtmux, works much better. If you inspect >>> the ffmpeg mp4 muxer they actually suggest that you use a different one as >>> it's unstable. >>> >>> Best, >>> Tim >>> >>> Sent from my iPhone >>> >>> On Dec 22, 2010, at 8:27 AM, michael wrote: >>> >>> I just ran it : >>> gst-launch -e v4l2src num-buffers=200 ! >>> video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! >>> ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 >>> And got a mp4 file , but it still cannot be playbacked, and give the >>> error info: >>> Playing record.mp4. >>> libavformat file format detected. >>> [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to >>> guess time_increment_bits >>> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >>> [mpeg4 @ 0x9376c10]looks like this file was encoded with >>> (divx4/(old)xvid/opendivx) -> forcing low_delay flag >>> [mpeg4 @ 0x9376c10]picture size invalid (0x0) >>> [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) >>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters >>> (Video: mpeg4, yuv420p) >>> LAVF_header: av_find_stream_info() failed >>> ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>> Quicktime/MOV file format detected. >>> [mov] Video stream found, -vid 0 >>> VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) >>> open: No such file or directory >>> [MGA] Couldn't open: /dev/mga_vid >>> open: No such file or directory >>> [MGA] Couldn't open: /dev/mga_vid >>> [VO_TDFXFB] Can't open /dev/fb0: Permission denied. >>> [VO_3DFX] Unable to open /dev/3dfx. >>> >>> ========================================================================== >>> Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family >>> Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) >>> >>> ========================================================================== >>> Audio: no sound >>> Starting playback... >>> VDec: vo config request - 640 x 480 (preferred colorspace: Planar YV12) >>> VDec: using Planar YV12 as output csp (no 0) >>> Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. >>> VO: [xv] 640x480 => 640x480 Planar YV12 >>> V: 0.0 200/200 ??% ??% ??,?% 0 0 >>> >>> Exiting... (End of file) >>> >>> But the error info not like the old one! >>> >>> On Wed, Dec 22, 2010 at 9:17 PM, liJin < >>> lijinsyam at gmail.com> wrote: >>> >>>> I think thats the issue... >>>> >>>> >>>> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>> framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink >>>> location=1.mp4 >>>> >>>> run with* -e *option >>>> >>>> >>>> >>>> On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi < >>>> Topi.Santakivi at digia.com> wrote: >>>> >>>>> On 12/22/2010 12:18 PM, michael wrote: >>>>> > Now I can record the video to a AVI file with mpeg4 encoding by : >>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink >>>>> > location=record.avi >>>>> > >>>>> > But I got two issues: >>>>> > 1: I want record as a MP4 file, I do it like this: >>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink >>>>> > location=record.avi >>>>> > I can record the video , but when I playback it by mplayer, I got a >>>>> error: >>>>> > Playing ./record.mp4. >>>>> > libavformat file format detected. >>>>> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >>>>> > LAVF_header: av_open_input_stream() failed >>>>> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>> >>>>> When recording, with the above gst-launch line, do you stop >>>>> the pipeline with Ctrl+C..? >>>>> >>>>> It could be that the file container doesn't get fully written >>>>> in case of such an interrupt and that's why the clip doesn't play. >>>>> >>>>> For example, if you use this pipeline and >>>>> stop it with Ctrl+C, the clip doesn't play: >>>>> >>>>> gst-launch-0.10 videotestsrc ! >>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>> ffmpegcolorspace >>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>> >>>>> But this pipeline produces a valid video file: >>>>> >>>>> gst-launch-0.10 videotestsrc num-buffers=200 ! >>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>> ffmpegcolorspace >>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>> >>>>> > >>>>> > 2: I want encode the video to H264, so I do like this: >>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >>>>> > location=record.avi >>>>> >>>>> How about adding the format to the caps, e.g . >>>>> v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >>>>> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >>>>> location=record.avi ? >>>>> >>>>> BR, >>>>> Topi >>>>> >>>>> > But when I run the pipeline I just get the error info like this: >>>>> > Setting pipeline to PAUSED ... >>>>> > Pipeline is live and does not need PREROLL ... >>>>> > Setting pipeline to PLAYING ... >>>>> > New clock: GstSystemClock >>>>> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>> Internal >>>>> > data flow error. >>>>> > Additional debug info: >>>>> > gstbasesrc.c(2550): gst_base_src_loop (): >>>>> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>> > streaming task paused, reason not-negotiated (-4) >>>>> > Execution ended after 1727716724 ns. >>>>> > Setting pipeline to PAUSED ... >>>>> > Setting pipeline to READY ... >>>>> > Setting pipeline to NULL ... >>>>> > Freeing pipeline ... >>>>> > >>>>> > Some one can help me fix the issues? Tks :) >>>>> > >>>>> > >>>>> > On 12/22/10, Timothy Braun < >>>>> braunsquared at gmail.com >>>>> > braunsquared at gmail.com>> wrote: >>>>> > > H264 is encoded with x264enc element. >>>>> > > >>>>> > > Sent from my iPhone >>>>> > > >>>>> > > On Dec 22, 2010, at 3:10 AM, wl2776 < >>>>> wl2776 at gmail.com >>>>> > wl2776 at gmail.com>> wrote: >>>>> > > >>>>> > >> >>>>> > >> >>>>> > >> michael young wrote: >>>>> > >>> >>>>> > >>> And now , I want to record the video to a MP4 file with the h264 >>>>> > encoding >>>>> > >>> format , How can I do ? >>>>> > >>> >>>>> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael < >>>>> mcygogo at gmail.com >>>>> > mcygogo at gmail.com>> wrote: >>>>> > >>> >>>>> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >>>>> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>> xvimagesink >>>>> > >>> >>>>> > >>> >>>>> > >> >>>>> > >> Looks like you can't, because h264 encoder is unavailable. >>>>> > >> gst-inspect | grep h264 >>>>> > >> doesn't show it. >>>>> > >> However, general idea is to encode, then mux, and then save to a >>>>> file. >>>>> > >> gst-launch v4l2src device=/dev/video0 ! >>>>> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>> ffenc_mpeg4 ! >>>>> > >> mp4mux !filesink location=/home/user/file. >>>>> > >> >>>>> > >> -- >>>>> > >> View this message in context: >>>>> > >> >>>>> > >>>>> >>>>> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >>>>> > >> Sent from the GStreamer-devel mailing list archive at Nabble.com >>>>> . >>>>> > >> >>>>> > >> >>>>> > >>>>> ------------------------------------------------------------------------------ >>>>> > >> Forrester recently released a report on the Return on Investment >>>>> > (ROI) of >>>>> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>> break-even >>>>> > >> within 7 months. Over 3 million businesses have gone Google with >>>>> Google >>>>> > >> Apps: >>>>> > >> an online email calendar, and document program that's accessible >>>>> > from your >>>>> > >> >>>>> > >> browser. Read the Forrester report: >>>>> >>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>> > >> _______________________________________________ >>>>> > >> gstreamer-devel mailing list >>>>> > >> >>>>> gstreamer-devel at lists.sourceforge.net >>>>> > >>>>> gstreamer-devel at lists.sourceforge.net> >>>>> > >> >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> > > >>>>> > > >>>>> > >>>>> ------------------------------------------------------------------------------ >>>>> > > Forrester recently released a report on the Return on Investment >>>>> (ROI) of >>>>> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>> break-even >>>>> > > within 7 months. Over 3 million businesses have gone Google with >>>>> Google >>>>> > > Apps: >>>>> > > an online email calendar, and document program that's accessible >>>>> from >>>>> > your >>>>> > > browser. Read the Forrester report: >>>>> >>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>> > > _______________________________________________ >>>>> > > gstreamer-devel mailing list >>>>> > > >>>>> gstreamer-devel at lists.sourceforge.net >>>>> > >>>>> gstreamer-devel at lists.sourceforge.net> >>>>> > > >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> > > >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>> of >>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>> break-even >>>>> within 7 months. Over 3 million businesses have gone Google with >>>>> Google Apps: >>>>> an online email calendar, and document program that's accessible from >>>>> your >>>>> browser. Read the Forrester report: >>>>> >>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> >>>>> gstreamer-devel at lists.sourceforge.net >>>>> >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Forrester recently released a report on the Return on Investment (ROI) >>>> of >>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>>> within 7 months. Over 3 million businesses have gone Google with Google >>>> Apps: >>>> an online email calendar, and document program that's accessible from >>>> your >>>> browser. Read the Forrester report: >>>> >>>> http://p.sf.net/sfu/googleapps-sfnew >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> >>>> gstreamer-devel at lists.sourceforge.net >>>> >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google >>> Apps: >>> an online email calendar, and document program that's accessible from >>> your >>> browser. Read the Forrester report: >>> >>> http://p.sf.net/sfu/googleapps-sfnew >>> >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google >>> Apps: >>> an online email calendar, and document program that's accessible from >>> your >>> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google >> Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google > Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Thu Dec 23 07:38:22 2010 From: mcygogo at gmail.com (michael) Date: Thu, 23 Dec 2010 14:38:22 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> Message-ID: Is there nobody know how to record video from camera to a MP4 container with the H264 encoding ? On Thu, Dec 23, 2010 at 9:46 AM, michael wrote: > neither qtmux nor mp4mux is not work well > > > On Wed, Dec 22, 2010 at 11:24 PM, Timothy Braun wrote: > >> Not qtmux, use mp4mux. >> >> >> On Wed, Dec 22, 2010 at 10:10 AM, michael wrote: >> >>> I have tested changing the muxer to qtmux, but it still not work >>> >>> >>> On Wed, Dec 22, 2010 at 10:58 PM, Timothy Braun wrote: >>> >>>> I was never able to get the ffmpeg mpeg4 mixer to work quite right. I >>>> found that mp4mux, an extension of qtmux, works much better. If you inspect >>>> the ffmpeg mp4 muxer they actually suggest that you use a different one as >>>> it's unstable. >>>> >>>> Best, >>>> Tim >>>> >>>> Sent from my iPhone >>>> >>>> On Dec 22, 2010, at 8:27 AM, michael wrote: >>>> >>>> I just ran it : >>>> gst-launch -e v4l2src num-buffers=200 ! >>>> video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! >>>> ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 >>>> And got a mp4 file , but it still cannot be playbacked, and give the >>>> error info: >>>> Playing record.mp4. >>>> libavformat file format detected. >>>> [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to >>>> guess time_increment_bits >>>> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >>>> [mpeg4 @ 0x9376c10]looks like this file was encoded with >>>> (divx4/(old)xvid/opendivx) -> forcing low_delay flag >>>> [mpeg4 @ 0x9376c10]picture size invalid (0x0) >>>> [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) >>>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters >>>> (Video: mpeg4, yuv420p) >>>> LAVF_header: av_find_stream_info() failed >>>> ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>> Quicktime/MOV file format detected. >>>> [mov] Video stream found, -vid 0 >>>> VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) >>>> open: No such file or directory >>>> [MGA] Couldn't open: /dev/mga_vid >>>> open: No such file or directory >>>> [MGA] Couldn't open: /dev/mga_vid >>>> [VO_TDFXFB] Can't open /dev/fb0: Permission denied. >>>> [VO_3DFX] Unable to open /dev/3dfx. >>>> >>>> ========================================================================== >>>> Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family >>>> Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) >>>> >>>> ========================================================================== >>>> Audio: no sound >>>> Starting playback... >>>> VDec: vo config request - 640 x 480 (preferred colorspace: Planar YV12) >>>> VDec: using Planar YV12 as output csp (no 0) >>>> Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. >>>> VO: [xv] 640x480 => 640x480 Planar YV12 >>>> V: 0.0 200/200 ??% ??% ??,?% 0 0 >>>> >>>> Exiting... (End of file) >>>> >>>> But the error info not like the old one! >>>> >>>> On Wed, Dec 22, 2010 at 9:17 PM, liJin < >>>> lijinsyam at gmail.com> wrote: >>>> >>>>> I think thats the issue... >>>>> >>>>> >>>>> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>> framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink >>>>> location=1.mp4 >>>>> >>>>> run with* -e *option >>>>> >>>>> >>>>> >>>>> On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi < >>>>> Topi.Santakivi at digia.com> wrote: >>>>> >>>>>> On 12/22/2010 12:18 PM, michael wrote: >>>>>> > Now I can record the video to a AVI file with mpeg4 encoding by : >>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink >>>>>> > location=record.avi >>>>>> > >>>>>> > But I got two issues: >>>>>> > 1: I want record as a MP4 file, I do it like this: >>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink >>>>>> > location=record.avi >>>>>> > I can record the video , but when I playback it by mplayer, I got a >>>>>> error: >>>>>> > Playing ./record.mp4. >>>>>> > libavformat file format detected. >>>>>> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >>>>>> > LAVF_header: av_open_input_stream() failed >>>>>> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>>> >>>>>> When recording, with the above gst-launch line, do you stop >>>>>> the pipeline with Ctrl+C..? >>>>>> >>>>>> It could be that the file container doesn't get fully written >>>>>> in case of such an interrupt and that's why the clip doesn't play. >>>>>> >>>>>> For example, if you use this pipeline and >>>>>> stop it with Ctrl+C, the clip doesn't play: >>>>>> >>>>>> gst-launch-0.10 videotestsrc ! >>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>> ffmpegcolorspace >>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>> >>>>>> But this pipeline produces a valid video file: >>>>>> >>>>>> gst-launch-0.10 videotestsrc num-buffers=200 ! >>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>> ffmpegcolorspace >>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>> >>>>>> > >>>>>> > 2: I want encode the video to H264, so I do like this: >>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >>>>>> > location=record.avi >>>>>> >>>>>> How about adding the format to the caps, e.g . >>>>>> v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >>>>>> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >>>>>> location=record.avi ? >>>>>> >>>>>> BR, >>>>>> Topi >>>>>> >>>>>> > But when I run the pipeline I just get the error info like this: >>>>>> > Setting pipeline to PAUSED ... >>>>>> > Pipeline is live and does not need PREROLL ... >>>>>> > Setting pipeline to PLAYING ... >>>>>> > New clock: GstSystemClock >>>>>> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>> Internal >>>>>> > data flow error. >>>>>> > Additional debug info: >>>>>> > gstbasesrc.c(2550): gst_base_src_loop (): >>>>>> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>> > streaming task paused, reason not-negotiated (-4) >>>>>> > Execution ended after 1727716724 ns. >>>>>> > Setting pipeline to PAUSED ... >>>>>> > Setting pipeline to READY ... >>>>>> > Setting pipeline to NULL ... >>>>>> > Freeing pipeline ... >>>>>> > >>>>>> > Some one can help me fix the issues? Tks :) >>>>>> > >>>>>> > >>>>>> > On 12/22/10, Timothy Braun < >>>>>> braunsquared at gmail.com >>>>>> > braunsquared at gmail.com>> wrote: >>>>>> > > H264 is encoded with x264enc element. >>>>>> > > >>>>>> > > Sent from my iPhone >>>>>> > > >>>>>> > > On Dec 22, 2010, at 3:10 AM, wl2776 < >>>>>> wl2776 at gmail.com >>>>>> > wl2776 at gmail.com>> wrote: >>>>>> > > >>>>>> > >> >>>>>> > >> >>>>>> > >> michael young wrote: >>>>>> > >>> >>>>>> > >>> And now , I want to record the video to a MP4 file with the >>>>>> h264 >>>>>> > encoding >>>>>> > >>> format , How can I do ? >>>>>> > >>> >>>>>> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael < >>>>>> mcygogo at gmail.com >>>>>> > mcygogo at gmail.com>> wrote: >>>>>> > >>> >>>>>> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >>>>>> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>> xvimagesink >>>>>> > >>> >>>>>> > >>> >>>>>> > >> >>>>>> > >> Looks like you can't, because h264 encoder is unavailable. >>>>>> > >> gst-inspect | grep h264 >>>>>> > >> doesn't show it. >>>>>> > >> However, general idea is to encode, then mux, and then save to a >>>>>> file. >>>>>> > >> gst-launch v4l2src device=/dev/video0 ! >>>>>> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>> ffenc_mpeg4 ! >>>>>> > >> mp4mux !filesink location=/home/user/file. >>>>>> > >> >>>>>> > >> -- >>>>>> > >> View this message in context: >>>>>> > >> >>>>>> > >>>>>> >>>>>> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >>>>>> > >> Sent from the GStreamer-devel mailing list archive at >>>>>> Nabble.com. >>>>>> > >> >>>>>> > >> >>>>>> > >>>>>> ------------------------------------------------------------------------------ >>>>>> > >> Forrester recently released a report on the Return on Investment >>>>>> > (ROI) of >>>>>> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>> break-even >>>>>> > >> within 7 months. Over 3 million businesses have gone Google >>>>>> with Google >>>>>> > >> Apps: >>>>>> > >> an online email calendar, and document program that's accessible >>>>>> > from your >>>>>> > >> >>>>>> > >> browser. Read the Forrester report: >>>>>> >>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>> > >> _______________________________________________ >>>>>> > >> gstreamer-devel mailing list >>>>>> > >> >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> > >>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>> > >> >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> > > >>>>>> > > >>>>>> > >>>>>> ------------------------------------------------------------------------------ >>>>>> > > Forrester recently released a report on the Return on Investment >>>>>> (ROI) of >>>>>> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>> break-even >>>>>> > > within 7 months. Over 3 million businesses have gone Google with >>>>>> Google >>>>>> > > Apps: >>>>>> > > an online email calendar, and document program that's accessible >>>>>> from >>>>>> > your >>>>>> > > browser. Read the Forrester report: >>>>>> >>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>> > > _______________________________________________ >>>>>> > > gstreamer-devel mailing list >>>>>> > > >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> > >>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>> > > >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> > > >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>>> of >>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>> break-even >>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>> Google Apps: >>>>>> an online email calendar, and document program that's accessible from >>>>>> your >>>>>> browser. Read the Forrester report: >>>>>> >>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>> of >>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>> break-even >>>>> within 7 months. Over 3 million businesses have gone Google with >>>>> Google Apps: >>>>> an online email calendar, and document program that's accessible from >>>>> your >>>>> browser. Read the Forrester report: >>>>> >>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> >>>>> gstreamer-devel at lists.sourceforge.net >>>>> >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Forrester recently released a report on the Return on Investment (ROI) >>>> of >>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>>> within 7 months. Over 3 million businesses have gone Google with Google >>>> Apps: >>>> an online email calendar, and document program that's accessible from >>>> your >>>> browser. Read the Forrester report: >>>> >>>> http://p.sf.net/sfu/googleapps-sfnew >>>> >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Forrester recently released a report on the Return on Investment (ROI) >>>> of >>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>>> within 7 months. Over 3 million businesses have gone Google with Google >>>> Apps: >>>> an online email calendar, and document program that's accessible from >>>> your >>>> browser. Read the Forrester report: >>>> http://p.sf.net/sfu/googleapps-sfnew >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google >>> Apps: >>> an online email calendar, and document program that's accessible from >>> your >>> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google >> Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From lijinsyam at gmail.com Thu Dec 23 07:42:59 2010 From: lijinsyam at gmail.com (liJin) Date: Thu, 23 Dec 2010 12:12:59 +0530 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> Message-ID: Hi gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, framerate=30/1' ! ffmpegcolorspace ! x264enc ! mp4mux ! filesink location=1.mp4 This pipeline working fine (4 me)......:) On Thu, Dec 23, 2010 at 12:08 PM, michael wrote: > Is there nobody know how to record video from camera to a MP4 container > with the H264 encoding ? > > > On Thu, Dec 23, 2010 at 9:46 AM, michael wrote: > >> neither qtmux nor mp4mux is not work well >> >> >> On Wed, Dec 22, 2010 at 11:24 PM, Timothy Braun wrote: >> >>> Not qtmux, use mp4mux. >>> >>> >>> On Wed, Dec 22, 2010 at 10:10 AM, michael wrote: >>> >>>> I have tested changing the muxer to qtmux, but it still not work >>>> >>>> >>>> On Wed, Dec 22, 2010 at 10:58 PM, Timothy Braun >>> > wrote: >>>> >>>>> I was never able to get the ffmpeg mpeg4 mixer to work quite right. I >>>>> found that mp4mux, an extension of qtmux, works much better. If you inspect >>>>> the ffmpeg mp4 muxer they actually suggest that you use a different one as >>>>> it's unstable. >>>>> >>>>> Best, >>>>> Tim >>>>> >>>>> Sent from my iPhone >>>>> >>>>> On Dec 22, 2010, at 8:27 AM, michael wrote: >>>>> >>>>> I just ran it : >>>>> gst-launch -e v4l2src num-buffers=200 ! >>>>> video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! >>>>> ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 >>>>> And got a mp4 file , but it still cannot be playbacked, and give the >>>>> error info: >>>>> Playing record.mp4. >>>>> libavformat file format detected. >>>>> [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to >>>>> guess time_increment_bits >>>>> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >>>>> [mpeg4 @ 0x9376c10]looks like this file was encoded with >>>>> (divx4/(old)xvid/opendivx) -> forcing low_delay flag >>>>> [mpeg4 @ 0x9376c10]picture size invalid (0x0) >>>>> [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) >>>>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters >>>>> (Video: mpeg4, yuv420p) >>>>> LAVF_header: av_find_stream_info() failed >>>>> ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>> Quicktime/MOV file format detected. >>>>> [mov] Video stream found, -vid 0 >>>>> VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) >>>>> open: No such file or directory >>>>> [MGA] Couldn't open: /dev/mga_vid >>>>> open: No such file or directory >>>>> [MGA] Couldn't open: /dev/mga_vid >>>>> [VO_TDFXFB] Can't open /dev/fb0: Permission denied. >>>>> [VO_3DFX] Unable to open /dev/3dfx. >>>>> >>>>> ========================================================================== >>>>> Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family >>>>> Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) >>>>> >>>>> ========================================================================== >>>>> Audio: no sound >>>>> Starting playback... >>>>> VDec: vo config request - 640 x 480 (preferred colorspace: Planar YV12) >>>>> VDec: using Planar YV12 as output csp (no 0) >>>>> Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. >>>>> VO: [xv] 640x480 => 640x480 Planar YV12 >>>>> V: 0.0 200/200 ??% ??% ??,?% 0 0 >>>>> >>>>> Exiting... (End of file) >>>>> >>>>> But the error info not like the old one! >>>>> >>>>> On Wed, Dec 22, 2010 at 9:17 PM, liJin < >>>>> lijinsyam at gmail.com> wrote: >>>>> >>>>>> I think thats the issue... >>>>>> >>>>>> >>>>>> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>> framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink >>>>>> location=1.mp4 >>>>>> >>>>>> run with* -e *option >>>>>> >>>>>> >>>>>> >>>>>> On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi < >>>>>> Topi.Santakivi at digia.com> wrote: >>>>>> >>>>>>> On 12/22/2010 12:18 PM, michael wrote: >>>>>>> > Now I can record the video to a AVI file with mpeg4 encoding by : >>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink >>>>>>> > location=record.avi >>>>>>> > >>>>>>> > But I got two issues: >>>>>>> > 1: I want record as a MP4 file, I do it like this: >>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink >>>>>>> > location=record.avi >>>>>>> > I can record the video , but when I playback it by mplayer, I got a >>>>>>> error: >>>>>>> > Playing ./record.mp4. >>>>>>> > libavformat file format detected. >>>>>>> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >>>>>>> > LAVF_header: av_open_input_stream() failed >>>>>>> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>>>> >>>>>>> When recording, with the above gst-launch line, do you stop >>>>>>> the pipeline with Ctrl+C..? >>>>>>> >>>>>>> It could be that the file container doesn't get fully written >>>>>>> in case of such an interrupt and that's why the clip doesn't play. >>>>>>> >>>>>>> For example, if you use this pipeline and >>>>>>> stop it with Ctrl+C, the clip doesn't play: >>>>>>> >>>>>>> gst-launch-0.10 videotestsrc ! >>>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>>> ffmpegcolorspace >>>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>>> >>>>>>> But this pipeline produces a valid video file: >>>>>>> >>>>>>> gst-launch-0.10 videotestsrc num-buffers=200 ! >>>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>>> ffmpegcolorspace >>>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>>> >>>>>>> > >>>>>>> > 2: I want encode the video to H264, so I do like this: >>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >>>>>>> > location=record.avi >>>>>>> >>>>>>> How about adding the format to the caps, e.g . >>>>>>> v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >>>>>>> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >>>>>>> location=record.avi ? >>>>>>> >>>>>>> BR, >>>>>>> Topi >>>>>>> >>>>>>> > But when I run the pipeline I just get the error info like this: >>>>>>> > Setting pipeline to PAUSED ... >>>>>>> > Pipeline is live and does not need PREROLL ... >>>>>>> > Setting pipeline to PLAYING ... >>>>>>> > New clock: GstSystemClock >>>>>>> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>>> Internal >>>>>>> > data flow error. >>>>>>> > Additional debug info: >>>>>>> > gstbasesrc.c(2550): gst_base_src_loop (): >>>>>>> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>>> > streaming task paused, reason not-negotiated (-4) >>>>>>> > Execution ended after 1727716724 ns. >>>>>>> > Setting pipeline to PAUSED ... >>>>>>> > Setting pipeline to READY ... >>>>>>> > Setting pipeline to NULL ... >>>>>>> > Freeing pipeline ... >>>>>>> > >>>>>>> > Some one can help me fix the issues? Tks :) >>>>>>> > >>>>>>> > >>>>>>> > On 12/22/10, Timothy Braun < >>>>>>> braunsquared at gmail.com >>>>>>> > braunsquared at gmail.com>> wrote: >>>>>>> > > H264 is encoded with x264enc element. >>>>>>> > > >>>>>>> > > Sent from my iPhone >>>>>>> > > >>>>>>> > > On Dec 22, 2010, at 3:10 AM, wl2776 < >>>>>>> wl2776 at gmail.com >>>>>>> > wl2776 at gmail.com>> wrote: >>>>>>> > > >>>>>>> > >> >>>>>>> > >> >>>>>>> > >> michael young wrote: >>>>>>> > >>> >>>>>>> > >>> And now , I want to record the video to a MP4 file with the >>>>>>> h264 >>>>>>> > encoding >>>>>>> > >>> format , How can I do ? >>>>>>> > >>> >>>>>>> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael < >>>>>>> mcygogo at gmail.com >>>>>>> > mcygogo at gmail.com>> wrote: >>>>>>> > >>> >>>>>>> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >>>>>>> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>>> xvimagesink >>>>>>> > >>> >>>>>>> > >>> >>>>>>> > >> >>>>>>> > >> Looks like you can't, because h264 encoder is unavailable. >>>>>>> > >> gst-inspect | grep h264 >>>>>>> > >> doesn't show it. >>>>>>> > >> However, general idea is to encode, then mux, and then save to >>>>>>> a file. >>>>>>> > >> gst-launch v4l2src device=/dev/video0 ! >>>>>>> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>>> ffenc_mpeg4 ! >>>>>>> > >> mp4mux !filesink location=/home/user/file. >>>>>>> > >> >>>>>>> > >> -- >>>>>>> > >> View this message in context: >>>>>>> > >> >>>>>>> > >>>>>>> >>>>>>> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >>>>>>> > >> Sent from the GStreamer-devel mailing list archive at >>>>>>> Nabble.com. >>>>>>> > >> >>>>>>> > >> >>>>>>> > >>>>>>> ------------------------------------------------------------------------------ >>>>>>> > >> Forrester recently released a report on the Return on >>>>>>> Investment >>>>>>> > (ROI) of >>>>>>> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>> break-even >>>>>>> > >> within 7 months. Over 3 million businesses have gone Google >>>>>>> with Google >>>>>>> > >> Apps: >>>>>>> > >> an online email calendar, and document program that's >>>>>>> accessible >>>>>>> > from your >>>>>>> > >> >>>>>>> > >> browser. Read the Forrester report: >>>>>>> >>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>> > >> _______________________________________________ >>>>>>> > >> gstreamer-devel mailing list >>>>>>> > >> >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> > >>>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>>> > >> >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> > > >>>>>>> > > >>>>>>> > >>>>>>> ------------------------------------------------------------------------------ >>>>>>> > > Forrester recently released a report on the Return on Investment >>>>>>> (ROI) of >>>>>>> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>> break-even >>>>>>> > > within 7 months. Over 3 million businesses have gone Google >>>>>>> with Google >>>>>>> > > Apps: >>>>>>> > > an online email calendar, and document program that's accessible >>>>>>> from >>>>>>> > your >>>>>>> > > browser. Read the Forrester report: >>>>>>> >>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>> > > _______________________________________________ >>>>>>> > > gstreamer-devel mailing list >>>>>>> > > >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> > >>>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>>> > > >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> > > >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------------ >>>>>>> Forrester recently released a report on the Return on Investment >>>>>>> (ROI) of >>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>> break-even >>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>> Google Apps: >>>>>>> an online email calendar, and document program that's accessible from >>>>>>> your >>>>>>> browser. Read the Forrester report: >>>>>>> >>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>>> of >>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>> break-even >>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>> Google Apps: >>>>>> an online email calendar, and document program that's accessible from >>>>>> your >>>>>> browser. Read the Forrester report: >>>>>> >>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>> of >>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>> break-even >>>>> within 7 months. Over 3 million businesses have gone Google with >>>>> Google Apps: >>>>> an online email calendar, and document program that's accessible from >>>>> your >>>>> browser. Read the Forrester report: >>>>> >>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>> >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> gstreamer-devel at lists.sourceforge.net >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>> of >>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>> break-even >>>>> within 7 months. Over 3 million businesses have gone Google with >>>>> Google Apps: >>>>> an online email calendar, and document program that's accessible from >>>>> your >>>>> browser. Read the Forrester report: >>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> gstreamer-devel at lists.sourceforge.net >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Forrester recently released a report on the Return on Investment (ROI) >>>> of >>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>>> within 7 months. Over 3 million businesses have gone Google with Google >>>> Apps: >>>> an online email calendar, and document program that's accessible from >>>> your >>>> browser. Read the Forrester report: >>>> http://p.sf.net/sfu/googleapps-sfnew >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google >>> Apps: >>> an online email calendar, and document program that's accessible from >>> your >>> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Thu Dec 23 07:52:08 2010 From: mcygogo at gmail.com (michael) Date: Thu, 23 Dec 2010 14:52:08 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> Message-ID: Really!!?? but it's not work on my book!! God,, this issue drives me crazy!! :( On Thu, Dec 23, 2010 at 2:42 PM, liJin wrote: > Hi > > > gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, > framerate=30/1' ! ffmpegcolorspace ! x264enc ! mp4mux ! filesink > location=1.mp4 > > This pipeline working fine (4 me)......:) > > > On Thu, Dec 23, 2010 at 12:08 PM, michael wrote: > >> Is there nobody know how to record video from camera to a MP4 container >> with the H264 encoding ? >> >> >> On Thu, Dec 23, 2010 at 9:46 AM, michael wrote: >> >>> neither qtmux nor mp4mux is not work well >>> >>> >>> On Wed, Dec 22, 2010 at 11:24 PM, Timothy Braun wrote: >>> >>>> Not qtmux, use mp4mux. >>>> >>>> >>>> On Wed, Dec 22, 2010 at 10:10 AM, michael wrote: >>>> >>>>> I have tested changing the muxer to qtmux, but it still not work >>>>> >>>>> >>>>> On Wed, Dec 22, 2010 at 10:58 PM, Timothy Braun < >>>>> braunsquared at gmail.com> wrote: >>>>> >>>>>> I was never able to get the ffmpeg mpeg4 mixer to work quite right. I >>>>>> found that mp4mux, an extension of qtmux, works much better. If you inspect >>>>>> the ffmpeg mp4 muxer they actually suggest that you use a different one as >>>>>> it's unstable. >>>>>> >>>>>> Best, >>>>>> Tim >>>>>> >>>>>> Sent from my iPhone >>>>>> >>>>>> On Dec 22, 2010, at 8:27 AM, michael wrote: >>>>>> >>>>>> I just ran it : >>>>>> gst-launch -e v4l2src num-buffers=200 ! >>>>>> video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! >>>>>> ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 >>>>>> And got a mp4 file , but it still cannot be playbacked, and give the >>>>>> error info: >>>>>> Playing record.mp4. >>>>>> libavformat file format detected. >>>>>> [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to >>>>>> guess time_increment_bits >>>>>> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >>>>>> [mpeg4 @ 0x9376c10]looks like this file was encoded with >>>>>> (divx4/(old)xvid/opendivx) -> forcing low_delay flag >>>>>> [mpeg4 @ 0x9376c10]picture size invalid (0x0) >>>>>> [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) >>>>>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters >>>>>> (Video: mpeg4, yuv420p) >>>>>> LAVF_header: av_find_stream_info() failed >>>>>> ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>>> Quicktime/MOV file format detected. >>>>>> [mov] Video stream found, -vid 0 >>>>>> VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) >>>>>> open: No such file or directory >>>>>> [MGA] Couldn't open: /dev/mga_vid >>>>>> open: No such file or directory >>>>>> [MGA] Couldn't open: /dev/mga_vid >>>>>> [VO_TDFXFB] Can't open /dev/fb0: Permission denied. >>>>>> [VO_3DFX] Unable to open /dev/3dfx. >>>>>> >>>>>> ========================================================================== >>>>>> Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family >>>>>> Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) >>>>>> >>>>>> ========================================================================== >>>>>> Audio: no sound >>>>>> Starting playback... >>>>>> VDec: vo config request - 640 x 480 (preferred colorspace: Planar >>>>>> YV12) >>>>>> VDec: using Planar YV12 as output csp (no 0) >>>>>> Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. >>>>>> VO: [xv] 640x480 => 640x480 Planar YV12 >>>>>> V: 0.0 200/200 ??