From gang.a.hu at intel.com Sat Jan 1 03:52:39 2011 From: gang.a.hu at intel.com (Hu, Gang A) Date: Sat, 1 Jan 2011 10:52:39 +0800 Subject: [gst-devel] How to add a complicate property to an element Message-ID: <1A42CE6F5F474C41B63392A5F80372B231D2AEB2@shsmsx501.ccr.corp.intel.com> Hi, I want to add a property to a camera source element. This property contains five field. How can create this property with one property id and extract it? For example, to add the ae_windows property to the v4l2src, the user can use the command gst-launch v4l2src ae_windows="width=49, height=60, centerx=20, centery=30, weight=3" ! xvimagesink. to specify the ae_windows. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From 21stninjagod at gmail.com Sat Jan 1 06:25:58 2011 From: 21stninjagod at gmail.com (gst_call_me_newbie()) Date: Fri, 31 Dec 2010 21:25:58 -0800 (PST) Subject: [gst-devel] From where can I download the plugins for webmmux and vp8enc Message-ID: <1293859558206-3169973.post@n4.nabble.com> I searched over the Internet & I was not able to find the links for downloading the "webmmux" and "vp8enc" plugins. Can anyone post the links. Thanks -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/From-where-can-I-download-the-plugins-for-webmmux-and-vp8enc-tp3169973p3169973.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From 21stninjagod at gmail.com Sun Jan 2 07:43:18 2011 From: 21stninjagod at gmail.com (gst_call_me_newbie()) Date: Sat, 1 Jan 2011 22:43:18 -0800 (PST) Subject: [gst-devel] Why do we call gst_object_unref() ? Message-ID: <1293950598145-3170699.post@n4.nabble.com> hi all, I checked the meaning of the function gst_object_unref() and got what it does but I could not get why do we call it while the program is in execution. for e.g this is a code snippet from the application developer manual provided at the gstreamer's site. [some code here.....] bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); gst_bus_add_watch (bus, my_bus_callback, loop); gst_object_unref (bus); [some other code here.....] so why would someone want to unref the bus when we still need it in our program. If I unref it and if somehow that would have been the last reference to the object than object would surely be destroyed. so why should I call it. Please enlighten me. Thanks -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Why-do-we-call-gst-object-unref-tp3170699p3170699.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From mgruenke at Tycoint.com Sun Jan 2 08:43:46 2011 From: mgruenke at Tycoint.com (Gruenke, Matt) Date: Sun, 2 Jan 2011 02:43:46 -0500 Subject: [gst-devel] Why do we call gst_object_unref() ? References: <1293950598145-3170699.post@n4.nabble.com> Message-ID: It's used to perform reference counting. Each use or user of an object should be counted. When that count goes to 0, then it means the object is no longer needed and the resources it occupies can be reclaimed. If you see an object being unref'd that should continue to exist, then you should check up on who is resposible for keeping a count on it, after the unref in question. The reason this is necessary during program execution is that numerous short-lived objects are created. If these are never deallocated, then a program's memory usage would continually grow. Ref-counting is sometimes described as a lesser form of garbage collection, and is commonly used in cases where there's not necessarilly a clear relationship between consumers and producers of objects. Be warned that ref counting fails in the case of circular references (weak refs can sometimes be used to avoid these scenarios). BTW, this is complicated by the concept of "parenting", where exclusive ownership of an object is asserted. I encourage you to look for details about all of this in the GLib docs, as I'm sure it must be described in there at length. Matt ________________________________ From: gst_call_me_newbie() [mailto:21stninjagod at gmail.com] Sent: Sun 1/2/2011 1:43 AM To: gstreamer-devel at lists.sourceforge.net Subject: [gst-devel] Why do we call gst_object_unref() ? hi all, I checked the meaning of the function gst_object_unref() and got what it does but I could not get why do we call it while the program is in execution. for e.g this is a code snippet from the application developer manual provided at the gstreamer's site. [some code here.....] bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); gst_bus_add_watch (bus, my_bus_callback, loop); gst_object_unref (bus); [some other code here.....] so why would someone want to unref the bus when we still need it in our program. If I unref it and if somehow that would have been the last reference to the object than object would surely be destroyed. so why should I call it. Please enlighten me. Thanks -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Why-do-we-call-gst-object-unref-tp3170699p3170699.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 t.i.m at zen.co.uk Sun Jan 2 12:54:14 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sun, 02 Jan 2011 11:54:14 +0000 Subject: [gst-devel] MXF-wrapped JPEG2000 decoding In-Reply-To: <4D1CAF84.7070809@ymagis.com> References: <4D1CAF84.7070809@ymagis.com> Message-ID: <1293969254.8553.8.camel@zingle> On Thu, 2010-12-30 at 17:12 +0100, Florian Agez wrote: > 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 First thing to try in such cases is: does it work with playbin? gst-launch-0.10 -v playbin2 uri=file:///path/to/foo.mxf If not, the output of that might already tell us something. > 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. You might need an ffmpegcolorspace ! videoscale in front of the autovideosink as well. Cheers -Tim From t.i.m at zen.co.uk Sun Jan 2 13:09:42 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sun, 02 Jan 2011 12:09:42 +0000 Subject: [gst-devel] From where can I download the plugins for webmmux and vp8enc In-Reply-To: <1293859558206-3169973.post@n4.nabble.com> References: <1293859558206-3169973.post@n4.nabble.com> Message-ID: <1293970182.8553.13.camel@zingle> On Fri, 2010-12-31 at 21:25 -0800, gst_call_me_newbie() wrote: > I searched over the Internet & I was not able to find the links for > downloading the "webmmux" and "vp8enc" plugins. Typing 'webmmux gstreamer' and 'vp8enc gstreamer' into your favourite search engine should point you towards gst-plugins-good and gst-plugins-bad. Since you did not mention what distro/OS you're running, we can't tell you the exact package names. > Can anyone post the links. Thanks Install the gstreamer -good and -bad plugins packages provided by your distro. Cheers -Tim From t.i.m at zen.co.uk Sun Jan 2 13:17:59 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sun, 02 Jan 2011 12:17:59 +0000 Subject: [gst-devel] Multiple Caps for udpsrc In-Reply-To: References: Message-ID: <1293970679.8553.21.camel@zingle> On Sat, 2011-01-01 at 02:48 +0530, Umakant Goyal wrote: > > 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. You can do this when setting caps on a capsfilter element ("foo/bar,x=5;mon/key,y=5"), but in this case, receiving rtp packets, setting the output caps of a source element, you don't want to restrict the caps allowed, but set them. And when you set caps, they need to be simple fixed caps. Incoming RTP packets are either audio or video or something else, but not both audio and video, so I don't quite see how what you ask makes sense. Maybe you need to provide more context.. Cheers -Tim From h at realh.co.uk Sun Jan 2 14:34:32 2011 From: h at realh.co.uk (Tony Houghton) Date: Sun, 02 Jan 2011 13:34:32 +0000 Subject: [gst-devel] Why do we call gst_object_unref() ? In-Reply-To: <1293950598145-3170699.post@n4.nabble.com> References: <1293950598145-3170699.post@n4.nabble.com> Message-ID: <4D207EE8.5000008@realh.co.uk> On 02/01/11 06:43, gst_call_me_newbie() wrote: > I checked the meaning of the function gst_object_unref() and got what it > does but I could not get why do we call it while the program is in > execution. > > for e.g this is a code snippet from the application developer manual > provided at the gstreamer's site. > > [some code here.....] > bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); > gst_bus_add_watch (bus, my_bus_callback, loop); > gst_object_unref (bus); > [some other code here.....] > > so why would someone want to unref the bus when we still need it in our > program. If I unref it and if somehow that would have been the last > reference to the object than object would surely be destroyed. > so why should I call it. Please enlighten me. Presumably the get functions such as gst_pipeline_get_bus() automatically add a reference to each object they return, so when the object is no longer needed in the context that requested it you have to unref it. From ylatuya at gmail.com Sun Jan 2 22:54:48 2011 From: ylatuya at gmail.com (Andoni Morales) Date: Sun, 2 Jan 2011 22:54:48 +0100 Subject: [gst-devel] Implementing a MPEGTS segmenter, what's the best approach? In-Reply-To: References: <20101231072336.GA20495@cooker.entropywave.com> Message-ID: 2010/12/31 David Zhao : > 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 That's probably the quickest way to produce segments for testing purpose but these fragments wouldn't follow the spec's recommendations[1]. " Transport Stream files MUST contain a single MPEG-2 Program. There SHOULD be a Program Association Table and a Program Map Table at the start of each file. A file that contains video SHOULD have at least one key frame and enough information to completely initialize a video decoder." A client can join the stream at any moment (downloading any of the available segments), so each segment must be independently decodable, which means you need to start each fragment with a PAT table, followed with a PMT table and starting with a keyframe. Andoni [1]http://tools.ietf.org/html/draft-pantos-http-live-streaming-05#page-4 >> >> >> >> 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 > > > ------------------------------------------------------------------------------ > 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 > > -- Andoni Morales Alastruey LongoMatch:The Digital Coach http://www.longomatch.ylatuya.es From david at davidzhao.com Mon Jan 3 01:27:25 2011 From: david at davidzhao.com (David Zhao) Date: Sun, 2 Jan 2011 16:27:25 -0800 Subject: [gst-devel] Implementing a MPEGTS segmenter, what's the best approach? In-Reply-To: References: <20101231072336.GA20495@cooker.entropywave.com> Message-ID: On Sun, Jan 2, 2011 at 1:54 PM, Andoni Morales wrote: > 2010/12/31 David Zhao : > > 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 > > That's probably the quickest way to produce segments for testing > purpose but these fragments wouldn't follow the spec's > recommendations[1]. > > " Transport Stream files MUST contain a single MPEG-2 Program. There > SHOULD be a Program Association Table and a Program Map Table at the > start of each file. A file that contains video SHOULD have at least > one key frame and enough information to completely initialize a video > decoder." > > A client can join the stream at any moment (downloading any of the > available segments), so each segment must be independently decodable, > which means you need to start each fragment with a PAT table, followed > with a PMT table and starting with a keyframe. > Thanks for the clarification here. Looks like the "correct" way to implement this is still to create a special purposed mpegts muxer that will write the PAT/PMT tables at the beginning of each TS segment. I noticed that the protocol says "SHOULD" instead of "MUST", perhaps the current implementation of iOS treat those tables as optional? > > Andoni > > [1]http://tools.ietf.org/html/draft-pantos-http-live-streaming-05#page-4 > >> > >> > >> > >> 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 > > > > > > > ------------------------------------------------------------------------------ > > 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 > > > > > > > > -- > Andoni Morales Alastruey > > LongoMatch:The Digital Coach > http://www.longomatch.ylatuya.es > > > ------------------------------------------------------------------------------ > 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 Mon Jan 3 04:28:57 2011 From: ds at entropywave.com (David Schleef) Date: Sun, 2 Jan 2011 22:28:57 -0500 Subject: [gst-devel] Implementing a MPEGTS segmenter, what's the best approach? In-Reply-To: References: <20101231072336.GA20495@cooker.entropywave.com> Message-ID: <20110103032857.GA5928@cooker.entropywave.com> On Sun, Jan 02, 2011 at 04:27:25PM -0800, David Zhao wrote: > On Sun, Jan 2, 2011 at 1:54 PM, Andoni Morales wrote: > > > 2010/12/31 David Zhao : > > > Thanks David! That's exactly what I was looking for. > > > > > > On Thu, Dec 30, 2010 at 11:23 PM, David Schleef > > >> gst-launch ... ! mpegtsmux ! multifilesink next-file=key-frame \ > > >> location=%05d.ts > > > > That's probably the quickest way to produce segments for testing > > purpose but these fragments wouldn't follow the spec's > > recommendations[1]. > > > > " Transport Stream files MUST contain a single MPEG-2 Program. There > > SHOULD be a Program Association Table and a Program Map Table at the > > start of each file. A file that contains video SHOULD have at least > > one key frame and enough information to completely initialize a video > > decoder." > > > > A client can join the stream at any moment (downloading any of the > > available segments), so each segment must be independently decodable, > > which means you need to start each fragment with a PAT table, followed > > with a PMT table and starting with a keyframe. > > > > Thanks for the clarification here. Looks like the "correct" way to implement > this is still to create a special purposed mpegts muxer that will write the > PAT/PMT tables at the beginning of each TS segment. > > I noticed that the protocol says "SHOULD" instead of "MUST", perhaps the > current implementation of iOS treat those tables as optional? It helps to have this commit: commit 6ab508dfa56f92c86c3b379b6cfe4ff28f2551e4 Author: David Schleef Date: Sun Jan 2 19:19:27 2011 -0800 multifilesink: send stream headers in key-frame mode However, the problems with the above pipeline are greatly exaggerated. In the worst case (without the patch), clients will start the video one keyframe late. The *actual* problems with using mpegtsmux ! multifilesink are that it doesn't write out the .m3u8 file for you, and that you have little control over keyframe placement. David From cmaster2 at hotmail.com Mon Jan 3 05:27:33 2011 From: cmaster2 at hotmail.com (lithium) Date: Sun, 2 Jan 2011 20:27:33 -0800 (PST) Subject: [gst-devel] Video freeze Message-ID: <1294028853954-3171570.post@n4.nabble.com> I've been writing a tv capture program. Basically by the end of it i'm hoping to have the gstreamer stuff setup somewhat like this: /* /----queue----xvimagesink //screenoutput v4l2src---------tee----- \----queue----filesink //screenshot \----queue----\ ----multiplexer ---filesink //recording /----queue----/ alsasrc----------tee----- \----queue----alsasink //playback from capture if needed */ At the moment i've implemented the top video branch and am now working on the screenshot one. I've been using a combination of gtk and gstreamer to write the program. When the program starts I call the 'openVideoPipeline' function (see below) and on clicking the screenshot button in my gtk window it runs the 'take_screenshot' function. The problem I am having is that when I press the screenshot button it causes the screen output to freeze. The file is not created by the filesink. I have found that when i click my button that causes the tv card options menu to be created and drawn it causes the screen output to resume playing and the file to be written. The only functions used from gstreamer in the tv card menu are 'gst_tuner_list_channels', 'gst_tuner_list_norms', and 'gst_tuner_get_norm'. It is very likely I am going about taking screenshots the wrong way, if someone has a better idea please suggest it. Otherwise it would be good if someone has any idea whats causing the video to freeze. I have tried setting the pipeline to playing again after unlinking the screenshot pads but it didn't change anything. int openVideoPipeline(videoIOStruct *videoIO) { //create elements videoIO->pipeline = gst_pipeline_new("tvVideo"); videoIO->vidSrc = gst_element_factory_make("v4l2src", "source"); GstElement *vidSink = gst_element_factory_make("xvimagesink", "sink"); videoIO->tee = gst_element_factory_make("tee", "splitter"); GstElement *vidQueue = gst_element_factory_make("queue", NULL); if (!videoIO->pipeline || !videoIO->vidSrc || !vidSink || !videoIO->tee || !vidQueue) return 0; //set arguments g_object_set(videoIO->vidSrc, "device", videoIO->devName, NULL); gst_bin_add_many(GST_BIN(videoIO->pipeline), videoIO->vidSrc, videoIO->tee, vidQueue, vidSink, NULL); videoIO->videoScreenPad = gst_element_get_request_pad(videoIO->tee, "src%d"); GstPad *staticOutputPad = gst_element_get_static_pad(vidQueue, "sink"); //link stuff gst_pad_link(videoIO->videoScreenPad, staticOutputPad); gst_object_unref(GST_OBJECT(videoIO->videoScreenPad)); gst_element_link(videoIO->vidSrc, videoIO->tee); gst_element_link(vidQueue, vidSink); if (!videoIO->pipeline) return 0; return 1; } void close_screenshot(GstElement *jpegEncode, videoIOStruct *videoIO) { GstPad *pad1 = GST_PAD(g_object_get_data(G_OBJECT(jpegEncode), "videoScreenshotPad")); GstPad *pad2 = GST_PAD(g_object_get_data(G_OBJECT(jpegEncode), "staticOutputPad")); gst_pad_unlink(pad1, pad2); } void take_screenshot(GtkWidget *widget, videoIOStruct *videoIO) { GstElement *jpegOutput = gst_element_factory_make("filesink", "jpegoutput"); GstElement *jpegQueue = gst_element_factory_make("queue", "jpgqueue"); GstElement *jpegEncode = gst_element_factory_make("jpegenc", "jpegencoder"); if (!jpegOutput || !jpegQueue || !jpegEncode || !videoIO->jpegPipeline) return; //set arguments g_object_set(jpegOutput, "location", "test.jpg", NULL); //link stuff gst_bin_add_many(GST_BIN(videoIO->pipeline), jpegEncode, jpegQueue, jpegOutput, NULL); gst_element_link_many(jpegQueue, jpegEncode, jpegOutput, NULL); //gst_element_set_state(videoIO->jpegPipeline, GST_STATE_PLAYING); g_signal_connect(jpegEncode, "frame-encoded", G_CALLBACK(close_screenshot), videoIO); GstPad *videoScreenshotPad = gst_element_get_request_pad(videoIO->tee, "src%d"); GstPad *staticOutputPad = gst_element_get_static_pad(jpegQueue, "sink"); g_object_set_data(G_OBJECT(jpegEncode), "videoScreenshotPad", videoScreenshotPad); g_object_set_data(G_OBJECT(jpegEncode), "staticOutputPad", staticOutputPad); gst_pad_link(videoScreenshotPad, staticOutputPad); } -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Video-freeze-tp3171570p3171570.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From david at davidzhao.com Mon Jan 3 07:07:58 2011 From: david at davidzhao.com (David Zhao) Date: Sun, 2 Jan 2011 22:07:58 -0800 Subject: [gst-devel] Implementing a MPEGTS segmenter, what's the best approach? In-Reply-To: <20110103032857.GA5928@cooker.entropywave.com> References: <20101231072336.GA20495@cooker.entropywave.com> <20110103032857.GA5928@cooker.entropywave.com> Message-ID: On Sun, Jan 2, 2011 at 7:28 PM, David Schleef wrote: > On Sun, Jan 02, 2011 at 04:27:25PM -0800, David Zhao wrote: > > On Sun, Jan 2, 2011 at 1:54 PM, Andoni Morales > wrote: > > > > > 2010/12/31 David Zhao : > > > > Thanks David! That's exactly what I was looking for. > > > > > > > > On Thu, Dec 30, 2010 at 11:23 PM, David Schleef > > > >> gst-launch ... ! mpegtsmux ! multifilesink next-file=key-frame \ > > > >> location=%05d.ts > > > > > > That's probably the quickest way to produce segments for testing > > > purpose but these fragments wouldn't follow the spec's > > > recommendations[1]. > > > > > > " Transport Stream files MUST contain a single MPEG-2 Program. There > > > SHOULD be a Program Association Table and a Program Map Table at the > > > start of each file. A file that contains video SHOULD have at least > > > one key frame and enough information to completely initialize a video > > > decoder." > > > > > > A client can join the stream at any moment (downloading any of the > > > available segments), so each segment must be independently decodable, > > > which means you need to start each fragment with a PAT table, followed > > > with a PMT table and starting with a keyframe. > > > > > > > Thanks for the clarification here. Looks like the "correct" way to > implement > > this is still to create a special purposed mpegts muxer that will write > the > > PAT/PMT tables at the beginning of each TS segment. > > > > I noticed that the protocol says "SHOULD" instead of "MUST", perhaps the > > current implementation of iOS treat those tables as optional? > > > It helps to have this commit: > > commit 6ab508dfa56f92c86c3b379b6cfe4ff28f2551e4 > Author: David Schleef > Date: Sun Jan 2 19:19:27 2011 -0800 > > multifilesink: send stream headers in key-frame mode > > However, the problems with the above pipeline are greatly exaggerated. > In the worst case (without the patch), clients will start the video > one keyframe late. > > The *actual* problems with using mpegtsmux ! multifilesink are that it > doesn't write out the .m3u8 file for you, and that you have little > control over keyframe placement. > > > David > > I've tested ts files generated by multifilesink (even before this commit) and it seems to work fine when it's not switching streams. When it does switch between variants, this commit may become necessary. I'll give this a shot, will report back with results. 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From suneets at gmail.com Mon Jan 3 09:35:06 2011 From: suneets at gmail.com (Suneet Shah) Date: Mon, 3 Jan 2011 00:35:06 -0800 Subject: [gst-devel] Achieving <400ms latency with x264enc Message-ID: Hello, I'm wondering if it is possible to achieve <400ms latency with streaming video using x264enc. I'm using gstreamer to stream video (server) to VLC (client) on another machine within the same local network. My pipeline is as follows (on the server machine): ximagesrc ! video/x-raw-rgb,framerate=10/1 ! videorate ! ffmpegcolorspace ! videoscale method=1 ! video/x-raw-yuv,width=480,height=300,framerate=5/1 ! x264enc byte-stream=true bitrate=195 tune=0x00000004 ! mpegtsmux ! rtpmp2tpay ! udpsink port=1500 host=192.168.2.197 Note that I'm using the tune=zerolatency (which sets several values, according to http://mewiki.project357.com/wiki/X264_Encoding_Suggestions) to achieve low latency. My client machine is simply running VLC with rtp://:1500 With this setting-- I'm seeing about 2.5 seconds of latency from an action taken on the server to being seen in the video stream by the client. If you've achieved low latency with x264enc, can you share your settings? Thanks! Suneet -------------- next part -------------- An HTML attachment was scrubbed... URL: From timothejahan at gmail.com Mon Jan 3 10:28:14 2011 From: timothejahan at gmail.com (timothe jahan) Date: Mon, 3 Jan 2011 10:28:14 +0100 Subject: [gst-devel] Achieving <400ms latency with x264enc In-Reply-To: References: Message-ID: Hi, I am not a gstreamer expert at all but I am working quite a lot on live streaming. I would advice to check the buffer settings of VLC. Usually the client always have some buffer to hide network ups and downs. A few seconds is common. Regards, Timothe On Mon, Jan 3, 2011 at 9:35 AM, Suneet Shah wrote: > Hello, > > I'm wondering if it is possible to achieve <400ms latency with streaming > video using x264enc. I'm using gstreamer to stream video (server) to VLC > (client) on another machine within the same local network. > > My pipeline is as follows (on the server machine): > > ximagesrc ! video/x-raw-rgb,framerate=10/1 ! videorate ! ffmpegcolorspace ! > videoscale method=1 ! video/x-raw-yuv,width=480,height=300,framerate=5/1 ! > x264enc byte-stream=true bitrate=195 tune=0x00000004 ! mpegtsmux ! > rtpmp2tpay ! udpsink port=1500 host=192.168.2.197 > > Note that I'm using the tune=zerolatency (which sets several values, > according to http://mewiki.project357.com/wiki/X264_Encoding_Suggestions) > to achieve low latency. > > My client machine is simply running VLC with rtp://:1500 > > With this setting-- I'm seeing about 2.5 seconds of latency from an action > taken on the server to being seen in the video stream by the client. > > If you've achieved low latency with x264enc, can you share your settings? > > Thanks! > Suneet > > > ------------------------------------------------------------------------------ > 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 grodriguez at ingelabs.com Mon Jan 3 11:38:09 2011 From: grodriguez at ingelabs.com (Guillermo Rodriguez Garcia) Date: Mon, 03 Jan 2011 11:38:09 +0100 Subject: [gst-devel] HTTP video source serves individual JPEG images Message-ID: <4D21A711.1060504@ingelabs.com> Hello all, I am trying to use gstreamer to