% ??% ??,?% 0 0 >>>>>> >>>>>> Exiting... (End of file) >>>>>> >>>>>> But the error info not like the old one! >>>>>> >>>>>> On Wed, Dec 22, 2010 at 9:17 PM, liJin < >>>>>> lijinsyam at gmail.com> wrote: >>>>>> >>>>>>> I think thats the issue... >>>>>>> >>>>>>> >>>>>>> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>> framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink >>>>>>> location=1.mp4 >>>>>>> >>>>>>> run with* -e *option >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi < >>>>>>> Topi.Santakivi at digia.com> wrote: >>>>>>> >>>>>>>> On 12/22/2010 12:18 PM, michael wrote: >>>>>>>> > Now I can record the video to a AVI file with mpeg4 encoding by : >>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink >>>>>>>> > location=record.avi >>>>>>>> > >>>>>>>> > But I got two issues: >>>>>>>> > 1: I want record as a MP4 file, I do it like this: >>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! >>>>>>>> filesink >>>>>>>> > location=record.avi >>>>>>>> > I can record the video , but when I playback it by mplayer, I got >>>>>>>> a error: >>>>>>>> > Playing ./record.mp4. >>>>>>>> > libavformat file format detected. >>>>>>>> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >>>>>>>> > LAVF_header: av_open_input_stream() failed >>>>>>>> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>>>>> >>>>>>>> When recording, with the above gst-launch line, do you stop >>>>>>>> the pipeline with Ctrl+C..? >>>>>>>> >>>>>>>> It could be that the file container doesn't get fully written >>>>>>>> in case of such an interrupt and that's why the clip doesn't play. >>>>>>>> >>>>>>>> For example, if you use this pipeline and >>>>>>>> stop it with Ctrl+C, the clip doesn't play: >>>>>>>> >>>>>>>> gst-launch-0.10 videotestsrc ! >>>>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>>>> ffmpegcolorspace >>>>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>>>> >>>>>>>> But this pipeline produces a valid video file: >>>>>>>> >>>>>>>> gst-launch-0.10 videotestsrc num-buffers=200 ! >>>>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>>>> ffmpegcolorspace >>>>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>>>> >>>>>>>> > >>>>>>>> > 2: I want encode the video to H264, so I do like this: >>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >>>>>>>> > location=record.avi >>>>>>>> >>>>>>>> How about adding the format to the caps, e.g . >>>>>>>> v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >>>>>>>> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >>>>>>>> location=record.avi ? >>>>>>>> >>>>>>>> BR, >>>>>>>> Topi >>>>>>>> >>>>>>>> > But when I run the pipeline I just get the error info like this: >>>>>>>> > Setting pipeline to PAUSED ... >>>>>>>> > Pipeline is live and does not need PREROLL ... >>>>>>>> > Setting pipeline to PLAYING ... >>>>>>>> > New clock: GstSystemClock >>>>>>>> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>>>> Internal >>>>>>>> > data flow error. >>>>>>>> > Additional debug info: >>>>>>>> > gstbasesrc.c(2550): gst_base_src_loop (): >>>>>>>> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>>>> > streaming task paused, reason not-negotiated (-4) >>>>>>>> > Execution ended after 1727716724 ns. >>>>>>>> > Setting pipeline to PAUSED ... >>>>>>>> > Setting pipeline to READY ... >>>>>>>> > Setting pipeline to NULL ... >>>>>>>> > Freeing pipeline ... >>>>>>>> > >>>>>>>> > Some one can help me fix the issues? Tks :) >>>>>>>> > >>>>>>>> > >>>>>>>> > On 12/22/10, Timothy Braun < >>>>>>>> braunsquared at gmail.com >>>>>>>> > braunsquared at gmail.com>> wrote: >>>>>>>> > > H264 is encoded with x264enc element. >>>>>>>> > > >>>>>>>> > > Sent from my iPhone >>>>>>>> > > >>>>>>>> > > On Dec 22, 2010, at 3:10 AM, wl2776 < >>>>>>>> wl2776 at gmail.com >>>>>>>> > wl2776 at gmail.com>> wrote: >>>>>>>> > > >>>>>>>> > >> >>>>>>>> > >> >>>>>>>> > >> michael young wrote: >>>>>>>> > >>> >>>>>>>> > >>> And now , I want to record the video to a MP4 file with the >>>>>>>> h264 >>>>>>>> > encoding >>>>>>>> > >>> format , How can I do ? >>>>>>>> > >>> >>>>>>>> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael < >>>>>>>> mcygogo at gmail.com >>>>>>>> > mcygogo at gmail.com>> wrote: >>>>>>>> > >>> >>>>>>>> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >>>>>>>> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>>>> xvimagesink >>>>>>>> > >>> >>>>>>>> > >>> >>>>>>>> > >> >>>>>>>> > >> Looks like you can't, because h264 encoder is unavailable. >>>>>>>> > >> gst-inspect | grep h264 >>>>>>>> > >> doesn't show it. >>>>>>>> > >> However, general idea is to encode, then mux, and then save to >>>>>>>> a file. >>>>>>>> > >> gst-launch v4l2src device=/dev/video0 ! >>>>>>>> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>>>> ffenc_mpeg4 ! >>>>>>>> > >> mp4mux !filesink location=/home/user/file. >>>>>>>> > >> >>>>>>>> > >> -- >>>>>>>> > >> View this message in context: >>>>>>>> > >> >>>>>>>> > >>>>>>>> >>>>>>>> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >>>>>>>> > >> Sent from the GStreamer-devel mailing list archive at >>>>>>>> Nabble.com. >>>>>>>> > >> >>>>>>>> > >> >>>>>>>> > >>>>>>>> ------------------------------------------------------------------------------ >>>>>>>> > >> Forrester recently released a report on the Return on >>>>>>>> Investment >>>>>>>> > (ROI) of >>>>>>>> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>> break-even >>>>>>>> > >> within 7 months. Over 3 million businesses have gone Google >>>>>>>> with Google >>>>>>>> > >> Apps: >>>>>>>> > >> an online email calendar, and document program that's >>>>>>>> accessible >>>>>>>> > from your >>>>>>>> > >> >>>>>>>> > >> browser. Read the Forrester report: >>>>>>>> >>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>> > >> _______________________________________________ >>>>>>>> > >> gstreamer-devel mailing list >>>>>>>> > >> >>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>> > >>>>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>>>> > >> >>>>>>>> >>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>> > > >>>>>>>> > > >>>>>>>> > >>>>>>>> ------------------------------------------------------------------------------ >>>>>>>> > > Forrester recently released a report on the Return on >>>>>>>> Investment (ROI) of >>>>>>>> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>> break-even >>>>>>>> > > within 7 months. Over 3 million businesses have gone Google >>>>>>>> with Google >>>>>>>> > > Apps: >>>>>>>> > > an online email calendar, and document program that's >>>>>>>> accessible from >>>>>>>> > your >>>>>>>> > > browser. Read the Forrester report: >>>>>>>> >>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>> > > _______________________________________________ >>>>>>>> > > gstreamer-devel mailing list >>>>>>>> > > >>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>> > >>>>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>>>> > > >>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>> > > >>>>>>>> >>>>>>>> >>>>>>>> ------------------------------------------------------------------------------ >>>>>>>> Forrester recently released a report on the Return on Investment >>>>>>>> (ROI) of >>>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>> break-even >>>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>>> Google Apps: >>>>>>>> an online email calendar, and document program that's accessible >>>>>>>> from your >>>>>>>> browser. Read the Forrester report: >>>>>>>> >>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>> _______________________________________________ >>>>>>>> gstreamer-devel mailing list >>>>>>>> >>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>> >>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------------ >>>>>>> Forrester recently released a report on the Return on Investment >>>>>>> (ROI) of >>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>> break-even >>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>> Google Apps: >>>>>>> an online email calendar, and document program that's accessible from >>>>>>> your >>>>>>> browser. Read the Forrester report: >>>>>>> >>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>>> of >>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>> break-even >>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>> Google Apps: >>>>>> an online email calendar, and document program that's accessible from >>>>>> your >>>>>> browser. Read the Forrester report: >>>>>> >>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>> >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>>> of >>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>> break-even >>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>> Google Apps: >>>>>> an online email calendar, and document program that's accessible from >>>>>> your >>>>>> browser. Read the Forrester report: >>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>> of >>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>> break-even >>>>> within 7 months. Over 3 million businesses have gone Google with >>>>> Google Apps: >>>>> an online email calendar, and document program that's accessible from >>>>> your >>>>> browser. Read the Forrester report: >>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> gstreamer-devel at lists.sourceforge.net >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Forrester recently released a report on the Return on Investment (ROI) >>>> of >>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>>> within 7 months. Over 3 million businesses have gone Google with Google >>>> Apps: >>>> an online email calendar, and document program that's accessible from >>>> your >>>> browser. Read the Forrester report: >>>> http://p.sf.net/sfu/googleapps-sfnew >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, >> and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Thu Dec 23 08:06:03 2010 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Thu, 23 Dec 2010 12:36:03 +0530 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> Message-ID: Make sure you have all latest Gstreamer core, plugins and supported library packages installed. On Thu, Dec 23, 2010 at 12:22 PM, michael wrote: > Really!!?? but it's not work on my book!! God,, this issue drives me > crazy!! :( > > > On Thu, Dec 23, 2010 at 2:42 PM, liJin wrote: > >> Hi >> >> >> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >> framerate=30/1' ! ffmpegcolorspace ! x264enc ! mp4mux ! filesink >> location=1.mp4 >> >> This pipeline working fine (4 me)......:) >> >> >> On Thu, Dec 23, 2010 at 12:08 PM, michael wrote: >> >>> Is there nobody know how to record video from camera to a MP4 container >>> with the H264 encoding ? >>> >>> >>> On Thu, Dec 23, 2010 at 9:46 AM, michael wrote: >>> >>>> neither qtmux nor mp4mux is not work well >>>> >>>> >>>> On Wed, Dec 22, 2010 at 11:24 PM, Timothy Braun >>> > wrote: >>>> >>>>> Not qtmux, use mp4mux. >>>>> >>>>> >>>>> On Wed, Dec 22, 2010 at 10:10 AM, michael wrote: >>>>> >>>>>> I have tested changing the muxer to qtmux, but it still not work >>>>>> >>>>>> >>>>>> On Wed, Dec 22, 2010 at 10:58 PM, Timothy Braun < >>>>>> braunsquared at gmail.com> wrote: >>>>>> >>>>>>> I was never able to get the ffmpeg mpeg4 mixer to work quite right. I >>>>>>> found that mp4mux, an extension of qtmux, works much better. If you inspect >>>>>>> the ffmpeg mp4 muxer they actually suggest that you use a different one as >>>>>>> it's unstable. >>>>>>> >>>>>>> Best, >>>>>>> Tim >>>>>>> >>>>>>> Sent from my iPhone >>>>>>> >>>>>>> On Dec 22, 2010, at 8:27 AM, michael wrote: >>>>>>> >>>>>>> I just ran it : >>>>>>> gst-launch -e v4l2src num-buffers=200 ! >>>>>>> video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! >>>>>>> ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 >>>>>>> And got a mp4 file , but it still cannot be playbacked, and give the >>>>>>> error info: >>>>>>> Playing record.mp4. >>>>>>> libavformat file format detected. >>>>>>> [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to >>>>>>> guess time_increment_bits >>>>>>> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >>>>>>> [mpeg4 @ 0x9376c10]looks like this file was encoded with >>>>>>> (divx4/(old)xvid/opendivx) -> forcing low_delay flag >>>>>>> [mpeg4 @ 0x9376c10]picture size invalid (0x0) >>>>>>> [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) >>>>>>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters >>>>>>> (Video: mpeg4, yuv420p) >>>>>>> LAVF_header: av_find_stream_info() failed >>>>>>> ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>>>> Quicktime/MOV file format detected. >>>>>>> [mov] Video stream found, -vid 0 >>>>>>> VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) >>>>>>> open: No such file or directory >>>>>>> [MGA] Couldn't open: /dev/mga_vid >>>>>>> open: No such file or directory >>>>>>> [MGA] Couldn't open: /dev/mga_vid >>>>>>> [VO_TDFXFB] Can't open /dev/fb0: Permission denied. >>>>>>> [VO_3DFX] Unable to open /dev/3dfx. >>>>>>> >>>>>>> ========================================================================== >>>>>>> Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family >>>>>>> Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) >>>>>>> >>>>>>> ========================================================================== >>>>>>> Audio: no sound >>>>>>> Starting playback... >>>>>>> VDec: vo config request - 640 x 480 (preferred colorspace: Planar >>>>>>> YV12) >>>>>>> VDec: using Planar YV12 as output csp (no 0) >>>>>>> Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. >>>>>>> VO: [xv] 640x480 => 640x480 Planar YV12 >>>>>>> V: 0.0 200/200 ??% ??% ??,?% 0 0 >>>>>>> >>>>>>> Exiting... (End of file) >>>>>>> >>>>>>> But the error info not like the old one! >>>>>>> >>>>>>> On Wed, Dec 22, 2010 at 9:17 PM, liJin < >>>>>>> lijinsyam at gmail.com> wrote: >>>>>>> >>>>>>>> I think thats the issue... >>>>>>>> >>>>>>>> >>>>>>>> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>> framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink >>>>>>>> location=1.mp4 >>>>>>>> >>>>>>>> run with* -e *option >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi < >>>>>>>> Topi.Santakivi at digia.com> wrote: >>>>>>>> >>>>>>>>> On 12/22/2010 12:18 PM, michael wrote: >>>>>>>>> > Now I can record the video to a AVI file with mpeg4 encoding by : >>>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! >>>>>>>>> filesink >>>>>>>>> > location=record.avi >>>>>>>>> > >>>>>>>>> > But I got two issues: >>>>>>>>> > 1: I want record as a MP4 file, I do it like this: >>>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! >>>>>>>>> filesink >>>>>>>>> > location=record.avi >>>>>>>>> > I can record the video , but when I playback it by mplayer, I got >>>>>>>>> a error: >>>>>>>>> > Playing ./record.mp4. >>>>>>>>> > libavformat file format detected. >>>>>>>>> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >>>>>>>>> > LAVF_header: av_open_input_stream() failed >>>>>>>>> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>>>>>> >>>>>>>>> When recording, with the above gst-launch line, do you stop >>>>>>>>> the pipeline with Ctrl+C..? >>>>>>>>> >>>>>>>>> It could be that the file container doesn't get fully written >>>>>>>>> in case of such an interrupt and that's why the clip doesn't play. >>>>>>>>> >>>>>>>>> For example, if you use this pipeline and >>>>>>>>> stop it with Ctrl+C, the clip doesn't play: >>>>>>>>> >>>>>>>>> gst-launch-0.10 videotestsrc ! >>>>>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>>>>> ffmpegcolorspace >>>>>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>>>>> >>>>>>>>> But this pipeline produces a valid video file: >>>>>>>>> >>>>>>>>> gst-launch-0.10 videotestsrc num-buffers=200 ! >>>>>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>>>>> ffmpegcolorspace >>>>>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>>>>> >>>>>>>>> > >>>>>>>>> > 2: I want encode the video to H264, so I do like this: >>>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >>>>>>>>> > location=record.avi >>>>>>>>> >>>>>>>>> How about adding the format to the caps, e.g . >>>>>>>>> v4l2src ! >>>>>>>>> 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >>>>>>>>> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >>>>>>>>> location=record.avi ? >>>>>>>>> >>>>>>>>> BR, >>>>>>>>> Topi >>>>>>>>> >>>>>>>>> > But when I run the pipeline I just get the error info like this: >>>>>>>>> > Setting pipeline to PAUSED ... >>>>>>>>> > Pipeline is live and does not need PREROLL ... >>>>>>>>> > Setting pipeline to PLAYING ... >>>>>>>>> > New clock: GstSystemClock >>>>>>>>> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>>>>> Internal >>>>>>>>> > data flow error. >>>>>>>>> > Additional debug info: >>>>>>>>> > gstbasesrc.c(2550): gst_base_src_loop (): >>>>>>>>> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>>>>> > streaming task paused, reason not-negotiated (-4) >>>>>>>>> > Execution ended after 1727716724 ns. >>>>>>>>> > Setting pipeline to PAUSED ... >>>>>>>>> > Setting pipeline to READY ... >>>>>>>>> > Setting pipeline to NULL ... >>>>>>>>> > Freeing pipeline ... >>>>>>>>> > >>>>>>>>> > Some one can help me fix the issues? Tks :) >>>>>>>>> > >>>>>>>>> > >>>>>>>>> > On 12/22/10, Timothy Braun < >>>>>>>>> braunsquared at gmail.com >>>>>>>>> > braunsquared at gmail.com>> wrote: >>>>>>>>> > > H264 is encoded with x264enc element. >>>>>>>>> > > >>>>>>>>> > > Sent from my iPhone >>>>>>>>> > > >>>>>>>>> > > On Dec 22, 2010, at 3:10 AM, wl2776 < >>>>>>>>> wl2776 at gmail.com >>>>>>>>> > wl2776 at gmail.com>> wrote: >>>>>>>>> > > >>>>>>>>> > >> >>>>>>>>> > >> >>>>>>>>> > >> michael young wrote: >>>>>>>>> > >>> >>>>>>>>> > >>> And now , I want to record the video to a MP4 file with the >>>>>>>>> h264 >>>>>>>>> > encoding >>>>>>>>> > >>> format , How can I do ? >>>>>>>>> > >>> >>>>>>>>> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael < >>>>>>>>> mcygogo at gmail.com >>>>>>>>> > mcygogo at gmail.com>> wrote: >>>>>>>>> > >>> >>>>>>>>> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 >>>>>>>>> ! >>>>>>>>> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>>>>> xvimagesink >>>>>>>>> > >>> >>>>>>>>> > >>> >>>>>>>>> > >> >>>>>>>>> > >> Looks like you can't, because h264 encoder is unavailable. >>>>>>>>> > >> gst-inspect | grep h264 >>>>>>>>> > >> doesn't show it. >>>>>>>>> > >> However, general idea is to encode, then mux, and then save >>>>>>>>> to a file. >>>>>>>>> > >> gst-launch v4l2src device=/dev/video0 ! >>>>>>>>> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>>>>> ffenc_mpeg4 ! >>>>>>>>> > >> mp4mux !filesink location=/home/user/file. >>>>>>>>> > >> >>>>>>>>> > >> -- >>>>>>>>> > >> View this message in context: >>>>>>>>> > >> >>>>>>>>> > >>>>>>>>> >>>>>>>>> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >>>>>>>>> > >> Sent from the GStreamer-devel mailing list archive at >>>>>>>>> Nabble.com. >>>>>>>>> > >> >>>>>>>>> > >> >>>>>>>>> > >>>>>>>>> ------------------------------------------------------------------------------ >>>>>>>>> > >> Forrester recently released a report on the Return on >>>>>>>>> Investment >>>>>>>>> > (ROI) of >>>>>>>>> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>>> break-even >>>>>>>>> > >> within 7 months. Over 3 million businesses have gone Google >>>>>>>>> with Google >>>>>>>>> > >> Apps: >>>>>>>>> > >> an online email calendar, and document program that's >>>>>>>>> accessible >>>>>>>>> > from your >>>>>>>>> > >> >>>>>>>>> > >> browser. Read the Forrester report: >>>>>>>>> >>>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>>> > >> _______________________________________________ >>>>>>>>> > >> gstreamer-devel mailing list >>>>>>>>> > >> >>>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>>> > >>>>>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>>>>> > >> >>>>>>>>> >>>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>>> > > >>>>>>>>> > > >>>>>>>>> > >>>>>>>>> ------------------------------------------------------------------------------ >>>>>>>>> > > Forrester recently released a report on the Return on >>>>>>>>> Investment (ROI) of >>>>>>>>> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>>> break-even >>>>>>>>> > > within 7 months. Over 3 million businesses have gone Google >>>>>>>>> with Google >>>>>>>>> > > Apps: >>>>>>>>> > > an online email calendar, and document program that's >>>>>>>>> accessible from >>>>>>>>> > your >>>>>>>>> > > browser. Read the Forrester report: >>>>>>>>> >>>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>>> > > _______________________________________________ >>>>>>>>> > > gstreamer-devel mailing list >>>>>>>>> > > >>>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>>> > >>>>>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>>>>> > > >>>>>>>>> >>>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>>> > > >>>>>>>>> >>>>>>>>> >>>>>>>>> ------------------------------------------------------------------------------ >>>>>>>>> Forrester recently released a report on the Return on Investment >>>>>>>>> (ROI) of >>>>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>>> break-even >>>>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>>>> Google Apps: >>>>>>>>> an online email calendar, and document program that's accessible >>>>>>>>> from your >>>>>>>>> browser. Read the Forrester report: >>>>>>>>> >>>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>>> _______________________________________________ >>>>>>>>> gstreamer-devel mailing list >>>>>>>>> >>>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>>> >>>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> ------------------------------------------------------------------------------ >>>>>>>> Forrester recently released a report on the Return on Investment >>>>>>>> (ROI) of >>>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>> break-even >>>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>>> Google Apps: >>>>>>>> an online email calendar, and document program that's accessible >>>>>>>> from your >>>>>>>> browser. Read the Forrester report: >>>>>>>> >>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>> _______________________________________________ >>>>>>>> gstreamer-devel mailing list >>>>>>>> >>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>> >>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------------ >>>>>>> Forrester recently released a report on the Return on Investment >>>>>>> (ROI) of >>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>> break-even >>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>> Google Apps: >>>>>>> an online email calendar, and document program that's accessible from >>>>>>> your >>>>>>> browser. Read the Forrester report: >>>>>>> >>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>> >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------------ >>>>>>> Forrester recently released a report on the Return on Investment >>>>>>> (ROI) of >>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>> break-even >>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>> Google Apps: >>>>>>> an online email calendar, and document program that's accessible from >>>>>>> your >>>>>>> browser. Read the Forrester report: >>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>>> of >>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>> break-even >>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>> Google Apps: >>>>>> an online email calendar, and document program that's accessible from >>>>>> your >>>>>> browser. Read the Forrester report: >>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>> of >>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>> break-even >>>>> within 7 months. Over 3 million businesses have gone Google with >>>>> Google Apps: >>>>> an online email calendar, and document program that's accessible from >>>>> your >>>>> browser. Read the Forrester report: >>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> gstreamer-devel at lists.sourceforge.net >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Learn how Oracle Real Application Clusters (RAC) One Node allows >>> customers >>> to consolidate database storage, standardize their database environment, >>> and, >>> should the need arise, upgrade to a full multi-node Oracle RAC database >>> without downtime or disruption >>> http://p.sf.net/sfu/oracle-sfdevnl >>> >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, >> and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From braunsquared at gmail.com Thu Dec 23 08:06:05 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Thu, 23 Dec 2010 02:06:05 -0500 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> Message-ID: <2F24B456-1BE7-411D-9206-580C811353C5@gmail.com> Try dropping the capsfilter and see if you get any results then. Sent from my iPhone On Dec 23, 2010, at 1:52 AM, michael wrote: > Really!!?? but it's not work on my book!! God,, this issue drives me crazy!! :( > > On Thu, Dec 23, 2010 at 2:42 PM, liJin wrote: > Hi > > > gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, framerate=30/1' ! ffmpegcolorspace ! x264enc ! mp4mux ! filesink location=1.mp4 > > This pipeline working fine (4 me)......:) > > > On Thu, Dec 23, 2010 at 12:08 PM, michael wrote: > Is there nobody know how to record video from camera to a MP4 container with the H264 encoding ? > > > On Thu, Dec 23, 2010 at 9:46 AM, michael wrote: > neither qtmux nor mp4mux is not work well > > > On Wed, Dec 22, 2010 at 11:24 PM, Timothy Braun wrote: > Not qtmux, use mp4mux. > > > On Wed, Dec 22, 2010 at 10:10 AM, michael wrote: > I have tested changing the muxer to qtmux, but it still not work > > > On Wed, Dec 22, 2010 at 10:58 PM, Timothy Braun wrote: > I was never able to get the ffmpeg mpeg4 mixer to work quite right. I found that mp4mux, an extension of qtmux, works much better. If you inspect the ffmpeg mp4 muxer they actually suggest that you use a different one as it's unstable. > > Best, > Tim > > Sent from my iPhone > > On Dec 22, 2010, at 8:27 AM, michael wrote: > >> I just ran it : >> gst-launch -e v4l2src num-buffers=200 ! video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 >> And got a mp4 file , but it still cannot be playbacked, and give the error info: >> Playing record.mp4. >> libavformat file format detected. >> [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to guess time_increment_bits >> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >> [mpeg4 @ 0x9376c10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag >> [mpeg4 @ 0x9376c10]picture size invalid (0x0) >> [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) >> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters (Video: mpeg4, yuv420p) >> LAVF_header: av_find_stream_info() failed >> ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >> Quicktime/MOV file format detected. >> [mov] Video stream found, -vid 0 >> VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) >> open: No such file or directory >> [MGA] Couldn't open: /dev/mga_vid >> open: No such file or directory >> [MGA] Couldn't open: /dev/mga_vid >> [VO_TDFXFB] Can't open /dev/fb0: Permission denied. >> [VO_3DFX] Unable to open /dev/3dfx. >> ========================================================================== >> Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family >> Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) >> ========================================================================== >> Audio: no sound >> Starting playback... >> VDec: vo config request - 640 x 480 (preferred colorspace: Planar YV12) >> VDec: using Planar YV12 as output csp (no 0) >> Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. >> VO: [xv] 640x480 => 640x480 Planar YV12 >> V: 0.0 200/200 ??% ??% ??,?% 0 0 >> >> Exiting... (End of file) >> >> But the error info not like the old one! >> >> On Wed, Dec 22, 2010 at 9:17 PM, liJin wrote: >> I think thats the issue... >> >> >> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=1.mp4 >> >> run with -e option >> >> >> >> On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi wrote: >> On 12/22/2010 12:18 PM, michael wrote: >> > Now I can record the video to a AVI file with mpeg4 encoding by : >> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink >> > location=record.avi >> > >> > But I got two issues: >> > 1: I want record as a MP4 file, I do it like this: >> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink >> > location=record.avi >> > I can record the video , but when I playback it by mplayer, I got a error: >> > Playing ./record.mp4. >> > libavformat file format detected. >> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >> > LAVF_header: av_open_input_stream() failed >> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >> >> When recording, with the above gst-launch line, do you stop >> the pipeline with Ctrl+C..? >> >> It could be that the file container doesn't get fully written >> in case of such an interrupt and that's why the clip doesn't play. >> >> For example, if you use this pipeline and >> stop it with Ctrl+C, the clip doesn't play: >> >> gst-launch-0.10 videotestsrc ! >> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace >> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >> >> But this pipeline produces a valid video file: >> >> gst-launch-0.10 videotestsrc num-buffers=200 ! >> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace >> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >> >> > >> > 2: I want encode the video to H264, so I do like this: >> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >> > location=record.avi >> >> How about adding the format to the caps, e.g . >> v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >> location=record.avi ? >> >> BR, >> Topi >> >> > But when I run the pipeline I just get the error info like this: >> > Setting pipeline to PAUSED ... >> > Pipeline is live and does not need PREROLL ... >> > Setting pipeline to PLAYING ... >> > New clock: GstSystemClock >> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Internal >> > data flow error. >> > Additional debug info: >> > gstbasesrc.c(2550): gst_base_src_loop (): >> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >> > streaming task paused, reason not-negotiated (-4) >> > Execution ended after 1727716724 ns. >> > Setting pipeline to PAUSED ... >> > Setting pipeline to READY ... >> > Setting pipeline to NULL ... >> > Freeing pipeline ... >> > >> > Some one can help me fix the issues? Tks :) >> > >> > >> > On 12/22/10, Timothy Braun > > > wrote: >> > > H264 is encoded with x264enc element. >> > > >> > > Sent from my iPhone >> > > >> > > On Dec 22, 2010, at 3:10 AM, wl2776 > > > wrote: >> > > >> > >> >> > >> >> > >> michael young wrote: >> > >>> >> > >>> And now , I want to record the video to a MP4 file with the h264 >> > encoding >> > >>> format , How can I do ? >> > >>> >> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael > > > wrote: >> > >>> >> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink >> > >>> >> > >>> >> > >> >> > >> Looks like you can't, because h264 encoder is unavailable. >> > >> gst-inspect | grep h264 >> > >> doesn't show it. >> > >> However, general idea is to encode, then mux, and then save to a file. >> > >> gst-launch v4l2src device=/dev/video0 ! >> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffenc_mpeg4 ! >> > >> mp4mux !filesink location=/home/user/file. >> > >> >> > >> -- >> > >> View this message in context: >> > >> >> > http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >> > >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> > >> >> > >> >> > ------------------------------------------------------------------------------ >> > >> Forrester recently released a report on the Return on Investment >> > (ROI) of >> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> > >> within 7 months. Over 3 million businesses have gone Google with Google >> > >> Apps: >> > >> an online email calendar, and document program that's accessible >> > from your >> > >> >> > >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> > >> _______________________________________________ >> > >> gstreamer-devel mailing list >> > >> gstreamer-devel at lists.sourceforge.net >> > >> > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > >> > > >> > ------------------------------------------------------------------------------ >> > > Forrester recently released a report on the Return on Investment (ROI) of >> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> > > within 7 months. Over 3 million businesses have gone Google with Google >> > > Apps: >> > > an online email calendar, and document program that's accessible from >> > your >> > > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> > > _______________________________________________ >> > > gstreamer-devel mailing list >> > > gstreamer-devel at lists.sourceforge.net >> > >> > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even > within 7 months. Over 3 million businesses have gone Google with Google Apps: > an online email calendar, and document program that's accessible from your > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Thu Dec 23 08:26:27 2010 From: mcygogo at gmail.com (michael) Date: Thu, 23 Dec 2010 15:26:27 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> Message-ID: My Gstreamer and Plugins version is : gst-ffmpeg-0.10.11 gst-plugins-bad-0.10.20 gst-plugins-base-0.10.30 gst-plugins-good-0.10.25 gst-plugins-ugly-0.10.16 gstreamer-0.10.30 So I think the version is ok, isn't it? and the all plugin can run, so I think the supported library packages is ready. On Thu, Dec 23, 2010 at 3:06 PM, sudarshan bisht wrote: > > Make sure you have all latest Gstreamer core, plugins and supported library > packages installed. > > > On Thu, Dec 23, 2010 at 12:22 PM, michael wrote: > >> Really!!?? but it's not work on my book!! God,, this issue drives me >> crazy!! :( >> >> >> On Thu, Dec 23, 2010 at 2:42 PM, liJin wrote: >> >>> Hi >>> >>> >>> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>> framerate=30/1' ! ffmpegcolorspace ! x264enc ! mp4mux ! filesink >>> location=1.mp4 >>> >>> This pipeline working fine (4 me)......:) >>> >>> >>> On Thu, Dec 23, 2010 at 12:08 PM, michael wrote: >>> >>>> Is there nobody know how to record video from camera to a MP4 container >>>> with the H264 encoding ? >>>> >>>> >>>> On Thu, Dec 23, 2010 at 9:46 AM, michael wrote: >>>> >>>>> neither qtmux nor mp4mux is not work well >>>>> >>>>> >>>>> On Wed, Dec 22, 2010 at 11:24 PM, Timothy Braun < >>>>> braunsquared at gmail.com> wrote: >>>>> >>>>>> Not qtmux, use mp4mux. >>>>>> >>>>>> >>>>>> On Wed, Dec 22, 2010 at 10:10 AM, michael wrote: >>>>>> >>>>>>> I have tested changing the muxer to qtmux, but it still not work >>>>>>> >>>>>>> >>>>>>> On Wed, Dec 22, 2010 at 10:58 PM, Timothy Braun < >>>>>>> braunsquared at gmail.com> wrote: >>>>>>> >>>>>>>> I was never able to get the ffmpeg mpeg4 mixer to work quite right. >>>>>>>> I found that mp4mux, an extension of qtmux, works much better. If you >>>>>>>> inspect the ffmpeg mp4 muxer they actually suggest that you use a different >>>>>>>> one as it's unstable. >>>>>>>> >>>>>>>> Best, >>>>>>>> Tim >>>>>>>> >>>>>>>> Sent from my iPhone >>>>>>>> >>>>>>>> On Dec 22, 2010, at 8:27 AM, michael wrote: >>>>>>>> >>>>>>>> I just ran it : >>>>>>>> gst-launch -e v4l2src num-buffers=200 ! >>>>>>>> video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! >>>>>>>> ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 >>>>>>>> And got a mp4 file , but it still cannot be playbacked, and give the >>>>>>>> error info: >>>>>>>> Playing record.mp4. >>>>>>>> libavformat file format detected. >>>>>>>> [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying >>>>>>>> to guess time_increment_bits >>>>>>>> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >>>>>>>> [mpeg4 @ 0x9376c10]looks like this file was encoded with >>>>>>>> (divx4/(old)xvid/opendivx) -> forcing low_delay flag >>>>>>>> [mpeg4 @ 0x9376c10]picture size invalid (0x0) >>>>>>>> [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) >>>>>>>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters >>>>>>>> (Video: mpeg4, yuv420p) >>>>>>>> LAVF_header: av_find_stream_info() failed >>>>>>>> ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>>>>> Quicktime/MOV file format detected. >>>>>>>> [mov] Video stream found, -vid 0 >>>>>>>> VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) >>>>>>>> open: No such file or directory >>>>>>>> [MGA] Couldn't open: /dev/mga_vid >>>>>>>> open: No such file or directory >>>>>>>> [MGA] Couldn't open: /dev/mga_vid >>>>>>>> [VO_TDFXFB] Can't open /dev/fb0: Permission denied. >>>>>>>> [VO_3DFX] Unable to open /dev/3dfx. >>>>>>>> >>>>>>>> ========================================================================== >>>>>>>> Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family >>>>>>>> Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) >>>>>>>> >>>>>>>> ========================================================================== >>>>>>>> Audio: no sound >>>>>>>> Starting playback... >>>>>>>> VDec: vo config request - 640 x 480 (preferred colorspace: Planar >>>>>>>> YV12) >>>>>>>> VDec: using Planar YV12 as output csp (no 0) >>>>>>>> Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. >>>>>>>> VO: [xv] 640x480 => 640x480 Planar YV12 >>>>>>>> V: 0.0 200/200 ??% ??% ??,?% 0 0 >>>>>>>> >>>>>>>> Exiting... (End of file) >>>>>>>> >>>>>>>> But the error info not like the old one! >>>>>>>> >>>>>>>> On Wed, Dec 22, 2010 at 9:17 PM, liJin < >>>>>>>> lijinsyam at gmail.com> wrote: >>>>>>>> >>>>>>>>> I think thats the issue... >>>>>>>>> >>>>>>>>> >>>>>>>>> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>>> framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink >>>>>>>>> location=1.mp4 >>>>>>>>> >>>>>>>>> run with* -e *option >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi < >>>>>>>>> Topi.Santakivi at digia.com> wrote: >>>>>>>>> >>>>>>>>>> On 12/22/2010 12:18 PM, michael wrote: >>>>>>>>>> > Now I can record the video to a AVI file with mpeg4 encoding by >>>>>>>>>> : >>>>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! >>>>>>>>>> filesink >>>>>>>>>> > location=record.avi >>>>>>>>>> > >>>>>>>>>> > But I got two issues: >>>>>>>>>> > 1: I want record as a MP4 file, I do it like this: >>>>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! >>>>>>>>>> filesink >>>>>>>>>> > location=record.avi >>>>>>>>>> > I can record the video , but when I playback it by mplayer, I >>>>>>>>>> got a error: >>>>>>>>>> > Playing ./record.mp4. >>>>>>>>>> > libavformat file format detected. >>>>>>>>>> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >>>>>>>>>> > LAVF_header: av_open_input_stream() failed >>>>>>>>>> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>>>>>>> >>>>>>>>>> When recording, with the above gst-launch line, do you stop >>>>>>>>>> the pipeline with Ctrl+C..? >>>>>>>>>> >>>>>>>>>> It could be that the file container doesn't get fully written >>>>>>>>>> in case of such an interrupt and that's why the clip doesn't play. >>>>>>>>>> >>>>>>>>>> For example, if you use this pipeline and >>>>>>>>>> stop it with Ctrl+C, the clip doesn't play: >>>>>>>>>> >>>>>>>>>> gst-launch-0.10 videotestsrc ! >>>>>>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>>>>>> ffmpegcolorspace >>>>>>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>>>>>> >>>>>>>>>> But this pipeline produces a valid video file: >>>>>>>>>> >>>>>>>>>> gst-launch-0.10 videotestsrc num-buffers=200 ! >>>>>>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>>>>>> ffmpegcolorspace >>>>>>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>>>>>> >>>>>>>>>> > >>>>>>>>>> > 2: I want encode the video to H264, so I do like this: >>>>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >>>>>>>>>> > location=record.avi >>>>>>>>>> >>>>>>>>>> How about adding the format to the caps, e.g . >>>>>>>>>> v4l2src ! >>>>>>>>>> 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >>>>>>>>>> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >>>>>>>>>> location=record.avi ? >>>>>>>>>> >>>>>>>>>> BR, >>>>>>>>>> Topi >>>>>>>>>> >>>>>>>>>> > But when I run the pipeline I just get the error info like this: >>>>>>>>>> > Setting pipeline to PAUSED ... >>>>>>>>>> > Pipeline is live and does not need PREROLL ... >>>>>>>>>> > Setting pipeline to PLAYING ... >>>>>>>>>> > New clock: GstSystemClock >>>>>>>>>> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>>>>>> Internal >>>>>>>>>> > data flow error. >>>>>>>>>> > Additional debug info: >>>>>>>>>> > gstbasesrc.c(2550): gst_base_src_loop (): >>>>>>>>>> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>>>>>> > streaming task paused, reason not-negotiated (-4) >>>>>>>>>> > Execution ended after 1727716724 ns. >>>>>>>>>> > Setting pipeline to PAUSED ... >>>>>>>>>> > Setting pipeline to READY ... >>>>>>>>>> > Setting pipeline to NULL ... >>>>>>>>>> > Freeing pipeline ... >>>>>>>>>> > >>>>>>>>>> > Some one can help me fix the issues? Tks :) >>>>>>>>>> > >>>>>>>>>> > >>>>>>>>>> > On 12/22/10, Timothy Braun < >>>>>>>>>> braunsquared at gmail.com >>>>>>>>>> > braunsquared at gmail.com>> >>>>>>>>>> wrote: >>>>>>>>>> > > H264 is encoded with x264enc element. >>>>>>>>>> > > >>>>>>>>>> > > Sent from my iPhone >>>>>>>>>> > > >>>>>>>>>> > > On Dec 22, 2010, at 3:10 AM, wl2776 < >>>>>>>>>> wl2776 at gmail.com >>>>>>>>>> > wl2776 at gmail.com>> wrote: >>>>>>>>>> > > >>>>>>>>>> > >> >>>>>>>>>> > >> >>>>>>>>>> > >> michael young wrote: >>>>>>>>>> > >>> >>>>>>>>>> > >>> And now , I want to record the video to a MP4 file with the >>>>>>>>>> h264 >>>>>>>>>> > encoding >>>>>>>>>> > >>> format , How can I do ? >>>>>>>>>> > >>> >>>>>>>>>> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael < >>>>>>>>>> mcygogo at gmail.com >>>>>>>>>> > mcygogo at gmail.com>> wrote: >>>>>>>>>> > >>> >>>>>>>>>> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 >>>>>>>>>> ! >>>>>>>>>> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>>>>>> xvimagesink >>>>>>>>>> > >>> >>>>>>>>>> > >>> >>>>>>>>>> > >> >>>>>>>>>> > >> Looks like you can't, because h264 encoder is unavailable. >>>>>>>>>> > >> gst-inspect | grep h264 >>>>>>>>>> > >> doesn't show it. >>>>>>>>>> > >> However, general idea is to encode, then mux, and then save >>>>>>>>>> to a file. >>>>>>>>>> > >> gst-launch v4l2src device=/dev/video0 ! >>>>>>>>>> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>>>>>> ffenc_mpeg4 ! >>>>>>>>>> > >> mp4mux !filesink location=/home/user/file. >>>>>>>>>> > >> >>>>>>>>>> > >> -- >>>>>>>>>> > >> View this message in context: >>>>>>>>>> > >> >>>>>>>>>> > >>>>>>>>>> >>>>>>>>>> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >>>>>>>>>> > >> Sent from the GStreamer-devel mailing list archive at >>>>>>>>>> Nabble.com. >>>>>>>>>> > >> >>>>>>>>>> > >> >>>>>>>>>> > >>>>>>>>>> ------------------------------------------------------------------------------ >>>>>>>>>> > >> Forrester recently released a report on the Return on >>>>>>>>>> Investment >>>>>>>>>> > (ROI) of >>>>>>>>>> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, >>>>>>>>>> and break-even >>>>>>>>>> > >> within 7 months. Over 3 million businesses have gone Google >>>>>>>>>> with Google >>>>>>>>>> > >> Apps: >>>>>>>>>> > >> an online email calendar, and document program that's >>>>>>>>>> accessible >>>>>>>>>> > from your >>>>>>>>>> > >> >>>>>>>>>> > >> browser. Read the Forrester report: >>>>>>>>>> >>>>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>>>> > >> _______________________________________________ >>>>>>>>>> > >> gstreamer-devel mailing list >>>>>>>>>> > >> >>>>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>>>> > >>>>>>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>>>>>> > >> >>>>>>>>>> >>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>>>> > > >>>>>>>>>> > > >>>>>>>>>> > >>>>>>>>>> ------------------------------------------------------------------------------ >>>>>>>>>> > > Forrester recently released a report on the Return on >>>>>>>>>> Investment (ROI) of >>>>>>>>>> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>>>> break-even >>>>>>>>>> > > within 7 months. Over 3 million businesses have gone Google >>>>>>>>>> with Google >>>>>>>>>> > > Apps: >>>>>>>>>> > > an online email calendar, and document program that's >>>>>>>>>> accessible from >>>>>>>>>> > your >>>>>>>>>> > > browser. Read the Forrester report: >>>>>>>>>> >>>>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>>>> > > _______________________________________________ >>>>>>>>>> > > gstreamer-devel mailing list >>>>>>>>>> > > >>>>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>>>> > >>>>>>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>>>>>> > > >>>>>>>>>> >>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>>>> > > >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> ------------------------------------------------------------------------------ >>>>>>>>>> Forrester recently released a report on the Return on Investment >>>>>>>>>> (ROI) of >>>>>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>>>> break-even >>>>>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>>>>> Google Apps: >>>>>>>>>> an online email calendar, and document program that's accessible >>>>>>>>>> from your >>>>>>>>>> browser. Read the Forrester report: >>>>>>>>>> >>>>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>>>> _______________________________________________ >>>>>>>>>> gstreamer-devel mailing list >>>>>>>>>> >>>>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>>>> >>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> ------------------------------------------------------------------------------ >>>>>>>>> Forrester recently released a report on the Return on Investment >>>>>>>>> (ROI) of >>>>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>>> break-even >>>>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>>>> Google Apps: >>>>>>>>> an online email calendar, and document program that's accessible >>>>>>>>> from your >>>>>>>>> browser. Read the Forrester report: >>>>>>>>> >>>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>>> _______________________________________________ >>>>>>>>> gstreamer-devel mailing list >>>>>>>>> >>>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>>> >>>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> ------------------------------------------------------------------------------ >>>>>>>> Forrester recently released a report on the Return on Investment >>>>>>>> (ROI) of >>>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>> break-even >>>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>>> Google Apps: >>>>>>>> an online email calendar, and document program that's accessible >>>>>>>> from your >>>>>>>> browser. Read the Forrester report: >>>>>>>> >>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> gstreamer-devel mailing list >>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> ------------------------------------------------------------------------------ >>>>>>>> Forrester recently released a report on the Return on Investment >>>>>>>> (ROI) of >>>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>> break-even >>>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>>> Google Apps: >>>>>>>> an online email calendar, and document program that's accessible >>>>>>>> from your >>>>>>>> browser. Read the Forrester report: >>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>> _______________________________________________ >>>>>>>> gstreamer-devel mailing list >>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------------ >>>>>>> Forrester recently released a report on the Return on Investment >>>>>>> (ROI) of >>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>> break-even >>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>> Google Apps: >>>>>>> an online email calendar, and document program that's accessible from >>>>>>> your >>>>>>> browser. Read the Forrester report: >>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>>> of >>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>> break-even >>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>> Google Apps: >>>>>> an online email calendar, and document program that's accessible from >>>>>> your >>>>>> browser. Read the Forrester report: >>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>>> >>>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Learn how Oracle Real Application Clusters (RAC) One Node allows >>>> customers >>>> to consolidate database storage, standardize their database environment, >>>> and, >>>> should the need arise, upgrade to a full multi-node Oracle RAC database >>>> without downtime or disruption >>>> http://p.sf.net/sfu/oracle-sfdevnl >>>> >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Learn how Oracle Real Application Clusters (RAC) One Node allows >>> customers >>> to consolidate database storage, standardize their database environment, >>> and, >>> should the need arise, upgrade to a full multi-node Oracle RAC database >>> without downtime or disruption >>> http://p.sf.net/sfu/oracle-sfdevnl >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, >> and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > -- > Regards, > > Sudarshan Bisht > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mcygogo at gmail.com Thu Dec 23 08:27:48 2010 From: mcygogo at gmail.com (michael) Date: Thu, 23 Dec 2010 15:27:48 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: <2F24B456-1BE7-411D-9206-580C811353C5@gmail.com> References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> <2F24B456-1BE7-411D-9206-580C811353C5@gmail.com> Message-ID: how to "dropping the capsfilter" ? tks:) On Thu, Dec 23, 2010 at 3:06 PM, Timothy Braun wrote: > Try dropping the capsfilter and see if you get any results then. > > Sent from my iPhone > > On Dec 23, 2010, at 1:52 AM, michael wrote: > > Really!!?? but it's not work on my book!! God,, this issue drives me > crazy!! :( > > On Thu, Dec 23, 2010 at 2:42 PM, liJin < > lijinsyam at gmail.com> wrote: > >> Hi >> >> >> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >> framerate=30/1' ! ffmpegcolorspace ! x264enc ! mp4mux ! filesink >> location=1.mp4 >> >> This pipeline working fine (4 me)......:) >> >> >> On Thu, Dec 23, 2010 at 12:08 PM, michael < >> mcygogo at gmail.com> wrote: >> >>> Is there nobody know how to record video from camera to a MP4 container >>> with the H264 encoding ? >>> >>> >>> On Thu, Dec 23, 2010 at 9:46 AM, michael < >>> mcygogo at gmail.com> wrote: >>> >>>> neither qtmux nor mp4mux is not work well >>>> >>>> >>>> On Wed, Dec 22, 2010 at 11:24 PM, Timothy Braun < >>>> braunsquared at gmail.com> wrote: >>>> >>>>> Not qtmux, use mp4mux. >>>>> >>>>> >>>>> On Wed, Dec 22, 2010 at 10:10 AM, michael < >>>>> mcygogo at gmail.com> wrote: >>>>> >>>>>> I have tested changing the muxer to qtmux, but it still not work >>>>>> >>>>>> >>>>>> On Wed, Dec 22, 2010 at 10:58 PM, Timothy Braun < >>>>>> braunsquared at gmail.com> wrote: >>>>>> >>>>>>> I was never able to get the ffmpeg mpeg4 mixer to work quite right. I >>>>>>> found that mp4mux, an extension of qtmux, works much better. If you inspect >>>>>>> the ffmpeg mp4 muxer they actually suggest that you use a different one as >>>>>>> it's unstable. >>>>>>> >>>>>>> Best, >>>>>>> Tim >>>>>>> >>>>>>> Sent from my iPhone >>>>>>> >>>>>>> On Dec 22, 2010, at 8:27 AM, michael < >>>>>>> mcygogo at gmail.com> wrote: >>>>>>> >>>>>>> I just ran it : >>>>>>> gst-launch -e v4l2src num-buffers=200 ! >>>>>>> video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! >>>>>>> ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 >>>>>>> And got a mp4 file , but it still cannot be playbacked, and give the >>>>>>> error info: >>>>>>> Playing record.mp4. >>>>>>> libavformat file format detected. >>>>>>> [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to >>>>>>> guess time_increment_bits >>>>>>> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >>>>>>> [mpeg4 @ 0x9376c10]looks like this file was encoded with >>>>>>> (divx4/(old)xvid/opendivx) -> forcing low_delay flag >>>>>>> [mpeg4 @ 0x9376c10]picture size invalid (0x0) >>>>>>> [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) >>>>>>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters >>>>>>> (Video: mpeg4, yuv420p) >>>>>>> LAVF_header: av_find_stream_info() failed >>>>>>> ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>>>> Quicktime/MOV file format detected. >>>>>>> [mov] Video stream found, -vid 0 >>>>>>> VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) >>>>>>> open: No such file or directory >>>>>>> [MGA] Couldn't open: /dev/mga_vid >>>>>>> open: No such file or directory >>>>>>> [MGA] Couldn't open: /dev/mga_vid >>>>>>> [VO_TDFXFB] Can't open /dev/fb0: Permission denied. >>>>>>> [VO_3DFX] Unable to open /dev/3dfx. >>>>>>> >>>>>>> ========================================================================== >>>>>>> Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family >>>>>>> Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) >>>>>>> >>>>>>> ========================================================================== >>>>>>> Audio: no sound >>>>>>> Starting playback... >>>>>>> VDec: vo config request - 640 x 480 (preferred colorspace: Planar >>>>>>> YV12) >>>>>>> VDec: using Planar YV12 as output csp (no 0) >>>>>>> Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. >>>>>>> VO: [xv] 640x480 => 640x480 Planar YV12 >>>>>>> V: 0.0 200/200 ??% ??% ??,?% 0 0 >>>>>>> >>>>>>> Exiting... (End of file) >>>>>>> >>>>>>> But the error info not like the old one! >>>>>>> >>>>>>> On Wed, Dec 22, 2010 at 9:17 PM, liJin < >>>>>>> lijinsyam at gmail.com> wrote: >>>>>>> >>>>>>>> I think thats the issue... >>>>>>>> >>>>>>>> >>>>>>>> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>> framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink >>>>>>>> location=1.mp4 >>>>>>>> >>>>>>>> run with* -e *option >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi < >>>>>>>> Topi.Santakivi at digia.com> wrote: >>>>>>>> >>>>>>>>> On 12/22/2010 12:18 PM, michael wrote: >>>>>>>>> > Now I can record the video to a AVI file with mpeg4 encoding by : >>>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! >>>>>>>>> filesink >>>>>>>>> > location=record.avi >>>>>>>>> > >>>>>>>>> > But I got two issues: >>>>>>>>> > 1: I want record as a MP4 file, I do it like this: >>>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! >>>>>>>>> filesink >>>>>>>>> > location=record.avi >>>>>>>>> > I can record the video , but when I playback it by mplayer, I got >>>>>>>>> a error: >>>>>>>>> > Playing ./record.mp4. >>>>>>>>> > libavformat file format detected. >>>>>>>>> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >>>>>>>>> > LAVF_header: av_open_input_stream() failed >>>>>>>>> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>>>>>>>> >>>>>>>>> When recording, with the above gst-launch line, do you stop >>>>>>>>> the pipeline with Ctrl+C..? >>>>>>>>> >>>>>>>>> It could be that the file container doesn't get fully written >>>>>>>>> in case of such an interrupt and that's why the clip doesn't play. >>>>>>>>> >>>>>>>>> For example, if you use this pipeline and >>>>>>>>> stop it with Ctrl+C, the clip doesn't play: >>>>>>>>> >>>>>>>>> gst-launch-0.10 videotestsrc ! >>>>>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>>>>> ffmpegcolorspace >>>>>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>>>>> >>>>>>>>> But this pipeline produces a valid video file: >>>>>>>>> >>>>>>>>> gst-launch-0.10 videotestsrc num-buffers=200 ! >>>>>>>>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! >>>>>>>>> ffmpegcolorspace >>>>>>>>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>>>>>>>> >>>>>>>>> > >>>>>>>>> > 2: I want encode the video to H264, so I do like this: >>>>>>>>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>>>>>>>> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >>>>>>>>> > location=record.avi >>>>>>>>> >>>>>>>>> How about adding the format to the caps, e.g . >>>>>>>>> v4l2src ! >>>>>>>>> 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >>>>>>>>> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >>>>>>>>> location=record.avi ? >>>>>>>>> >>>>>>>>> BR, >>>>>>>>> Topi >>>>>>>>> >>>>>>>>> > But when I run the pipeline I just get the error info like this: >>>>>>>>> > Setting pipeline to PAUSED ... >>>>>>>>> > Pipeline is live and does not need PREROLL ... >>>>>>>>> > Setting pipeline to PLAYING ... >>>>>>>>> > New clock: GstSystemClock >>>>>>>>> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>>>>> Internal >>>>>>>>> > data flow error. >>>>>>>>> > Additional debug info: >>>>>>>>> > gstbasesrc.c(2550): gst_base_src_loop (): >>>>>>>>> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>>>>>>>> > streaming task paused, reason not-negotiated (-4) >>>>>>>>> > Execution ended after 1727716724 ns. >>>>>>>>> > Setting pipeline to PAUSED ... >>>>>>>>> > Setting pipeline to READY ... >>>>>>>>> > Setting pipeline to NULL ... >>>>>>>>> > Freeing pipeline ... >>>>>>>>> > >>>>>>>>> > Some one can help me fix the issues? Tks :) >>>>>>>>> > >>>>>>>>> > >>>>>>>>> > On 12/22/10, Timothy Braun < >>>>>>>>> braunsquared at gmail.com >>>>>>>>> > >>>>>>>>> braunsquared at gmail.com>> wrote: >>>>>>>>> > > H264 is encoded with x264enc element. >>>>>>>>> > > >>>>>>>>> > > Sent from my iPhone >>>>>>>>> > > >>>>>>>>> > > On Dec 22, 2010, at 3:10 AM, wl2776 < >>>>>>>>> wl2776 at gmail.com >>>>>>>>> > wl2776 at gmail.com>> >>>>>>>>> wrote: >>>>>>>>> > > >>>>>>>>> > >> >>>>>>>>> > >> >>>>>>>>> > >> michael young wrote: >>>>>>>>> > >>> >>>>>>>>> > >>> And now , I want to record the video to a MP4 file with the >>>>>>>>> h264 >>>>>>>>> > encoding >>>>>>>>> > >>> format , How can I do ? >>>>>>>>> > >>> >>>>>>>>> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael < >>>>>>>>> mcygogo at gmail.com >>>>>>>>> > >>>>>>>>> mcygogo at gmail.com>> wrote: >>>>>>>>> > >>> >>>>>>>>> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 >>>>>>>>> ! >>>>>>>>> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>>>>> xvimagesink >>>>>>>>> > >>> >>>>>>>>> > >>> >>>>>>>>> > >> >>>>>>>>> > >> Looks like you can't, because h264 encoder is unavailable. >>>>>>>>> > >> gst-inspect | grep h264 >>>>>>>>> > >> doesn't show it. >>>>>>>>> > >> However, general idea is to encode, then mux, and then save >>>>>>>>> to a file. >>>>>>>>> > >> gst-launch v4l2src device=/dev/video0 ! >>>>>>>>> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! >>>>>>>>> ffenc_mpeg4 ! >>>>>>>>> > >> mp4mux !filesink location=/home/user/file. >>>>>>>>> > >> >>>>>>>>> > >> -- >>>>>>>>> > >> View this message in context: >>>>>>>>> > >> >>>>>>>>> > >>>>>>>>> >>>>>>>>> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >>>>>>>>> > >> Sent from the GStreamer-devel mailing list archive at >>>>>>>>> Nabble.com. >>>>>>>>> > >> >>>>>>>>> > >> >>>>>>>>> > >>>>>>>>> ------------------------------------------------------------------------------ >>>>>>>>> > >> Forrester recently released a report on the Return on >>>>>>>>> Investment >>>>>>>>> > (ROI) of >>>>>>>>> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>>> break-even >>>>>>>>> > >> within 7 months. Over 3 million businesses have gone Google >>>>>>>>> with Google >>>>>>>>> > >> Apps: >>>>>>>>> > >> an online email calendar, and document program that's >>>>>>>>> accessible >>>>>>>>> > from your >>>>>>>>> > >> >>>>>>>>> > >> browser. Read the Forrester report: >>>>>>>>> >>>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>>> > >> _______________________________________________ >>>>>>>>> > >> gstreamer-devel mailing list >>>>>>>>> > >> >>>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>>> > >>>>>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>>>>> > >> >>>>>>>>> >>>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>>> > > >>>>>>>>> > > >>>>>>>>> > >>>>>>>>> ------------------------------------------------------------------------------ >>>>>>>>> > > Forrester recently released a report on the Return on >>>>>>>>> Investment (ROI) of >>>>>>>>> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>>> break-even >>>>>>>>> > > within 7 months. Over 3 million businesses have gone Google >>>>>>>>> with Google >>>>>>>>> > > Apps: >>>>>>>>> > > an online email calendar, and document program that's >>>>>>>>> accessible from >>>>>>>>> > your >>>>>>>>> > > browser. Read the Forrester report: >>>>>>>>> >>>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>>> > > _______________________________________________ >>>>>>>>> > > gstreamer-devel mailing list >>>>>>>>> > > >>>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>>> > >>>>>>>>> gstreamer-devel at lists.sourceforge.net> >>>>>>>>> > > >>>>>>>>> >>>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>>> > > >>>>>>>>> >>>>>>>>> >>>>>>>>> ------------------------------------------------------------------------------ >>>>>>>>> Forrester recently released a report on the Return on Investment >>>>>>>>> (ROI) of >>>>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>>> break-even >>>>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>>>> Google Apps: >>>>>>>>> an online email calendar, and document program that's accessible >>>>>>>>> from your >>>>>>>>> browser. Read the Forrester report: >>>>>>>>> >>>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>>> _______________________________________________ >>>>>>>>> gstreamer-devel mailing list >>>>>>>>> >>>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>>> >>>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> ------------------------------------------------------------------------------ >>>>>>>> Forrester recently released a report on the Return on Investment >>>>>>>> (ROI) of >>>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>>> break-even >>>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>>> Google Apps: >>>>>>>> an online email calendar, and document program that's accessible >>>>>>>> from your >>>>>>>> browser. Read the Forrester report: >>>>>>>> >>>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>>> _______________________________________________ >>>>>>>> gstreamer-devel mailing list >>>>>>>> >>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>> >>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------------ >>>>>>> Forrester recently released a report on the Return on Investment >>>>>>> (ROI) of >>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>> break-even >>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>> Google Apps: >>>>>>> an online email calendar, and document program that's accessible from >>>>>>> your >>>>>>> browser. Read the Forrester report: >>>>>>> >>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>> >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------------ >>>>>>> Forrester recently released a report on the Return on Investment >>>>>>> (ROI) of >>>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>>> break-even >>>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>>> Google Apps: >>>>>>> an online email calendar, and document program that's accessible from >>>>>>> your >>>>>>> browser. Read the Forrester report: >>>>>>> >>>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>>> of >>>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>>> break-even >>>>>> within 7 months. Over 3 million businesses have gone Google with >>>>>> Google Apps: >>>>>> an online email calendar, and document program that's accessible from >>>>>> your >>>>>> browser. Read the Forrester report: >>>>>> >>>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Forrester recently released a report on the Return on Investment (ROI) >>>>> of >>>>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and >>>>> break-even >>>>> within 7 months. Over 3 million businesses have gone Google with >>>>> Google Apps: >>>>> an online email calendar, and document program that's accessible from >>>>> your >>>>> browser. Read the Forrester report: >>>>> >>>>> http://p.sf.net/sfu/googleapps-sfnew >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> >>>>> gstreamer-devel at lists.sourceforge.net >>>>> >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Learn how Oracle Real Application Clusters (RAC) One Node allows >>> customers >>> to consolidate database storage, standardize their database environment, >>> and, >>> should the need arise, upgrade to a full multi-node Oracle RAC database >>> without downtime or disruption >>> http://p.sf.net/sfu/oracle-sfdevnl >>> >>> _______________________________________________ >>> gstreamer-devel mailing list >>> >>> gstreamer-devel at lists.sourceforge.net >>> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, >> and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From braunsquared at gmail.com Thu Dec 23 08:39:18 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Thu, 23 Dec 2010 02:39:18 -0500 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <2F0F8526-FA1F-4A1F-9107-E88EAD3ADDE4@gmail.com> <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> <2F24B456-1BE7-411D-9206-580C811353C5@gmail.com> Message-ID: Drop this part of the statement: > 'video/x-raw-yuv,width=320,height=240, framerate=30/1' Sent from my iPhone On Dec 23, 2010, at 2:27 AM, michael wrote: > how to "dropping the capsfilter" ? tks:) > > On Thu, Dec 23, 2010 at 3:06 PM, Timothy Braun wrote: > Try dropping the capsfilter and see if you get any results then. > > Sent from my iPhone > > On Dec 23, 2010, at 1:52 AM, michael wrote: > >> Really!!?? but it's not work on my book!! God,, this issue drives me crazy!! :( >> >> On Thu, Dec 23, 2010 at 2:42 PM, liJin wrote: >> Hi >> >> >> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, framerate=30/1' ! ffmpegcolorspace ! x264enc ! mp4mux ! filesink location=1.mp4 >> >> This pipeline working fine (4 me)......:) >> >> >> On Thu, Dec 23, 2010 at 12:08 PM, michael wrote: >> Is there nobody know how to record video from camera to a MP4 container with the H264 encoding ? >> >> >> On Thu, Dec 23, 2010 at 9:46 AM, michael wrote: >> neither qtmux nor mp4mux is not work well >> >> >> On Wed, Dec 22, 2010 at 11:24 PM, Timothy Braun wrote: >> Not qtmux, use mp4mux. >> >> >> On Wed, Dec 22, 2010 at 10:10 AM, michael wrote: >> I have tested changing the muxer to qtmux, but it still not work >> >> >> On Wed, Dec 22, 2010 at 10:58 PM, Timothy Braun wrote: >> I was never able to get the ffmpeg mpeg4 mixer to work quite right. I found that mp4mux, an extension of qtmux, works much better. If you inspect the ffmpeg mp4 muxer they actually suggest that you use a different one as it's unstable. >> >> Best, >> Tim >> >> Sent from my iPhone >> >> On Dec 22, 2010, at 8:27 AM, michael wrote: >> >>> I just ran it : >>> gst-launch -e v4l2src num-buffers=200 ! video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=record.mp4 >>> And got a mp4 file , but it still cannot be playbacked, and give the error info: >>> Playing record.mp4. >>> libavformat file format detected. >>> [mpeg4 @ 0x9376c10]hmm, seems the headers are not complete, trying to guess time_increment_bits >>> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >>> [mpeg4 @ 0x9376c10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag >>> [mpeg4 @ 0x9376c10]picture size invalid (0x0) >>> [mpeg4 @ 0x9376c10]get_buffer() failed (-1 0 0 (nil)) >>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9375640]Could not find codec parameters (Video: mpeg4, yuv420p) >>> LAVF_header: av_find_stream_info() failed >>> ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>> Quicktime/MOV file format detected. >>> [mov] Video stream found, -vid 0 >>> VIDEO: [mp4v] 640x480 24bpp inf fps 0.0 kbps ( 0.0 kbyte/s) >>> open: No such file or directory >>> [MGA] Couldn't open: /dev/mga_vid >>> open: No such file or directory >>> [MGA] Couldn't open: /dev/mga_vid >>> [VO_TDFXFB] Can't open /dev/fb0: Permission denied. >>> [VO_3DFX] Unable to open /dev/3dfx. >>> ========================================================================== >>> Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family >>> Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) >>> ========================================================================== >>> Audio: no sound >>> Starting playback... >>> VDec: vo config request - 640 x 480 (preferred colorspace: Planar YV12) >>> VDec: using Planar YV12 as output csp (no 0) >>> Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. >>> VO: [xv] 640x480 => 640x480 Planar YV12 >>> V: 0.0 200/200 ??% ??% ??,?