stream video (sort of) from a HTTP source that serves individual JPEG images. This is not a MJPEG stream -- instead, a new HTTP request is required for each frame that is to be retrieved. This pipeline works for MJPEG: gst-launch souphttpsrc location=http://user:pass at hostname/path ! jpegdec ! xvimagesink If I try to use the same pipeline for an HTTP source that serves individual JPEG images, only one image is shown (since the HTTP response actually contains just one single image). Is there any way to have souphttpsrc make additional requests periodically from the specified location? Is there any other way to achieve this? Thanks, GRodriguez From janne.grunau at collabora.co.uk Mon Jan 3 12:11:10 2011 From: janne.grunau at collabora.co.uk (Janne Grunau) Date: Mon, 03 Jan 2011 12:11:10 +0100 Subject: [gst-devel] MpegTS playback bug. In-Reply-To: References: Message-ID: <4D21AECE.5040907@collabora.co.uk> On 29.12.2010 20:28, 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? Reproduced, probably with a very different FFmpeg version. Gstreamer feeds the mpeg4 decoder (ffmpeg's in my) data it doesn't understand. I see lot's of decoding errors. The same FFmpeg command with mpeg2video as -vcodec works as expected. It looks like GStreamer and FFmpeg disagree about the bitstream format. Janne From vaisaari at gmail.com Mon Jan 3 12:12:39 2011 From: vaisaari at gmail.com (vaisaari) Date: Mon, 3 Jan 2011 03:12:39 -0800 (PST) Subject: [gst-devel] Getting converted frame from playbin2 element Message-ID: <1294053159060-3171830.post@n4.nabble.com> Hi all, I'm trying to implement a function that returns the current frame of playbin2 element as a QImage (Qt data type). When I call the function (for the same video), sometimes it works perfectly, sometimes parts of the frame (bottom rows) are missing. What am I doing wrong? Should I somehow be able to wait for the "convert-frame" operation to finish, or what? You can find my example code below. Thank you for your help! QImage Video::getFrame() { GstBuffer *buffer; GstCaps *caps; caps = gst_caps_new_simple("video/x-raw-rgb","bpp", G_TYPE_INT, 24,"depth", G_TYPE_INT, 24,NULL); g_signal_emit_by_name(player, "convert-frame", caps, &buffer); unsigned char* data=(unsigned char *) GST_BUFFER_DATA (buffer); QImage img(data, width, height, QImage::Format_RGB888); return img; } -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Getting-converted-frame-from-playbin2-element-tp3171830p3171830.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From florian.agez at ymagis.com Mon Jan 3 15:50:58 2011 From: florian.agez at ymagis.com (Florian Agez) Date: Mon, 03 Jan 2011 15:50:58 +0100 Subject: [gst-devel] MXF-wrapped JPEG2000 decoding In-Reply-To: <1293969254.8553.8.camel@zingle> References: <4D1CAF84.7070809@ymagis.com> <1293969254.8553.8.camel@zingle> Message-ID: <4D21E252.8060903@ymagis.com> Le 02/01/2011 12:54, Tim-Philipp M?ller a ?crit : > First thing to try in such cases is: does it work with playbin? > gst-launch-0.10 -v playbin2 uri=file:///path/to/foo.mxf > > If not, the output of that might already tell us something. > > You might need an ffmpegcolorspace ! videoscale in front of the > autovideosink as well. > > Cheers > -Tim I tried these, they fail with the exact same error messages (full output log for the "playbin" command line is attached). I also tried running the command line with GST_DEBUG=2 as advised by Edward, the output of that didn't look really helpful to me either: 0:00:00.114181342 2907 0x98f1f00 WARN mxf mxfmetadata.c:482:mxf_metadata_new: No handler for type 0x015a found -- using generic metadata parser 0:00:00.114224084 2907 0x98f1f00 WARN mxfdemux mxfdemux.c:1244:gst_mxf_demux_handle_metadata: Unknown or unhandled metadata of type 0x015a 0:00:01.249151118 2907 0x98f1f00 WARN mxfdemux mxfdemux.c:2797:gst_mxf_demux_loop: error: Internal data stream error. 0:00:01.249210483 2907 0x98f1f00 WARN mxfdemux mxfdemux.c:2797:gst_mxf_demux_loop: error: stream stopped, reason not-negotiated So I'm going to do as Edward said and file a bug report for this. Thank you both for your help! -- Florian -------------- next part -------------- A non-text attachment was scrubbed... Name: playbin.log Type: text/x-log Size: 4128 bytes Desc: not available URL: From t.i.m at zen.co.uk Mon Jan 3 16:07:30 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 03 Jan 2011 15:07:30 +0000 Subject: [gst-devel] MXF-wrapped JPEG2000 decoding In-Reply-To: <4D21E252.8060903@ymagis.com> References: <4D1CAF84.7070809@ymagis.com> <1293969254.8553.8.camel@zingle> <4D21E252.8060903@ymagis.com> Message-ID: <1294067251.3220.0.camel@zingle> On Mon, 2011-01-03 at 15:50 +0100, Florian Agez wrote: > mxfdemux.c:2797:gst_mxf_demux_loop: error: stream stopped, reason > not-negotiated > > So I'm going to do as Edward said and file a bug report for this. Please do. If you could make the file (or another sample file that triggers this) available, that'd be very helpful, thanks! Cheers -Tim From wmiller at sdr.com Mon Jan 3 17:48:08 2011 From: wmiller at sdr.com (Wes Miller) Date: Mon, 3 Jan 2011 08:48:08 -0800 (PST) Subject: [gst-devel] Extracting properties from element causes hang In-Reply-To: References: <1293726221815-3168471.post@n4.nabble.com> Message-ID: <1294073288460-3172213.post@n4.nabble.com> Rohit, I tried your suggestion and found that it has exactly the same problem and a couple of others. The value I'm after is in gstrtpbin: latency : Default amount of ms to buffer in the jitterbuffers flags: readable, writable Unsigned Integer. Range: 0 - 4294967295 Default: 200 Current: 200 I tried skipping over "latency" and hung on "do-lost": do-lost : Send an event downstream when a packet is lost flags: readable, writable Boolean. Default: false Current: false Which, at least, confirms its not related to the datatype of the queried value. Other suggestions happily accepted! Wes p.s. the "other" problems using g_object_get() are that it returns different datatypes for the values than does g_object_get_property(). For example, the "name" property is returned as a gchararray not as the expected G_TYPE_STRING. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Extracting-properties-from-element-causes-hang-tp3168471p3172213.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bilboed at gmail.com Mon Jan 3 18:35:17 2011 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 03 Jan 2011 18:35:17 +0100 Subject: [gst-devel] Extracting properties from element causes hang In-Reply-To: <1294073288460-3172213.post@n4.nabble.com> References: <1293726221815-3168471.post@n4.nabble.com> <1294073288460-3172213.post@n4.nabble.com> Message-ID: <1294076117.23475.4.camel@localhost> On Mon, 2011-01-03 at 08:48 -0800, Wes Miller wrote: gstrtpbin holds an internal lock when adding those pads. That same lock is used to query those properties. > > Wes > > p.s. the "other" problems using g_object_get() are that it returns different > datatypes for the values than does g_object_get_property(). For example, > the "name" property is returned as a gchararray not as the expected > G_TYPE_STRING. And what would the difference between a gchar array (gchar*) and a G_TYPE_STRING (gchar*) be ? > From t.i.m at zen.co.uk Mon Jan 3 18:38:02 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 03 Jan 2011 17:38:02 +0000 Subject: [gst-devel] Extracting properties from element causes hang In-Reply-To: <1294073288460-3172213.post@n4.nabble.com> References: <1293726221815-3168471.post@n4.nabble.com> <1294073288460-3172213.post@n4.nabble.com> Message-ID: <1294076282.3220.8.camel@zingle> On Mon, 2011-01-03 at 08:48 -0800, Wes Miller wrote: > Which, at least, confirms its not related to the datatype of the queried > value. It won't depend on the data type, but on the locks taken by the implementation. > Other suggestions happily accepted! Run your code in gdb. When it hangs, interrupt it and get the stack trace (thread apply all bt). That should show you exactly where it hangs (ie. what lock it tries to take ususally). Then you can look at the implementation .. > p.s. the "other" problems using g_object_get() are that it returns different > datatypes for the values than does g_object_get_property(). For example, > the "name" property is returned as a gchararray not as the expected > G_TYPE_STRING. Not quite sure what the problem is here or what you expect for G_TYPE_STRING if not a char array. Cheers -Tim From gibrovacco at gmail.com Mon Jan 3 18:45:45 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Mon, 3 Jan 2011 19:45:45 +0200 Subject: [gst-devel] Need Advice!!! In-Reply-To: <1293457213712-3164997.post@n4.nabble.com> References: <1293457213712-3164997.post@n4.nabble.com> Message-ID: Hi, extremely in late, hopefully you've already fixed the implementation, btw here's my pov. On Mon, Dec 27, 2010 at 3:40 PM, Anirudh wrote: > > 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. if the server uses a session initiation protocol (I assume you're using RTP) it may be possible to drop such a requirement. > > 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. for the receiving side, you can use uridecodebin: http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-uridecodebin.html then you can listen for the pad-added signal and dynamically connect encoder, payloader and rtpbin/udpsink. In case you want to use a session initiation protocol for the sending part, you may be lucky and find an already existing implementation (e.g. http://people.freedesktop.org/~wtay/) or implement it by your own. Regards > > 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. > > ------------------------------------------------------------------------------ > 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 wmiller at sdr.com Mon Jan 3 19:39:54 2011 From: wmiller at sdr.com (Wes Miller) Date: Mon, 3 Jan 2011 10:39:54 -0800 (PST) Subject: [gst-devel] Why do we call gst_object_unref() ? In-Reply-To: <4D207EE8.5000008@realh.co.uk> References: <1293950598145-3170699.post@n4.nabble.com> <4D207EE8.5000008@realh.co.uk> Message-ID: <1294079994902-3172399.post@n4.nabble.com> I'm going to give you a slightly different view of this answer. This explanation is slightly less correct, but once i figured it out, the whole unref stuff made more sense. This is a simple approach and there are probably dozens of special cases I am skipping over. First, let's say you have a program running in a local (your) thread. You want to do some Gstreamer stuff, so you create the pipeline, elements, etc. by calling gst_init() and various "create" routines. Here's the stricky part; whatever you create will not be in your local thread, It will be in some other thread (probably) spawned by the gst-Init() call. Think of this as saying your code remains "over here" and the pipeline, bus, loop, etc, are created "over there". The "create" functions return pointers that tell you not where your creations are located, but where they are described. They point at structures (located in "over there's heap) that holds the details about the pipeline or whatever you created. These structures have a refcount var in them that is set to 1 meaning somebody (you) are using those structures and the stuff they describe. Useful Note: When a pipeline is created, it creates a bus. The bus is an object "over there" but you have no idea where! A GstBus structure is also created to describe the bus itself and to hold the refcount for the bus. It is set to one showing the pipeline holds a ref to the bus. Hold that thought and see the example below. When you want to access something "over there" you must use the pointer you got from the create function or get a new pointer by asking "over there" to create a pointer/structure for you. When you are done with something, some object, "over there', you must dereference it. We do this with the g_object_unerf() function, which reduces the refcount by 1. It is possible for something to have many references. When the reference count goes to zero, the structure and the object it describes are garbage collected. Example: In your local code you create a GstBus* pointer. When you call bus = gst_pipeline_get_bus(), the pointer to the structure describing the bus is returned.. Note that, in this case, the structure doesn't have to be created. It already exists. It will be the same one created when the pipeline created the bus. It will hold a refcount of 2, one for you and one for the pipeline. Now, do whatever you need to with the bus over there, say, assign it a callback routine for the on_pad_added signal. Notice what you are really doing. You are NOT directly touching the bus. Gst_bus_add_watch() uses the information in the "bus" structure and does its magic indirectly. You set the callback pointer into the bus object itself, not its descriptive structure. finally, you need to unref the bus. You no longer need to know anything about the bus. It knows what you wanted it to know. So, use g_object_unref() to remove one ref from the structure. The refcount will decrease by one and will now equal one, the ref held by the pipeline. Had the refcount gone to zero, the bus and structure would have been garbage collected and gone out of scope. The ref held by the pipeline keeps them "alive". At the end of your program. deref-ing the pipeline will cause it to deref the bus automatically. This can be messy and confusing. Be sure to read the description of every g_ and gst_ call you make to see if it tells you to unref something. Hope this helps, Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Why-do-we-call-gst-object-unref-tp3170699p3172399.