% 0 0 >>> >>> Exiting... (End of file) >>> >>> But the error info not like the old one! >>> >>> On Wed, Dec 22, 2010 at 9:17 PM, liJin wrote: >>> I think thats the issue... >>> >>> >>> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=1.mp4 >>> >>> run with -e option >>> >>> >>> >>> On Wed, Dec 22, 2010 at 6:01 PM, Santakivi Topi wrote: >>> On 12/22/2010 12:18 PM, michael wrote: >>> > Now I can record the video to a AVI file with mpeg4 encoding by : >>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4! avimux! filesink >>> > location=record.avi >>> > >>> > But I got two issues: >>> > 1: I want record as a MP4 file, I do it like this: >>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>> > framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux! filesink >>> > location=record.avi >>> > I can record the video , but when I playback it by mplayer, I got a error: >>> > Playing ./record.mp4. >>> > libavformat file format detected. >>> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8741640]moov atom not found >>> > LAVF_header: av_open_input_stream() failed >>> > ISO: File Type Major Brand: ISO/IEC 14496-1 (MPEG-4 system) v2 >>> >>> When recording, with the above gst-launch line, do you stop >>> the pipeline with Ctrl+C..? >>> >>> It could be that the file container doesn't get fully written >>> in case of such an interrupt and that's why the clip doesn't play. >>> >>> For example, if you use this pipeline and >>> stop it with Ctrl+C, the clip doesn't play: >>> >>> gst-launch-0.10 videotestsrc ! >>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace >>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>> >>> But this pipeline produces a valid video file: >>> >>> gst-launch-0.10 videotestsrc num-buffers=200 ! >>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' ! ffmpegcolorspace >>> ! ffenc_mpeg4 ! mp4mux ! filesink location=test.mp4 >>> >>> > >>> > 2: I want encode the video to H264, so I do like this: >>> > gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240, >>> > framerate=30/1' ! ffmpegcolorspace ! x264enc! avimux ! filesink >>> > location=record.avi >>> >>> How about adding the format to the caps, e.g . >>> v4l2src ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >>> framerate=30/1' ! fmpegcolorspace ! x264enc! avimux ! filesink >>> location=record.avi ? >>> >>> BR, >>> Topi >>> >>> > But when I run the pipeline I just get the error info like this: >>> > Setting pipeline to PAUSED ... >>> > Pipeline is live and does not need PREROLL ... >>> > Setting pipeline to PLAYING ... >>> > New clock: GstSystemClock >>> > ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Internal >>> > data flow error. >>> > Additional debug info: >>> > gstbasesrc.c(2550): gst_base_src_loop (): >>> > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>> > streaming task paused, reason not-negotiated (-4) >>> > Execution ended after 1727716724 ns. >>> > Setting pipeline to PAUSED ... >>> > Setting pipeline to READY ... >>> > Setting pipeline to NULL ... >>> > Freeing pipeline ... >>> > >>> > Some one can help me fix the issues? Tks :) >>> > >>> > >>> > On 12/22/10, Timothy Braun >> > > wrote: >>> > > H264 is encoded with x264enc element. >>> > > >>> > > Sent from my iPhone >>> > > >>> > > On Dec 22, 2010, at 3:10 AM, wl2776 >> > > wrote: >>> > > >>> > >> >>> > >> >>> > >> michael young wrote: >>> > >>> >>> > >>> And now , I want to record the video to a MP4 file with the h264 >>> > encoding >>> > >>> format , How can I do ? >>> > >>> >>> > >>> On Wed, Dec 22, 2010 at 2:26 PM, michael >> > > wrote: >>> > >>> >>> > >>>> tks, this is work: gst-launch v4l2src device=/dev/video0 ! >>> > >>>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! xvimagesink >>> > >>> >>> > >>> >>> > >> >>> > >> Looks like you can't, because h264 encoder is unavailable. >>> > >> gst-inspect | grep h264 >>> > >> doesn't show it. >>> > >> However, general idea is to encode, then mux, and then save to a file. >>> > >> gst-launch v4l2src device=/dev/video0 ! >>> > >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffenc_mpeg4 ! >>> > >> mp4mux !filesink location=/home/user/file. >>> > >> >>> > >> -- >>> > >> View this message in context: >>> > >> >>> > http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >>> > >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >>> > >> >>> > >> >>> > ------------------------------------------------------------------------------ >>> > >> Forrester recently released a report on the Return on Investment >>> > (ROI) of >>> > >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> > >> within 7 months. Over 3 million businesses have gone Google with Google >>> > >> Apps: >>> > >> an online email calendar, and document program that's accessible >>> > from your >>> > >> >>> > >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >>> > >> _______________________________________________ >>> > >> gstreamer-devel mailing list >>> > >> gstreamer-devel at lists.sourceforge.net >>> > >>> > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> > > >>> > > >>> > ------------------------------------------------------------------------------ >>> > > Forrester recently released a report on the Return on Investment (ROI) of >>> > > Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> > > within 7 months. Over 3 million businesses have gone Google with Google >>> > > Apps: >>> > > an online email calendar, and document program that's accessible from >>> > your >>> > > browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >>> > > _______________________________________________ >>> > > gstreamer-devel mailing list >>> > > gstreamer-devel at lists.sourceforge.net >>> > >>> > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> > > >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google Apps: >>> an online email calendar, and document program that's accessible from your >>> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google Apps: >>> an online email calendar, and document program that's accessible from your >>> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million businesses have gone Google with Google Apps: >>> an online email calendar, and document program that's accessible from your >>> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost savings, and break-even >> within 7 months. Over 3 million businesses have gone Google with Google Apps: >> an online email calendar, and document program that's accessible from your >> browser. Read the Forrester report: http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From Topi.Santakivi at digia.com Thu Dec 23 08:30:42 2010 From: Topi.Santakivi at digia.com (Santakivi Topi) Date: Thu, 23 Dec 2010 09:30:42 +0200 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> <2F24B456-1BE7-411D-9206-580C811353C5@gmail.com> Message-ID: <4D12FAA2.3000002@digia.com> On 12/23/2010 09:27 AM, michael wrote: > how to "dropping the capsfilter" ? tks:) Just remove the 'video/x-raw-yuv,width=320,height=240, framerate=30/1' BR, Topi > > On Thu, Dec 23, 2010 at 3:06 PM, Timothy Braun > wrote: > > Try dropping the capsfilter and see if you get any results then. > > Sent from my iPhone > > On Dec 23, 2010, at 1:52 AM, michael > wrote: > >> Really!!?? but it's not work on my book!! God,, this issue >> drives me crazy!! :( >> >> On Thu, Dec 23, 2010 at 2:42 PM, liJin >> <lijinsyam at gmail.com >> > wrote: >> >> Hi >> >> >> gst-launch -e v4l2src ! 'video/x-raw-yuv,width=320,height=240, >> framerate=30/1' ! ffmpegcolorspace ! x264enc ! mp4mux ! >> filesink location=1.mp4 >> >> This pipeline working fine (4 me)......:) >> >> >> On Thu, Dec 23, 2010 at 12:08 PM, michael >> <mcygogo at gmail.com >> > wrote: >> >> Is there nobody know how to record video from camera to a >> MP4 container with the H264 encoding ? >> >> >> On Thu, Dec 23, 2010 at 9:46 AM, michael >> <mcygogo at gmail.com >> > wrote: >> >> neither qtmux nor mp4mux is not work well >> >> >> On Wed, Dec 22, 2010 at 11:24 PM, Timothy Braun >> <braunsquared at gmail.com >> > wrote: >> >> Not qtmux, use mp4mux. >> >> >> On Wed, Dec 22, 2010 at 10:10 AM, michael >> <mcygogo at gmail.com >> > wrote: >> >> I have tested changing the muxer to qtmux, but >> it still not work >> >> >> On Wed, Dec 22, 2010 at 10:58 PM, Timothy >> Braun >> <braunsquared at gmail.com >> > wrote: >> >> I was never able to get the ffmpeg mpeg4 >> mixer to work quite right. I found that >> mp4mux, an extension of qtmux, works much >> better. If you inspect the ffmpeg mp4 >> muxer they actually suggest that you use a >> different one as it's unstable. >> >> Best, >> Tim >> >> Sent from my iPhone >> >> On Dec 22, 2010, at 8:27 AM, michael >> <mcygogo at gmail.com >> > wrote: >> >>> I just ran it : >>> gst-launch -e v4l2src num-buffers=200 ! >>> video/x-raw-yuv,width=640,height=480,framerate=30/1 >>> ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux >>> ! filesink location=record.mp4 >>> And got a mp4 file , but it still cannot >>> be playbacked, and give the error info: >>> Playing record.mp4. >>> libavformat file format detected. >>> [mpeg4 @ 0x9376c10]hmm, seems the headers >>> are not complete, trying to guess >>> time_increment_bits >>> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >>> [mpeg4 @ 0x9376c10]looks like this file >>> was encoded with >>> (divx4/(old)xvid/opendivx) -> forcing >>> low_delay flag >>> [mpeg4 @ 0x9376c10]picture size invalid (0x0) >>> [mpeg4 @ 0x9376c10]get_buffer() failed >>> (-1 0 0 (nil)) >>> [mov,mp4,m4a,3gp,3g2,mj2 @ >>> 0x9375640]Could not find codec parameters >>> (Video: mpeg4, yuv420p) >>> LAVF_header: av_find_stream_info() failed >>> ISO: File Type Major Brand: ISO/IEC >>> 14496-1 (MPEG-4 system) v2 >>> Quicktime/MOV file format detected. >>> [mov] Video stream found, -vid 0 >>> VIDEO: [mp4v] 640x480 24bpp inf fps >>> 0.0 kbps ( 0.0 kbyte/s) >>> open: No such file or directory >>> [MGA] Couldn't open: /dev/mga_vid >>> open: No such file or directory >>> [MGA] Couldn't open: /dev/mga_vid >>> [VO_TDFXFB] Can't open /dev/fb0: >>> Permission denied. >>> [VO_3DFX] Unable to open /dev/3dfx. >>> ========================================================================== >>> Opening video decoder: [ffmpeg] FFmpeg's >>> libavcodec codec family >>> Selected video codec: [ffodivx] vfm: >>> ffmpeg (FFmpeg MPEG-4) >>> ========================================================================== >>> Audio: no sound >>> Starting playback... >>> VDec: vo config request - 640 x 480 >>> (preferred colorspace: Planar YV12) >>> VDec: using Planar YV12 as output csp (no 0) >>> Movie-Aspect is 1.33:1 - prescaling to >>> correct movie aspect. >>> VO: [xv] 640x480 => 640x480 Planar YV12 >>> V: 0.0 200/200 ??% ??% ??,?% 0 0 >>> >>> Exiting... (End of file) >>> >>> But the error info not like the old one! >>> >>> On Wed, Dec 22, 2010 at 9:17 PM, liJin >>> <lijinsyam at gmail.com >>> > wrote: >>> >>> I think thats the issue... >>> >>> >>> gst-launch -e v4l2src ! >>> 'video/x-raw-yuv,width=320,height=240, framerate=30/1' >>> ! ffmpegcolorspace ! ffenc_mpeg4 ! >>> mp4mux ! filesink location=1.mp4 >>> >>> run with*-e *option >>> >>> >>> >>> On Wed, Dec 22, 2010 at 6:01 PM, >>> Santakivi Topi >>> <Topi.Santakivi at digia.com >>> > wrote: >>> >>> On 12/22/2010 12:18 PM, michael >>> wrote: >>> > Now I can record the video to a >>> AVI file with mpeg4 encoding by : >>> > gst-launch v4l2src ! >>> 'video/x-raw-yuv,width=320,height=240, >>> > framerate=30/1' ! >>> ffmpegcolorspace ! ffenc_mpeg4! >>> avimux! filesink >>> > location=record.avi >>> > >>> > But I got two issues: >>> > 1: I want record as a MP4 file, >>> I do it like this: >>> > gst-launch v4l2src ! >>> 'video/x-raw-yuv,width=320,height=240, >>> > framerate=30/1' ! >>> ffmpegcolorspace ! ffenc_mpeg4 ! >>> mp4mux! filesink >>> > location=record.avi >>> > I can record the video , but >>> when I playback it by mplayer, I >>> got a error: >>> > Playing ./record.mp4. >>> > libavformat file format detected. >>> > [mov,mp4,m4a,3gp,3g2,mj2 @ >>> 0x8741640]moov atom not found >>> > LAVF_header: >>> av_open_input_stream() failed >>> > ISO: File Type Major Brand: >>> ISO/IEC 14496-1 (MPEG-4 system) v2 >>> >>> When recording, with the above >>> gst-launch line, do you stop >>> the pipeline with Ctrl+C..? >>> >>> It could be that the file >>> container doesn't get fully written >>> in case of such an interrupt and >>> that's why the clip doesn't play. >>> >>> For example, if you use this >>> pipeline and >>> stop it with Ctrl+C, the clip >>> doesn't play: >>> >>> gst-launch-0.10 videotestsrc ! >>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' >>> ! ffmpegcolorspace >>> ! ffenc_mpeg4 ! mp4mux ! filesink >>> location=test.mp4 >>> >>> But this pipeline produces a >>> valid video file: >>> >>> gst-launch-0.10 videotestsrc >>> num-buffers=200 ! >>> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' >>> ! ffmpegcolorspace >>> ! ffenc_mpeg4 ! mp4mux ! filesink >>> location=test.mp4 >>> >>> > >>> > 2: I want encode the video to >>> H264, so I do like this: >>> > gst-launch v4l2src ! >>> 'video/x-raw-yuv,width=320,height=240, >>> > framerate=30/1' ! >>> ffmpegcolorspace ! x264enc! >>> avimux ! filesink >>> > location=record.avi >>> >>> How about adding the format to >>> the caps, e.g . >>> v4l2src ! >>> 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >>> framerate=30/1' ! fmpegcolorspace >>> ! x264enc! avimux ! filesink >>> location=record.avi ? >>> >>> BR, >>> Topi >>> >>> > But when I run the pipeline I >>> just get the error info like this: >>> > Setting pipeline to PAUSED ... >>> > Pipeline is live and does not >>> need PREROLL ... >>> > Setting pipeline to PLAYING ... >>> > New clock: GstSystemClock >>> > ERROR: from element >>> /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>> Internal >>> > data flow error. >>> > Additional debug info: >>> > gstbasesrc.c(2550): >>> gst_base_src_loop (): >>> > >>> /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >>> > streaming task paused, reason >>> not-negotiated (-4) >>> > Execution ended after >>> 1727716724 ns. >>> > Setting pipeline to PAUSED ... >>> > Setting pipeline to READY ... >>> > Setting pipeline to NULL ... >>> > Freeing pipeline ... >>> > >>> > Some one can help me fix the >>> issues? Tks :) >>> > >>> > >>> > On 12/22/10, Timothy Braun >>> <braunsquared at gmail.com >>> >>> > >>> braunsquared at gmail.com >>> >> >>> wrote: >>> > > H264 is encoded with x264enc >>> element. >>> > > >>> > > Sent from my iPhone >>> > > >>> > > On Dec 22, 2010, at 3:10 AM, >>> wl2776 >>> <wl2776 at gmail.com >>> >>> > >>> wl2776 at gmail.com >>> >> wrote: >>> > > >>> > >> >>> > >> >>> > >> michael young wrote: >>> > >>> >>> > >>> And now , I want to record >>> the video to a MP4 file with the h264 >>> > encoding >>> > >>> format , How can I do ? >>> > >>> >>> > >>> On Wed, Dec 22, 2010 at >>> 2:26 PM, michael >>> <mcygogo at gmail.com >>> >>> > >>> mcygogo at gmail.com >>> >> wrote: >>> > >>> >>> > >>>> tks, this is work: >>> gst-launch v4l2src >>> device=/dev/video0 ! >>> > >>>> >>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' >>> ! xvimagesink >>> > >>> >>> > >>> >>> > >> >>> > >> Looks like you can't, >>> because h264 encoder is unavailable. >>> > >> gst-inspect | grep h264 >>> > >> doesn't show it. >>> > >> However, general idea is to >>> encode, then mux, and then save >>> to a file. >>> > >> gst-launch v4l2src >>> device=/dev/video0 ! >>> > >> >>> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' >>> ! ffenc_mpeg4 ! >>> > >> mp4mux !filesink >>> location=/home/user/file. >>> > >> >>> > >> -- >>> > >> View this message in context: >>> > >> >>> > >>> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >>> > >> Sent from the >>> GStreamer-devel mailing list >>> archive at >>> Nabble.com >>> . >>> > >> >>> > >> >>> > >>> ------------------------------------------------------------------------------ >>> > >> Forrester recently released >>> a report on the Return on Investment >>> > (ROI) of >>> > >> Google Apps. They found a >>> 300% ROI, 38%-56% cost savings, >>> and break-even >>> > >> within 7 months. Over 3 >>> million businesses have gone >>> Google with Google >>> > >> Apps: >>> > >> an online email calendar, >>> and document program that's >>> accessible >>> > from your >>> > >> >>> > >> browser. Read the Forrester >>> report: >>> http://p.sf.net/sfu/googleapps-sfnew >>> > >> >>> _______________________________________________ >>> > >> gstreamer-devel mailing list >>> > >> >>> gstreamer-devel at lists.sourceforge.net >>> >>> > >>> gstreamer-devel at lists.sourceforge.net >>> > >>> > >> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> > > >>> > > >>> > >>> ------------------------------------------------------------------------------ >>> > > Forrester recently released a >>> report on the Return on >>> Investment (ROI) of >>> > > Google Apps. They found a >>> 300% ROI, 38%-56% cost savings, >>> and break-even >>> > > within 7 months. Over 3 >>> million businesses have gone >>> Google with Google >>> > > Apps: >>> > > an online email calendar, and >>> document program that's >>> accessible from >>> > your >>> > > browser. Read the Forrester >>> report: >>> http://p.sf.net/sfu/googleapps-sfnew >>> > > >>> _______________________________________________ >>> > > gstreamer-devel mailing list >>> > > >>> gstreamer-devel at lists.sourceforge.net >>> >>> > >>> gstreamer-devel at lists.sourceforge.net >>> > >>> > > >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> > > >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a >>> report on the Return on >>> Investment (ROI) of >>> Google Apps. They found a 300% >>> ROI, 38%-56% cost savings, and >>> break-even >>> within 7 months. Over 3 million >>> businesses have gone Google with >>> Google Apps: >>> an online email calendar, and >>> document program that's >>> accessible from your >>> browser. Read the Forrester >>> report: >>> http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report >>> on the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, >>> 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million >>> businesses have gone Google with >>> Google Apps: >>> an online email calendar, and >>> document program that's accessible >>> from your >>> browser. Read the Forrester report: >>> http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> ------------------------------------------------------------------------------ >>> Forrester recently released a report on >>> the Return on Investment (ROI) of >>> Google Apps. They found a 300% ROI, >>> 38%-56% cost savings, and break-even >>> within 7 months. Over 3 million >>> businesses have gone Google with Google Apps: >>> an online email calendar, and document >>> program that's accessible from your >>> browser. Read the Forrester report: >>> http://p.sf.net/sfu/googleapps-sfnew >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on >> the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, >> 38%-56% cost savings, and break-even >> within 7 months. Over 3 million >> businesses have gone Google with Google Apps: >> an online email calendar, and document >> program that's accessible from your >> browser. Read the Forrester report: >> http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the >> Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% >> cost savings, and break-even >> within 7 months. Over 3 million businesses >> have gone Google with Google Apps: >> an online email calendar, and document program >> that's accessible from your >> browser. Read the Forrester report: >> http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report on the Return >> on Investment (ROI) of >> Google Apps. They found a 300% ROI, 38%-56% cost >> savings, and break-even >> within 7 months. Over 3 million businesses have >> gone Google with Google Apps: >> an online email calendar, and document program >> that's accessible from your >> browser. Read the Forrester report: >> http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node >> allows customers >> to consolidate database storage, standardize their >> database environment, and, >> should the need arise, upgrade to a full multi-node Oracle >> RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node >> allows customers >> to consolidate database storage, standardize their database >> environment, and, >> should the need arise, upgrade to a full multi-node Oracle RAC >> database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows >> customers >> to consolidate database storage, standardize their database >> environment, and, >> should the need arise, upgrade to a full multi-node Oracle RAC >> database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows > customers > to consolidate database storage, standardize their database > environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From yuanqing.cai at tieto.com Thu Dec 23 09:34:26 2010 From: yuanqing.cai at tieto.com (Cai Yuanqing) Date: Thu, 23 Dec 2010 16:34:26 +0800 Subject: [gst-devel] how to create a pipeline to record video from camera? In-Reply-To: References: <1293005445306-3160261.post@n4.nabble.com> <4D11EFA7.8080300@digia.com> <8C8F3C72-A489-4D23-B84E-4BF96B6B246E@gmail.com> Message-ID: <4D130992.60103@tieto.com> Hi Michael: I tried command below and it works well: $ gst-launch-0.10 -e v4l2src num-buffers=200 ! 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffmpegcolorspace ! ffenc_mpeg4 ! mp4mux ! filesink location=test1.mp4 Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock Got EOS from element "pipeline0". Execution ended after 6682641000 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) ========================================================================== Audio: no sound Starting playback... Movie-Aspect is 1.33:1 - prescaling to correct movie aspect. VO: [xv] 640x480 => 640x480 Planar YV12 V: 0.4 0/ 0 5% 0% 0.0% 0 0 Exiting... (End of file) Seems that you just can record MP4 file but can't playback. maybe you can try the command below to playback it in command line: $ gst-launch-0.10 filesrc location=test1.mp4 ! ffdemux_mov_mp4_m4a_3gp_3g2_mj2 ! ffdec_mpeg4 ! 'video/x-raw-yuv,width=640,height=480,framerate=30/1' ! ffmpegcolorspace ! xvimagesink BTW,you could use gst-inspect-0.10 as much as possible to guess and try the elements you want. On 12/23/2010 03:26 PM, michael wrote: > My Gstreamer and Plugins version is : > gst-ffmpeg-0.10.11 > gst-plugins-bad-0.10.20 > gst-plugins-base-0.10.30 > gst-plugins-good-0.10.25 > gst-plugins-ugly-0.10.16 > gstreamer-0.10.30 > So I think the version is ok, isn't it? > and the all plugin can run, so I think the supported library packages > is ready. > > On Thu, Dec 23, 2010 at 3:06 PM, sudarshan bisht > > wrote: > > > Make sure you have all latest Gstreamer core, plugins and > supported library packages installed. > > > On Thu, Dec 23, 2010 at 12:22 PM, michael > wrote: > > Really!!?? but it's not work on my book!! God,, this issue > drives me crazy!! :( > > > On Thu, Dec 23, 2010 at 2:42 PM, liJin > wrote: > > Hi > > > gst-launch -e v4l2src ! > 'video/x-raw-yuv,width=320,height=240, framerate=30/1' ! > ffmpegcolorspace ! x264enc ! mp4mux ! filesink location=1.mp4 > > This pipeline working fine (4 me)......:) > > > On Thu, Dec 23, 2010 at 12:08 PM, michael > > wrote: > > Is there nobody know how to record video from camera > to a MP4 container with the H264 encoding ? > > > On Thu, Dec 23, 2010 at 9:46 AM, michael > > wrote: > > neither qtmux nor mp4mux is not work well > > > On Wed, Dec 22, 2010 at 11:24 PM, Timothy Braun > > wrote: > > Not qtmux, use mp4mux. > > > On Wed, Dec 22, 2010 at 10:10 AM, michael > > > wrote: > > I have tested changing the muxer to qtmux, > but it still not work > > > On Wed, Dec 22, 2010 at 10:58 PM, Timothy > Braun > wrote: > > I was never able to get the ffmpeg > mpeg4 mixer to work quite right. I > found that mp4mux, an extension of > qtmux, works much better. If you > inspect the ffmpeg mp4 muxer they > actually suggest that you use a > different one as it's unstable. > > Best, > Tim > > Sent from my iPhone > > On Dec 22, 2010, at 8:27 AM, michael > > wrote: > >> I just ran it : >> gst-launch -e v4l2src num-buffers=200 >> ! >> video/x-raw-yuv,width=640,height=480,framerate=30/1 >> ! ffmpegcolorspace ! ffenc_mpeg4 ! >> mp4mux ! filesink location=record.mp4 >> And got a mp4 file , but it still >> cannot be playbacked, and give the >> error info: >> Playing record.mp4. >> libavformat file format detected. >> [mpeg4 @ 0x9376c10]hmm, seems the >> headers are not complete, trying to >> guess time_increment_bits >> [mpeg4 @ 0x9376c10]my guess is 5 bits ;) >> [mpeg4 @ 0x9376c10]looks like this >> file was encoded with >> (divx4/(old)xvid/opendivx) -> forcing >> low_delay flag >> [mpeg4 @ 0x9376c10]picture size >> invalid (0x0) >> [mpeg4 @ 0x9376c10]get_buffer() >> failed (-1 0 0 (nil)) >> [mov,mp4,m4a,3gp,3g2,mj2 @ >> 0x9375640]Could not find codec >> parameters (Video: mpeg4, yuv420p) >> LAVF_header: av_find_stream_info() failed >> ISO: File Type Major Brand: ISO/IEC >> 14496-1 (MPEG-4 system) v2 >> Quicktime/MOV file format detected. >> [mov] Video stream found, -vid 0 >> VIDEO: [mp4v] 640x480 24bpp inf >> fps 0.0 kbps ( 0.0 kbyte/s) >> open: No such file or directory >> [MGA] Couldn't open: /dev/mga_vid >> open: No such file or directory >> [MGA] Couldn't open: /dev/mga_vid >> [VO_TDFXFB] Can't open /dev/fb0: >> Permission denied. >> [VO_3DFX] Unable to open /dev/3dfx. >> ========================================================================== >> Opening video decoder: [ffmpeg] >> FFmpeg's libavcodec codec family >> Selected video codec: [ffodivx] vfm: >> ffmpeg (FFmpeg MPEG-4) >> ========================================================================== >> Audio: no sound >> Starting playback... >> VDec: vo config request - 640 x 480 >> (preferred colorspace: Planar YV12) >> VDec: using Planar YV12 as output csp >> (no 0) >> Movie-Aspect is 1.33:1 - prescaling >> to correct movie aspect. >> VO: [xv] 640x480 => 640x480 Planar YV12 >> V: 0.0 200/200 ??% ??% ??,?% 0 0 >> >> Exiting... (End of file) >> >> But the error info not like the old one! >> >> On Wed, Dec 22, 2010 at 9:17 PM, >> liJin > > wrote: >> >> I think thats the issue... >> >> >> gst-launch -e v4l2src ! >> 'video/x-raw-yuv,width=320,height=240, >> framerate=30/1' ! >> ffmpegcolorspace ! ffenc_mpeg4 ! >> mp4mux ! filesink location=1.mp4 >> >> run with* -e *option >> >> >> >> On Wed, Dec 22, 2010 at 6:01 PM, >> Santakivi Topi >> > > wrote: >> >> On 12/22/2010 12:18 PM, >> michael wrote: >> > Now I can record the video >> to a AVI file with mpeg4 >> encoding by : >> > gst-launch v4l2src ! >> 'video/x-raw-yuv,width=320,height=240, >> > framerate=30/1' ! >> ffmpegcolorspace ! >> ffenc_mpeg4! avimux! filesink >> > location=record.avi >> > >> > But I got two issues: >> > 1: I want record as a MP4 >> file, I do it like this: >> > gst-launch v4l2src ! >> 'video/x-raw-yuv,width=320,height=240, >> > framerate=30/1' ! >> ffmpegcolorspace ! >> ffenc_mpeg4 ! mp4mux! filesink >> > location=record.avi >> > I can record the video , >> but when I playback it by >> mplayer, I got a error: >> > Playing ./record.mp4. >> > libavformat file format >> detected. >> > [mov,mp4,m4a,3gp,3g2,mj2 @ >> 0x8741640]moov atom not found >> > LAVF_header: >> av_open_input_stream() failed >> > ISO: File Type Major Brand: >> ISO/IEC 14496-1 (MPEG-4 >> system) v2 >> >> When recording, with the >> above gst-launch line, do you >> stop >> the pipeline with Ctrl+C..? >> >> It could be that the file >> container doesn't get fully >> written >> in case of such an interrupt >> and that's why the clip >> doesn't play. >> >> For example, if you use this >> pipeline and >> stop it with Ctrl+C, the clip >> doesn't play: >> >> gst-launch-0.10 videotestsrc ! >> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' >> ! ffmpegcolorspace >> ! ffenc_mpeg4 ! mp4mux ! >> filesink location=test.mp4 >> >> But this pipeline produces a >> valid video file: >> >> gst-launch-0.10 videotestsrc >> num-buffers=200 ! >> 'video/x-raw-yuv,width=320,height=240,framerate=30/1' >> ! ffmpegcolorspace >> ! ffenc_mpeg4 ! mp4mux ! >> filesink location=test.mp4 >> >> > >> > 2: I want encode the video >> to H264, so I do like this: >> > gst-launch v4l2src ! >> 'video/x-raw-yuv,width=320,height=240, >> > framerate=30/1' ! >> ffmpegcolorspace ! x264enc! >> avimux ! filesink >> > location=record.avi >> >> How about adding the format >> to the caps, e.g . >> v4l2src ! >> 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240, >> framerate=30/1' ! >> fmpegcolorspace ! x264enc! >> avimux ! filesink >> location=record.avi ? >> >> BR, >> Topi >> >> > But when I run the pipeline >> I just get the error info >> like this: >> > Setting pipeline to PAUSED ... >> > Pipeline is live and does >> not need PREROLL ... >> > Setting pipeline to PLAYING ... >> > New clock: GstSystemClock >> > ERROR: from element >> /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >> Internal >> > data flow error. >> > Additional debug info: >> > gstbasesrc.c(2550): >> gst_base_src_loop (): >> > >> /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: >> > streaming task paused, >> reason not-negotiated (-4) >> > Execution ended after >> 1727716724 ns. >> > Setting pipeline to PAUSED ... >> > Setting pipeline to READY ... >> > Setting pipeline to NULL ... >> > Freeing pipeline ... >> > >> > Some one can help me fix >> the issues? Tks :) >> > >> > >> > On 12/22/10, Timothy Braun >> > >> > >> >> >> wrote: >> > > H264 is encoded with >> x264enc element. >> > > >> > > Sent from my iPhone >> > > >> > > On Dec 22, 2010, at 3:10 >> AM, wl2776 > >> > > >> >> wrote: >> > > >> > >> >> > >> >> > >> michael young wrote: >> > >>> >> > >>> And now , I want to >> record the video to a MP4 >> file with the h264 >> > encoding >> > >>> format , How can I do ? >> > >>> >> > >>> On Wed, Dec 22, 2010 at >> 2:26 PM, michael >> > >> > > >> >> wrote: >> > >>> >> > >>>> tks, this is work: >> gst-launch v4l2src >> device=/dev/video0 ! >> > >>>> >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' >> ! xvimagesink >> > >>> >> > >>> >> > >> >> > >> Looks like you can't, >> because h264 encoder is >> unavailable. >> > >> gst-inspect | grep h264 >> > >> doesn't show it. >> > >> However, general idea is >> to encode, then mux, and then >> save to a file. >> > >> gst-launch v4l2src >> device=/dev/video0 ! >> > >> >> 'video/x-raw-yuv,width=640,height=480,framerate=30/1' >> ! ffenc_mpeg4 ! >> > >> mp4mux !filesink >> location=/home/user/file. >> > >> >> > >> -- >> > >> View this message in >> context: >> > >> >> > >> http://gstreamer-devel.966125.n4.nabble.com/how-to-create-a-pipeline-to-record-video-from-camera-tp3134399p3160261.html >> > >> Sent from the >> GStreamer-devel mailing list >> archive at Nabble.com >> . >> > >> >> > >> >> > >> ------------------------------------------------------------------------------ >> > >> Forrester recently >> released a report on the >> Return on Investment >> > (ROI) of >> > >> Google Apps. They found >> a 300% ROI, 38%-56% cost >> savings, and break-even >> > >> within 7 months. Over 3 >> million businesses have gone >> Google with Google >> > >> Apps: >> > >> an online email >> calendar, and document >> program that's accessible >> > from your >> > >> >> > >> browser. Read the >> Forrester report: >> http://p.sf.net/sfu/googleapps-sfnew >> > >> >> _______________________________________________ >> > >> gstreamer-devel mailing list >> > >> >> gstreamer-devel at lists.sourceforge.net >> >> > >> > > >> > >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > >> > > >> > >> ------------------------------------------------------------------------------ >> > > Forrester recently >> released a report on the >> Return on Investment (ROI) of >> > > Google Apps. They found a >> 300% ROI, 38%-56% cost >> savings, and break-even >> > > within 7 months. Over 3 >> million businesses have gone >> Google with Google >> > > Apps: >> > > an online email calendar, >> and document program that's >> accessible from >> > your >> > > browser. Read the >> Forrester report: >> http://p.sf.net/sfu/googleapps-sfnew >> > > >> _______________________________________________ >> > > gstreamer-devel mailing list >> > > >> gstreamer-devel at lists.sourceforge.net >> >> > >> > > >> > > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > >> >> ------------------------------------------------------------------------------ >> Forrester recently released a >> report on the Return on >> Investment (ROI) of >> Google Apps. They found a >> 300% ROI, 38%-56% cost >> savings, and break-even >> within 7 months. Over 3 >> million businesses have gone >> Google with Google Apps: >> an online email calendar, and >> document program that's >> accessible from your >> browser. Read the Forrester >> report: >> http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a >> report on the Return on >> Investment (ROI) of >> Google Apps. They found a 300% >> ROI, 38%-56% cost savings, and >> break-even >> within 7 months. Over 3 million >> businesses have gone Google with >> Google Apps: >> an online email calendar, and >> document program that's >> accessible from your >> browser. Read the Forrester >> report: >> http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> ------------------------------------------------------------------------------ >> Forrester recently released a report >> on the Return on Investment (ROI) of >> Google Apps. They found a 300% ROI, >> 38%-56% cost savings, and break-even >> within 7 months. Over 3 million >> businesses have gone Google with >> Google Apps: >> an online email calendar, and >> document program that's accessible >> from your >> browser. Read the Forrester report: >> http://p.sf.net/sfu/googleapps-sfnew >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Forrester recently released a report > on the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, > 38%-56% cost savings, and break-even > within 7 months. Over 3 million > businesses have gone Google with > Google Apps: > an online email calendar, and document > program that's accessible from your > browser. Read the Forrester report: > http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on > the Return on Investment (ROI) of > Google Apps. They found a 300% ROI, > 38%-56% cost savings, and break-even > within 7 months. Over 3 million > businesses have gone Google with Google Apps: > an online email calendar, and document > program that's accessible from your > browser. Read the Forrester report: > http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Forrester recently released a report on the > Return on Investment (ROI) of > Google Apps. They found a 300% ROI, 38%-56% > cost savings, and break-even > within 7 months. Over 3 million businesses > have gone Google with Google Apps: > an online email calendar, and document program > that's accessible from your > browser. Read the Forrester report: > http://p.sf.net/sfu/googleapps-sfnew > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One > Node allows customers > to consolidate database storage, standardize their > database environment, and, > should the need arise, upgrade to a full multi-node > Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node > allows customers > to consolidate database storage, standardize their > database environment, and, > should the need arise, upgrade to a full multi-node Oracle > RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node > allows customers > to consolidate database storage, standardize their database > environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC > database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > Regards, > > Sudarshan Bisht > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows > customers > to consolidate database storage, standardize their database > environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC > database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- B.R Cai Yuanqing From gibrovacco at gmail.com Thu Dec 23 10:13:28 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Thu, 23 Dec 2010 11:13:28 +0200 Subject: [gst-devel] decodebin & gapless In-Reply-To: References: Message-ID: Hi, On Thu, Dec 23, 2010 at 12:25 AM, Leon Bogaert wrote: > Hi all, > > Could anyone give me a hand with how to use decodebin and make gapless playback work? I've tried searching for some example code but I couldn't really find anything. you've not googled enough (maybe you're using Bing?): http://sourceforge.net/mailarchive/message.php?msg_id=26763211 Regards > > Thanks in advance! > > Regards, > Leon > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From wl2776 at gmail.com Thu Dec 23 12:59:36 2010 From: wl2776 at gmail.com (wl2776) Date: Thu, 23 Dec 2010 03:59:36 -0800 (PST) Subject: [gst-devel] Problems in framestepping in mpeg-2. Message-ID: <1293105576543-3161957.post@n4.nabble.com> Hi all. I have some problems with framestepping. I. the position, returned by gst_element_query_position always remains the same after frame steps in MPEG-2 files. Since playbin2 responds with the same value to the _query_position, I've tried to develop a workaround by calculating the position by myself from the duration in the step-done message. Here the second problem has come. II. Suppose the following scenario. 1. I create playbin2. 2. load a mpeg2-PS file into it. 3. set the pipeline to PAUSED state (the previous was NULL) 4. seek to any position, say 6.96 sec (the pipeline is still in paused mode). 5. issue a new-step event. I observe that the new frame displayed is the next frame. However the 'duration' field in the step-done message contains the value of about 1 second of the time passed. I've switched on the debug output and have found that when I issue new-step event, mpegpsdemux sends new segment, beginning from the next SCR in the video file, which appears, say 7.58 sec. videosink catches this new segment and takes its start position into account when calculates step-done fields. Therefore I'm getting abnormally large amount of passed time. How can I fix these? I know that mpegpsdemux lacks the support, therefore I'm going to fix it by myself. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Problems-in-framestepping-in-mpeg-2-tp3161957p3161957.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From prabhu.fort at gmail.com Thu Dec 23 15:17:06 2010 From: prabhu.fort at gmail.com (prabhulinga Swamy B S) Date: Thu, 23 Dec 2010 19:47:06 +0530 Subject: [gst-devel] Problem with Pipeline udpsrc -> mux -> filesink In-Reply-To: <4D135915.5090402@globaledgesoft.com> References: <4D135915.5090402@globaledgesoft.com> Message-ID: Hi all, I'm able to send, receive and playback H264 video file using rtph264pay, rtph264depay and ximagesink elements across different hosts. I'm using following pipeline. host_send $ gst-launch filesrc location=./Cam111-20101126-193700.avi ! \ avidemux ! h264parse ! rtph264pay ! udpsink host=172.16.10.32 port=3000 host_rec $ gst-launch udpsrc port=3000 \ caps="application/x-rtp,media=(string)video,encoding-string=(string)H264" ! \ rtph264depay ! h264parse ! ffdec_h264 ! ffmpegcolorspace ! ximagesink The above combination of pipelines works correctly. The problem is, when I try to save the video into any container format( say, .avi or .mp4), udpsrc element reports "Internal data flow error". I tried following pipeline, host_rec $ gst-launch udpsrc port=3000 \ caps="application/x-rtp,media=(string)video,encoding-string=(string)H264" ! \ rtph264depay ! h264parse ! avimux ! filesink location=test.avi Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error. Additional debug info: gstbasesrc.c(2507): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: streaming task paused, reason not-negotiated (-4) Execution ended after 73780416 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... I wonder how it affects changing the sink element. I tried changing the properties of 'h264parse' and tried setting the full caps for 'udpsrc'. same problem persists. Can anybody guide me regarding this. -- Regards, Prabhu B S From braunsquared at gmail.com Thu Dec 23 16:08:13 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Thu, 23 Dec 2010 10:08:13 -0500 Subject: [gst-devel] Problem with Pipeline udpsrc -> mux -> filesink In-Reply-To: References: <4D135915.5090402@globaledgesoft.com> Message-ID: Take a look at what h264parse's src pad caps are and make sure they are compatible with avimux's sink pad caps. If not, you are going to need to add elements to convert the stream to something avimux is happy with. It will likely be an encoder and colorspace transform of some nature. If you set GST_DEBUG=2 you will get some helpful info as well. - Tim Sent from my iPhone On Dec 23, 2010, at 9:17 AM, prabhulinga Swamy B S wrote: > Hi all, > > I'm able to send, receive and playback H264 video file using rtph264pay, > rtph264depay and ximagesink elements across different hosts. I'm using > following pipeline. > > host_send $ gst-launch filesrc location=./Cam111-20101126-193700.avi ! \ > avidemux ! h264parse ! rtph264pay ! udpsink > host=172.16.10.32 port=3000 > > host_rec $ gst-launch udpsrc port=3000 \ > > caps="application/x-rtp,media=(string)video,encoding-string=(string)H264" ! > \ > rtph264depay ! h264parse ! ffdec_h264 ! ffmpegcolorspace ! > ximagesink > > The above combination of pipelines works correctly. > > The problem is, when I try to save the video into any container format( > say, .avi or .mp4), > udpsrc element reports "Internal data flow error". > > I tried following pipeline, > > host_rec $ gst-launch udpsrc port=3000 \ > > caps="application/x-rtp,media=(string)video,encoding-string=(string)H264" ! > \ > rtph264depay ! h264parse ! avimux ! filesink > location=test.avi > > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal > data flow error. > Additional debug info: > gstbasesrc.c(2507): gst_base_src_loop (): > /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: > streaming task paused, reason not-negotiated (-4) > Execution ended after 73780416 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > Freeing pipeline ... > > > I wonder how it affects changing the sink element. I tried changing > the properties of 'h264parse' and tried setting the full caps for 'udpsrc'. > same problem persists. > > Can anybody guide me regarding this. > > -- > > Regards, > Prabhu B S > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From radivojejovanovic at gmail.com Sat Dec 25 00:21:54 2010 From: radivojejovanovic at gmail.com (Radivoje Jovanovic) Date: Fri, 24 Dec 2010 15:21:54 -0800 Subject: [gst-devel] Video and audio speed issue? Message-ID: Hello, If I try playing just a sound from my video with following pipeline: gst-launch-0.10 filesrc location=MPEG4_MOVIE ! qtdemux name=t ! queue ! faad ! alsasink sound plays just fine and song duration is 38 seconds. If I play video only using following pipeline: gst-launch-0.10 filesrc location=MPEG4_MOVIE ! qtdemux name=t ! queue ! mpeg4dec ! mySink video plays just fine in duration of 38 seconds. However, if I play both of them like: gst-launch-0.10 filesrc location=MPEG4_MOVIE ! qtdemux name=t ! queue ! mpeg4dec ! mySink t. ! queue ! faad ! alsasink everything is much faster and the whole thing is over in 26 seconds. Any idea what might be causing this? Cheers Ogi -------------- next part -------------- An HTML attachment was scrubbed... URL: From shamun.toha at gmail.com Sat Dec 25 01:29:14 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sat, 25 Dec 2010 01:29:14 +0100 Subject: [gst-devel] bind failed -1: Cannot assign requested address (99) Message-ID: In same pc if i test this it works, now i included this in remote server and then it keeps giving this same error. Any idea suggestion kindly. server (fedora): ====== [root at ip-10-224-63-198 ~]# gst-launch -v filesrc location=/tmp/top.wav ! wavparse ! audioconvert ! speexenc ! rtpspeexpay ! udpsink auto-multicast=true port=4444 client (fedora): ============== [sun at example ~]$ gst-launch -v udpsrc multicast-group=xx.xx.122.66 port=4444 caps = "application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110, ssrc=(uint)90483196, clock-base=(uint)4057897018, seqnum-base=(uint)43738" ! rtpspeexdepay ! speexdec ! audioconvert ! autoaudiosink Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could not get/set settings from/on resource. Additional debug info: gstudpsrc.c(909): gst_udpsrc_start (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: bind failed -1: Cannot assign requested address (99) Setting pipeline to NULL ... /GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps = NULL Freeing pipeline ... From shamun.toha at gmail.com Sat Dec 25 12:24:16 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sat, 25 Dec 2010 12:24:16 +0100 Subject: [gst-devel] ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could not get/set settings from/on resource Message-ID: Tried multicast-iface still same. Please suggest. CentOS and Fedora Remote server: =========================== [root at www tmp]# gst-launch -v filesrc location=/tmp/top.wav ! wavparse ! audioconvert ! speexenc ! rtpspeexpay ! udpsink host=xxxx.64.12 auto-multicast=true port=4444 Setting pipeline to PAUSED ... Pipeline is PREROLLING ... /pipeline0/audioconvert0.src: caps = audio/x-raw-int, rate=(int)44100, channels=(int)1, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16 /pipeline0/audioconvert0.sink: caps = audio/x-raw-int, endianness=(int)1234, channels=(int)2, width=(int)16, depth=(int)16, signed=(boolean)true, rate=(int)44100 /pipeline0/speexenc0: last-message = "Warning: speex is optimized for 8, 16 and 32 KHz" /pipeline0/speexenc0.sink: caps = audio/x-raw-int, rate=(int)44100, channels=(int)1, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16 /pipeline0/speexenc0.src: caps = audio/x-speex, rate=(int)44100, channels=(int)1, streamheader=(buffer)< 5370656578202020312e302e35000000000000000000000000000000010000005000000044ac0000020000000400000001000000ffffffff800200000000000001000000000000000000000000000000, 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > /pipeline0/rtpspeexpay0.sink: caps = audio/x-speex, rate=(int)44100, channels=(int)1, streamheader=(buffer)< 5370656578202020312e302e35000000000000000000000000000000010000005000000044ac0000020000000400000001000000ffffffff800200000000000001000000000000000000000000000000, 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > /pipeline0/rtpspeexpay0.src: caps = application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110, ssrc=(guint)1634646208, clock-base=(guint)298586317, seqnum-base=(guint)13594 /pipeline0/udpsink0.sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110, ssrc=(guint)1634646208, clock-base=(guint)298586317, seqnum-base=(guint)13594 Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock Fedora Local ========== [sun at example ~]$ gst-launch -v udpsrc multicast-iface=xxx64.12 port=4444 caps ="application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110, ssrc=(guint)3450810950, clock-base=(guint)985102105, seqnum-base=(guint)33680" ! rtpspeexdepay ! speexdec ! audioconvert ! autoaudiosink Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock ^CCaught interrupt -- handling interrupt. Interrupt: Stopping pipeline ... Execution ended after 2995661460 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... /GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps = NULL Setting pipeline to NULL ... Freeing pipeline ... NOR [sun at example ~]$ gst-launch -v udpsrc multicast-group=xxxx.64.12 port=4444 caps ="application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110, ssrc=(guint)3450810950, clock-base=(guint)985102105, seqnum-base=(guint)33680" ! rtpspeexdepay ! speexdec ! audioconvert ! autoaudiosink Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could not get/set settings from/on resource. Additional debug info: gstudpsrc.c(909): gst_udpsrc_start (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: bind failed -1: Cannot assign requested address (99) Setting pipeline to NULL ... /GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps = NULL Freeing pipeline ... [sun at example ~]$ -------------- next part -------------- An HTML attachment was scrubbed... URL: From donny.viszneki at gmail.com Sat Dec 25 14:12:37 2010 From: donny.viszneki at gmail.com (Donny Viszneki) Date: Sat, 25 Dec 2010 08:12:37 -0500 Subject: [gst-devel] bind failed -1: Cannot assign requested address (99) In-Reply-To: References: Message-ID: strace the process! merry christmas! On Fri, Dec 24, 2010 at 7:29 PM, Shamun toha md wrote: > In same pc if i test this it works, now i included this in remote > server and then it keeps giving this same error. Any idea suggestion > kindly. > > server (fedora): > ====== > [root at ip-10-224-63-198 ~]# gst-launch -v filesrc location=/tmp/top.wav > ! wavparse ! audioconvert ! speexenc ! rtpspeexpay ! udpsink > auto-multicast=true port=4444 > > client (fedora): > ============== > [sun at example ~]$ gst-launch -v udpsrc multicast-group=xx.xx.122.66 > port=4444 caps = "application/x-rtp, media=(string)audio, > clock-rate=(int)44100, encoding-name=(string)SPEEX, > encoding-params=(string)1, payload=(int)110, ssrc=(uint)90483196, > clock-base=(uint)4057897018, seqnum-base=(uint)43738" ! rtpspeexdepay > ! speexdec ! audioconvert ! autoaudiosink > Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could > not get/set settings from/on resource. > Additional debug info: > gstudpsrc.c(909): gst_udpsrc_start (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: > bind failed -1: Cannot assign requested address (99) > Setting pipeline to NULL ... > /GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps = NULL > Freeing pipeline ... > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- http://codebad.com/ From shamun.toha at gmail.com Sat Dec 25 15:13:57 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sat, 25 Dec 2010 15:13:57 +0100 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 99 In-Reply-To: References: Message-ID: I tried now with 3 Peer involved, still not working, Please kindly advise, all selinux/iptables/firewalls are shutdown, but still no luck: Peer 1 (acting as server on CentOS 5.4): ------- PLAYING NICELY ================================================= [root at www tmp]# gst-launch -v filesrc location=/tmp/top.wav ! wavparse ! audioconvert ! speexenc ! rtpspeexpay ! udpsink host=xxxx.64.12 auto-multicast=true port=4444 Setting pipeline to PAUSED ... Pipeline is PREROLLING ... /pipeline0/audioconvert0.src: caps = audio/x-raw-int, rate=(int)44100, channels=(int)1, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16 /pipeline0/audioconvert0.sink: caps = audio/x-raw-int, endianness=(int)1234, channels=(int)2, width=(int)16, depth=(int)16, signed=(boolean)true, rate=(int)44100 /pipeline0/speexenc0: last-message = "Warning: speex is optimized for 8, 16 and 32 KHz" /pipeline0/speexenc0.sink: caps = audio/x-raw-int, rate=(int)44100, channels=(int)1, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16 /pipeline0/speexenc0.src: caps = audio/x-speex, rate=(int)44100, channels=(int)1, streamheader=(buffer)< 5370656578202020312e302e35000000000000000000000000000000010000005000000044ac0000020000000400000001000000ffffffff800200000000000001000000000000000000000000000000, 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > /pipeline0/rtpspeexpay0.sink: caps = audio/x-speex, rate=(int)44100, channels=(int)1, streamheader=(buffer)< 5370656578202020312e302e35000000000000000000000000000000010000005000000044ac0000020000000400000001000000ffffffff800200000000000001000000000000000000000000000000, 1f000000456e636f6465642077697468204753747265616d6572205370656578656e630000000001 > /pipeline0/rtpspeexpay0.src: caps = application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110, ssrc=(guint)3586096616, clock-base=(guint)3098534659, seqnum-base=(guint)60242 /pipeline0/udpsink0.sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110, ssrc=(guint)3586096616, clock-base=(guint)3098534659, seqnum-base=(guint)60242 Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock Peer 2 (listener on Fedora -> Amazon EC2): -------- FAILED ============================================== [root at ip-10-224-63-198 tmp]# gst-launch -v udpsrc multicast-group=xxxx.64.12 port=4444 caps ="application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110, ssrc=(guint)3450810950, clock-base=(guint)985102105, seqnum-base=(guint)33680" ! rtpspeexdepay ! speexdec ! audioconvert ! fakesink Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could not get/set settings from/on resource. Additional debug info: gstudpsrc.c(909): gst_udpsrc_start (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: bind failed -1: Cannot assign requested address (99) Setting pipeline to NULL ... /GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps = NULL Freeing pipeline ... Peer 3 (listener on Fedora -> MyLaptop):: -------- FAILED ============================================== [sun at example tmp]$ gst-launch -v udpsrc multicast-group=xxxx.64.12 port=4444 caps ="application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110, ssrc=(guint)3450810950, clock-base=(guint)985102105, seqnum-base=(guint)33680" ! rtpspeexdepay ! speexdec ! audioconvert ! autoaudiosink Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could not get/set settings from/on resource. Additional debug info: gstudpsrc.c(909): gst_udpsrc_start (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: bind failed -1: Cannot assign requested address (99) Setting pipeline to NULL ... /GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps = NULL Freeing pipeline ... What is causing this ? In local network it works, why not Remotely its working like the same way ?? > 7. ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: > Could not get/set settings from/on resource (Shamun toha md) > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shamun.toha at gmail.com Sat Dec 25 17:40:53 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sat, 25 Dec 2010 17:40:53 +0100 Subject: [gst-devel] gstreamer-devel Digest, Vol 55, Issue 99 In-Reply-To: References: Message-ID: Forget it, i know understand whats going on. lol hahahaha it works. -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Sun Dec 26 18:13:19 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sun, 26 Dec 2010 17:13:19 +0000 Subject: [gst-devel] Compilation error in "[gst-cvs] gst-plugins-bad: camerabin: fix for latest GTK+ API changes" ? In-Reply-To: <4D120055.9040303@nokia.com> References: <20101220193458.518EE10057@kemper.freedesktop.org> <4D120055.9040303@nokia.com> Message-ID: <1293383599.462.0.camel@zingle> On Wed, 2010-12-22 at 15:42 +0200, Markus Vartiainen wrote: > I just skimmed through latest commits and could not help noticing that > there is a typo in the variable "store" (it is written "sotre"): > > > +#if GTK_CHECK_VERSION (2, 91, 6) > > + GtkListStore *sotre; > > + GtkTreeIter iter; > > +#endif > (...) > To me it seems like this would not even compile, if the #if statements > resolve to true, since there is no other variable "store" defined in the > scope. Thanks, should be fixed in git now. Cheers -Tim From anirudhtomer at gmail.com Sun Dec 26 19:10:01 2010 From: anirudhtomer at gmail.com (Anirudh) Date: Sun, 26 Dec 2010 10:10:01 -0800 (PST) Subject: [gst-devel] What are the types of the parameters that can be passed to gst_init ? Message-ID: <1293387001551-3164362.post@n4.nabble.com> Hi everyone, I am a newbie in the field of Gstreamer and I have just started reading the official manual for application developers and its damn! interesting. I encountered that gst_init() function is used to initialize the Gstreamer. (fine) but I do not know what kind of parameters can be passed to it. They have given an example where they use it like : gst_init(&argc,&argv) but its not given what are the values/can be the possible values of argc and argv. (that I should pass) please enlighten me!!! thanks in advance. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-are-the-types-of-the-parameters-that-can-be-passed-to-gst-init-tp3164362p3164362.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Sun Dec 26 19:18:12 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sun, 26 Dec 2010 18:18:12 +0000 Subject: [gst-devel] What are the types of the parameters that can be passed to gst_init ? In-Reply-To: <1293387001551-3164362.post@n4.nabble.com> References: <1293387001551-3164362.post@n4.nabble.com> Message-ID: <1293387492.462.2.camel@zingle> On Sun, 2010-12-26 at 10:10 -0800, Anirudh wrote: > I am a newbie in the field of Gstreamer and I have just started reading the > official manual for application developers and its damn! interesting. > > I encountered that gst_init() function is used to initialize the Gstreamer. > (fine) > but I do not know what kind of parameters can be passed to it. > > They have given an example where they use it like : gst_init(&argc,&argv) > but its not given what are the values/can be the possible values of argc and > argv. (that I should pass) The options from 'gst-inspect-0.10 --help-gst' basically, ie.: GStreamer Options --gst-version Print the GStreamer version --gst-fatal-warnings Make all warnings fatal --gst-debug-help Print available debug categories and exit --gst-debug-level=LEVEL Default debug level from 1 (only error) to 5 (anything) or 0 for no output --gst-debug=LIST Comma-separated list of category_name:level pairs to set specific levels for the individual categories. Example: GST_AUTOPLUG:5,GST_ELEMENT_*:3 --gst-debug-no-color Disable colored debugging output --gst-debug-disable Disable debugging --gst-plugin-spew Enable verbose plugin loading diagnostics --gst-plugin-path=PATHS Colon-separated paths containing plugins --gst-plugin-load=PLUGINS Comma-separated list of plugins to preload in addition to the list stored in environment variable GST_PLUGIN_PATH --gst-disable-segtrap Disable trapping of segmentation faults during plugin loading --gst-disable-registry-update Disable updating the registry --gst-disable-registry-fork Disable spawning a helper process while scanning the registry However, the idea is basically that you just pass in the command line arguments a passed by the user. If you want to enable/disable any of these things from your program, you should just use the corresponding GStreamer API to enable/disable it (where available). Cheers -Tim From anirudhtomer at gmail.com Sun Dec 26 19:37:52 2010 From: anirudhtomer at gmail.com (Anirudh) Date: Sun, 26 Dec 2010 10:37:52 -0800 (PST) Subject: [gst-devel] What are the types of the parameters that can be passed to gst_init ? In-Reply-To: <1293387492.462.2.camel@zingle> References: <1293387001551-3164362.post@n4.nabble.com> <1293387492.462.2.camel@zingle> Message-ID: <1293388672874-3164378.post@n4.nabble.com> thanks for the reply :) okay so u mean that I can pass the parameters via command line and later use them as parameters for API's in my program. yeah! but I wanted to know what should I pass...Gstreamer manual says that u can pass NULL as a parameter as well. So what are consequences of passing NULL. gst_init(NULL,NULL); Suppose I don't want to pass the parameters from command line then it would be something like gst_init(param1,param2); then what should be param1 and param2...can anyone give an example for clarifying this doubt. something like ./a.out xyz pqr yeah! some value...but what should be the XYZ or PQR & what will be its consequences after gst_init(&argc,&argv) uses them to initialize Gstreamer -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-are-the-types-of-the-parameters-that-can-be-passed-to-gst-init-tp3164362p3164378.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Sun Dec 26 20:02:37 2010 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sun, 26 Dec 2010 19:02:37 +0000 Subject: [gst-devel] What are the types of the parameters that can be passed to gst_init ? In-Reply-To: <1293388672874-3164378.post@n4.nabble.com> References: <1293387001551-3164362.post@n4.nabble.com> <1293387492.462.2.camel@zingle> <1293388672874-3164378.post@n4.nabble.com> Message-ID: <1293390157.462.7.camel@zingle> On Sun, 2010-12-26 at 10:37 -0800, Anirudh wrote: > but I wanted to know what should I pass...Gstreamer manual says that u > can pass NULL as a parameter as well. So what are consequences of > passing NULL. gst_init(NULL,NULL); What do you think the consequences of that are? Simply that you can't set extra options via command line arguments. > Suppose I don't want to pass the parameters from command line then it would > be something like > gst_init(param1,param2); > > then what should be param1 and param2...can anyone give an example for > clarifying this doubt. > > something like ./a.out xyz pqr > yeah! some value...but what should be the XYZ or PQR & what will be its > consequences after gst_init(&argc,&argv) uses them to initialize Gstreamer There's nothing particularly GStreamer-specific here, it's just an extension of how main() works in C, or how gtk_init() works. The reason the addresses of argv/argc are passed is so that arguments that have been handled by GStreamer can be removed. It's quite easy to play around with this a bit in code to see how it works. Cheers -Tim From braunsquared at gmail.com Sun Dec 26 20:19:35 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Sun, 26 Dec 2010 14:19:35 -0500 Subject: [gst-devel] What are the types of the parameters that can be passed to gst_init ? In-Reply-To: <1293390157.462.7.camel@zingle> References: <1293387001551-3164362.post@n4.nabble.com> <1293387492.462.2.camel@zingle> <1293388672874-3164378.post@n4.nabble.com> <1293390157.462.7.camel@zingle> Message-ID: I would suggest looking at the application developers guide on the gstreamer website. It covers all of this with code examples. You don't have to be concerned with overlapping arguments since all of the gstreamer library args begin with '--gst-' and I don't believe they have a short form. Best, Tim Sent from my iPhone On Dec 26, 2010, at 2:02 PM, Tim-Philipp M?ller wrote: > On Sun, 2010-12-26 at 10:37 -0800, Anirudh wrote: > >> but I wanted to know what should I pass...Gstreamer manual says that u >> can pass NULL as a parameter as well. So what are consequences of >> passing NULL. gst_init(NULL,NULL); > > What do you think the consequences of that are? > > Simply that you can't set extra options via command line arguments. > > >> Suppose I don't want to pass the parameters from command line then it would >> be something like >> gst_init(param1,param2); >> >> then what should be param1 and param2...can anyone give an example for >> clarifying this doubt. >> >> something like ./a.out xyz pqr >> yeah! some value...but what should be the XYZ or PQR & what will be its >> consequences after gst_init(&argc,&argv) uses them to initialize Gstreamer > > There's nothing particularly GStreamer-specific here, it's just an > extension of how main() works in C, or how gtk_init() works. The reason > the addresses of argv/argc are passed is so that arguments that have > been handled by GStreamer can be removed. > > It's quite easy to play around with this a bit in code to see how it > works. > > Cheers > -Tim > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From anirudhtomer at gmail.com Sun Dec 26 20:36:55 2010 From: anirudhtomer at gmail.com (Anirudh) Date: Sun, 26 Dec 2010 11:36:55 -0800 (PST) Subject: [gst-devel] What are the types of the parameters that can be passed to gst_init ? In-Reply-To: References: <1293387001551-3164362.post@n4.nabble.com> <1293387492.462.2.camel@zingle> <1293388672874-3164378.post@n4.nabble.com> <1293390157.462.7.camel@zingle> Message-ID: <1293392215041-3164419.post@n4.nabble.com> thanks to Tim Muller and Timothy Braun. after digging it a bit more I got the meaning of your replies. looking back it feels like what I dumb doubt it was...( as I said I am a novice ) doesn't matter its the first time for me & I still have a long long way to go. thanks guys. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-are-the-types-of-the-parameters-that-can-be-passed-to-gst-init-tp3164362p3164419.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From umakantgoyal1 at gmail.com Mon Dec 27 07:21:11 2010 From: umakantgoyal1 at gmail.com (Umakant Goyal) Date: Mon, 27 Dec 2010 11:51:11 +0530 Subject: [gst-devel] Pipeline Crashes when it recieves comfort noise/dtmf packets Message-ID: Hi All, I am making pipeline using gstrtpbin, rtppcmudepay, udpsrc and fakesink elements to receive RTP audio stream from network. Everything goes fine when i receive audio(pcmu) rtp packets with payload type 0 and encdoing-name PCMU. But my pipeline breaks when i receive CN (comfort noise) rtp packets with payload type 13 and dtmf packets with payload type 101. I have tried lot to ignore these packets but i did not succeed to find out the solution of my problem. Please help me by telling the possible solutions of my problem. I am also wondering why udpsrc element is receiving rtp messages with payload 101 and 13 because i have set the caps for udpsrc as given below: media --> audio, payload --> 0, clock-rate --> 8000 encoding-name --> PCMU, As per my understanding if i would set the above given caps to udpsrc then it should ignore other rtp packets except with payload 0 and encoding-name PCMU. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From gibrovacco at gmail.com Mon Dec 27 08:46:28 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Mon, 27 Dec 2010 09:46:28 +0200 Subject: [gst-devel] Video and audio speed issue? In-Reply-To: References: Message-ID: Hi, On Sat, Dec 25, 2010 at 1:21 AM, Radivoje Jovanovic wrote: > Hello, > If I try playing just a sound from my video with following pipeline: > gst-launch-0.10 filesrc location=MPEG4_MOVIE ! qtdemux name=t !? queue ! > faad ! alsasink > sound plays just fine and song duration is 38 seconds. > If I play video only using following pipeline: > gst-launch-0.10 filesrc location=MPEG4_MOVIE ! qtdemux name=t ! queue? ! > mpeg4dec ! mySink > video plays just fine in duration of 38 seconds. > However, if I play both of them like: > gst-launch-0.10 filesrc location=MPEG4_MOVIE ! qtdemux name=t ! queue? ! > mpeg4dec ! mySink t. ! queue ! faad !? alsasink > everything is much faster and the whole thing is over in 26 seconds. > Any idea what might be causing this? there may be several reasons for such a behavior, ranging from an incorrect clip to wrongly behaving elements in the pipeline. Some suggestions: - replace your sink elements with fakesinks and enable gst-launch verbosity by using its -v option. - add some identity elements after the tee and set its check-perfect proerty to "true". Maybe you can post something here (use pastebin/txt attachments for more than 10 lines please) after running those tests. Regards > Cheers > Ogi > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From gibrovacco at gmail.com Mon Dec 27 09:16:00 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Mon, 27 Dec 2010 10:16:00 +0200 Subject: [gst-devel] Pipeline Crashes when it recieves comfort noise/dtmf packets In-Reply-To: References: Message-ID: Hi, it's not too clear how you're trying to get your data on the receiving pipeline. Can you post the structure of the pipeline you're using? Some more comments below. On Mon, Dec 27, 2010 at 8:21 AM, Umakant Goyal wrote: > Hi All, > > I am making pipeline using gstrtpbin, rtppcmudepay, udpsrc and fakesink > elements to receive RTP audio stream from network. Everything goes fine when > i receive audio(pcmu) rtp packets with payload type 0 and encdoing-name > PCMU. But my pipeline breaks when i receive CN (comfort noise) rtp packets > with payload type 13 and dtmf packets with payload type 101. I have tried > lot to ignore these packets but i did not succeed to find out the solution > of my problem. Please help me by telling the possible solutions of my > problem. can you collect a backtrace of the crash? It should be fairly easy with the aid of GDB. > > I am also wondering why udpsrc element is receiving rtp messages with > payload 101 and 13 because i have set the caps for udpsrc as given below: > > media --> audio, > payload --> 0, > clock-rate --> 8000 > encoding-name --> PCMU, do you mean that the source pad in the downstream element (which I suppose being a rtppcmudepay) is receiving buffer with those payloads? How did you detect such a behaviour? As a further note, if your sending side is using the same port for PCMU and DTMF there may something wrong in its configuration. > > As per my understanding if i would set the above given caps to udpsrc then > it should ignore other rtp packets except with payload 0 and encoding-name > PCMU. > > Thanks > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From umakantgoyal1 at gmail.com Mon Dec 27 10:47:20 2010 From: umakantgoyal1 at gmail.com (Umakant Goyal) Date: Mon, 27 Dec 2010 15:17:20 +0530 Subject: [gst-devel] Pipeline Crashes when it recieves comfort noise/dtmf packets In-Reply-To: References: Message-ID: Hi, Thanks for your response My pipeline is as below: fakesink <---- rtppcmudepay <---- gstrtpbin <--- udpsrc do you mean that the source pad in the downstream element (which I suppose being a rtppcmudepay) is receiving buffer with those payloads? How did you detect such a behaviour? ---->Logs shows that gstrtbin is receiving rtp packets with different payload and looking for element for linking. As a further note, if your sending side is using the same port for PCMU and DTMF there may something wrong in its configuration. ---->No, This behaviour is right as per standard. Sender can use same port for RTP Audio (PCMU) and DTMF (RFC 2833)/ Thanks On Mon, Dec 27, 2010 at 1:46 PM, Marco Ballesio wrote: > Hi, > > it's not too clear how you're trying to get your data on the receiving > pipeline. Can you post the structure of the pipeline you're using? > Some more comments below. > > On Mon, Dec 27, 2010 at 8:21 AM, Umakant Goyal > wrote: > > Hi All, > > > > I am making pipeline using gstrtpbin, rtppcmudepay, udpsrc and fakesink > > elements to receive RTP audio stream from network. Everything goes fine > when > > i receive audio(pcmu) rtp packets with payload type 0 and encdoing-name > > PCMU. But my pipeline breaks when i receive CN (comfort noise) rtp > packets > > with payload type 13 and dtmf packets with payload type 101. I have tried > > lot to ignore these packets but i did not succeed to find out the > solution > > of my problem. Please help me by telling the possible solutions of my > > problem. > > can you collect a backtrace of the crash? It should be fairly easy > with the aid of GDB. > > > > > I am also wondering why udpsrc element is receiving rtp messages with > > payload 101 and 13 because i have set the caps for udpsrc as given below: > > > > media --> audio, > > payload --> 0, > > clock-rate --> 8000 > > encoding-name --> PCMU, > > do you mean that the source pad in the downstream element (which I > suppose being a rtppcmudepay) is receiving buffer with those payloads? > How did you detect such a behaviour? > > As a further note, if your sending side is using the same port for > PCMU and DTMF there may something wrong in its configuration. > > > > > As per my understanding if i would set the above given caps to udpsrc > then > > it should ignore other rtp packets except with payload 0 and > encoding-name > > PCMU. > > > > Thanks > > > ------------------------------------------------------------------------------ > > Learn how Oracle Real Application Clusters (RAC) One Node allows > customers > > to consolidate database storage, standardize their database environment, > > and, > > should the need arise, upgrade to a full multi-node Oracle RAC database > > without downtime or disruption > > http://p.sf.net/sfu/oracle-sfdevnl > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From anirudhtomer at gmail.com Mon Dec 27 11:52:26 2010 From: anirudhtomer at gmail.com (Anirudh) Date: Mon, 27 Dec 2010 02:52:26 -0800 (PST) Subject: [gst-devel] What is the purpose of gst audioconvert plugin Message-ID: <1293447146580-3164871.post@n4.nabble.com> So I was going thru the gst manual and I found that in an example for playing an ogg file they were using an audio converter after decoding the ogg format into raw format. the pipeline they were using is as follows: file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output so my question is what is the purpose of audioconverter when I have already decoded it. I mean!!! to which format am I converting it. So if I have decoded it into raw format, I think I can directly play it. please enlighten me!!! THanks in advance -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-is-the-purpose-of-gst-audioconvert-plugin-tp3164871p3164871.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From smcnam at gmail.com Mon Dec 27 12:22:38 2010 From: smcnam at gmail.com (Sean McNamara) Date: Mon, 27 Dec 2010 06:22:38 -0500 Subject: [gst-devel] What is the purpose of gst audioconvert plugin In-Reply-To: <1293447146580-3164871.post@n4.nabble.com> References: <1293447146580-3164871.post@n4.nabble.com> Message-ID: Hi, On Mon, Dec 27, 2010 at 5:52 AM, Anirudh wrote: > > So I was going thru the gst manual and I found that in an example for playing > an ogg file they were using an audio converter after decoding the ogg format > into raw format. > > the pipeline they were using is as follows: > > file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output > > so my question is what is the purpose of audioconverter when I have already > decoded it. > I mean!!! to which format am I converting it. The audioconvert element's purpose is to act both as a negotiator and a sort of policeman. The negotiator part is this: the audioconvert element is extremely flexible as to the audio formats it will accept on either "end" of the element (the source pad and the sink pad). So it helps two plugins potentially expecting an incompatible format to "talk" to one another seamlessly. The policeman part is this: the audioconvert element *makes sure* that the audio flowing from its sink pad to its source pad is going to be converted to the format that the source pad's client expects. It has fairly good error checking if it is unable to do that for some reason, and it is fairly good at performing most conversions without any loss of data (except where data loss is inevitable, such as stereo to mono). Unfortunately, the audioconvert element does not (that I am aware) perform sample rate conversions. For that, you need the `audioresample' element. Its goals and function are vaguely similar to audioconvert, except that, rather than converting sample format parameters such as encoding, bit depth, and number of channels, instead it converts between sample rates. You can think of the element chain `audioconvert ! audioresample' as being a single unit that performs full-scope sample format negotiation and conversion for all of the standard parameters of a PCM sample. The reason why audioconvert is necessary in that case is that the alsasink element (as well as many other elements) are not able to unconditionally accept all sample formats. Here's a simple example. Let's say you have a PCI sound card that accepts sample formats that are 8 or 16-bits wide, signed little endian, stereo or up to 5.1 channels, and sample rates from 8000 Hz to 48000 Hz. If your alsasink element does not specify an ALSA device string that goes through the ALSA `plug' PCM plugin, then ALSA is incapable of supporting sample formats that your soundcard does not natively support in the hardware. So if you pass a 192 KHz sample of 32-bit floats in 8.1 channel to the above stated soundcard, you will get an error in the alsasink element. Not to overgeneralize, but the basic point is that not all "uncompressed" or "decoded" audio is created equal, and you can not take for granted that any two GStreamer elements supporting uncompressed PCM audio will necessarily be able to natively support all possible sample formats. The solution is to use extra CPU power to do the needed conversions with the audioconvert and audioresample elements. HTH, Sean > So if I have decoded it into raw format, I think I can directly play it. > > please enlighten me!!! THanks in advance > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-is-the-purpose-of-gst-audioconvert-plugin-tp3164871p3164871.