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wmiller at sdr.com Mon Jan 3 19:44:36 2011 From: wmiller at sdr.com (Wes Miller) Date: Mon, 3 Jan 2011 10:44:36 -0800 (PST) Subject: [gst-devel] How to add a complicate property to an element In-Reply-To: <1A42CE6F5F474C41B63392A5F80372B231D2AEB2@shsmsx501.ccr.corp.intel.com> References: <1A42CE6F5F474C41B63392A5F80372B231D2AEB2@shsmsx501.ccr.corp.intel.com> Message-ID: <1294080276223-3172409.post@n4.nabble.com> You aren't adding properties to v4l2src, they already exist in the element and you are just setting values into them. See gst-inspect v4l2src for details. You'd need a rewritten version of your element if you need your property to be settable and to have it actually do something. Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-add-a-complicate-property-to-an-element-tp3169945p3172409.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From c.r.n.a at wanadoo.fr Mon Jan 3 19:59:12 2011 From: c.r.n.a at wanadoo.fr (Nicolas) Date: Mon, 03 Jan 2011 19:59:12 +0100 Subject: [gst-devel] Need help for rezising video. Message-ID: <4D221C80.10601@wanadoo.fr> Hello, I have created a plugin that displays a video in an evas image. Here is my problem: When I resize the picture I am calling "evas_object_image_fill_set", it works correctly, however the consumption of CPU is just horrible, more than 120% ! If I leave it in normal size, the consumption of CPU is normal. I want to know if there is a way to resize the video (via gstreamer) in my plugin. Have you any idea? May be I could create a function "buffer alloc" (gstbase_sink_class->buffer_alloc = mybufferalloc) and modify the buffer in it ? I know that there is "videoscale" but I do not know how to use it inside the plugin. Thank you in advance for your reply, Sincerely, Nicolas. Ps: sorry for bad english. From h at realh.co.uk Mon Jan 3 20:32:30 2011 From: h at realh.co.uk (Tony Houghton) Date: Mon, 03 Jan 2011 19:32:30 +0000 Subject: [gst-devel] Recording TS from a DVB card Message-ID: <4D22244E.2040509@realh.co.uk> [Apologies if you get more than one copy, my first attempt hasn't appeared] I want to record MPEG TS streams from my DVB cards to disc. I want to be able to record more than one thing at a time from a single dvr device so so far I've written code to save packets to a buffer and use callback functions to selectively notify "clients" which have registered their interest in packets with certain pids. I want to sound out GStreamer's capabilities in this area to see whether it's the best thing to handle the dumping of the partial streams to disc. AIUI recorded TS files should start with a PAT and PMT and the first video frame should be an I-frame. Although writing my own code to handle that shouldn't be difficult, does GStreamer already have code to make that easy? The main reason I'm considering GStreamer is because I also want to add live streaming at some point and using gst's RTSP extension will probably be the most sensible way to achieve that. But I want the ability to "rewind" as well as pause live TV, and then be able to seek forwards to catch up. Can GStreamer already do all that, seamlessly handling the changeover between disk and memory, or should I write my own code for it and wrap it in a gst source plugin for RTSP streaming? -- TH * http://www.realh.co.uk From mgruenke at Tycoint.com Mon Jan 3 20:36:02 2011 From: mgruenke at Tycoint.com (Gruenke, Matt) Date: Mon, 3 Jan 2011 14:36:02 -0500 Subject: [gst-devel] Why do we call gst_object_unref() ? In-Reply-To: <1294079994902-3172399.post@n4.nabble.com> References: <1293950598145-3170699.post@n4.nabble.com><4D207EE8.5000008@realh.co.uk> <1294079994902-3172399.post@n4.nabble.com> Message-ID: Agreed about the point of always checking the docs. It occurred to me that I should have mentioned this. Not only must you always check whether the caller is responsible for unreffing an object returned by a function, but note that some functions cause ownership or refcount transfer of objects passed as parameters. Therefore, don't forget to check what the docs say about each parameter, in addition to the return types. Over time, you'll start to notice some patterns. For instance, macros - which are in ALL_CAPS - tend not to affect refcounts or ownership. BTW, parenting is only valid for objects derived from GObject. There's another baseclass in use, called GstMiniObject. These are used for lighter-weight objects where refcounting is still desirable, like buffers. Not only can these not be parented (or weak ref'd), but they also must be unref'd using a function specific to that type (e.g. gst_buffer_unref()), rather than gst_object_unref(). Matt -----Original Message----- From: Wes Miller [mailto:wmiller at sdr.com] Sent: Monday, January 03, 2011 13:40 To: gstreamer-devel at lists.sourceforge.net Subject: Re: [gst-devel] Why do we call gst_object_unref() ? This can be messy and confusing. Be sure to read the description of every g_ and gst_ call you make to see if it tells you to unref something. From lijinsyam at gmail.com Mon Jan 3 21:03:59 2011 From: lijinsyam at gmail.com (liJin) Date: Tue, 4 Jan 2011 01:33:59 +0530 Subject: [gst-devel] Need help for rezising video. In-Reply-To: <4D221C80.10601@wanadoo.fr> References: <4D221C80.10601@wanadoo.fr> Message-ID: Hi.. If you are looking for a plugin to resize the image ..i think capsfilter will help you.. Reagrds Lijin On Tue, Jan 4, 2011 at 12:29 AM, Nicolas wrote: > Hello, > > I have created a plugin that displays a video in an evas image. > > Here is my problem: > When I resize the picture I am calling "evas_object_image_fill_set", it > works correctly, however the consumption of CPU is just horrible, more > than 120% ! > If I leave it in normal size, the consumption of CPU is normal. > > I want to know if there is a way to resize the video (via gstreamer) in > my plugin. > Have you any idea? > May be I could create a function "buffer alloc" > (gstbase_sink_class->buffer_alloc = mybufferalloc) and modify the buffer > in it ? > > I know that there is "videoscale" but I do not know how to use it inside > the plugin. > > Thank you in advance for your reply, > Sincerely, > Nicolas. > > Ps: sorry for bad english. > > > ------------------------------------------------------------------------------ > 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 shiva.varma at gmail.com Mon Jan 3 21:31:09 2011 From: shiva.varma at gmail.com (shiva varma) Date: Mon, 3 Jan 2011 12:31:09 -0800 (PST) Subject: [gst-devel] How to extract G711 u-law audio from RTP packets Message-ID: <1294086669869-3172578.post@n4.nabble.com> Hi I have to extract G711 u-law audio from a pcap file. The pcap file has RTP packets with the RTP payload conforming to RFC 3551. I do not see any rtp depay's by description, which can extract G711 u-law audio from RTP packets. Can someone point me to a way of getting this done. Thanks Shiva -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-extract-G711-u-law-audio-from-RTP-packets-tp3172578p3172578.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wmiller at sdr.com Mon Jan 3 21:52:25 2011 From: wmiller at sdr.com (Wes Miller) Date: Mon, 3 Jan 2011 12:52:25 -0800 (PST) Subject: [gst-devel] Extracting properties from element causes hang In-Reply-To: <1294076282.3220.8.camel@zingle> References: <1293726221815-3168471.post@n4.nabble.com> <1294073288460-3172213.post@n4.nabble.com> <1294076282.3220.8.camel@zingle> Message-ID: <1294087945202-3172615.post@n4.nabble.com> I stink at gdb. I'll see where I get but I'm light on hope. I guess I expected a string to be a string object with a length field and a pointer to the data and gchararray to be either a char xyz[234] or the "g" representation of a chararray. I hadn't expected any difference in what wouyld be returned. FWIW, the code I copied from gst-inspect sees them differently. With g_object_get_property() this runs as expected: switch( G_VALUE_TYPE( &value )) { case G_TYPE_STRING: // String But with g_object_get() I get to the bailaout case for Unknown datatype, which prints Unknown datatype: gchararray (64). The 64 appears to be the type index of gchararray. So, I assume I'm missing the obvious and that they really are the same, but Tim's code caught it. Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Extracting-properties-from-element-causes-hang-tp3168471p3172615.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From c.r.n.a at wanadoo.fr Mon Jan 3 22:06:59 2011 From: c.r.n.a at wanadoo.fr (Nicolas) Date: Mon, 03 Jan 2011 22:06:59 +0100 Subject: [gst-devel] Need help for rezising video. Message-ID: <4D223A73.9050802@wanadoo.fr> Hi Lijin, Thanks for your reply. But how capsfilter can help me ? I only see in the doc: capsfilter ? Pass data without modification, limiting formats Regards, Nicolas. From mgruenke at Tycoint.com Mon Jan 3 23:57:01 2011 From: mgruenke at Tycoint.com (Gruenke, Matt) Date: Mon, 3 Jan 2011 17:57:01 -0500 Subject: [gst-devel] GObjectClass.dispose() and GstElement Message-ID: Based on my reading of the Plugin Writer's Guide & the GstElement reference docs, elements are supposed to complete all of their deallocation on the READY -> NULL transition. However, it's always bothered me that there's a case where you could still get a memory leak. Suppose an element's GObjectClass.set_property() gets called while the element is still in the NULL state. If this property accepts a string or some other type of heap-allocated object, the element is probably going to create a copy or add a ref. Now, suppose some problem occurs and the element is disposed before ever going to the READY state. There will be no transition back to NULL. Therefore, the only possibility of deallocating the value is by overriding GstObjetClass.dispose(). I concede that this isn't normally an issue, since the size of any such properties is likely to be very small and many applications will either exit or typically repeat this sequence a limited number of times. However, it's quite possible that some sort of automated recording application might repeat this sequence somewhat rapidly and for an indefinite period of time. Then, there's the whole philosophical issue of avoiding even small leaks, which turns into a very practical issue when trying to use tools to find slightly more serious leaks. I know you can usually work around this, through the use of filtering, but it's inconvenient and not always satisfactory... let's please try to avoid a flame war about this. Would it be so bad for GstObjectClass and GstElementClass to at least provide a dispose() hook? I'm aware that I could just save the old value & replace with my own dispose(), but something feels wrong about having the derived type dispose of its ancestors. It's also not a very user-friendly way to do it, since there are numerous ways the derived type can mess things up (i.e. not saving the old value or failing to call it last). Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: From rohitratri at gmail.com Tue Jan 4 05:06:33 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Tue, 4 Jan 2011 09:36:33 +0530 Subject: [gst-devel] Need help for rezising video. In-Reply-To: <4D223A73.9050802@wanadoo.fr> References: <4D223A73.9050802@wanadoo.fr> Message-ID: You will have to create a capsfilter element and set the relevant caps on it (with the frame size you require). Then follow it up with a video scale element. (...->decoder->capsfilter->videoscale->...->sink) Check out this doc here - http://pygstdocs.berlios.de/pygst-tutorial/capabilities.html Its pygst but the concept is the same. Rohit On Tue, Jan 4, 2011 at 2:36 AM, Nicolas wrote: > Hi Lijin, > > Thanks for your reply. > > But how capsfilter can help me ? > > I only see in the doc: > capsfilter ? Pass data without modification, limiting formats > > Regards, > Nicolas. > > > ------------------------------------------------------------------------------ > 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 Tue Jan 4 06:52:42 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Tue, 4 Jan 2011 11:22:42 +0530 Subject: [gst-devel] When to call g_thread_init(), again... and again... Message-ID: Hi, Can somebody explain when exactly the api g_thread_init() should be called? And does a "threaded" program exactly mean(in the context of the documentation here - http://library.gnome.org/devel/glib/2.26/glib-Threads.html#g-thread-init)? If I've a gst pipeline which obviously runs in a different thread from my application thread, should I be calling g_thread_init()? I see that this discussion has already taken place here - http://mail.gnome.org/archives/gtk-devel-list/2008-August/msg00110.html but I didn't see a convincing answer/solution being discussed. Thanks Rohit -------------- next part -------------- An HTML attachment was scrubbed... URL: From lijinsyam at gmail.com Tue Jan 4 07:00:03 2011 From: lijinsyam at gmail.com (liJin) Date: Tue, 4 Jan 2011 11:30:03 +0530 Subject: [gst-devel] Need help for rezising video. In-Reply-To: References: <4D223A73.9050802@wanadoo.fr> Message-ID: Hi... This is how i used the capsfilter vflt=gst_element_factory_make("capsfilter","imagefilter"); if(!vflt) { g_print("\n Image Filter element could not be created \n"); return 0; } g_object_set(G_OBJECT(vflt),"caps",gst_caps_new_simple("image/jpeg", "width",G_TYPE_INT,320, "height",G_TYPE_INT,240, "framerate",GST_TYPE_FRACTION,30,1, NULL),NULL); I think this will help you... On Tue, Jan 4, 2011 at 9:36 AM, Rohit Atri wrote: > You will have to create a capsfilter element and set the relevant caps on > it (with the frame size you require). Then follow it up with a video scale > element. (...->decoder->capsfilter->videoscale->...->sink) > Check out this doc here - > http://pygstdocs.berlios.de/pygst-tutorial/capabilities.html > > Its pygst > but the concept is the same. > > Rohit > > > On Tue, Jan 4, 2011 at 2:36 AM, Nicolas wrote: > >> Hi Lijin, >> >> Thanks for your reply. >> >> But how capsfilter can help me ? >> >> I only see in the doc: >> capsfilter ? Pass data without modification, limiting formats >> >> Regards, >> Nicolas. >> >> >> ------------------------------------------------------------------------------ >> 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 lane at brooks.nu Tue Jan 4 07:23:46 2011 From: lane at brooks.nu (Lane Brooks) Date: Mon, 03 Jan 2011 23:23:46 -0700 Subject: [gst-devel] Video transitions using gnloperation Message-ID: <4D22BCF2.90006@brooks.nu> I have a gnlcomposition containing two video sources. I would like to apply a smpte transition from one source to the next. I have tried overlapping the two gnlsources by 3 seconds. I then put a 'smpte' element in a 'gnloperation' and add the 'gnloperation' to the 'gnlcompostion'. The addition of the gnloperation doesn't seem to do anything. The resulting video is the same regardless of whether I add the gnloperation or not. The result is always that the second video takes over during the period of overlap. Here is code I have (in python): comp = gst.element_factory_make("gnlcomposition") src1 = gst.element_factory_make("gnlfilesource") src2 = gst.element_factory_make("gnlfilesource") comp.add(src1) src1.set_property("location", "/test1.mp4") src1.set_property("start", 0 * gst.SECOND) src1.set_property("duration", 5 * gst.SECOND) src1.set_property("media-start", 0 * gst.SECOND) src1.set_property("media-duration", 5 * gst.SECOND) comp.add(src2) src2.set_property("location", "/test2.mp4") src2.set_property("start", 2 * gst.SECOND) src2.set_property("duration", 5 * gst.SECOND) src2.set_property("media-start", 0 * gst.SECOND) src2.set_property("media-duration", 5 * gst.SECOND) op = gst.element_factory_make("gnloperation") tran = gst.element_factory_make("smpte") tran.set_property("type", 234) tran.set_property("duration", 3 * gst.SECOND) op.add(tran) comp.add(op) Anyone know if using an smpte transition is possible, and if so, what I am doing wrong? Thanks, Lane From wim.taymans at gmail.com Tue Jan 4 10:23:55 2011 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 04 Jan 2011 10:23:55 +0100 Subject: [gst-devel] How to extract G711 u-law audio from RTP packets In-Reply-To: <1294086669869-3172578.post@n4.nabble.com> References: <1294086669869-3172578.post@n4.nabble.com> Message-ID: <1294133035.28802.58.camel@meany> On Mon, 2011-01-03 at 12:31 -0800, shiva varma wrote: > Hi > I have to extract G711 u-law audio from a pcap file. The pcap file has RTP > packets with the RTP payload conforming to RFC 3551. I do not see any rtp > depay's by description, which can extract G711 u-law audio from RTP packets. > Can someone point me to a way of getting this done. rtppcmudepay Wim > > Thanks > Shiva > From remy.perrier at endirectv.com Tue Jan 4 12:26:52 2011 From: remy.perrier at endirectv.com (=?ISO-8859-1?Q?R=E9my_PERRIER?=) Date: Tue, 04 Jan 2011 12:26:52 +0100 Subject: [gst-devel] Gstreamer Installation with Mingw Message-ID: <4D2303FC.1040304@endirectv.com> Hi all, I am trying to install gstreamer on Win32 thanks to mingw. All installation steps seems to succeed however examples or some gstreamer commands do not work. I always get thoses error messages : zae at zae~ $ gst-inspect zae at zae~ $ (gst-inspect-0.10:6008): GLib-GObject-WARNING **: cannot register existing type `GstObject' (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: assertion `initial ization_value != 0' failed (gst-inspect-0.10:6008): GLib-GObject-CRITICAL **: g_type_register_static: asser tion `parent_type > 0' failed (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: assertion `initial ization_value != 0' failed (gst-inspect-0.10:6008): GLib-GObject-CRITICAL **: g_type_register_static: asser tion `parent_type > 0' failed (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: assertion `initial ization_value != 0' failed _/My configuration :/_ *OS*: win XP *Gstreamer version *: 0.10.31 *GLIB version* : 2.26.0 *gst plugin base *: 0.10.31 I would like to know where are my mistakes. Thanks, remy -------------- next part -------------- An HTML attachment was scrubbed... URL: From henriette.depoulpiquet at gmail.com Tue Jan 4 12:47:53 2011 From: henriette.depoulpiquet at gmail.com (henriette de poulpiquet) Date: Tue, 4 Jan 2011 12:47:53 +0100 Subject: [gst-devel] install gst-plugins-gl-0.10.2 Message-ID: Hi! I'me not sure where to post my question, hope i'me in the right place! so my problem is that my gst-inspect indicate that gl plugin is blacklisted, I'me quite disappointed and can't understand why /usr/local/lib/gstreamer-0.10/libgstopengl.so can' t be found since libgstopengl.so do exists at this place (I checked it). what should I do? thanks for your answer! Henriette i'me working on a Linux: Debian 32 bits Release: 2.6.26-2-686 i686 gstreamer-0.10.31 gst-plugins-base-0.10.31 gst-plugins-good-0.10.26 From bisht.sudarshan at gmail.com Tue Jan 4 12:57:50 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 4 Jan 2011 13:57:50 +0200 Subject: [gst-devel] install gst-plugins-gl-0.10.2 In-Reply-To: References: Message-ID: Did you do " export GST_PLUGIN_PATH=/usr/local/lib/gstreamer-0.10" ? On Tue, Jan 4, 2011 at 1:47 PM, henriette de poulpiquet < henriette.depoulpiquet at gmail.com> wrote: > Hi! > > I'me not sure where to post my question, hope i'me in the right place! > > so my problem is that my gst-inspect indicate that gl plugin is > blacklisted, I'me quite disappointed and can't understand why > /usr/local/lib/gstreamer-0.10/libgstopengl.so can' t be found since > libgstopengl.so do exists at this place (I checked it). > > what should I do? > > thanks for your answer! > > > Henriette > > i'me working on a > Linux: Debian 32 bits > Release: 2.6.26-2-686 > i686 > gstreamer-0.10.31 > gst-plugins-base-0.10.31 > gst-plugins-good-0.10.26 > > > ------------------------------------------------------------------------------ > 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 tiagomatos at gmail.com Tue Jan 4 13:01:24 2011 From: tiagomatos at gmail.com (=?UTF-8?Q?Rui_Tiago_Ca=C3=A7=C3=A3o_Matos?=) Date: Tue, 4 Jan 2011 12:01:24 +0000 Subject: [gst-devel] When to call g_thread_init(), again... and again... In-Reply-To: References: Message-ID: On 4 January 2011 05:52, Rohit Atri wrote: > Can somebody explain when exactly the api g_thread_init() should be called? Only if you would like to provide glib with a GThreadFunctions[1] other than the default. > And does a "threaded" program exactly mean(in the context of the > documentation here > -?http://library.gnome.org/devel/glib/2.26/glib-Threads.html#g-thread-init)? > If I've a gst pipeline which obviously runs in a different thread from my > application thread, should I be calling g_thread_init()? As documented there: "Please note that since version 2.24 the GObject initialization function g_type_init() initializes threads (with a NULL argument), so most applications, including those using Gtk+ will run with threads enabled." Since GStreamer builds upon the GType system[2] it initializes GType and thus also the glib threading stuff when you call gst_init(). Rui [1] http://library.gnome.org/devel/glib/2.26/glib-Threads.html#GThreadFunctions [2] http://library.gnome.org/devel/gobject/2.26/gobject-Type-Information.html From henriette.depoulpiquet at gmail.com Tue Jan 4 13:05:44 2011 From: henriette.depoulpiquet at gmail.com (henriette de poulpiquet) Date: Tue, 4 Jan 2011 13:05:44 +0100 Subject: [gst-devel] install gst-plugins-gl-0.10.2 In-Reply-To: References: Message-ID: yes I did... On 1/4/11, sudarshan bisht wrote: > Did you do " export GST_PLUGIN_PATH=/usr/local/lib/gstreamer-0.10" ? > > On Tue, Jan 4, 2011 at 1:47 PM, henriette de poulpiquet < > henriette.depoulpiquet at gmail.com> wrote: > >> Hi! >> >> I'me not sure where to post my question, hope i'me in the right place! >> >> so my problem is that my gst-inspect indicate that gl plugin is >> blacklisted, I'me quite disappointed and can't understand why >> /usr/local/lib/gstreamer-0.10/libgstopengl.so can' t be found since >> libgstopengl.so do exists at this place (I checked it). >> >> what should I do? >> >> thanks for your answer! >> >> >> Henriette >> >> i'me working on a >> Linux: Debian 32 bits >> Release: 2.6.26-2-686 >> i686 >> gstreamer-0.10.31 >> gst-plugins-base-0.10.31 >> gst-plugins-good-0.10.26 >> >> >> ------------------------------------------------------------------------------ >> 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 > From bradley.goldsmith at gmail.com Tue Jan 4 13:35:35 2011 From: bradley.goldsmith at gmail.com (bcg) Date: Tue, 4 Jan 2011 04:35:35 -0800 (PST) Subject: [gst-devel] Play audio from a memory buffer? Message-ID: <1294144535806-3173448.post@n4.nabble.com> Hi All, This should almost be an FAQ but it's not in there so here goes: I have a pipeline thus: file-source -> decodebin -> audioresample -> converter -> audio_output Using location (or fd if I change file_source to a fdsrc) I can happily play most audio I throw at it (including mp3s) providing its in a file. What I would like to do is play from a memory location with the mp3 data already in it. I am assuming I would need to use a fakesrc with a callback to get my data into the pipeline - I've tried lots of variations on this and gotten nowhere. Can I just replace the file-source with the fakesrc, load the data and go (if this is possible couple someone provide an example)? Or do I have to worry about queues and caps? If so, can someone provide an example? Will decodebin suffice or will I have to be more specific? Great software by the way, just tricky to use with mp3s (I know why...). Thanks, Brad -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Play-audio-from-a-memory-buffer-tp3173448p3173448.