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From anirudhtomer at gmail.com Mon Dec 27 12:40:59 2010 From: anirudhtomer at gmail.com (Anirudh) Date: Mon, 27 Dec 2010 03:40:59 -0800 (PST) Subject: [gst-devel] What is the purpose of gst audioconvert plugin In-Reply-To: References: <1293447146580-3164871.post@n4.nabble.com> Message-ID: <1293450059351-3164890.post@n4.nabble.com> thanks that was really a nice article. I got the use of audioconvert. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-is-the-purpose-of-gst-audioconvert-plugin-tp3164871p3164890.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From weian.chen at intel.com Mon Dec 27 13:15:43 2010 From: weian.chen at intel.com (Chen, Weian) Date: Mon, 27 Dec 2010 20:15:43 +0800 Subject: [gst-devel] Can we use gst-launch to do video capture using camerabin? In-Reply-To: <1293450059351-3164890.post@n4.nabble.com> References: <1293447146580-3164871.post@n4.nabble.com> <1293450059351-3164890.post@n4.nabble.com> Message-ID: Hi there, Can we use command line (gst-launch) to do video capture using camerabin? I know we can do view-finder very easily. Thanks, Weian From thiagossantos at gmail.com Mon Dec 27 13:21:54 2010 From: thiagossantos at gmail.com (Thg.) Date: Mon, 27 Dec 2010 09:21:54 -0300 Subject: [gst-devel] Can we use gst-launch to do video capture using camerabin? In-Reply-To: References: <1293447146580-3164871.post@n4.nabble.com> <1293450059351-3164890.post@n4.nabble.com> Message-ID: On Mon, Dec 27, 2010 at 9:15 AM, Chen, Weian wrote: > Hi there, > > Can we use command line (gst-launch) to do video capture using camerabin? I > know we can do view-finder very easily. > You need to use capture-start signal to do video capture with camerabin, so you can't do that from gst-launch. There is, however, an example application on -bad/tests/examples/camerabin that allows you to use it and do video/image capture from command line. > > Thanks, > Weian > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Thiago Sousa Santos -------------- next part -------------- An HTML attachment was scrubbed... URL: From anirudhtomer at gmail.com Mon Dec 27 14:40:13 2010 From: anirudhtomer at gmail.com (Anirudh) Date: Mon, 27 Dec 2010 05:40:13 -0800 (PST) Subject: [gst-devel] Need Advice!!! Message-ID: <1293457213712-3164997.post@n4.nabble.com> Hi everyone, ( I am not asking anyone to make the project for me but I am asking to show me the right path ) CONTEXT: I am a bachelor's student and I am working on a project where I have to transcode a multimedia stream coming from a server on my machine, as well as stream it again on the fly to another machine. Server ----------------> My machine --------------------------------> client machine actual stream (transcode on the fly) stream transcoded I can't store the incoming stream on my machine because the client is waiting for it so I have to transcode it on the fly.(live streaming) For SIMPLICITY I am assuming the i/p stream is in ".mp3" format (only audio) and output stream is "vorbis". I am supposed to write the transcoder on my own & not use existing ones. WHAT I DONE TILL NOW: Its the first time I have heard of GStreamer and I am going through the manuals for application developer and understanding the basics and the manuals are really good. I have gone through almost 50 pages. MY PROBLEM: I want to know how should I proceed with it. So my source would be URL (coz as I said my i/p would be coming from a streaming server) and sink would be a URL again (if u think I am wrong please correct me here). How should I move on. right now I'll be trying to play a stream from localhost using a small code and see how it works and then I'll jump onto making some complex pipeline structure to solve my problem. Thanks for reading my post!!!. I'll be really thankful if someone can give an advice to me on what approach should I follow. What should be my Learning Flow ? thanks in advance for the help. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Need-Advice-tp3164997p3164997.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From weian.chen at intel.com Mon Dec 27 14:40:48 2010 From: weian.chen at intel.com (Chen, Weian) Date: Mon, 27 Dec 2010 21:40:48 +0800 Subject: [gst-devel] Can we use gst-launch to do video capture using camerabin? In-Reply-To: References: <1293447146580-3164871.post@n4.nabble.com> <1293450059351-3164890.post@n4.nabble.com> Message-ID: Thanks for the information. Weian From: Thg. [mailto:thiagossantos at gmail.com] Sent: Monday, December 27, 2010 8:22 PM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] Can we use gst-launch to do video capture using camerabin? On Mon, Dec 27, 2010 at 9:15 AM, Chen, Weian > wrote: Hi there, Can we use command line (gst-launch) to do video capture using camerabin? I know we can do view-finder very easily. You need to use capture-start signal to do video capture with camerabin, so you can't do that from gst-launch. There is, however, an example application on -bad/tests/examples/camerabin that allows you to use it and do video/image capture from command line. Thanks, Weian ------------------------------------------------------------------------------ Learn how Oracle Real Application Clusters (RAC) One Node allows customers to consolidate database storage, standardize their database environment, and, should the need arise, upgrade to a full multi-node Oracle RAC database without downtime or disruption http://p.sf.net/sfu/oracle-sfdevnl _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Thiago Sousa Santos -------------- next part -------------- An HTML attachment was scrubbed... URL: From anirudhtomer at gmail.com Mon Dec 27 17:23:13 2010 From: anirudhtomer at gmail.com (Anirudh) Date: Mon, 27 Dec 2010 08:23:13 -0800 (PST) Subject: [gst-devel] How to play a file lying on localhost using gst-launch ? Message-ID: <1293466993918-3165195.post@n4.nabble.com> I am having a file on my localhost server at the address: http://localhost/htdocs/test.ogg I am trying to play it using tcpclientsrc (test.ogg has just audio so I need not to worry about the video part) anirudh at anirudh-laptop:~$ gst-launch-0.10 tcpclientsrc protocol=none host=localhost port=80 ! oggdemux ! vorbisdec ! audioconvert ! alsasink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... and then it keeps waiting. I know that I am doing a mistake, and I know that without giving the URL of the file I can not play it but I don't know how to give the URL because tcpclientsrc has just 3 properties (protocol,host,port) which I can set and none of them is for setting the URL. So I want to know how to play that file using command line. please help me out on this issue. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-play-a-file-lying-on-localhost-using-gst-launch-tp3165195p3165195.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wim.taymans at gmail.com Mon Dec 27 17:27:55 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 27 Dec 2010 17:27:55 +0100 Subject: [gst-devel] How to play a file lying on localhost using gst-launch ? In-Reply-To: <1293466993918-3165195.post@n4.nabble.com> References: <1293466993918-3165195.post@n4.nabble.com> Message-ID: <1293467275.28802.9.camel@meany> On Mon, 2010-12-27 at 08:23 -0800, Anirudh wrote: > I am having a file on my localhost server at the address: > http://localhost/htdocs/test.ogg > try gst-launch-0.10 playbin2 uri=http://localhost/htdocs/test.ogg Wim > I am trying to play it using tcpclientsrc (test.ogg has just audio so I need > not to worry about the video part) > > anirudh at anirudh-laptop:~$ gst-launch-0.10 tcpclientsrc protocol=none > host=localhost port=80 ! oggdemux ! vorbisdec ! audioconvert ! alsasink > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > > and then it keeps waiting. I know that I am doing a mistake, and I know that > without giving the URL of the file I can not play it but I don't know how to > give the URL because tcpclientsrc has just 3 properties (protocol,host,port) > which I can set and none of them is for setting the URL. > > So I want to know how to play that file using command line. > please help me out on this issue. > > From anirudhtomer at gmail.com Mon Dec 27 17:43:25 2010 From: anirudhtomer at gmail.com (Anirudh) Date: Mon, 27 Dec 2010 08:43:25 -0800 (PST) Subject: [gst-devel] How to play a file lying on localhost using gst-launch ? In-Reply-To: <1293467275.28802.9.camel@meany> References: <1293466993918-3165195.post@n4.nabble.com> <1293467275.28802.9.camel@meany> Message-ID: <1293468205232-3165225.post@n4.nabble.com> wow! a reply from Wim Taymans...thanks for the reply. well!!! I tried to run that command u gave & it played the file but I didn't get how it worked internally. I was trying to run it by making a pipeline structure like I mentioned in my question. Since I am a newbie in field of GStreamer if I manage to run the file using the way I was trying to do earlier (this way) anirudh at anirudh-laptop:~$ gst-launch-0.10 tcpclientsrc protocol=none host=localhost port=80 ! oggdemux ! vorbisdec ! audioconvert ! alsasink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... then I will probably understand a bit more.I have been thru around 60 pages of the manual & I am able to understand the basics of Gstreamer and this pipeline structure is right now more clearer in my mind. So can u help me out on that ? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-play-a-file-lying-on-localhost-using-gst-launch-tp3165195p3165225.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From vaibhav.d.ingale at gmail.com Mon Dec 27 19:04:53 2010 From: vaibhav.d.ingale at gmail.com (Vaibhav Ingale) Date: Mon, 27 Dec 2010 23:34:53 +0530 Subject: [gst-devel] How to play a file lying on localhost using gst-launch ? In-Reply-To: <1293468205232-3165225.post@n4.nabble.com> References: <1293466993918-3165195.post@n4.nabble.com> <1293467275.28802.9.camel@meany> <1293468205232-3165225.post@n4.nabble.com> Message-ID: you may be interested in this: ( for video ) $ gst-launch-0.10 souphttpsrc location= http://localhost/htdocs/test.ogg ! oggdemux ! theoradec ! xvimagesink -v Kind Regards, Vaibhav Ingale | +91-9226938504 vaibhav.d.ingale at gmail.com http://vaibhav-ingale.blogspot.com/ On Mon, Dec 27, 2010 at 10:13 PM, Anirudh wrote: > > wow! a reply from Wim Taymans...thanks for the reply. > > well!!! I tried to run that command u gave & it played the file but I > didn't > get how it worked internally. I was trying to run it by making a pipeline > structure like I mentioned in my question. Since I am a newbie in field of > GStreamer if I manage to run the file using the way I was trying to do > earlier > (this way) > anirudh at anirudh-laptop:~$ gst-launch-0.10 tcpclientsrc protocol=none > host=localhost port=80 ! oggdemux ! vorbisdec ! audioconvert ! alsasink > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > > > then I will probably understand a bit more.I have been thru around 60 pages > of the manual & I am able to understand the basics of Gstreamer and this > pipeline structure is right now more clearer in my mind. So can u help me > out on that ? > > > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/How-to-play-a-file-lying-on-localhost-using-gst-launch-tp3165195p3165225.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From member at linkedin.com Tue Dec 28 04:07:23 2010 From: member at linkedin.com (Vipin Tiwari via LinkedIn) Date: Tue, 28 Dec 2010 03:07:23 +0000 (UTC) Subject: [gst-devel] Invitation to connect on LinkedIn Message-ID: <841907897.20919907.1293505643280.JavaMail.app@ela4-bed35.prod> LinkedIn ------------Vipin Tiwari requested to add you as a connection on LinkedIn: ------------------------------------------ Terry, I'd like to add you to my professional network on LinkedIn. - Vipin Accept invitation from Vipin Tiwari http://www.linkedin.com/e/8uvs3p-gi87v219-3o/I6odceFIR0nwoItulatqxeuuuYZHsrOdOOozdxXIoOow9a7uxmbiOMp/blk/I2523465960_2/1BpC5vrmRLoRZcjkkZt5YCpnlOt3RApnhMpmdzgmhxrSNBszYOnP0SejkSd3cOdj99bRh5cSBypPtCbPwQc3AQdPgSc38LrCBxbOYWrSlI/EML_comm_afe/ View invitation from Vipin Tiwari http://www.linkedin.com/e/8uvs3p-gi87v219-3o/I6odceFIR0nwoItulatqxeuuuYZHsrOdOOozdxXIoOow9a7uxmbiOMp/blk/I2523465960_2/39vc3oVdjoQcP8RcAALqnpPbOYWrSlI/svi/ ------------------------------------------ DID YOU KNOW your LinkedIn profile helps you control your public image when people search for you? Setting your profile as public means your LinkedIn profile will come up when people enter your name in leading search engines. Take control of your image! http://www.linkedin.com/e/8uvs3p-gi87v219-3o/ewp/inv-22/ -- (c) 2010, LinkedIn Corporation -------------- next part -------------- An HTML attachment was scrubbed... URL: From anirudhtomer at gmail.com Tue Dec 28 16:13:56 2010 From: anirudhtomer at gmail.com (Anirudh) Date: Tue, 28 Dec 2010 07:13:56 -0800 (PST) Subject: [gst-devel] How to play a file lying on localhost using gst-launch ? In-Reply-To: References: <1293466993918-3165195.post@n4.nabble.com> <1293467275.28802.9.camel@meany> <1293468205232-3165225.post@n4.nabble.com> Message-ID: <1293549236263-3166076.post@n4.nabble.com> as I said I had an ogg file with audio stream only I can play it using this command line anirudh at anirudh-laptop:~ $gst-launch-0.10 souphttpsrc location=http://localhost/htdocs/test.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-play-a-file-lying-on-localhost-using-gst-launch-tp3165195p3166076.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wl2776 at gmail.com Tue Dec 28 17:00:18 2010 From: wl2776 at gmail.com (wl2776) Date: Tue, 28 Dec 2010 08:00:18 -0800 (PST) Subject: [gst-devel] How to play a file lying on localhost using gst-launch ? In-Reply-To: <1293468205232-3165225.post@n4.nabble.com> References: <1293466993918-3165195.post@n4.nabble.com> <1293467275.28802.9.camel@meany> <1293468205232-3165225.post@n4.nabble.com> Message-ID: <1293552018432-3166116.post@n4.nabble.com> playbin2 discovers that you give it an HTTP url. Then it searches through its element registry and discovers what source elements can handle http protocol, finds one, creates its instance, sets its location property to the uri value, Then it lets the source element to connect to the HTTP server, send GET request, receive response, output received data and so on. Anirudh wrote: > > well!!! I tried to run that command u gave & it played the file but I > didn't get how it worked internally. I was trying to run it by making a > pipeline structure like I mentioned in my question. Since I am a newbie in > field of GStreamer if I manage to run the file using the way I was trying > to do earlier > (this way) > anirudh at anirudh-laptop:~$ gst-launch-0.10 tcpclientsrc protocol=none > host=localhost port=80 ! oggdemux ! vorbisdec ! audioconvert ! alsasink > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > > > then I will probably understand a bit more.I have been thru around 60 > pages of the manual & I am able to understand the basics of Gstreamer and > this pipeline structure is right now more clearer in my mind. So can u > help me out on that ? > tcpclientsrc will not do since it expects raw data to come from the TCP socket, but the HTTP server won't send them out without HTTP request. You should also learn about the OSI model and where TCP and HTTP protocols are in it. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-play-a-file-lying-on-localhost-using-gst-launch-tp3165195p3166116.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wl2776 at gmail.com Tue Dec 28 17:27:59 2010 From: wl2776 at gmail.com (wl2776) Date: Tue, 28 Dec 2010 08:27:59 -0800 (PST) Subject: [gst-devel] How to play a file lying on localhost using gst-launch ? In-Reply-To: <1293552018432-3166116.post@n4.nabble.com> References: <1293466993918-3165195.post@n4.nabble.com> <1293467275.28802.9.camel@meany> <1293468205232-3165225.post@n4.nabble.com> <1293552018432-3166116.post@n4.nabble.com> Message-ID: <1293553679771-3166146.post@n4.nabble.com> wl2776 wrote: > > playbin2 discovers that you give it an HTTP url. > Then it searches through its element registry and discovers what source > elements can handle http protocol, finds one, creates its instance, sets > its location property to the uri value, > Well, to be more specific, it creates the uridecodebin element, gives that uri to it, it creates the source element and the decodebin element, which builds the rest of the decoding chain, basing on data, received from the source element. > Then it lets the source element to connect to the HTTP server, send GET > request, receive response, output received data and so on. > > > Anirudh wrote: >> >> well!!! I tried to run that command u gave & it played the file but I >> didn't get how it worked internally. I was trying to run it by making a >> pipeline structure like I mentioned in my question. Since I am a newbie >> in field of GStreamer if I manage to run the file using the way I was >> trying to do earlier >> (this way) >> anirudh at anirudh-laptop:~$ gst-launch-0.10 tcpclientsrc protocol=none >> host=localhost port=80 ! oggdemux ! vorbisdec ! audioconvert ! alsasink >> Setting pipeline to PAUSED ... >> Pipeline is PREROLLING ... >> >> >> then I will probably understand a bit more.I have been thru around 60 >> pages of the manual & I am able to understand the basics of Gstreamer and >> this pipeline structure is right now more clearer in my mind. So can u >> help me out on that ? >> > > tcpclientsrc will not do since it expects raw data to come from the TCP > socket, but the HTTP server won't send them out without HTTP request. > > You should also learn about the OSI model and where TCP and HTTP protocols > are in it. > -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-play-a-file-lying-on-localhost-using-gst-launch-tp3165195p3166146.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From radivojejovanovic at gmail.com Tue Dec 28 19:27:19 2010 From: radivojejovanovic at gmail.com (Radivoje Jovanovic) Date: Tue, 28 Dec 2010 10:27:19 -0800 Subject: [gst-devel] Alsasink problem? Message-ID: Hi, I have ALSA driver for my card and it works fine when I try playing sound through aplay or watch movies through gstreamer, but if I try to do: gst-launch-0.10 filesrc location=FILE_NAME.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink it comes back: Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstAlsaSink:alsasink0: Could not get/set settings from/on resource. Additional debug info: gstalsasink.c(519): set_hwparams (): /GstPipeline:pipeline0/GstAlsaSink:alsasink0: Unable to set hw params for playback: Invalid argument ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... So it comes back that alsa cannot setup correctly even though alsasink works correctly when I play MPEG movies. any thoughts? Cheers Ogi -------------- next part -------------- An HTML attachment was scrubbed... URL: From umakantgoyal1 at gmail.com Tue Dec 28 21:02:04 2010 From: umakantgoyal1 at gmail.com (Umakant Goyal) Date: Wed, 29 Dec 2010 01:32:04 +0530 Subject: [gst-devel] EOS for Element Not for Entire Pipeline Message-ID: Hi All, Can i receive EOS message from any specific element rather than from pipeline? What i know app receives EOS receives message when there is no activity in pipeline but i want to receive this event when element of my pipeline finishes its task but other elements of pipeline are still doing work. Is it possible? e.g if my pipeline is receiving and sending data to network, when my sending elements finishes its task then i want EOS stream for sending side elements even though my receiving elements are still receiving data from network. Thanks in Advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From michael at panicking.kicks-ass.org Tue Dec 28 21:10:07 2010 From: michael at panicking.kicks-ass.org (Michael Trimarchi) Date: Tue, 28 Dec 2010 21:10:07 +0100 Subject: [gst-devel] Alsasink problem? In-Reply-To: References: Message-ID: <4D1A441F.1000801@panicking.kicks-ass.org> Hi On 12/28/2010 07:27 PM, Radivoje Jovanovic wrote: > Hi, > I have ALSA driver for my card and it works fine when I try playing sound > through aplay or watch movies through gstreamer, but if I try to do: > gst-launch-0.10 filesrc location=FILE_NAME.ogg ! oggdemux ! vorbisdec ! > audioconvert ! audioresample ! alsasink > it comes back: > Pipeline is PREROLLING ... > ERROR: from element /GstPipeline:pipeline0/GstAlsaSink:alsasink0: Could not > get/set settings from/on resource. > Additional debug info: > gstalsasink.c(519): set_hwparams (): > /GstPipeline:pipeline0/GstAlsaSink:alsasink0: > Unable to set hw params for playback: Invalid argument > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... is this an embedded system. Can you try to use alsamixer, aplay and arecord to test if your driver is working correctly? Michael Trimarchi > > So it comes back that alsa cannot setup correctly even though alsasink works > correctly when I play MPEG movies. > any thoughts? > Cheers > Ogi > > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > > > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From radivojejovanovic at gmail.com Tue Dec 28 22:30:59 2010 From: radivojejovanovic at gmail.com (Radivoje Jovanovic) Date: Tue, 28 Dec 2010 13:30:59 -0800 Subject: [gst-devel] Alsasink problem? In-Reply-To: <4D1A441F.1000801@panicking.kicks-ass.org> References: <4D1A441F.1000801@panicking.kicks-ass.org> Message-ID: Yes this is an embedded system. As I stated in my previous post aplay, alsamixer and playing MPEG4 movies through gstreamer works just fine. Problem occurs when I try playing ogg files through the pipeline specified. Cheers Ogi On Tue, Dec 28, 2010 at 12:10 PM, Michael Trimarchi < michael at panicking.kicks-ass.org> wrote: > Hi > > On 12/28/2010 07:27 PM, Radivoje Jovanovic wrote: > > Hi, > > I have ALSA driver for my card and it works fine when I try playing sound > > through aplay or watch movies through gstreamer, but if I try to do: > > gst-launch-0.10 filesrc location=FILE_NAME.ogg ! oggdemux ! vorbisdec ! > > audioconvert ! audioresample ! alsasink > > it comes back: > > Pipeline is PREROLLING ... > > ERROR: from element /GstPipeline:pipeline0/GstAlsaSink:alsasink0: Could > not > > get/set settings from/on resource. > > Additional debug info: > > gstalsasink.c(519): set_hwparams (): > > /GstPipeline:pipeline0/GstAlsaSink:alsasink0: > > Unable to set hw params for playback: Invalid argument > > ERROR: pipeline doesn't want to preroll. > > Setting pipeline to NULL ... > > is this an embedded system. Can you try to use alsamixer, aplay and arecord > to test if your driver > is working correctly? > Michael Trimarchi > > > > > So it comes back that alsa cannot setup correctly even though alsasink > works > > correctly when I play MPEG movies. > > any thoughts? > > Cheers > > Ogi > > > > > > > > > > > ------------------------------------------------------------------------------ > > Learn how Oracle Real Application Clusters (RAC) One Node allows > customers > > to consolidate database storage, standardize their database environment, > and, > > should the need arise, upgrade to a full multi-node Oracle RAC database > > without downtime or disruption > > http://p.sf.net/sfu/oracle-sfdevnl > > > > > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Tue Dec 28 23:15:45 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 28 Dec 2010 23:15:45 +0100 Subject: [gst-devel] EOS for Element Not for Entire Pipeline In-Reply-To: References: Message-ID: <1293574545.28802.14.camel@meany> On Wed, 2010-12-29 at 01:32 +0530, Umakant Goyal wrote: > Hi All, > > Can i receive EOS message from any specific element rather than from > pipeline? > What i know app receives EOS receives message when there is no > activity in pipeline but i want to receive this event when element of > my pipeline finishes its task but other elements of pipeline are still > doing work. Is it possible? e.g if my pipeline is receiving and > sending data to network, when my sending elements finishes its task > then i want EOS stream for sending side elements even though my > receiving elements are still receiving data from network. > You can use the message-forward property on GstBin, see the docs of that property for how to use it. Wim > Thanks in Advance > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From rohitratri at gmail.com Wed Dec 29 04:42:13 2010 From: rohitratri at gmail.com (Rohit Atri) Date: Wed, 29 Dec 2010 09:12:13 +0530 Subject: [gst-devel] Packets received even after pipeline is PAUSE'd Message-ID: Hi, I've a pipeline with the following elements : filesrc -> qtdemux -> appsink. In my use-case, I collect 'N' number of packets, pause the pipeline, seek back to a particular packet('s timestamp), and play again. I've noticed that even after I receive the state-changed message (to PAUSE'd state) for the pipeline, I continue to receive packets and if let alone I receive them till EOS(have tested my pipeline with a 40s video only mpeg4 medium and also a two-hr long movie). I'm sure this is not the expected behavior. Can someone explain why this could be happening? Could it be the flush that is done on seek? Thanks Rohit -------------- next part -------------- An HTML attachment was scrubbed... URL: From pa at ufsoft.org Wed Dec 29 07:59:45 2010 From: pa at ufsoft.org (Pedro Algarvio, aka, s0undt3ch) Date: Tue, 28 Dec 2010 22:59:45 -0800 (PST) Subject: [gst-devel] gstrreamer and optparse In-Reply-To: <4B1DF5DC.9040603@inattendu.org> References: <4B1DF5DC.9040603@inattendu.org> Message-ID: <1293605985515-3166825.post@n4.nabble.com> I'm having the same issue. Pedro Algarvio. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/gstrreamer-and-optparse-tp966512p3166825.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From rui.chen at tieto.com Wed Dec 29 08:18:05 2010 From: rui.chen at tieto.com (chen rui) Date: Wed, 29 Dec 2010 15:18:05 +0800 Subject: [gst-devel] how to let the flag "resend-latest" work in output-selector ? Message-ID: <4D1AE0AD.1030409@tieto.com> Hi all, I modify the example in gst-plugins-bad/tests/icles/output-selector-test.c and create new pieline like this: filesink / videotestsrc -- output-selector \ autovideosink I think if i set "resend-latest" flag to TRUE, when i play my file, it will fluency. But it does not, the same as resend-latest = FALSE. So how to use let the flag work? here is my test base on output-selector-test.c --- test.c 2010-12-29 14:39:27.875179000 +0800 +++ ../gst-plugins-bad/tests/icles/output-selector-test.c 2010-12-23 13:52:09.355683000 +0800 @@ -75,7 +75,7 @@ gint main (gint argc, gchar * argv[]) { - GstElement *pipeline, *src, *toverlay, *osel, *sink1, *sink2, *c1, *c2, *c0, *jpegenc, *avi, *filesink; + GstElement *pipeline, *src, *toverlay, *osel, *sink1, *sink2, *c1, *c2, *c0; GstPad *sinkpad; GstBus *bus; @@ -93,30 +93,28 @@ c2 = gst_element_factory_make ("ffmpegcolorspace", NULL); sink1 = gst_element_factory_make ("autovideosink", "sink1"); sink2 = gst_element_factory_make ("autovideosink", "sink2"); - avi = gst_element_factory_make ("avimux", NULL); - filesink = gst_element_factory_make ("filesink", NULL); - jpegenc = gst_element_factory_make ("jpegenc", NULL); if (!pipeline || !src || !c0 || !toverlay || !osel || !c1 || !c2 || !sink1 || - !sink2 || !avi || !filesink || !jpegenc ) { + !sink2) { g_print ("missing element\n"); return -1; } /* add them to bin */ gst_bin_add_many (GST_BIN (pipeline), src, c0, toverlay, osel, c1, sink1, c2, - jpegenc, avi, filesink, NULL); + sink2, NULL); /* set properties */ g_object_set (G_OBJECT (src), "is-live", TRUE, NULL); g_object_set (G_OBJECT (src), "do-timestamp", TRUE, NULL); g_object_set (G_OBJECT (src), "num-buffers", NUM_VIDEO_BUFFERS, NULL); g_object_set (G_OBJECT (osel), "resend-latest", TRUE, NULL); - g_object_set (G_OBJECT (filesink), "location", "test.avi", NULL); /* handle deferred properties */ g_signal_connect (G_OBJECT (sink1), "element-added", G_CALLBACK (on_bin_element_added), NULL); + g_signal_connect (G_OBJECT (sink2), "element-added", + G_CALLBACK (on_bin_element_added), NULL); /* link src ! timeoverlay ! osel */ if (!gst_element_link_many (src, c0, toverlay, osel, NULL)) { @@ -147,7 +145,7 @@ } gst_object_unref (sinkpad); - if (!gst_element_link_many (c2, jpegenc, avi, filesink, NULL)) { + if (!gst_element_link (c2, sink2)) { g_print ("linking output 2 failed\n"); return -1; } B.R Chen Rui From phil at base-art.net Wed Dec 29 09:31:59 2010 From: phil at base-art.net (Philippe Normand) Date: Wed, 29 Dec 2010 09:31:59 +0100 Subject: [gst-devel] gstrreamer and optparse In-Reply-To: <1293605985515-3166825.post@n4.nabble.com> References: <4B1DF5DC.9040603@inattendu.org> <1293605985515-3166825.post@n4.nabble.com> Message-ID: <1293611519.3028.10.camel@espinkinglis> On Tue, 2010-12-28 at 22:59 -0800, Pedro Algarvio, aka, s0undt3ch wrote: > I'm having the same issue. > > Pedro Algarvio. A workaround would be to import gst after you parsed the options with optparse. Philippe From wmiller at sdr.com Wed Dec 29 14:44:17 2010 From: wmiller at sdr.com (Wes Miller) Date: Wed, 29 Dec 2010 05:44:17 -0800 (PST) Subject: [gst-devel] What is the purpose of gst audioconvert plugin In-Reply-To: References: <1293447146580-3164871.post@n4.nabble.com> Message-ID: <1293630257660-3167118.post@n4.nabble.com> Sean, This is a wonderful explanation; one that need to be moved to some featured place on the wiki or, better, a new doc on the website where all of the elements can get equal treatment. So, Sean, Wim, et. al, what is the method to undertake such a task. Though I know better than to ever volunteer, I volunteer to collect the explanations and turn them into some sort of document (good excuse to learn LaTEX?). I'll begin here with a request to all published element authors to write a similar explanation and perhaps add any extra bits you think useful. One I would consider helpful is an explanation of why your element is -bad or -ugly -- especially if it's a licensing issue. Details about the properties would be helpful too. I still have no clue what half the encoding/compression parms do with mpeg. Another idea, in cases where the element is otherwise LGPL but it uses, say, some licensed video encoding, whether someone licensed to use that technology can use your element as if it were LGPL. I hear a lot of "ask your lawyer" responses coming. Happy New Year, Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-is-the-purpose-of-gst-audioconvert-plugin-tp3164871p3167118.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From suneets at gmail.com Wed Dec 29 16:42:56 2010 From: suneets at gmail.com (Suneet Shah) Date: Wed, 29 Dec 2010 07:42:56 -0800 Subject: [gst-devel] How to stream video from a desktop to an Android device? Message-ID: Hello, My name is Suneet and I'm working on a proof of concept project to get back up to speed with development (I haven't done any serious development in 3 years ;) ). I have some questions about what gstreamer plugins I should look at to accomplish this. My project is this: - Stream a Linux desktop to an Android device as video - Add simple control layer --> A click on the android device should result in a click on the Linux desktop I've done a bunch of research on gstreamer and I think it is perfect for what I want to do. So far, I've figured out how to capture the desktop and record it to a video file (thanks to looking at the Istanbul application). Now I need to figure out: - How to encode the video as h.264 - How to stream the video to an android device I've chosen h.264 because Android has native capabilities for playing H.264 streams, so this should keep the Android part of it fairly simple. I'm planning on using simple socket programming to pass messages between Android and the desktop for the control layer. I'm using Python and pygst to write the Linux server app. My questions are: - Does my approach seem sound and should it work? - What gstreamer plugins should I look at to stream the video? - What plugins should I look at to encode H.264? - Will I need to use another tool, like a streaming media server (Darwin or otherwise) to stream the video, or can I accomplish it completely through a gstreamer pipeline? - Are there any relevant examples I should look at? Thanks in advance for your time. Suneet -------------- next part -------------- An HTML attachment was scrubbed... URL: From rohitratri at gmail.com Wed Dec 29 18:13:33 2010 From: rohitratri at gmail.com (Rohit Atri) Date: Wed, 29 Dec 2010 22:43:33 +0530 Subject: [gst-devel] Packets received even after pipeline is PAUSE'd In-Reply-To: References: Message-ID: Hi Guys, Any comments on the below query? Thanks Rohit On Wed, Dec 29, 2010 at 9:12 AM, Rohit Atri wrote: > Hi, > > I've a pipeline with the following elements : filesrc -> qtdemux -> > appsink. In my use-case, I collect 'N' number of packets, pause the > pipeline, seek back to a particular packet('s timestamp), and play again. > > I've noticed that even after I receive the state-changed message (to > PAUSE'd state) for the pipeline, I continue to receive packets and if let > alone I receive them till EOS(have tested my pipeline with a 40s video only > mpeg4 medium and also a two-hr long movie). I'm sure this is not the > expected behavior. > > Can someone explain why this could be happening? > > Could it be the flush that is done on seek? > > Thanks > Rohit > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Wed Dec 29 18:18:50 2010 From: wim.taymans at gmail.com (Wim Taymans) Date: Wed, 29 Dec 2010 18:18:50 +0100 Subject: [gst-devel] Packets received even after pipeline is PAUSE'd In-Reply-To: References: Message-ID: <1293643130.28802.21.camel@meany> On Wed, 2010-12-29 at 22:43 +0530, Rohit Atri wrote: > Hi Guys, > > > Any comments on the below query? Sounds like an unlikely bug, it needs more information to figure out what's going on. Are you setting the PAUSED state on the pipeline? If unsure, open a bug and provide a debug log. Wim > > > Thanks > Rohit > > On Wed, Dec 29, 2010 at 9:12 AM, Rohit Atri > wrote: > Hi, > > > I've a pipeline with the following elements : filesrc -> > qtdemux -> appsink. In my use-case, I collect 'N' number of > packets, pause the pipeline, seek back to a particular > packet('s timestamp), and play again. > > > I've noticed that even after I receive the state-changed > message (to PAUSE'd state) for the pipeline, I continue to > receive packets and if let alone I receive them till EOS(have > tested my pipeline with a 40s video only mpeg4 medium and also > a two-hr long movie). I'm sure this is not the expected > behavior. > > > Can someone explain why this could be happening? > > > Could it be the flush that is done on seek? > > > Thanks > Rohit > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From rohitratri at gmail.com Wed Dec 29 18:38:00 2010 From: rohitratri at gmail.com (Rohit Atri) Date: Wed, 29 Dec 2010 23:08:00 +0530 Subject: [gst-devel] Packets received even after pipeline is PAUSE'd In-Reply-To: <1293643130.28802.21.camel@meany> References: <1293643130.28802.21.camel@meany> Message-ID: Yes I set the *paused *on the pipeline. I'll take some logs and open a bug as suggested. Thanks Rohit On Wed, Dec 29, 2010 at 10:48 PM, Wim Taymans wrote: > On Wed, 2010-12-29 at 22:43 +0530, Rohit Atri wrote: > > Hi Guys, > > > > > > Any comments on the below query? > > Sounds like an unlikely bug, it needs more information to figure out > what's going on. Are you setting the PAUSED state on the pipeline? > > If unsure, open a bug and provide a debug log. > > Wim > > > > > > > Thanks > > Rohit > > > > On Wed, Dec 29, 2010 at 9:12 AM, Rohit Atri > > wrote: > > Hi, > > > > > > I've a pipeline with the following elements : filesrc -> > > qtdemux -> appsink. In my use-case, I collect 'N' number of > > packets, pause the pipeline, seek back to a particular > > packet('s timestamp), and play again. > > > > > > I've noticed that even after I receive the state-changed > > message (to PAUSE'd state) for the pipeline, I continue to > > receive packets and if let alone I receive them till EOS(have > > tested my pipeline with a 40s video only mpeg4 medium and also > > a two-hr long movie). I'm sure this is not the expected > > behavior. > > > > > > Can someone explain why this could be happening? > > > > > > Could it be the flush that is done on seek? > > > > > > Thanks > > Rohit > > > > > > > ------------------------------------------------------------------------------ > > Learn how Oracle Real Application Clusters (RAC) One Node allows > customers > > to consolidate database storage, standardize their database environment, > and, > > should the need arise, upgrade to a full multi-node Oracle RAC database > > without downtime or disruption > > http://p.sf.net/sfu/oracle-sfdevnl > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From umakantgoyal1 at gmail.com Wed Dec 29 19:30:40 2010 From: umakantgoyal1 at gmail.com (Umakant Goyal) Date: Thu, 30 Dec 2010 00:00:40 +0530 Subject: [gst-devel] EOS for Element Not for Entire Pipeline In-Reply-To: <1293574545.28802.14.camel@meany> References: <1293574545.28802.14.camel@meany> Message-ID: Hi, I have set the message-forward property for my bin using function g_object_set (mybin, "message-forward", 1, NULL); But still i am not getting EOS stream event for my sending pipepline when it completes playing stream. I got message in gstreamer logs *"message-forward*" only once my sending stream elements complete its task. Please help me. Thanks On Wed, Dec 29, 2010 at 3:45 AM, Wim Taymans wrote: > On Wed, 2010-12-29 at 01:32 +0530, Umakant Goyal wrote: > > Hi All, > > > > Can i receive EOS message from any specific element rather than from > > pipeline? > > What i know app receives EOS receives message when there is no > > activity in pipeline but i want to receive this event when element of > > my pipeline finishes its task but other elements of pipeline are still > > doing work. Is it possible? e.g if my pipeline is receiving and > > sending data to network, when my sending elements finishes its task > > then i want EOS stream for sending side elements even though my > > receiving elements are still receiving data from network. > > > You can use the message-forward property on GstBin, see the docs of that > property for how to use it. > > Wim > > > Thanks in Advance > > > ------------------------------------------------------------------------------ > > Learn how Oracle Real Application Clusters (RAC) One Node allows > customers > > to consolidate database storage, standardize their database environment, > and, > > should the need arise, upgrade to a full multi-node Oracle RAC database > > without downtime or disruption > > http://p.sf.net/sfu/oracle-sfdevnl > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From i.s.tarasov at gmail.com Wed Dec 29 20:28:47 2010 From: i.s.tarasov at gmail.com (Igor Tarasov) Date: Wed, 29 Dec 2010 22:28:47 +0300 Subject: [gst-devel] MpegTS playback bug. Message-ID: Hello, I found a problem to playback MpegTS video file with gstreamer. The image is broken, looks like a random pixels. I create this mpegts file using ffmpeg: ffmpeg -i somefile.avi -f mpegts -vcodec mpeg4 -acodec mp2 -ac 2 -ab 128k -s 720x432 -r 30 -re -b 10240k -threads 2 test.ts Seems stream 0 codec frame rate differs from container frame rate: 23.98 (65535/2733) -> 23.98 (10000000/417083) Input #0, avi, from 'ohotniki.avi': Metadata: ISFT : VirtualDubMod 1.5.10.2 (build 2542/release) INAM : Chasseurs de Dragons (2008) IART : vladaudi (BDRip 720) ICOP : HQ-ViDEO ICMT : 11.11.2008 Duration: 01:22:15.13, start: 0.000000, bitrate: 2537 kb/s Stream #0.0: Video: mpeg4, yuv420p, 720x384 [PAR 1:1 DAR 15:8], PAR 128:135 DAR 16:9, 23.98 tbr, 23.98 tbn, 23.98 tbc Stream #0.1: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s When I'm trying to play - I have the result I described above: gst-launch playbin uri=file:///home/foo/bar/somefile.ts But both VLC and MPlayer playback this file perfectly. Is it a gstreamer bug? Or something wrong with ffmpeg which creates this file? Thanks, Igor. From adam at 8planes.com Wed Dec 29 22:01:53 2010 From: adam at 8planes.com (Adam Duston) Date: Wed, 29 Dec 2010 16:01:53 -0500 Subject: [gst-devel] hiring gstreamer python dev for small-ish task Message-ID: I'm a python developer who can't spend the time to learn the gstreamer library right now. So I'd like to hire a Python GStreamer developer to write a script that can be run in the following form: join_videos It should join the two videos together side-by-side and combine audio. All videos are exactly the same size and each pair to be joined has the same duration. I can output the video files using any arbitrary codecs, so don't worry about the video/audio formats. Tell me if I'm wrong, but I think this is a pretty quick task for anyone who's fluent in the library. Please send your estimate off-list if you're interested. Thanks! Adam -- 8planes.com adam at 8planes.com 312-375-9879 Skype: aduston From pa at ufsoft.org Wed Dec 29 23:32:21 2010 From: pa at ufsoft.org (Pedro Algarvio, aka, s0undt3ch) Date: Wed, 29 Dec 2010 14:32:21 -0800 (PST) Subject: [gst-devel] gstrreamer and optparse In-Reply-To: <1293611519.3028.10.camel@espinkinglis> References: <4B1DF5DC.9040603@inattendu.org> <1293605985515-3166825.post@n4.nabble.com> <1293611519.3028.10.camel@espinkinglis> Message-ID: <4D1BB6ED.3000503@ufsoft.org> On 29/12/10 08:58, Philippe Normand [via GStreamer-devel] wrote: > On Tue, 2010-12-28 at 22:59 -0800, Pedro Algarvio, aka, s0undt3ch wrote: >> I'm having the same issue. >> >> Pedro Algarvio. > > A workaround would be to import gst after you parsed the options with > optparse. > > Philippe > > > ------------------------------------------------------------------------------ > > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------ > View message @ > http://gstreamer-devel.966125.n4.nabble.com/gstrreamer-and-optparse-tp966512p3166872.html > To unsubscribe from gstrreamer and optparse, click here > . > That's what I'm currently doing, but this does not seem the proper behaviour of a library. Regards, Pedro Algarvio -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/gstrreamer-and-optparse-tp966512p3167724.html Sent from the GStreamer-devel mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: From yuanqing.cai at tieto.com Thu Dec 30 03:42:53 2010 From: yuanqing.cai at tieto.com (Cai Yuanqing) Date: Thu, 30 Dec 2010 10:42:53 +0800 Subject: [gst-devel] How to stream video from a desktop to an Android device? In-Reply-To: References: Message-ID: <4D1BF1AD.30604@tieto.com> Hi Suneet: On 12/29/2010 11:42 PM, Suneet Shah wrote: > Hello, > > My name is Suneet and I'm working on a proof of concept project to get > back up to speed with development (I haven't done any serious > development in 3 years ;) ). I have some questions about what > gstreamer plugins I should look at to accomplish this. > > My project is this: > > * Stream a Linux desktop to an Android device as video > * Add simple control layer --> A click on the android device > should result in a click on the Linux desktop > > I've done a bunch of research on gstreamer and I think it is perfect > for what I want to do. So far, I've figured out how to capture the > desktop and record it to a video file (thanks to looking at the > Istanbul application). > > Now I need to figure out: > > * How to encode the video as h.264 > * How to stream the video to an android device > > I've chosen h.264 because Android has native capabilities for playing > H.264 streams, so this should keep the Android part of it fairly > simple. I'm planning on using simple socket programming to pass > messages between Android and the desktop for the control layer. I'm > using Python and pygst to write the Linux server app. > > My questions are: > > * Does my approach seem sound and should it work? > * What gstreamer plugins should I look at to stream the video? > * What plugins should I look at to encode H.264? > * Will I need to use another tool, like a streaming media server > (Darwin or otherwise) to stream the video, or can I accomplish > it completely through a gstreamer pipeline? > * Are there any relevant examples I should look at? > > Thanks in advance for your time. > > Suneet Sounds so great :-) In my opinion: 1.It should work. 2.Maybe you can try a gstreamer plugin named "rfbsrc" for streaming video. 3.You can gst-inspect-0.10 | grep enc to find any codec you want,and in this case,rfbsrc can fetch video stream from VNC server as RGB format,so you can just encode the stream to any format Android supported. 4.Maybe you can setup your Linux Desktop as a VNC server,you can just implement a gstreamer pipeline on the Android side(like rfbsrc -> capfilter -> videorate -> videoencoder -> surfaceflingersink). Thanks. -- B.R Cai Yuanqing From praveen.pande1 at gmail.com Thu Dec 30 07:16:02 2010 From: praveen.pande1 at gmail.com (praveen pandey) Date: Thu, 30 Dec 2010 11:46:02 +0530 Subject: [gst-devel] EOS for Element Not for Entire Pipeline In-Reply-To: References: <1293574545.28802.14.camel@meany> Message-ID: but why do you want to use EOS for a specific element rather then for full pipeline(which is the general case).? you can rather use other notification events for that particular element. BR, Praveen Pandey On Thu, Dec 30, 2010 at 12:00 AM, Umakant Goyal wrote: > Hi, > > I have set the message-forward property for my bin using function > g_object_set (mybin, "message-forward", 1, NULL); > But still i am not getting EOS stream event for my sending pipepline when > it completes playing stream. > I got message in gstreamer logs *"message-forward*" only once my sending > stream elements complete its task. > Please help me. > > Thanks > > > On Wed, Dec 29, 2010 at 3:45 AM, Wim Taymans wrote: > >> On Wed, 2010-12-29 at 01:32 +0530, Umakant Goyal wrote: >> > Hi All, >> > >> > Can i receive EOS message from any specific element rather than from >> > pipeline? >> > What i know app receives EOS receives message when there is no >> > activity in pipeline but i want to receive this event when element of >> > my pipeline finishes its task but other elements of pipeline are still >> > doing work. Is it possible? e.g if my pipeline is receiving and >> > sending data to network, when my sending elements finishes its task >> > then i want EOS stream for sending side elements even though my >> > receiving elements are still receiving data from network. >> > >> You can use the message-forward property on GstBin, see the docs of that >> property for how to use it. >> >> Wim >> >> > Thanks in Advance >> > >> ------------------------------------------------------------------------------ >> > Learn how Oracle Real Application Clusters (RAC) One Node allows >> customers >> > to consolidate database storage, standardize their database environment, >> and, >> > should the need arise, upgrade to a full multi-node Oracle RAC database >> > without downtime or disruption >> > http://p.sf.net/sfu/oracle-sfdevnl >> > _______________________________________________ gstreamer-devel mailing >> list gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, >> and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From praveen.pande1 at gmail.com Thu Dec 30 07:19:28 2010 From: praveen.pande1 at gmail.com (praveen pandey) Date: Thu, 30 Dec 2010 11:49:28 +0530 Subject: [gst-devel] Packets received even after pipeline is PAUSE'd In-Reply-To: References: <1293643130.28802.21.camel@meany> Message-ID: What is the input file format you are feeding to filesrc ? BR, Praveen Pandey On Wed, Dec 29, 2010 at 11:08 PM, Rohit Atri wrote: > Yes I set the *paused *on the pipeline. I'll take some logs and open a bug > as suggested. > > Thanks > Rohit > > > On Wed, Dec 29, 2010 at 10:48 PM, Wim Taymans wrote: > >> On Wed, 2010-12-29 at 22:43 +0530, Rohit Atri wrote: >> > Hi Guys, >> > >> > >> > Any comments on the below query? >> >> Sounds like an unlikely bug, it needs more information to figure out >> what's going on. Are you setting the PAUSED state on the pipeline? >> >> If unsure, open a bug and provide a debug log. >> >> Wim >> >> > >> > >> > Thanks >> > Rohit >> > >> > On Wed, Dec 29, 2010 at 9:12 AM, Rohit Atri >> > wrote: >> > Hi, >> > >> > >> > I've a pipeline with the following elements : filesrc -> >> > qtdemux -> appsink. In my use-case, I collect 'N' number of >> > packets, pause the pipeline, seek back to a particular >> > packet('s timestamp), and play again. >> > >> > >> > I've noticed that even after I receive the state-changed >> > message (to PAUSE'd state) for the pipeline, I continue to >> > receive packets and if let alone I receive them till EOS(have >> > tested my pipeline with a 40s video only mpeg4 medium and also >> > a two-hr long movie). I'm sure this is not the expected >> > behavior. >> > >> > >> > Can someone explain why this could be happening? >> > >> > >> > Could it be the flush that is done on seek? >> > >> > >> > Thanks >> > Rohit >> > >> > >> > >> ------------------------------------------------------------------------------ >> > Learn how Oracle Real Application Clusters (RAC) One Node allows >> customers >> > to consolidate database storage, standardize their database environment, >> and, >> > should the need arise, upgrade to a full multi-node Oracle RAC database >> > without downtime or disruption >> > http://p.sf.net/sfu/oracle-sfdevnl >> > _______________________________________________ gstreamer-devel mailing >> list gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, >> and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From praveen.pande1 at gmail.com Thu Dec 30 07:21:06 2010 From: praveen.pande1 at gmail.com (praveen pandey) Date: Thu, 30 Dec 2010 11:51:06 +0530 Subject: [gst-devel] Alsasink problem? In-Reply-To: References: <4D1A441F.1000801@panicking.kicks-ass.org> Message-ID: does this pipeline works for you gst-launch audiotestsrc ! alsasink -v BR, Praveen Pandey On Wed, Dec 29, 2010 at 3:00 AM, Radivoje Jovanovic < radivojejovanovic at gmail.com> wrote: > Yes this is an embedded system. As I stated in my previous post aplay, > alsamixer and playing MPEG4 movies through gstreamer works just fine. > Problem occurs when I try playing ogg files through the pipeline specified. > Cheers > Ogi > > > On Tue, Dec 28, 2010 at 12:10 PM, Michael Trimarchi < > michael at panicking.kicks-ass.org> wrote: > >> Hi >> >> On 12/28/2010 07:27 PM, Radivoje Jovanovic wrote: >> > Hi, >> > I have ALSA driver for my card and it works fine when I try playing >> sound >> > through aplay or watch movies through gstreamer, but if I try to do: >> > gst-launch-0.10 filesrc location=FILE_NAME.ogg ! oggdemux ! vorbisdec ! >> > audioconvert ! audioresample ! alsasink >> > it comes back: >> > Pipeline is PREROLLING ... >> > ERROR: from element /GstPipeline:pipeline0/GstAlsaSink:alsasink0: Could >> not >> > get/set settings from/on resource. >> > Additional debug info: >> > gstalsasink.c(519): set_hwparams (): >> > /GstPipeline:pipeline0/GstAlsaSink:alsasink0: >> > Unable to set hw params for playback: Invalid argument >> > ERROR: pipeline doesn't want to preroll. >> > Setting pipeline to NULL ... >> >> is this an embedded system. Can you try to use alsamixer, aplay and >> arecord to test if your driver >> is working correctly? >> Michael Trimarchi >> >> > >> > So it comes back that alsa cannot setup correctly even though alsasink >> works >> > correctly when I play MPEG movies. >> > any thoughts? >> > Cheers >> > Ogi >> > >> > >> > >> > >> > >> ------------------------------------------------------------------------------ >> > Learn how Oracle Real Application Clusters (RAC) One Node allows >> customers >> > to consolidate database storage, standardize their database environment, >> and, >> > should the need arise, upgrade to a full multi-node Oracle RAC database >> > without downtime or disruption >> > http://p.sf.net/sfu/oracle-sfdevnl >> > >> > >> > >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, >> and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From umakantgoyal1 at gmail.com Thu Dec 30 07:41:24 2010 From: umakantgoyal1 at gmail.com (Umakant Goyal) Date: Thu, 30 Dec 2010 12:11:24 +0530 Subject: [gst-devel] EOS for Element Not for Entire Pipeline In-Reply-To: References: <1293574545.28802.14.camel@meany> Message-ID: What all notificaton events i can receive from pipeline when one ends finish its processing? On Thu, Dec 30, 2010 at 11:46 AM, praveen pandey wrote: > but why do you want to use EOS for a specific element rather then for full > pipeline(which is the general case).? > > you can rather use other notification events for that particular element. > > BR, > Praveen Pandey > > > On Thu, Dec 30, 2010 at 12:00 AM, Umakant Goyal wrote: > >> Hi, >> >> I have set the message-forward property for my bin using function >> g_object_set (mybin, "message-forward", 1, NULL); >> But still i am not getting EOS stream event for my sending pipepline when >> it completes playing stream. >> I got message in gstreamer logs *"message-forward*" only once my sending >> stream elements complete its task. >> Please help me. >> >> Thanks >> >> >> On Wed, Dec 29, 2010 at 3:45 AM, Wim Taymans wrote: >> >>> On Wed, 2010-12-29 at 01:32 +0530, Umakant Goyal wrote: >>> > Hi All, >>> > >>> > Can i receive EOS message from any specific element rather than from >>> > pipeline? >>> > What i know app receives EOS receives message when there is no >>> > activity in pipeline but i want to receive this event when element of >>> > my pipeline finishes its task but other elements of pipeline are still >>> > doing work. Is it possible? e.g if my pipeline is receiving and >>> > sending data to network, when my sending elements finishes its task >>> > then i want EOS stream for sending side elements even though my >>> > receiving elements are still receiving data from network. >>> > >>> You can use the message-forward property on GstBin, see the docs of that >>> property for how to use it. >>> >>> Wim >>> >>> > Thanks in Advance >>> > >>> ------------------------------------------------------------------------------ >>> > Learn how Oracle Real Application Clusters (RAC) One Node allows >>> customers >>> > to consolidate database storage, standardize their database >>> environment, and, >>> > should the need arise, upgrade to a full multi-node Oracle RAC database >>> > without downtime or disruption >>> > http://p.sf.net/sfu/oracle-sfdevnl >>> > _______________________________________________ gstreamer-devel mailing >>> list gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Learn how Oracle Real Application Clusters (RAC) One Node allows >>> customers >>> to consolidate database storage, standardize their database environment, >>> and, >>> should the need arise, upgrade to a full multi-node Oracle RAC database >>> without downtime or disruption >>> http://p.sf.net/sfu/oracle-sfdevnl >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, >> and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From glassesbamboo at gmail.com Thu Dec 30 16:22:46 2010 From: glassesbamboo at gmail.com (Jiahao Wong) Date: Thu, 30 Dec 2010 23:22:46 +0800 Subject: [gst-devel] no block videomixer2 Message-ID: Hi, I mix some live streamer by videomixer2 element, but some media stream may delay or lost packet in Internet, videomixer2 will "WAIT" all pads buffer arrived but I want it process streamer in not block mode. How to modify videomixer, Please give me some hint on it. Thank you. From wmiller at sdr.com Thu Dec 30 17:23:41 2010 From: wmiller at sdr.com (Wes Miller) Date: Thu, 30 Dec 2010 08:23:41 -0800 (PST) Subject: [gst-devel] Extracting properties from element causes hang Message-ID: <1293726221815-3168471.post@n4.nabble.com> Elsewhere I have posted to this forum a routine based on gst-inspect that queries the various properties and ancestry of a declared element. I have used this routine today to extract information about a gstrtpbin element is a running pipeline (the inspection point is in the on_pad_added(0 callback). The program hangs as the third property is queried (latency). I have read other code that discusses the lock on elements and wonder if I am stuck behind some mutex or if there is some other reason a running element would block a g_object_get_property() (I think that what's blocking)? And, of course, any suggestions for a method to make this work? Warmly, Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Extracting-properties-from-element-causes-hang-tp3168471p3168471.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From florian.agez at ymagis.com Thu Dec 30 17:12:52 2010 From: florian.agez at ymagis.com (Florian Agez) Date: Thu, 30 Dec 2010 17:12:52 +0100 Subject: [gst-devel] MXF-wrapped JPEG2000 decoding Message-ID: <4D1CAF84.7070809@ymagis.com> Hi, I'm currently trying to make a very simple pipeline that would take an MXF file with JPEG2000 video essence as input and display it. For now, the pipeline looks something like this: gst-launch filesrc location="source.mxf" ! mxfdemux name=d d. ! jp2kdec ! autovideosink But that won't take me anywhere, I only get this error message: ERROR : element /GstPipeline:pipeline0/GstMXFDemux:d : Internal data stream error. Additional debug info: mxfdemux.c(2797): gst_mxf_demux_loop (): /GstPipeline:pipeline0/GstMXFDemux:d: stream stopped, reason not-negotiated From what I've experimented so far, it looks like mxfdemux has no problem at all unwrapping the JPEG2000 code stream (if I just link the demuxer to a filesink, I get a nice concatenation of valid JPEG2000 code streams), but I can't think of a reason why jp2kdec would refuse to handle this raw code stream. Adding to the confusion, when I try to rewrap the code stream into an MJ2 file, I also seem to get a corrupt file. I don't really know where to look now... Any ideas? Thanks! -- Florian From bilboed at gmail.com Thu Dec 30 18:05:15 2010 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 30 Dec 2010 18:05:15 +0100 Subject: [gst-devel] MXF-wrapped JPEG2000 decoding In-Reply-To: <4D1CAF84.7070809@ymagis.com> References: <4D1CAF84.7070809@ymagis.com> Message-ID: <1293728715.2527.4.camel@deumeu> On Thu, 2010-12-30 at 17:12 +0100, Florian Agez wrote: > Hi, > > I'm currently trying to make a very simple pipeline that would take an > MXF file with JPEG2000 video essence as input and display it. For now, > the pipeline looks something like this: > > gst-launch filesrc location="source.mxf" ! mxfdemux name=d d. ! > jp2kdec ! autovideosink > > But that won't take me anywhere, I only get this error message: > > ERROR : element /GstPipeline:pipeline0/GstMXFDemux:d : Internal > data stream error. > Additional debug info: > mxfdemux.c(2797): gst_mxf_demux_loop (): > /GstPipeline:pipeline0/GstMXFDemux:d: > stream stopped, reason not-negotiated > Do you get any more detailed warnings when running the same command line with "GST_DEBUG=2 gst-launch...." ? If nothing 'useful' comes up, please file a bug report with full compressed log (GST_DEBUG=5 gst-launch .... > log 2>&1). Bonus points for quicker debugging if you also provide (a link to) a sample file which fails. Edward > > From what I've experimented so far, it looks like mxfdemux has no > problem at all unwrapping the JPEG2000 code stream (if I just link the > demuxer to a filesink, I get a nice concatenation of valid JPEG2000 code > streams), but I can't think of a reason why jp2kdec would refuse to > handle this raw code stream. Adding to the confusion, when I try to > rewrap the code stream into an MJ2 file, I also seem to get a corrupt file. > > I don't really know where to look now... Any ideas? > > Thanks! > From gibrovacco at gmail.com Thu Dec 30 18:26:22 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Thu, 30 Dec 2010 19:26:22 +0200 Subject: [gst-devel] MpegTS playback bug. In-Reply-To: References: Message-ID: Hi, On Wed, Dec 29, 2010 at 9:28 PM, Igor Tarasov wrote: > Hello, > > I found a problem to playback MpegTS video file with gstreamer. The > image is broken, looks like a random pixels. > > I create this mpegts file using ffmpeg: > > ffmpeg -i somefile.avi -f mpegts ?-vcodec mpeg4 -acodec mp2 -ac 2 -ab > 128k -s 720x432 -r 30 -re -b 10240k -threads 2 test.ts > > Seems stream 0 codec frame rate differs from container frame rate: > 23.98 (65535/2733) -> 23.98 (10000000/417083) > Input #0, avi, from 'ohotniki.avi': > ?Metadata: > ? ?ISFT ? ? ? ? ? ?: VirtualDubMod 1.5.10.2 (build 2542/release) > ? ?INAM ? ? ? ? ? : Chasseurs de Dragons (2008) > ? ?IART ? ? ? ? ? ?: vladaudi (BDRip 720) > ? ?ICOP ? ? ? ? ? ?: HQ-ViDEO > ? ?ICMT ? ? ? ? ? ?: 11.11.2008 > ?Duration: 01:22:15.13, start: 0.000000, bitrate: 2537 kb/s > ? ?Stream #0.0: Video: mpeg4, yuv420p, 720x384 [PAR 1:1 DAR 15:8], > PAR 128:135 DAR 16:9, 23.98 tbr, 23.98 tbn, 23.98 tbc > ? ?Stream #0.1: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s > > When I'm trying to play - I have the result I described above: > gst-launch playbin ?uri=file:///home/foo/bar/somefile.ts > > But both VLC and MPlayer playback this file perfectly. > > Is it a gstreamer bug? Or something wrong with ffmpeg which creates this file? > Can you collect some debug setting e.g. GST_DEBUG = 3 or more (and attach the output somewhere to pastebin)? Regards > Thanks, > Igor. > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From shamun.toha at gmail.com Thu Dec 30 18:53:58 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Thu, 30 Dec 2010 18:53:58 +0100 Subject: [gst-devel] Gstreamer vs Mac (Logic studio) can i build it ? Message-ID: Please have a look Mac (Logic studio). ex: http://www.apple.com/logicstudio/whats-new/200plus-features.html Mac logic studio (200 features): http://www.apple.com/logicstudio/whats-new/200plus-features.html With Gstreamer can i build similar staffs (well i cant be alone Mac, but atleast i can try :=) for learning and research ) ?: - I am trying use Gstreamer to build something similar like Logic Studio, i dont know how i can have the data to visualize the freqency from Microphone and Sound card and render it - To apply my own realtime metering/graphs etc, study/debug sound - Preamp create etc Please advise, what is Gstreamer vs Logic Studio ? How to achieve that quality with Gstreamer. Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From david at davidzhao.com Fri Dec 31 05:35:21 2010 From: david at davidzhao.com (David Zhao) Date: Thu, 30 Dec 2010 20:35:21 -0800 Subject: [gst-devel] Implementing a MPEGTS segmenter, what's the best approach? Message-ID: Hello, I'm trying to build a GStreamer plugin that would produce segments of MPEGTS files, each with a fixed duration. This is to create TS segments for the Apple HTTP live streaming protocol. I'm fairly new to developing for GStreamer and have a basic understanding of how it works. The approach I had in mind is to create a plugin that includes both mpegtsmux and filesink. Within that plugin, it could produce individual ts files when certain duration is accumulated. Is this feasible? Is there another way to do this without having to replicate code that's already in mpegtsmux and filesink? Thanks, -David -------------- next part -------------- An HTML attachment was scrubbed... URL: From rohitratri at gmail.com Fri Dec 31 06:15:29 2010 From: rohitratri at gmail.com (Rohit Atri) Date: Fri, 31 Dec 2010 10:45:29 +0530 Subject: [gst-devel] Packets received even after pipeline is PAUSE'd In-Reply-To: References: <1293643130.28802.21.camel@meany> Message-ID: I'm feeding an mpeg4 stream (xxx.mp4 is the file format btw) as mentioned in my earlier post. Have noticed it with multiple files and not just one. Thanks Rohit On Thu, Dec 30, 2010 at 11:49 AM, praveen pandey wrote: > What is the input file format you are feeding to filesrc ? > > BR, > Praveen Pandey > > On Wed, Dec 29, 2010 at 11:08 PM, Rohit Atri wrote: > >> Yes I set the *paused *on the pipeline. I'll take some logs and open a >> bug as suggested. >> >> Thanks >> Rohit >> >> >> On Wed, Dec 29, 2010 at 10:48 PM, Wim Taymans wrote: >> >>> On Wed, 2010-12-29 at 22:43 +0530, Rohit Atri wrote: >>> > Hi Guys, >>> > >>> > >>> > Any comments on the below query? >>> >>> Sounds like an unlikely bug, it needs more information to figure out >>> what's going on. Are you setting the PAUSED state on the pipeline? >>> >>> If unsure, open a bug and provide a debug log. >>> >>> Wim >>> >>> > >>> > >>> > Thanks >>> > Rohit >>> > >>> > On Wed, Dec 29, 2010 at 9:12 AM, Rohit Atri >>> > wrote: >>> > Hi, >>> > >>> > >>> > I've a pipeline with the following elements : filesrc -> >>> > qtdemux -> appsink. In my use-case, I collect 'N' number of >>> > packets, pause the pipeline, seek back to a particular >>> > packet('s timestamp), and play again. >>> > >>> > >>> > I've noticed that even after I receive the state-changed >>> > message (to PAUSE'd state) for the pipeline, I continue to >>> > receive packets and if let alone I receive them till EOS(have >>> > tested my pipeline with a 40s video only mpeg4 medium and also >>> > a two-hr long movie). I'm sure this is not the expected >>> > behavior. >>> > >>> > >>> > Can someone explain why this could be happening? >>> > >>> > >>> > Could it be the flush that is done on seek? >>> > >>> > >>> > Thanks >>> > Rohit >>> > >>> > >>> > >>> ------------------------------------------------------------------------------ >>> > Learn how Oracle Real Application Clusters (RAC) One Node allows >>> customers >>> > to consolidate database storage, standardize their database >>> environment, and, >>> > should the need arise, upgrade to a full multi-node Oracle RAC database >>> > without downtime or disruption >>> > http://p.sf.net/sfu/oracle-sfdevnl >>> > _______________________________________________ gstreamer-devel mailing >>> list gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Learn how Oracle Real Application Clusters (RAC) One Node allows >>> customers >>> to consolidate database storage, standardize their database environment, >>> and, >>> should the need arise, upgrade to a full multi-node Oracle RAC database >>> without downtime or disruption >>> http://p.sf.net/sfu/oracle-sfdevnl >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >> >> >> >> ------------------------------------------------------------------------------ >> Learn how Oracle Real Application Clusters (RAC) One Node allows customers >> to consolidate database storage, standardize their database environment, >> and, >> should the need arise, upgrade to a full multi-node Oracle RAC database >> without downtime or disruption >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rohitratri at gmail.com Fri Dec 31 06:20:35 2010 From: rohitratri at gmail.com (Rohit Atri) Date: Fri, 31 Dec 2010 10:50:35 +0530 Subject: [gst-devel] Extracting properties from element causes hang In-Reply-To: <1293726221815-3168471.post@n4.nabble.com> References: <1293726221815-3168471.post@n4.nabble.com> Message-ID: Have you tried using g_object_get( *)* instead? Not directly related to the issue but documentation recommends the usage of this one. Rohit * * * * On Thu, Dec 30, 2010 at 9:53 PM, Wes Miller wrote: > > Elsewhere I have posted to this forum a routine based on gst-inspect that > queries the various properties and ancestry of a declared element. > > I have used this routine today to extract information about a gstrtpbin > element is a running pipeline (the inspection point is in the > on_pad_added(0 > callback). The program hangs as the third property is queried (latency). > > I have read other code that discusses the lock on elements and wonder if I > am stuck behind some mutex or if there is some other reason a running > element would block a g_object_get_property() (I think that what's > blocking)? > > And, of course, any suggestions for a method to make this work? > > Warmly, > > Wes > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/Extracting-properties-from-element-causes-hang-tp3168471p3168471.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ds at entropywave.com Fri Dec 31 08:23:36 2010 From: ds at entropywave.com (David Schleef) Date: Fri, 31 Dec 2010 02:23:36 -0500 Subject: [gst-devel] Implementing a MPEGTS segmenter, what's the best approach? In-Reply-To: References: Message-ID: <20101231072336.GA20495@cooker.entropywave.com> On Thu, Dec 30, 2010 at 08:35:21PM -0800, David Zhao wrote: > Hello, > > I'm trying to build a GStreamer plugin that would produce segments of MPEGTS > files, each with a fixed duration. This is to create TS segments for the > Apple HTTP live streaming protocol. I'm fairly new to developing for > GStreamer and have a basic understanding of how it works. > > The approach I had in mind is to create a plugin that includes both > mpegtsmux and filesink. Within that plugin, it could produce individual ts > files when certain duration is accumulated. > > Is this feasible? Is there another way to do this without having to > replicate code that's already in mpegtsmux and filesink? gst-launch ... ! mpegtsmux ! multifilesink next-file=key-frame \ location=%05d.ts David From yiliangb at gmail.com Fri Dec 31 08:51:08 2010 From: yiliangb at gmail.com (yiliang) Date: Thu, 30 Dec 2010 23:51:08 -0800 (PST) Subject: [gst-devel] How to compile multiverse variant of plugin library? Message-ID: <1293781868795-3169168.post@n4.nabble.com> Hi, I want to compile just the multiverse variant of ulgy plugins on Ubuntu 10.10. How should I do it? The multiverse variant source downloaded from ubuntu is identical to that of universe variant. Thanks! Yiliang -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-compile-multiverse-variant-of-plugin-library-tp3169168p3169168.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From Topi.Santakivi at digia.com Fri Dec 31 09:00:22 2010 From: Topi.Santakivi at digia.com (Santakivi Topi) Date: Fri, 31 Dec 2010 10:00:22 +0200 Subject: [gst-devel] How to compile multiverse variant of plugin library? In-Reply-To: <1293781868795-3169168.post@n4.nabble.com> References: <1293781868795-3169168.post@n4.nabble.com> Message-ID: <4D1D8D96.8060509@digia.com> On 12/31/2010 09:51 AM, yiliang wrote: > > Hi, > > I want to compile just the multiverse variant of ulgy plugins on Ubuntu > 10.10. How should I do it? The multiverse variant source downloaded from > ubuntu is identical to that of universe variant. Take a look at the Debian rules file. For the multiverse variant: DEB_CONFIGURE_EXTRA_FLAGS += \ --disable-examples \ --enable-debug \ --with-package-name="GStreamer Ugly Multiverse Plugins (Ubuntu)" \ --with-package-origin="https://launchpad.net/distros/ubuntu/+source/gst-plugins-ugly-multiverse$(gst_abi)" \ --disable-gtk-doc \ --disable-asfdemux \ --disable-dvdlpcmdec \ --disable-dvdsub \ --disable-iec958 \ --disable-mpegaudioparse \ --disable-mpegstream \ --disable-realmedia BR, Topi > > Thanks! > Yiliang > > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/How-to-compile-multiverse-variant-of-plugin-library-tp3169168p3169168.