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bilboed at gmail.com Tue Jan 4 13:39:42 2011 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 04 Jan 2011 13:39:42 +0100 Subject: [gst-devel] Video transitions using gnloperation In-Reply-To: <4D22BCF2.90006@brooks.nu> References: <4D22BCF2.90006@brooks.nu> Message-ID: <1294144782.23475.5.camel@localhost> On Mon, 2011-01-03 at 23:23 -0700, Lane Brooks wrote: > I have a gnlcomposition containing two video sources. I would like to > apply a smpte transition from one source to the next. > > I have tried overlapping the two gnlsources by 3 seconds. I then put a > 'smpte' element in a 'gnloperation' and add the 'gnloperation' to the > 'gnlcompostion'. The addition of the gnloperation doesn't seem to do > anything. The resulting video is the same regardless of whether I add > the gnloperation or not. The result is always that the second video > takes over during the period of overlap. Here is code I have (in python): > > comp = gst.element_factory_make("gnlcomposition") > src1 = gst.element_factory_make("gnlfilesource") > src2 = gst.element_factory_make("gnlfilesource") > > comp.add(src1) > src1.set_property("location", "/test1.mp4") > src1.set_property("start", 0 * gst.SECOND) > src1.set_property("duration", 5 * gst.SECOND) > src1.set_property("media-start", 0 * gst.SECOND) > src1.set_property("media-duration", 5 * gst.SECOND) > > comp.add(src2) > src2.set_property("location", "/test2.mp4") > src2.set_property("start", 2 * gst.SECOND) > src2.set_property("duration", 5 * gst.SECOND) > src2.set_property("media-start", 0 * gst.SECOND) > src2.set_property("media-duration", 5 * gst.SECOND) > > op = gst.element_factory_make("gnloperation") > tran = gst.element_factory_make("smpte") > tran.set_property("type", 234) > tran.set_property("duration", 3 * gst.SECOND) > op.add(tran) Set the [media-]{start|duration} propertie son op also > comp.add(op) > > Anyone know if using an smpte transition is possible, and if so, what I > am doing wrong? > > Thanks, > Lane > > > > ------------------------------------------------------------------------------ > 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 remy.perrier at endirectv.com Tue Jan 4 13:45:46 2011 From: remy.perrier at endirectv.com (=?ISO-8859-1?Q?R=E9my_PERRIER?=) Date: Tue, 04 Jan 2011 13:45:46 +0100 Subject: [gst-devel] Gstreamer Installation with Mingw In-Reply-To: <4D2303FC.1040304@endirectv.com> References: <4D2303FC.1040304@endirectv.com> Message-ID: <4D23167A.7050706@endirectv.com> ok it s fine. In deed I forget to add GST_PLUGIN_PATH. Thanks, remy Le 04/01/2011 12:26, R?my PERRIER a ?crit : > Hi all, > > > I am trying to install gstreamer on Win32 thanks to mingw. > All installation steps seems to succeed however examples or some > gstreamer commands do not work. > > I always get thoses error messages : > > zae at zae~ > $ gst-inspect > > zae at zae~ > $ > (gst-inspect-0.10:6008): GLib-GObject-WARNING **: cannot register > existing type > `GstObject' > > (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: > assertion `initial > ization_value != 0' failed > > (gst-inspect-0.10:6008): GLib-GObject-CRITICAL **: > g_type_register_static: asser > tion `parent_type > 0' failed > > (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: > assertion `initial > ization_value != 0' failed > > (gst-inspect-0.10:6008): GLib-GObject-CRITICAL **: > g_type_register_static: asser > tion `parent_type > 0' failed > > (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: > assertion `initial > ization_value != 0' failed > > > _/My configuration :/_ > > *OS*: win XP > *Gstreamer version *: 0.10.31 > *GLIB version* : 2.26.0 > *gst plugin base *: 0.10.31 > > > I would like to know where are my mistakes. > Thanks, > > > remy > > > > > ------------------------------------------------------------------------------ > 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 w3cing at gmail.com Tue Jan 4 13:49:37 2011 From: w3cing at gmail.com (wecing) Date: Tue, 4 Jan 2011 20:49:37 +0800 Subject: [gst-devel] Gstreamer Installation with Mingw In-Reply-To: <4D23167A.7050706@endirectv.com> References: <4D2303FC.1040304@endirectv.com> <4D23167A.7050706@endirectv.com> Message-ID: HAHA. On Tue, Jan 4, 2011 at 8:45 PM, R?my PERRIER wrote: > ok it s fine. > In deed I forget to add GST_PLUGIN_PATH. > > Thanks, > > remy > > > Le 04/01/2011 12:26, R?my PERRIER a ?crit : > > Hi all, > > > I am trying to install gstreamer on Win32 thanks to mingw. > All installation steps seems to succeed however examples or some > gstreamer commands do not work. > > I always get thoses error messages : > > zae at zae~ > $ gst-inspect > > zae at zae~ > $ > (gst-inspect-0.10:6008): GLib-GObject-WARNING **: cannot register existing > type > `GstObject' > > (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: assertion > `initial > ization_value != 0' failed > > (gst-inspect-0.10:6008): GLib-GObject-CRITICAL **: g_type_register_static: > asser > tion `parent_type > 0' failed > > (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: assertion > `initial > ization_value != 0' failed > > (gst-inspect-0.10:6008): GLib-GObject-CRITICAL **: g_type_register_static: > asser > tion `parent_type > 0' failed > > (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: assertion > `initial > ization_value != 0' failed > > > *My configuration :* > > *OS*: win XP > *Gstreamer version *: 0.10.31 > *GLIB version* : 2.26.0 > *gst plugin base *: 0.10.31 > > > I would like to know where are my mistakes. > Thanks, > > > remy > > > > > ------------------------------------------------------------------------------ > 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 disruptionhttp://p.sf.net/sfu/oracle-sfdevnl > > > _______________________________________________ > gstreamer-devel mailing listgstreamer-devel at lists.sourceforge.nethttps://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 markus.t.vartiainen at nokia.com Tue Jan 4 15:43:03 2011 From: markus.t.vartiainen at nokia.com (Markus Vartiainen) Date: Tue, 4 Jan 2011 16:43:03 +0200 Subject: [gst-devel] Gstreamer Installation with Mingw In-Reply-To: References: <4D2303FC.1040304@endirectv.com> <4D23167A.7050706@endirectv.com> Message-ID: <4D2331F7.5030703@nokia.com> No reason to laugh at others' mistakes - we all make them. Thanks to original poster, who humbly posted not only the problem, but also the (obvious, in hindsight) solution to it, there is now another small piece of useful documentation in public archives for other GStreamer users to find. Regards, Markus ext wecing wrote: > HAHA. > > On Tue, Jan 4, 2011 at 8:45 PM, R?my PERRIER > wrote: > > ok it s fine. > In deed I forget to add GST_PLUGIN_PATH. > > Thanks, > > remy > > > Le 04/01/2011 12:26, R?my PERRIER a ?crit : >> Hi all, >> >> >> I am trying to install gstreamer on Win32 thanks to mingw. >> All installation steps seems to succeed however examples or some >> gstreamer commands do not work. >> >> I always get thoses error messages : >> >> zae at zae~ >> $ gst-inspect >> >> zae at zae~ >> $ >> (gst-inspect-0.10:6008): GLib-GObject-WARNING **: cannot register >> existing type >> `GstObject' >> >> (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: >> assertion `initial >> ization_value != 0' failed >> >> (gst-inspect-0.10:6008): GLib-GObject-CRITICAL **: >> g_type_register_static: asser >> tion `parent_type > 0' failed >> >> (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: >> assertion `initial >> ization_value != 0' failed >> >> (gst-inspect-0.10:6008): GLib-GObject-CRITICAL **: >> g_type_register_static: asser >> tion `parent_type > 0' failed >> >> (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: >> assertion `initial >> ization_value != 0' failed >> >> >> _/My configuration :/_ >> >> *OS*: win XP >> *Gstreamer version *: 0.10.31 >> *GLIB version* : 2.26.0 >> *gst plugin base *: 0.10.31 >> >> >> I would like to know where are my mistakes. >> Thanks, >> >> >> remy >> >> >> >> >> ------------------------------------------------------------------------------ >> 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 t.i.m at zen.co.uk Tue Jan 4 15:55:09 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 04 Jan 2011 14:55:09 +0000 Subject: [gst-devel] Play audio from a memory buffer? In-Reply-To: <1294144535806-3173448.post@n4.nabble.com> References: <1294144535806-3173448.post@n4.nabble.com> Message-ID: <1294152909.19493.22.camel@zingle> On Tue, 2011-01-04 at 04:35 -0800, bcg wrote: Hi, > This should almost be an FAQ but it's not in there so here goes: > > I have a pipeline thus: > > file-source -> decodebin -> audioresample -> converter -> audio_output > > Using location (or fd if I change file_source to a fdsrc) I can happily play > most audio I throw at it (including mp3s) providing its in a file. What I > would like to do is play from a memory location with the mp3 data already in > it. > > I am assuming I would need to use a fakesrc with a callback to get my data > into the pipeline - I've tried lots of variations on this and gotten > nowhere. > > Can I just replace the file-source with the fakesrc, load the data and go > (if this is possible couple someone provide an example)? Or do I have to > worry about queues and caps? If so, can someone provide an example? Will > decodebin suffice or will I have to be more specific? Use appsrc. Cheers -Tim From radivojejovanovic at gmail.com Tue Jan 4 17:09:29 2011 From: radivojejovanovic at gmail.com (Radivoje Jovanovic) Date: Tue, 4 Jan 2011 08:09:29 -0800 Subject: [gst-devel] Alsasink problem? In-Reply-To: References: <4D1A441F.1000801@panicking.kicks-ass.org> Message-ID: gst-launch audiotestsrc ! alsasink -v works no problem. I have realized that if I play the song using virtual device instead of the real device the thing works just fine. I have made the virtual device as: pcm.mydev { type plug slave { pcm default rate 32000 } } My driver supports all the rates from 8KHz to 48KHz, but it seems to me that the rate in this song is different than what my driver supports. Is this rate conversion suppose to happen in ALSA automatically or I have to use ALSA through virtual device in order to get unsupported parameters to work on my driver? If so, is it possible to specify multiple rates in my virtual device or I have to have one rate only? Cheers, Ogi On Wed, Dec 29, 2010 at 10:21 PM, praveen pandey wrote: > 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 >> >> > > > ------------------------------------------------------------------------------ > 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 w3cing at gmail.com Tue Jan 4 17:20:00 2011 From: w3cing at gmail.com (wecing) Date: Wed, 5 Jan 2011 00:20:00 +0800 Subject: [gst-devel] Gstreamer Installation with Mingw In-Reply-To: <4D2331F7.5030703@nokia.com> References: <4D2303FC.1040304@endirectv.com> <4D23167A.7050706@endirectv.com> <4D2331F7.5030703@nokia.com> Message-ID: I didn't mean that. "HAHA" means "take it easy, dude". On Tue, Jan 4, 2011 at 10:43 PM, Markus Vartiainen < markus.t.vartiainen at nokia.com> wrote: > No reason to laugh at others' mistakes - we all make them. > > Thanks to original poster, who humbly posted not only the problem, but > also the (obvious, in hindsight) solution to it, there is now another > small piece of useful documentation in public archives for other > GStreamer users to find. > > Regards, > > Markus > > > ext wecing wrote: > > HAHA. > > > > On Tue, Jan 4, 2011 at 8:45 PM, R?my PERRIER > > wrote: > > > > ok it s fine. > > In deed I forget to add GST_PLUGIN_PATH. > > > > Thanks, > > > > remy > > > > > > Le 04/01/2011 12:26, R?my PERRIER a ?crit : > >> Hi all, > >> > >> > >> I am trying to install gstreamer on Win32 thanks to mingw. > >> All installation steps seems to succeed however examples or some > >> gstreamer commands do not work. > >> > >> I always get thoses error messages : > >> > >> zae at zae~ > >> $ gst-inspect > >> > >> zae at zae~ > >> $ > >> (gst-inspect-0.10:6008): GLib-GObject-WARNING **: cannot register > >> existing type > >> `GstObject' > >> > >> (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: > >> assertion `initial > >> ization_value != 0' failed > >> > >> (gst-inspect-0.10:6008): GLib-GObject-CRITICAL **: > >> g_type_register_static: asser > >> tion `parent_type > 0' failed > >> > >> (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: > >> assertion `initial > >> ization_value != 0' failed > >> > >> (gst-inspect-0.10:6008): GLib-GObject-CRITICAL **: > >> g_type_register_static: asser > >> tion `parent_type > 0' failed > >> > >> (gst-inspect-0.10:6008): GLib-CRITICAL **: g_once_init_leave: > >> assertion `initial > >> ization_value != 0' failed > >> > >> > >> _/My configuration :/_ > >> > >> *OS*: win XP > >> *Gstreamer version *: 0.10.31 > >> *GLIB version* : 2.26.0 > >> *gst plugin base *: 0.10.31 > >> > >> > >> I would like to know where are my mistakes. > >> Thanks, > >> > >> > >> remy > >> > >> > >> > >> > >> > ------------------------------------------------------------------------------ > >> 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 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 henriette.depoulpiquet at gmail.com Tue Jan 4 18:08:45 2011 From: henriette.depoulpiquet at gmail.com (henriette de poulpiquet) Date: Tue, 4 Jan 2011 18:08:45 +0100 Subject: [gst-devel] install gst-plugins-gl-0.10.2 In-Reply-To: References: Message-ID: maybe my Glib version is important? Glib 2.26.1 .. no one ever had this problem? anyway, thanks Sudarshan for your help! Henriette On 1/4/11, henriette de poulpiquet wrote: > yes I did... > > > On 1/4/11, sudarshan bisht wrote: >> Did you do " export GST_PLUGIN_PATH=/usr/local/lib/gstreamer-0.10" ? >> >> On Tue, Jan 4, 2011 at 1:47 PM, henriette de poulpiquet < >> henriette.depoulpiquet at gmail.com> wrote: >> >>> Hi! >>> >>> I'me not sure where to post my question, hope i'me in the right place! >>> >>> so my problem is that my gst-inspect indicate that gl plugin is >>> blacklisted, I'me quite disappointed and can't understand why >>> /usr/local/lib/gstreamer-0.10/libgstopengl.so can' t be found since >>> libgstopengl.so do exists at this place (I checked it). >>> >>> what should I do? >>> >>> thanks for your answer! >>> >>> >>> Henriette >>> >>> i'me working on a >>> Linux: Debian 32 bits >>> Release: 2.6.26-2-686 >>> i686 >>> gstreamer-0.10.31 >>> gst-plugins-base-0.10.31 >>> gst-plugins-good-0.10.26 >>> >>> >>> ------------------------------------------------------------------------------ >>> 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 >> > From lfarkas at lfarkas.org Tue Jan 4 17:56:56 2011 From: lfarkas at lfarkas.org (Farkas Levente) Date: Tue, 04 Jan 2011 17:56:56 +0100 Subject: [gst-devel] is gstreamer git hacked? Message-ID: <4D235158.5090000@lfarkas.org> hi, i just read the docs: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/pwg/html/chapter-building-signals.html then click the link "GObject documentation" :-) imho it's not totally normal... -- Levente "Si vis pacem para bellum!" From james.hodapp at gmail.com Tue Jan 4 18:25:46 2011 From: james.hodapp at gmail.com (Jim Hodapp) Date: Tue, 4 Jan 2011 12:25:46 -0500 Subject: [gst-devel] GstXML API Message-ID: <85073A53-8A84-425E-BF26-E8014C4BF010@GMAIL.COM> I am curious if anyone can give a good explanation as to why the GstXML API in GStreamer was deprecated. If I wanted to write a general dynamic pipeline generator, what would be the recommended way to capture the different type of pipelines in a data file? Should I create my own XML definition and libxml2 instead of using GstXML? I'm looking for best practice suggestions here that will be maintainable going forward with new GStreamer releases. Thanks, Jim From lane at brooks.nu Tue Jan 4 18:01:08 2011 From: lane at brooks.nu (Lane Brooks) Date: Tue, 04 Jan 2011 10:01:08 -0700 Subject: [gst-devel] Video transitions using gnloperation In-Reply-To: <1294144782.23475.5.camel@localhost> References: <4D22BCF2.90006@brooks.nu> <1294144782.23475.5.camel@localhost> Message-ID: <4D235254.3040103@brooks.nu> On 01/04/2011 05:39 AM, Edward Hervey wrote: > On Mon, 2011-01-03 at 23:23 -0700, Lane Brooks wrote: >> I have a gnlcomposition containing two video sources. I would like to >> apply a smpte transition from one source to the next. >> >> I have tried overlapping the two gnlsources by 3 seconds. I then put a >> 'smpte' element in a 'gnloperation' and add the 'gnloperation' to the >> 'gnlcompostion'. The addition of the gnloperation doesn't seem to do >> anything. The resulting video is the same regardless of whether I add >> the gnloperation or not. The result is always that the second video >> takes over during the period of overlap. Here is code I have (in python): >> >> comp = gst.element_factory_make("gnlcomposition") >> src1 = gst.element_factory_make("gnlfilesource") >> src2 = gst.element_factory_make("gnlfilesource") >> >> comp.add(src1) >> src1.set_property("location", "/test1.mp4") >> src1.set_property("start", 0 * gst.SECOND) >> src1.set_property("duration", 5 * gst.SECOND) >> src1.set_property("media-start", 0 * gst.SECOND) >> src1.set_property("media-duration", 5 * gst.SECOND) >> >> comp.add(src2) >> src2.set_property("location", "/test2.mp4") >> src2.set_property("start", 2 * gst.SECOND) >> src2.set_property("duration", 5 * gst.SECOND) >> src2.set_property("media-start", 0 * gst.SECOND) >> src2.set_property("media-duration", 5 * gst.SECOND) >> >> op = gst.element_factory_make("gnloperation") >> tran = gst.element_factory_make("smpte") >> tran.set_property("type", 234) >> tran.set_property("duration", 3 * gst.SECOND) >> op.add(tran) > > Set the [media-]{start|duration} propertie son op also I had already tried this as well. op.set_property("start", 2 * gst.SECOND) op.set_property("duration", 3 * gst.SECOND) op.set_property("media-start", 0 * gst.SECOND) op.set_property("media-duration", 3 * gst.SECOND) I am not sure what I should put the media-start at, but I have tried both 0 and 2 and neither works. Any other ideas? > >> comp.add(op) >> >> Anyone know if using an smpte transition is possible, and if so, what I >> am doing wrong? >> >> Thanks, >> Lane From braunsquared at gmail.com Tue Jan 4 18:33:44 2011 From: braunsquared at gmail.com (Timothy Braun) Date: Tue, 4 Jan 2011 12:33:44 -0500 Subject: [gst-devel] Video transitions using gnloperation In-Reply-To: <4D235254.3040103@brooks.nu> References: <4D22BCF2.90006@brooks.nu> <1294144782.23475.5.camel@localhost> <4D235254.3040103@brooks.nu> Message-ID: Make sure the gnl input sources have different priorities and set the gnloperations priority to 0. Sent from my iPhone On Jan 4, 2011, at 12:01 PM, Lane Brooks wrote: > On 01/04/2011 05:39 AM, Edward Hervey wrote: >> On Mon, 2011-01-03 at 23:23 -0700, Lane Brooks wrote: >>> I have a gnlcomposition containing two video sources. I would like to >>> apply a smpte transition from one source to the next. >>> >>> I have tried overlapping the two gnlsources by 3 seconds. I then put a >>> 'smpte' element in a 'gnloperation' and add the 'gnloperation' to the >>> 'gnlcompostion'. The addition of the gnloperation doesn't seem to do >>> anything. The resulting video is the same regardless of whether I add >>> the gnloperation or not. The result is always that the second video >>> takes over during the period of overlap. Here is code I have (in python): >>> >>> comp = gst.element_factory_make("gnlcomposition") >>> src1 = gst.element_factory_make("gnlfilesource") >>> src2 = gst.element_factory_make("gnlfilesource") >>> >>> comp.add(src1) >>> src1.set_property("location", "/test1.mp4") >>> src1.set_property("start", 0 * gst.SECOND) >>> src1.set_property("duration", 5 * gst.SECOND) >>> src1.set_property("media-start", 0 * gst.SECOND) >>> src1.set_property("media-duration", 5 * gst.SECOND) >>> >>> comp.add(src2) >>> src2.set_property("location", "/test2.mp4") >>> src2.set_property("start", 2 * gst.SECOND) >>> src2.set_property("duration", 5 * gst.SECOND) >>> src2.set_property("media-start", 0 * gst.SECOND) >>> src2.set_property("media-duration", 5 * gst.SECOND) >>> >>> op = gst.element_factory_make("gnloperation") >>> tran = gst.element_factory_make("smpte") >>> tran.set_property("type", 234) >>> tran.set_property("duration", 3 * gst.SECOND) >>> op.add(tran) >> >> Set the [media-]{start|duration} propertie son op also > > I had already tried this as well. > > op.set_property("start", 2 * gst.SECOND) > op.set_property("duration", 3 * gst.SECOND) > op.set_property("media-start", 0 * gst.SECOND) > op.set_property("media-duration", 3 * gst.SECOND) > > I am not sure what I should put the media-start at, but I have tried > both 0 and 2 and neither works. Any other ideas? > > >> >>> comp.add(op) >>> >>> Anyone know if using an smpte transition is possible, and if so, what I >>> am doing wrong? >>> >>> Thanks, >>> Lane > > ------------------------------------------------------------------------------ > 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 bilboed at gmail.com Tue Jan 4 18:35:03 2011 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 04 Jan 2011 18:35:03 +0100 Subject: [gst-devel] is gstreamer git hacked? In-Reply-To: <4D235158.5090000@lfarkas.org> References: <4D235158.5090000@lfarkas.org> Message-ID: <1294162503.23475.6.camel@localhost> On Tue, 2011-01-04 at 17:56 +0100, Farkas Levente wrote: > hi, > i just read the docs: > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/pwg/html/chapter-building-signals.html > then click the link "GObject documentation" :-) > imho it's not totally normal... The content of where it points to was completely out of our control. I wouldn't call that "being hacked" :) Changing it to something more up-to-date. Edward > From bisht.sudarshan at gmail.com Tue Jan 4 18:54:32 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 4 Jan 2011 23:24:32 +0530 Subject: [gst-devel] install gst-plugins-gl-0.10.2 In-Reply-To: References: Message-ID: No idea about Glib. But just a guess again, check LD_LIBRARY_PATH and other environment variables. Since your Gstreamer installation is there in /usr/local/lib not in /usr/lib. On Tue, Jan 4, 2011 at 10:38 PM, henriette de poulpiquet < henriette.depoulpiquet at gmail.com> wrote: > maybe my Glib version is important? > Glib 2.26.1 > > .. no one ever had this problem? > > anyway, thanks Sudarshan for your help! > > Henriette > > On 1/4/11, henriette de poulpiquet > wrote: > > yes I did... > > > > > > On 1/4/11, sudarshan bisht wrote: > >> Did you do " export GST_PLUGIN_PATH=/usr/local/lib/gstreamer-0.10" > ? > >> > >> On Tue, Jan 4, 2011 at 1:47 PM, henriette de poulpiquet < > >> henriette.depoulpiquet at gmail.com> wrote: > >> > >>> Hi! > >>> > >>> I'me not sure where to post my question, hope i'me in the right place! > >>> > >>> so my problem is that my gst-inspect indicate that gl plugin is > >>> blacklisted, I'me quite disappointed and can't understand why > >>> /usr/local/lib/gstreamer-0.10/libgstopengl.so can' t be found since > >>> libgstopengl.so do exists at this place (I checked it). > >>> > >>> what should I do? > >>> > >>> thanks for your answer! > >>> > >>> > >>> Henriette > >>> > >>> i'me working on a > >>> Linux: Debian 32 bits > >>> Release: 2.6.26-2-686 > >>> i686 > >>> gstreamer-0.10.31 > >>> gst-plugins-base-0.10.31 > >>> gst-plugins-good-0.10.26 > >>> > >>> > >>> > ------------------------------------------------------------------------------ > >>> 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 > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From lane at brooks.nu Tue Jan 4 18:56:55 2011 From: lane at brooks.nu (Lane Brooks) Date: Tue, 04 Jan 2011 10:56:55 -0700 Subject: [gst-devel] Video transitions using gnloperation In-Reply-To: References: <4D22BCF2.90006@brooks.nu> <1294144782.23475.5.camel@localhost> <4D235254.3040103@brooks.nu> Message-ID: <4D235F67.2010200@brooks.nu> On 01/04/2011 10:33 AM, Timothy Braun wrote: > Make sure the gnl input sources have different priorities and set the gnloperations priority to 0. Thanks for the idea. I tried setting the priority of the op to 0, src1 to 1, and src2 to 2. Now when I run it the stream hangs after 2 seconds, which is when the gnloperation and src2 are starting up. I did some testing and the hanging only occurs if I set the priority on src2 to anything other than 0. I can set the priorities on the op and src1 without hanging (it still doesn't work, however). Here is my code again with the updates: comp = gst.element_factory_make("gnlcomposition") src1 = gst.element_factory_make("gnlfilesource") src2 = gst.element_factory_make("gnlfilesource") comp.add(src1) src1.set_property("location", "/test1.mp4") src1.set_property("start", 0 * gst.SECOND) src1.set_property("duration", 5 * gst.SECOND) src1.set_property("media-start", 0 * gst.SECOND) src1.set_property("media-duration", 5 * gst.SECOND) src1.set_property("priority", 1) comp.add(src2) src2.set_property("location", "/test2.mp4") src2.set_property("start", 2 * gst.SECOND) src2.set_property("duration", 5 * gst.SECOND) src2.set_property("media-start", 0 * gst.SECOND) src2.set_property("media-duration", 5 * gst.SECOND) src2.set_property("priority", 2) op = gst.element_factory_make("gnloperation") tran = gst.element_factory_make("smpte") tran.set_property("type", 234) tran.set_property("duration", 3 * gst.SECOND) op.add(tran) op.set_property("start", 2 * gst.SECOND) op.set_property("duration", 3 * gst.SECOND) op.set_property("media-start", 0 * gst.SECOND) op.set_property("media-duration", 3 * gst.SECOND) op.set_property("priority", 0) comp.add(op) I can get the smpte transition to work correctly on these two streams when not using the gnonlin elements, but that is not very interesting as I have to run the two streams in parallel. Any other ideas? Lane > On Jan 4, 2011, at 12:01 PM, Lane Brooks wrote: > >> On 01/04/2011 05:39 AM, Edward Hervey wrote: >>> On Mon, 2011-01-03 at 23:23 -0700, Lane Brooks wrote: >>>> I have a gnlcomposition containing two video sources. I would like to >>>> apply a smpte transition from one source to the next. >>>> >>>> I have tried overlapping the two gnlsources by 3 seconds. I then put a >>>> 'smpte' element in a 'gnloperation' and add the 'gnloperation' to the >>>> 'gnlcompostion'. The addition of the gnloperation doesn't seem to do >>>> anything. The resulting video is the same regardless of whether I add >>>> the gnloperation or not. The result is always that the second video >>>> takes over during the period of overlap. Here is code I have (in python): >>>> >>>> comp = gst.element_factory_make("gnlcomposition") >>>> src1 = gst.element_factory_make("gnlfilesource") >>>> src2 = gst.element_factory_make("gnlfilesource") >>>> >>>> comp.add(src1) >>>> src1.set_property("location", "/test1.mp4") >>>> src1.set_property("start", 0 * gst.SECOND) >>>> src1.set_property("duration", 5 * gst.SECOND) >>>> src1.set_property("media-start", 0 * gst.SECOND) >>>> src1.set_property("media-duration", 5 * gst.SECOND) >>>> >>>> comp.add(src2) >>>> src2.set_property("location", "/test2.mp4") >>>> src2.set_property("start", 2 * gst.SECOND) >>>> src2.set_property("duration", 5 * gst.SECOND) >>>> src2.set_property("media-start", 0 * gst.SECOND) >>>> src2.set_property("media-duration", 5 * gst.SECOND) >>>> >>>> op = gst.element_factory_make("gnloperation") >>>> tran = gst.element_factory_make("smpte") >>>> tran.set_property("type", 234) >>>> tran.set_property("duration", 3 * gst.SECOND) >>>> op.add(tran) >>> >>> Set the [media-]{start|duration} propertie son op also >> >> I had already tried this as well. >> >> op.set_property("start", 2 * gst.SECOND) >> op.set_property("duration", 3 * gst.SECOND) >> op.set_property("media-start", 0 * gst.SECOND) >> op.set_property("media-duration", 3 * gst.SECOND) >> >> I am not sure what I should put the media-start at, but I have tried >> both 0 and 2 and neither works. Any other ideas? >> >> >>> >>>> comp.add(op) >>>> >>>> Anyone know if using an smpte transition is possible, and if so, what I >>>> am doing wrong? >>>> >>>> Thanks, >>>> Lane From lane at brooks.nu Tue Jan 4 18:56:55 2011 From: lane at brooks.nu (Lane Brooks) Date: Tue, 04 Jan 2011 10:56:55 -0700 Subject: [gst-devel] Video transitions using gnloperation In-Reply-To: References: <4D22BCF2.90006@brooks.nu> <1294144782.23475.5.camel@localhost> <4D235254.3040103@brooks.nu> Message-ID: <4D235F67.2010200@brooks.nu> On 01/04/2011 10:33 AM, Timothy Braun wrote: > Make sure the gnl input sources have different priorities and set the gnloperations priority to 0. Thanks for the idea. I tried setting the priority of the op to 0, src1 to 1, and src2 to 2. Now when I run it the stream hangs after 2 seconds, which is when the gnloperation and src2 are starting up. I did some testing and the hanging only occurs if I set the priority on src2 to anything other than 0. I can set the priorities on the op and src1 without hanging (it still doesn't work, however). Here is my code again with the updates: comp = gst.element_factory_make("gnlcomposition") src1 = gst.element_factory_make("gnlfilesource") src2 = gst.element_factory_make("gnlfilesource") comp.add(src1) src1.set_property("location", "/test1.mp4") src1.set_property("start", 0 * gst.SECOND) src1.set_property("duration", 5 * gst.SECOND) src1.set_property("media-start", 0 * gst.SECOND) src1.set_property("media-duration", 5 * gst.SECOND) src1.set_property("priority", 1) comp.add(src2) src2.set_property("location", "/test2.mp4") src2.set_property("start", 2 * gst.SECOND) src2.set_property("duration", 5 * gst.SECOND) src2.set_property("media-start", 0 * gst.SECOND) src2.set_property("media-duration", 5 * gst.SECOND) src2.set_property("priority", 2) op = gst.element_factory_make("gnloperation") tran = gst.element_factory_make("smpte") tran.set_property("type", 234) tran.set_property("duration", 3 * gst.SECOND) op.add(tran) op.set_property("start", 2 * gst.SECOND) op.set_property("duration", 3 * gst.SECOND) op.set_property("media-start", 0 * gst.SECOND) op.set_property("media-duration", 3 * gst.SECOND) op.set_property("priority", 0) comp.add(op) I can get the smpte transition to work correctly on these two streams when not using the gnonlin elements, but that is not very interesting as I have to run the two streams in parallel. Any other ideas? Lane > On Jan 4, 2011, at 12:01 PM, Lane Brooks wrote: > >> On 01/04/2011 05:39 AM, Edward Hervey wrote: >>> On Mon, 2011-01-03 at 23:23 -0700, Lane Brooks wrote: >>>> I have a gnlcomposition containing two video sources. I would like to >>>> apply a smpte transition from one source to the next. >>>> >>>> I have tried overlapping the two gnlsources by 3 seconds. I then put a >>>> 'smpte' element in a 'gnloperation' and add the 'gnloperation' to the >>>> 'gnlcompostion'. The addition of the gnloperation doesn't seem to do >>>> anything. The resulting video is the same regardless of whether I add >>>> the gnloperation or not. The result is always that the second video >>>> takes over during the period of overlap. Here is code I have (in python): >>>> >>>> comp = gst.element_factory_make("gnlcomposition") >>>> src1 = gst.element_factory_make("gnlfilesource") >>>> src2 = gst.element_factory_make("gnlfilesource") >>>> >>>> comp.add(src1) >>>> src1.set_property("location", "/test1.mp4") >>>> src1.set_property("start", 0 * gst.SECOND) >>>> src1.set_property("duration", 5 * gst.SECOND) >>>> src1.set_property("media-start", 0 * gst.SECOND) >>>> src1.set_property("media-duration", 5 * gst.SECOND) >>>> >>>> comp.add(src2) >>>> src2.set_property("location", "/test2.mp4") >>>> src2.set_property("start", 2 * gst.SECOND) >>>> src2.set_property("duration", 5 * gst.SECOND) >>>> src2.set_property("media-start", 0 * gst.SECOND) >>>> src2.set_property("media-duration", 5 * gst.SECOND) >>>> >>>> op = gst.element_factory_make("gnloperation") >>>> tran = gst.element_factory_make("smpte") >>>> tran.set_property("type", 234) >>>> tran.set_property("duration", 3 * gst.SECOND) >>>> op.add(tran) >>> >>> Set the [media-]{start|duration} propertie son op also >> >> I had already tried this as well. >> >> op.set_property("start", 2 * gst.SECOND) >> op.set_property("duration", 3 * gst.SECOND) >> op.set_property("media-start", 0 * gst.SECOND) >> op.set_property("media-duration", 3 * gst.SECOND) >> >> I am not sure what I should put the media-start at, but I have tried >> both 0 and 2 and neither works. Any other ideas? >> >> >>> >>>> comp.add(op) >>>> >>>> Anyone know if using an smpte transition is possible, and if so, what I >>>> am doing wrong? >>>> >>>> Thanks, >>>> Lane From braunsquared at gmail.com Tue Jan 4 19:08:24 2011 From: braunsquared at gmail.com (Timothy Braun) Date: Tue, 4 Jan 2011 13:08:24 -0500 Subject: [gst-devel] Video transitions using gnloperation In-Reply-To: References: <4D22BCF2.90006@brooks.nu> <1294144782.23475.5.camel@localhost> <4D235254.3040103@brooks.nu> <4D235F67.2010200@brooks.nu> Message-ID: ---------- Forwarded message ---------- From: Timothy Braun Date: Tue, Jan 4, 2011 at 1:08 PM Subject: Re: Video transitions using gnloperation To: Lane Brooks What do you get when you set GST_DEBUG=2? The pipeline is most likely freezing because of a caps filter or a pad issue. What is the pipeline connected to for output? On Tue, Jan 4, 2011 at 12:56 PM, Lane Brooks wrote: > On 01/04/2011 10:33 AM, Timothy Braun wrote: > >> Make sure the gnl input sources have different priorities and set the >> gnloperations priority to 0. >> > > Thanks for the idea. I tried setting the priority of the op to 0, src1 to > 1, and src2 to 2. Now when I run it the stream hangs after 2 seconds, which > is when the gnloperation and src2 are starting up. I did some testing and > the hanging only occurs if I set the priority on src2 to anything other than > 0. I can set the priorities on the op and src1 without hanging (it still > doesn't work, however). Here is my code again with the updates: > > > comp = gst.element_factory_make("gnlcomposition") > src1 = gst.element_factory_make("gnlfilesource") > src2 = gst.element_factory_make("gnlfilesource") > > comp.add(src1) > src1.set_property("location", "/test1.mp4") > src1.set_property("start", 0 * gst.SECOND) > src1.set_property("duration", 5 * gst.SECOND) > src1.set_property("media-start", 0 * gst.SECOND) > src1.set_property("media-duration", 5 * gst.SECOND) > src1.set_property("priority", 1) > > > comp.add(src2) > src2.set_property("location", "/test2.mp4") > src2.set_property("start", 2 * gst.SECOND) > src2.set_property("duration", 5 * gst.SECOND) > src2.set_property("media-start", 0 * gst.SECOND) > src2.set_property("media-duration", 5 * gst.SECOND) > src2.set_property("priority", 2) > > > op = gst.element_factory_make("gnloperation") > tran = gst.element_factory_make("smpte") > tran.set_property("type", 234) > tran.set_property("duration", 3 * gst.SECOND) > op.add(tran) > op.set_property("start", 2 * gst.SECOND) > op.set_property("duration", 3 * gst.SECOND) > op.set_property("media-start", 0 * gst.SECOND) > op.set_property("media-duration", 3 * gst.SECOND) > op.set_property("priority", 0) > comp.add(op) > > I can get the smpte transition to work correctly on these two streams when > not using the gnonlin elements, but that is not very interesting as I have > to run the two streams in parallel. > > Any other ideas? > > Lane > > > > On Jan 4, 2011, at 12:01 PM, Lane Brooks wrote: >> >> On 01/04/2011 05:39 AM, Edward Hervey wrote: >>> >>>> On Mon, 2011-01-03 at 23:23 -0700, Lane Brooks wrote: >>>> >>>>> I have a gnlcomposition containing two video sources. I would like to >>>>> apply a smpte transition from one source to the next. >>>>> >>>>> I have tried overlapping the two gnlsources by 3 seconds. I then put a >>>>> 'smpte' element in a 'gnloperation' and add the 'gnloperation' to the >>>>> 'gnlcompostion'. The addition of the gnloperation doesn't seem to do >>>>> anything. The resulting video is the same regardless of whether I add >>>>> the gnloperation or not. The result is always that the second video >>>>> takes over during the period of overlap. Here is code I have (in >>>>> python): >>>>> >>>>> comp = gst.element_factory_make("gnlcomposition") >>>>> src1 = gst.element_factory_make("gnlfilesource") >>>>> src2 = gst.element_factory_make("gnlfilesource") >>>>> >>>>> comp.add(src1) >>>>> src1.set_property("location", "/test1.mp4") >>>>> src1.set_property("start", 0 * gst.SECOND) >>>>> src1.set_property("duration", 5 * gst.SECOND) >>>>> src1.set_property("media-start", 0 * gst.SECOND) >>>>> src1.set_property("media-duration", 5 * gst.SECOND) >>>>> >>>>> comp.add(src2) >>>>> src2.set_property("location", "/test2.mp4") >>>>> src2.set_property("start", 2 * gst.SECOND) >>>>> src2.set_property("duration", 5 * gst.SECOND) >>>>> src2.set_property("media-start", 0 * gst.SECOND) >>>>> src2.set_property("media-duration", 5 * gst.SECOND) >>>>> >>>>> op = gst.element_factory_make("gnloperation") >>>>> tran = gst.element_factory_make("smpte") >>>>> tran.set_property("type", 234) >>>>> tran.set_property("duration", 3 * gst.SECOND) >>>>> op.add(tran) >>>>> >>>> >>>> Set the [media-]{start|duration} propertie son op also >>>> >>> >>> I had already tried this as well. >>> >>> op.set_property("start", 2 * gst.SECOND) >>> op.set_property("duration", 3 * gst.SECOND) >>> op.set_property("media-start", 0 * gst.SECOND) >>> op.set_property("media-duration", 3 * gst.SECOND) >>> >>> I am not sure what I should put the media-start at, but I have tried >>> both 0 and 2 and neither works. Any other ideas? >>> >>> >>> >>>> comp.add(op) >>>>> >>>>> Anyone know if using an smpte transition is possible, and if so, what I >>>>> am doing wrong? >>>>> >>>>> Thanks, >