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From gibrovacco at gmail.com Fri Dec 31 11:58:29 2010 From: gibrovacco at gmail.com (Marco Ballesio) Date: Fri, 31 Dec 2010 12:58:29 +0200 Subject: [gst-devel] Gstreamer vs Mac (Logic studio) can i build it ? In-Reply-To: References: Message-ID: Hi, On Thu, Dec 30, 2010 at 7:53 PM, Shamun toha md wrote: > Please have a look Mac (Logic studio). ex: > http://www.apple.com/logicstudio/whats-new/200plus-features.html > > Mac logic studio (200 features): > http://www.apple.com/logicstudio/whats-new/200plus-features.html > With Gstreamer can i build similar staffs (well i cant be alone Mac, but > atleast i can try :=) for learning and research ) ?: > - I am trying use Gstreamer to build something similar like Logic Studio, i > dont know how i can have the data to visualize the freqency from Microphone > and Sound card and render it > - To apply my own realtime metering/graphs etc, study/debug sound > - Preamp create etc > Please advise, what is Gstreamer vs Logic Studio ? How to achieve that > quality with Gstreamer. Use the Google, Luke: http://www.jokosher.org/ and (pretty old but maybe interesting): http://wavemixer.sourceforge.net/ Regards > Thanks & Regards > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From 21stninjagod at gmail.com Fri Dec 31 18:18:45 2010 From: 21stninjagod at gmail.com (gst_call_me_newbie()) Date: Fri, 31 Dec 2010 09:18:45 -0800 (PST) Subject: [gst-devel] How to find the stream meta-data using Gstreamer ? Message-ID: <1293815925946-3169634.post@n4.nabble.com> I am using souphttpsrc as my source. I want to get the stream meta-data like Codec(what are the audio/video codecs present in the stream), type (audio/video stream) etc. I checked that using playbin/playbin2 I can find the properties. LINK: http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-playbin.html http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-playbin.html but the problem is I have to transcode the incoming stream from a http URL to another format. So I don't think playbin2 is of any use for me. (CORRECT ME IF I AM WRONG HERE) whatever may be the way, I just want to transcode incoming stream for which I should know the incoming stream format. So how would I get the stream meta data from "souphttpsrc" or should I go for playbin2 but then I am doubtful if I would be able to transcode stream using it. Thanks Mark -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-find-the-stream-meta-data-using-Gstreamer-tp3169634p3169634.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From braunsquared at gmail.com Fri Dec 31 18:28:02 2010 From: braunsquared at gmail.com (Timothy Braun) Date: Fri, 31 Dec 2010 12:28:02 -0500 Subject: [gst-devel] How to find the stream meta-data using Gstreamer ? In-Reply-To: <1293815925946-3169634.post@n4.nabble.com> References: <1293815925946-3169634.post@n4.nabble.com> Message-ID: playbin and decodebin both setup the decoders, using typefind, for you and provide you with a raw decoded video and audio stream. What are your goals for the output of the stream? Tim On Fri, Dec 31, 2010 at 12:18 PM, gst_call_me_newbie() < 21stninjagod at gmail.com> wrote: > > I am using souphttpsrc as my source. I want to get the stream meta-data > like > Codec(what are the audio/video codecs present in the stream), type > (audio/video stream) etc. > > I checked that using playbin/playbin2 I can find the properties. > LINK: > > http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-playbin.html > > http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-playbin.html > > but the problem is I have to transcode the incoming stream from a http URL > to another format. > So I don't think playbin2 is of any use for me. (CORRECT ME IF I AM WRONG > HERE) > > whatever may be the way, I just want to transcode incoming stream for which > I should know the incoming stream format. > So how would I get the stream meta data from "souphttpsrc" > or should I go for playbin2 but then I am doubtful if I would be able to > transcode stream using it. > > Thanks > Mark > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/How-to-find-the-stream-meta-data-using-Gstreamer-tp3169634p3169634.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From 21stninjagod at gmail.com Fri Dec 31 18:38:10 2010 From: 21stninjagod at gmail.com (gst_call_me_newbie()) Date: Fri, 31 Dec 2010 09:38:10 -0800 (PST) Subject: [gst-devel] How to find the stream meta-data using Gstreamer ? In-Reply-To: References: <1293815925946-3169634.post@n4.nabble.com> Message-ID: <1293817090516-3169652.post@n4.nabble.com> oh! I didn't knew that. I just checked it using gst-inpect decodebin (its a generic decoder). Thanks I am going to transcode that/those(audio+video) raw stream(s) to ogg container format and stream over to client machine which is present in the LAN. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-find-the-stream-meta-data-using-Gstreamer-tp3169634p3169652.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From w3cing at gmail.com Fri Dec 31 18:45:36 2010 From: w3cing at gmail.com (wecing) Date: Sat, 1 Jan 2011 01:45:36 +0800 Subject: [gst-devel] How to mix the subtitle layer with the video in my gtk program? Message-ID: I'm trying to write a video player that can make the subtitles "fly". So, I need to define my own output function. (In fact I'm not sure where should I put this question -- cairo-dev, or gst-devel? No offence, however.) Thanks. wecing. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david at davidzhao.com Fri Dec 31 20:42:16 2010 From: david at davidzhao.com (David Zhao) Date: Fri, 31 Dec 2010 11:42:16 -0800 Subject: [gst-devel] Implementing a MPEGTS segmenter, what's the best approach? In-Reply-To: <20101231072336.GA20495@cooker.entropywave.com> References: <20101231072336.GA20495@cooker.entropywave.com> Message-ID: Thanks David! That's exactly what I was looking for. On Thu, Dec 30, 2010 at 11:23 PM, David Schleef wrote: > On Thu, Dec 30, 2010 at 08:35:21PM -0800, David Zhao wrote: > > Hello, > > > > I'm trying to build a GStreamer plugin that would produce segments of > MPEGTS > > files, each with a fixed duration. This is to create TS segments for the > > Apple HTTP live streaming protocol. I'm fairly new to developing for > > GStreamer and have a basic understanding of how it works. > > > > The approach I had in mind is to create a plugin that includes both > > mpegtsmux and filesink. Within that plugin, it could produce individual > ts > > files when certain duration is accumulated. > > > > Is this feasible? Is there another way to do this without having to > > replicate code that's already in mpegtsmux and filesink? > > gst-launch ... ! mpegtsmux ! multifilesink next-file=key-frame \ > location=%05d.ts > > > > David > > > > ------------------------------------------------------------------------------ > Learn how Oracle Real Application Clusters (RAC) One Node allows customers > to consolidate database storage, standardize their database environment, > and, > should the need arise, upgrade to a full multi-node Oracle RAC database > without downtime or disruption > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From umakantgoyal1 at gmail.com Fri Dec 31 22:18:49 2010 From: umakantgoyal1 at gmail.com (Umakant Goyal) Date: Sat, 1 Jan 2011 02:48:49 +0530 Subject: [gst-devel] Multiple Caps for udpsrc Message-ID: Hi All, Can we set more than one caps for udpsrc? like "application/x-rtp, media=audio, clock-rate=8000, media=video, clock-rate=90000" if yes, then please share how it can be done. Thanks in Advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From keevon at gmail.com Sun Dec 12 07:50:14 2010 From: keevon at gmail.com (KEVIN WEST) Date: Sun, 12 Dec 2010 06:50:14 -0000 Subject: [gst-devel] gstdshowdecwrapper hangs on windows without ffmpeg Message-ID: We are running into issues trying to get gstreamer working with directshow without using the ffmpeg libs. We were able to transcode an mpeg1 file using the below pipeline, but we soon realized we were using libgstmpeg2dec.dll. Once we deleted that, we verified everything was going through directshow, but no files will complete. From stepping into the source it looks like the directshow plugin is being loaded, and then quickly unloaded, after which is simply hangs. It looks like it's hanging during the decode process, so we aren't even able to get the YUV2 output. Has anyone been able to get a valid directshow transcoding pipeline to work? Here is our (somewhat modified) pipeline: gst-launch.exe --gst-debug-level=3 --gst-plugin-path="plugins" --seek=0 uridecodebin name=dec uri= dec. ! videoscale ! videorate ! video/x-raw-yuv ! x264enc ! mux. dec. ! audioconvert ! audioresample ! audio/x-raw-int ! faac ! mux. flvmux streamable=true name= mux ! filesink location="output.flv" -------------- next part -------------- An HTML attachment was scrubbed... URL: From puneeth.mv at globaledgesoft.com Mon Dec 13 13:58:32 2010 From: puneeth.mv at globaledgesoft.com (Puneeth) Date: Mon, 13 Dec 2010 12:58:32 -0000 Subject: [gst-devel] How do u set caps for udpsrc Message-ID: <1292244042611-3085367.post@n4.nabble.com> Hi All I m new to gstreamer please help me to solve this bug.. sender: gst-launch -v alsasrc ! lamemp3enc ! rtpmpapay ! udpsink port=6969 host=192.168.0.200 Receiver: gst-launch -v udpsrc port=6969 caps="application/x-rtp, media=(string)audio, clock-rate=(int)90000, encoding-name=(string)MPA, payload=(int)96" ! rtpmpadepay ! mp3parse ! ffmux_mp3 ! filesink location=file.mp3 when i m using this in command line i m able to capture raw pcm data and encoding to mp3 format and sending and receiving through rtp and saving to file but when i m trying with program it is giving segmentation fault while setting capabilities using g_object_set Here is the code where at the receiver side...any suggestions will be greatly appreciated... #include #include #define caps "application/x-rtp,media=(string)audio,clock-rate=(int)90000,encoding-name=(string)MPA,payload=(int)96" static gboolean bus_call (GstBus *bus, GstMessage *msg, gpointer data) { GMainLoop *loop = (GMainLoop *) data; switch (GST_MESSAGE_TYPE (msg)) { case GST_MESSAGE_EOS: g_print ("End of stream\n"); g_main_loop_quit (loop); break; case GST_MESSAGE_ERROR: { gchar *debug; GError *error; gst_message_parse_error (msg, &error, &debug); g_free (debug); g_printerr ("Error: %s\n", error->message); g_error_free (error); g_main_loop_quit (loop); break; } default: break; } return TRUE; } static void on_pad_added (GstElement *element, GstPad *pad, gpointer data) { GstPad *sinkpad; GstElement *decoder = (GstElement *) data; /* We can now link this pad with the vorbis-decoder sink pad */ g_print ("Dynamic pad created, linking demuxer/decoder\n"); sinkpad = gst_element_get_static_pad (decoder, "sink"); gst_pad_link (pad, sinkpad); gst_object_unref (sinkpad); } int main (int argc, char *argv[]) { GMainLoop *loop; GstElement *pipeline, *rtp, *muxer, *parse, *sink, *udp; GstBus *bus; /* Initialisation */ gst_init (&argc, &argv); loop = g_main_loop_new (NULL, FALSE); /* Check input arguments */ if (argc != 2) { g_printerr ("Usage: %s \n", argv[0]); return -1; } /* Create gstreamer elements */ pipeline = gst_pipeline_new ("audio-player"); udp = gst_element_factory_make ("udpsrc", "udp-src"); rtp = gst_element_factory_make ("rtpmpadepay", "rtpmpa-depay"); parse = gst_element_factory_make ("mp3parse", "mp3-parse"); muxer = gst_element_factory_make ("ffmux_mp3", "ffmux-mp3"); sink = gst_element_factory_make ("filesink", "file-sink"); if (!pipeline || !udp || !rtp || !parse || !muxer || !sink) { g_printerr ("One element could not be created. Exiting.\n"); return -1; } /* Set up the pipeline */ /* we set the input filename to the source element */ g_object_set (G_OBJECT (sink), "location", argv[1], NULL); g_object_set (G_OBJECT (udp), "typefind", TRUE, "port", 3000, "caps", caps, NULL); /* we add a message handler */ bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); gst_bus_add_watch (bus, bus_call, loop); gst_object_unref (bus); /* we add all elements into the pipeline */ /* file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output */ gst_bin_add_many (GST_BIN (pipeline), udp, rtp, parse, muxer, sink, NULL); // g_signal_connect (muxer, "pad-added", G_CALLBACK (on_pad_added), decoder); gst_element_link_many ( udp, rtp, parse, muxer, sink, NULL); /* note that the demuxer will be linked to the decoder dynamically. The reason is that Ogg may contain various streams (for example audio and video). The source pad(s) will be created at run time, by the demuxer when it detects the amount and nature of streams. Therefore we connect a callback function which will be executed when the "pad-added" is emitted.*/ /* Set the pipeline to "playing" state*/ g_print ("Now playing: %s\n", argv[1]); gst_element_set_state (pipeline, GST_STATE_PLAYING); /* Iterate */ g_print ("Running...\n"); g_main_loop_run (loop); /* Out of the main loop, clean up nicely */ g_print ("Returned, stopping playback\n"); gst_element_set_state (pipeline, GST_STATE_NULL); g_print ("Deleting pipeline\n"); gst_object_unref (GST_OBJECT (pipeline)); return 0; } -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-do-u-set-caps-for-udpsrc-tp3085367p3085367.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From os_developer at yahoo.com Mon Dec 13 22:15:05 2010 From: os_developer at yahoo.com (os_developer) Date: Mon, 13 Dec 2010 21:15:05 -0000 Subject: [gst-devel] how to use appsrc in pipeline to rtsp-server Message-ID: <1292274899859-3086064.post@n4.nabble.com> What is the proper way to pass a GstBin or GstElement as the pipeline instead of using a launch string? I'm trying to use a pipeline as: appsrc ! ffmpegcolorspace ! x264enc ! rtph264pay But it isn't clear to me exactly what is required for implementing my own version of GstRTSPMediaFactoryClass create_pipeline and get_element. I create my own pipeline element (m_pipeline), then add and link the plugins, and return the m_pipeline pointer in create_pipeline and get_element. However, it seems however I tweak things, I always end up with errors relating to either not having a top-level bin, or not being able to pre-roll, etc. When I try to hook up the appsrc callbacks using gst_app_src_set_callbacks(), my callbacks are never called. Without the callbacks, I am using a loop where I pass frames of data to appsrc, but only the first 3 seem to go through before it errors saying it is full, but nothing ever appears on the client end of the stream. I know my pipeline is valid because I can use it in a stand-alone app that doesn't use the rtsp-server. I am assuming I'm missing some configuration somewhere but can't seem to figure out what it is. If I just a multifilesrc in place of appsrc and pass a launch string using the ...set_launch() function, it works great. Any ideas on where I could look for examples of overriding create_pipeline and the other methods in the factory? Thanks, Andrea -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/how-to-use-appsrc-in-pipeline-to-rtsp-server-tp3086064p3086064.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From faruk.namli at gmail.com Tue Dec 14 16:47:40 2010 From: faruk.namli at gmail.com (frknml) Date: Tue, 14 Dec 2010 15:47:40 -0000 Subject: [gst-devel] Live Streaming Message-ID: <1292341654412-3087339.post@n4.nabble.com> Hi Everyone; I'm trying to design real time video streamer.Firstly i tried it in my local system with my webcam.The senario is,I'm sending my webcam capture via udp at the same time i'm receiving this data from my local network.I could succeed to show my webcam capture but I couldn't overcome the latency and the video is not smooth.I wrote my command below,please advise me how can i handle these problems. Receiver gst-launch udpsrc buffer-size=16777216 port=1234 ! ffdemux_mpegts ! ffdec_mpeg2video ! autovideosink Sender gst-launch v4l2src ! video/x-raw-yuv,width=320,height=240,framerate=\(fraction\)30/1 ! ffenc_mpeg2video ! ffmux_mpegts ! udpsink host="10.0.0.217" port=1234 Faruk -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Live-Streaming-tp3087339p3087339.html Sent from the GStreamer-devel mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: From willy at payservices.com Tue Dec 14 22:25:48 2010 From: willy at payservices.com (Willy DANENBERG) Date: Tue, 14 Dec 2010 21:25:48 -0000 Subject: [gst-devel] Looking for a trainer in Gstreamer Message-ID: Hi, We are setting now a new server using Python / Gstreamer and Flumotion software for a Video On Demand system. We have some issues to work out and need someone who has experience in the above and is interested to teach for a few hours and / or give support for that server on a long term base. A kind reply will be highly appreciated. With My Kindest Regards, Willy This E-mail / letter is confidential and may also be privileged. If you are not the intended recipient, please notify us immediately. You should not copy it or use it for any other purpose, nor disclose its contents to any other person. All our transactions and correspondence are subject our general terms and conditions. Copy available on request. -------------- next part -------------- An HTML attachment was scrubbed... URL: From chiyeh.chen at movial.com Wed Dec 15 09:35:32 2010 From: chiyeh.chen at movial.com (chiyeh.chen) Date: Wed, 15 Dec 2010 08:35:32 -0000 Subject: [gst-devel] Converting .264 elemntary stream to .Mp4 In-Reply-To: References: <1290595763181-3057059.post@n4.nabble.com> Message-ID: <1292401042354-3088557.post@n4.nabble.com> Use ffmpeg maybe is good idea. Gstreamer's component convert file seems have problem. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Converting-264-elemntary-stream-to-Mp4-tp3057059p3088557.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From rohitratri at gmail.com Wed Dec 15 17:01:37 2010 From: rohitratri at gmail.com (rohitratri at gmail.com) Date: Wed, 15 Dec 2010 16:01:37 -0000 Subject: [gst-devel] video frame type identification Message-ID: <1292428891718-3089291.post@n4.nabble.com> Hi, Is there a way to identify the type of video frame within an encoded packet returned by qtdemux? May be some caps I can check on the GstBuffer return by qtdemux? Note - I use appsink to pull buffers out of the pipeline and process them - need to know the type of frame for processing the encoded packet Thanks Rohit -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/video-frame-type-identification-tp3089291p3089291.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From 632992753 at qq.com Mon Dec 20 03:08:21 2010 From: 632992753 at qq.com (citywolf) Date: Mon, 20 Dec 2010 02:08:21 -0000 Subject: [gst-devel] Example of Video player Message-ID: <1292810894762-3095045.post@n4.nabble.com> I can use the command below to play video file: gst-launch playbin uri=file:////home/jinkui/Downloads/sample.m4v but the command below would be error: gst-launch filesrc location=sample.m4v ! ffdec_mpeg4 ! ffmpegcolorspace ! ximagesink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/ffdec_mpeg4:ffdec_mpeg40: Internal GStreamer error: negotiation problem. Please file a bug at http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer. Additional debug info: gstffmpegdec.c(2729): gst_ffmpegdec_chain (): /GstPipeline:pipeline0/ffdec_mpeg4:ffdec_mpeg40: ffdec_mpeg4: input format was not set before data start ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... can someone give me a example of video player about m4v files using ffdec_mpeg4? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Example-of-Video-player-tp3095045p3095045.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From krkrams27 at gmail.com Mon Dec 20 10:24:36 2010 From: krkrams27 at gmail.com (rams k) Date: Mon, 20 Dec 2010 09:24:36 -0000 Subject: [gst-devel] Playbin for multiple video sink support Message-ID: Hi, I want to use playbin for renderign MVC streams. Im decoding using h264 mvc decoder plugin and sending multiple stream outputs by creating multiple src pads. ACTUAL: gst-launch location= ! name=d d.src_0 ! queue ! xvimagesink d.src_1 ! queue ! xvimagesink d.src_2 ! queue ! xvimagesink ! ..........................d.src_8 ! queue ! xvimagesink -----It renders 8 video streams ,in 8 different windows USING PLAYBIN: gst-launch playbin uri=file://gst-launch playbin uri=file:// ---Here it is playing only one window... Please help me in rendering multiple videos using playbin? rgds, RK -------------- next part -------------- An HTML attachment was scrubbed... URL: From krkrams27 at gmail.com Mon Dec 20 10:56:47 2010 From: krkrams27 at gmail.com (rams k) Date: Mon, 20 Dec 2010 09:56:47 -0000 Subject: [gst-devel] ogg pipeline In-Reply-To: References: Message-ID: Yes,I can play with playbin. But ,I want to know the pipeline to play the ogg file using oggdemux,theoradec,and vorbisdec. $gst-launch filesrc location= ! oggdemux name=d d.video_00 ! queue ! xvimagesink d.audio_00 ! queue ! alsasink (gst-launch-0.10:24567): GLib-WARNING **: g_set_prgname() called multiple times Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstOggDemux:d: Internal data stream error. Additional debug info: gstoggdemux.c(3251): gst_ogg_demux_loop (): /GstPipeline:pipeline0/GstOggDemux:d: stream stopped, reason not-linked ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... rgds, Rk On Tue, Dec 7, 2010 at 12:58 AM, Michael Smith wrote: > You have an ogg file. You're using a quicktime demuxer. It should not > be a surprise that it says that it's an invalid file. > > Try this instead: > gst-launch-0.10 playbin2 uri=file:///path/to/file.ogg > > Mike > > > On Thu, Dec 2, 2010 at 10:15 PM, rams k wrote: > > Hi, > > > > Can anybody let me know the correct pipeline to play the > ogg(theora+vorbis) > > file. > > > > Im using the following pipeline but it is givng the error. > > > > $gst-launch filesrc location= ! qtdemux name=d d.video_00 ! > queue > > ! theoradec ! xvimagesink d.audio_00 ! queue ! vorbisdec ! audioconvert > ! > > audioresample ! alsasink > > > > (gst-launch-0.10:23114): GLib-WARNING **: g_set_prgname() called multiple > > times > > Setting pipeline to PAUSED ... > > Pipeline is PREROLLING ... > > ERROR: from element /GstPipeline:pipeline0/GstQTDemux:d: This file is > > invalid and cannot be played. > > Additional debug info: > > qtdemux.c(420): gst_qtdemux_pull_atom (): > > /GstPipeline:pipeline0/GstQTDemux:d: > > atom has bogus size 1332176723 > > ERROR: pipeline doesn't want to preroll. > > Setting pipeline to NULL ... > > Freeing pipeline ... > > > > > > rgds, > > KR > > > > > ------------------------------------------------------------------------------ > > What happens now with your Lotus Notes apps - do you make another costly > > upgrade, or settle for being marooned without product support? Time to > move > > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > > use, and manage than apps on traditional platforms. Sign up for the Lotus > > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > ------------------------------------------------------------------------------ > What happens now with your Lotus Notes apps - do you make another costly > upgrade, or settle for being marooned without product support? Time to move > off Lotus Notes and onto the cloud with Force.com, apps are easier to > build, > use, and manage than apps on traditional platforms. Sign up for the Lotus > Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From krkrams27 at gmail.com Mon Dec 20 12:15:11 2010 From: krkrams27 at gmail.com (rk) Date: Mon, 20 Dec 2010 11:15:11 -0000 Subject: [gst-devel] Playbin for multiple video sink support Message-ID: <1292843703954-3095368.post@n4.nabble.com> Hi, I want to add support for multiple video streams (mvc) in the playbin. How to link multiple src pads of the decoder element to the multiple video sinks in the xvimagesink Curretnly Im trying to modiffy the gst-plugins-base-0.10.31/gst/playback/gstplaybin.c file rgds, RK -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Playbin-for-multiple-video-sink-support-tp3095368p3095368.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From sfaizanh at yahoo.com Tue Dec 21 23:57:59 2010 From: sfaizanh at yahoo.com (sfaizanh) Date: Tue, 21 Dec 2010 22:57:59 -0000 Subject: [gst-devel] how to encode to jpeg plz help Message-ID: <1292972273013-3159848.post@n4.nabble.com> hi i have just found out a bug in my program, when i was coding on my pc i got different camera, now when i switched my application to laptop it dosent run because of camera here is my code and gst command line: Command Line: gst-launch v4l2src ! image/jpeg,width=320,height=240,framerate=\(fraction\)20/1 ! ffmpegcolorspace ! tee name=t ! udpsink host=127.0.0.1 port=5000 t. ! queue ! jpegdec ! xvimagesink C code GstCaps *caps; // Outgoing Pipelines videoPipeline = gst_pipeline_new("videoPipeline"); audioPipeline = gst_pipeline_new("audioPipeline"); /* create video elements */ src = gst_element_factory_make("v4l2src", "video-src"); splitter = gst_element_factory_make("tee", "video-splitter"); sink = gst_element_factory_make("xvimagesink", "video-localsink"); videoStream = gst_element_factory_make("udpsink", "video-transmitter"); queue1 = gst_element_factory_make("queue","queue1"); queue2 = gst_element_factory_make("queue","queue2"); videoDec = gst_element_factory_make("jpegdec", "decoder"); videoFilter = gst_element_factory_make ("ffmpegcolorspace", "filter"); if (!videoPipeline) { printf("GStreamer Error: Could not create pipeline\n"); } if(!src || !videoStream || !sink || !videoDec || !videoFilter) { printf("GStreamer Error: Video Elements: Could not create element\n"); } if (!splitter) { printf("GStreamer Error: Splitter Element: Could not create video splitter\n"); } if (!videoStream) { printf("Stream Creation failure.\n"); } g_object_set(G_OBJECT(src),"device","/dev/video0",NULL); caps = gst_caps_new_simple ("image/jpeg", "width", G_TYPE_INT, 320, "height", G_TYPE_INT, 240, "framerate",GST_TYPE_FRACTION,100,1, NULL); // UDP Connection g_object_set(G_OBJECT(videoStream),"host",ipaddr, "port", outputPort, NULL); gst_bin_add_many(GST_BIN(videoPipeline),src, splitter, sink, videoDec, videoFilter, videoStream, queue1, queue2, NULL); if(!gst_element_link_many(src, splitter, NULL)) { printf("Source -> Splitter Link failure\n"); } pad1 = gst_element_get_request_pad(splitter, "src%d"); pad2 = gst_element_get_request_pad(splitter, "src%d"); gst_element_link(splitter, queue1); gst_element_link(splitter, queue2); if(!gst_element_link_filtered (queue1, videoStream, caps)) { printf("Link between queue and stream failed.\n"); } if (!gst_element_link_many(queue2, videoDec, videoFilter, sink, NULL)) { printf("queue -> Local Src link failed\n"); } now i want to encode it to jpep because my camera dosent support image/jpeg but it supports video/x-raw-yuv, so i figured out the command line gst-launch v4l2src ! video/x-raw-yuv,width=320,height=240,framerate=\(fraction\)5/1 ! ffmpegcolorspace ! jpegenc ! tee name=t ! udpsink host=127.0.0.1 port=5000 t. ! queue ! jpegdec ! xvimagesink now can any body explain me how to enccode it :S im stuck in code :S -queue -> updsink Source -> encode.to.jpeg -> filter -> tee -queue -> decode.jpeg > xvimagesink -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/how-to-encode-to-jpeg-plz-help-tp3159848p3159848.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From julien38813 at hotmail.com Wed Dec 22 09:16:10 2010 From: julien38813 at hotmail.com (julien38813) Date: Wed, 22 Dec 2010 08:16:10 -0000 Subject: [gst-devel] decoding H.264 using an hardware codec Message-ID: <1293005763894-3160269.post@n4.nabble.com> Hello, I would like to write a plug-in which decodes an H264 video stream using the hardware codec Intel Graphics Media Accelerator 500 of my Intel processor. So I have found some pay solutions: http://www.fluendo.com fluendo , or http://download.intel.com/design/intarch/PAPERS/324294.pdf Media Infrastructure Accelerator (MI-X) . Does anyone know an free solution, or the appropriate Gstreamer pipeline? Any idea why this might happen? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/decoding-H-264-using-an-hardware-codec-tp3160269p3160269.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From alex at lindusembedded.com Thu Dec 23 15:50:18 2010 From: alex at lindusembedded.com (Alex Gonzalez) Date: Thu, 23 Dec 2010 14:50:18 -0000 Subject: [gst-devel] Camara capture and theora encryption. Message-ID: Hi, I am trying to record from a camera to a theora ogg file. My pipeline looks like: gst-launch --gst-debug=theoraenc:5 v4l2src device=/dev/video0 num-buffers=200 ! 'video/x-raw-yuv,width=640,height=480,framerate=30/1,format=(fourcc)UYVY' ! ffmpegcolorspace ! 'video/x-raw-yuv,width=640,height=480,framerate=30/1,format= (fourcc)I420' ! tee name=t_vid ! theoraenc bitrate=32 ! queue ! oggmux ! filesink location=output.ogg sync=false My camera will only provide UYVY, which then is converted ti I420 which is supposedly accepted by theora, and onto disk. It is also previewed on the display. The error displayed is: 0:00:00.193561889 1021 0x15050 DEBUG theoraenc theoraenc.c:334:gst_theora_enc_init: keyframe_frequency_force is 0, granule shift is 32 0:00:00.338650519 1021 0x15050 WARN theoraenc theoraenc.c:433:theora_enc_sink_getcaps: no supported formats found. Encoder disabled? WARNING: erroneous pipeline: could not link t_vid to theoraenc0 Anything wrong with the capsfilter that upsets theora? Any other idea? Thanks, Alex PS: From gst-inspect theoraenc, Pad Templates: SINK template: 'sink' Availability: Always Capabilities: video/x-raw-yuv format: { I420, Y42B, Y444 } framerate: [ 0/1, 2147483647/1 ] width: [ 1, 2147483647 ] height: [ 1, 2147483647 ] SRC template: 'src' Availability: Always Capabilities: video/x-theora From umakantgoyal1 at gmail.com Sat Dec 25 13:44:16 2010 From: umakantgoyal1 at gmail.com (qwerty-1) Date: Sat, 25 Dec 2010 12:44:16 -0000 Subject: [gst-devel] Pipelline Crash when it recieves comfort noise packets Message-ID: <1293281049784-3163733.post@n4.nabble.com> Hi All, I am making pipeline using gstrtpbin, rtppcmudepay, udpsrc and fakesink elements to receive RTP audio stream from network. Everything goes fine when i receive audio(pcmu) rtp packets with payload type 0 and encdoing-name PCMU. But my pipeline breaks when i receive CN (comfort noise) rtp packets with payload type 13 and dtmf packets with payload type 101. I have tried lot to ignore these packets but i did not succeed to find out the solution of my problem. Please help me by telling the possible solutions of my problem. I am also wondering why udpsrc element is receiving rtp messages with payload 101 and 13 because i have set the caps for udpsrc as given below: media --> audio, payload --> 0, clock-rate --> 8000 encoding-name --> PCMU, As per my understanding if i would set the above given caps to udpsrc then it should ignore other rtp packets except with payload 0 and encoding-name PCMU. Thanks -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Pipelline-Crash-when-it-recieves-comfort-noise-packets-tp3163733p3163733.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From kaija at ms2.hinet.net Sat Dec 25 15:35:15 2010 From: kaija at ms2.hinet.net (kaija) Date: Sat, 25 Dec 2010 14:35:15 -0000 Subject: [gst-devel] dynamically remove custom branch from tee and videomixer Message-ID: <1293287707152-3163782.post@n4.nabble.com> I try to create a picture in picture channel to the main pipeline. I read this post and succeeded to create PIP channel dynamically. (http://gstreamer-devel.966125.n4.nabble.com/Dynamically-adding-and-removing-branches-of-a-tee-td973635.html#a973637) However, while I attempt to release the PIP channel while the main pipeline is still running, the main pipeline stops to refresh video frames. I cannot figure out the reason why main pipeline stops. Could anyone help me to solve the problem? Here is my pipeline looks like: / (main pipeline)queue - videomixer - videoscale - videocrop - caps - videosink tee / \ (pip branch ) queue - videoscale - videocrop - caps ------ mix -----/ the following code shows how I unlink pip branch from tee and videomixer void CPiPChannel::Unlink(void) { gst_pad_set_blocked_async(m_pVideoTeeSrcPad, TRUE, (GstPadBlockCallback) PadBlockHandler, m_pVideoTee); gst_element_remove_pad(m_pVideoMixer, m_pVideoMixerSinkPad); } void CPiPChannel::PadBlockHandler(GstPad* pPad, gboolean bBlocked, gpointer pData) { GstPad* pPeer = NULL; GstEvent* pEvent = NULL; GstElement* pTee = NULL; if(bBlocked) { pPeer = gst_pad_get_peer(pPad); pEvent = gst_event_new_eos(); pTee = (CPiPChannel*) pData; if(gst_pad_unlink(pPad, pPeer); dprintf("unlink pad success"); else dprintf("unlink pad failed"); gst_element_set_state(pBin, GST_STATE_READY); gst_element_set_state(pBin, GST_STATE_NULL); gst_pad_push_event(pPeer, pEvent); gst_pad_set_blocked_async(pPad, FALSE, (GstPadBlockCallback) PadBlockHandler, pData); gst_element_remove_pad(pTee, pPad); gst_object_unref(pPad); } else { dprintf("unblocked"); } } -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/dynamically-remove-custom-branch-from-tee-and-videomixer-tp3163782p3163782.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From umakantgoyal1 at gmail.com Sat Dec 25 22:26:44 2010 From: umakantgoyal1 at gmail.com (qwerty-1) Date: Sat, 25 Dec 2010 21:26:44 -0000 Subject: [gst-devel] Single Pipelne for rtp audio and dtmf (rfc 2833) Message-ID: <1293312398219-3163961.post@n4.nabble.com> Hi All, How can i receive rtp audio and dtmf (rfc 2833) using single pipeline? how can i force he platform to receive audio and dtmf both by setting the caps of udpsrc? Thanks. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Single-Pipelne-for-rtp-audio-and-dtmf-rfc-2833-tp3163961p3163961.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From emil at stinetworks.eu Mon Dec 27 16:15:29 2010 From: emil at stinetworks.eu (stojnovsky) Date: Mon, 27 Dec 2010 15:15:29 -0000 Subject: [gst-devel] matroska Message-ID: <1293462920665-3165093.post@n4.nabble.com> Hi All, I'm really newer with Gstreamer. I start to make one simple test, play webm movie. When I try to play only video or audio it works, but not work audio+video. when I start: gst-launch-0.10 -v filesrc location=bilqna.webm ! matroskademux name=d d.audio_00 ! vorbisdec ! audioconvert ! pulsesink d.video_00 ! vp8dec ! xvimagesink I see maybe first frame of video and the picture is stay freeze and audio isn't start and in console i get: ERROR: pipeline doesn't want to preroll. when i start: gst-launch-0.10 -v filesrc location=bilqna.webm ! matroskademux name=d d.audio_00 ! vorbisdec ! audioconvert ! pulsesink filesrc location=bilqna.webm ! matroskademux ! vp8dec ! xvimagesink I have video and audio but i think this is not a good solution. So, can you help me, what is wrong in my first command, or maybe this is a bug, or i miss do something when install my environment ? Thanks in advance. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/matroska-tp3165093p3165093.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From krkrams27 at gmail.com Tue Dec 28 06:51:10 2010 From: krkrams27 at gmail.com (rams k) Date: Tue, 28 Dec 2010 11:21:10 +0530 Subject: [gst-devel] how to modify playbin Message-ID: Hi, I want to add support for playing multiple video streams (h264 mvc file) in the playbin by creating multiple sinks. How to link multiple src pads of the decoder element to the multiple video sinks in the xvimagesink of playbin Curretnly Im trying to modiffy the gst-plugins-base-0.10.31/gst/playback/gstplaybin.c file and want to use the playbin for playing multiple videos MVC file rgds, RK -------------- next part -------------- An HTML attachment was scrubbed... URL: From krkrams27 at gmail.com Thu Dec 30 15:43:39 2010 From: krkrams27 at gmail.com (rk) Date: Thu, 30 Dec 2010 06:43:39 -0800 (PST) Subject: [gst-devel] playbin multiple sink Message-ID: Hi, Im trying to play MVC file through PLAYBIN with the mvc plugin. What changes I need to do to auto-support h264 MVC file suppport in playbin. I would like to know regarding adding multiple video sinks in the playbin. rgds, RK -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/playbin-multiple-sink-tp3168335p3168335.html Sent from the GStreamer-devel mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: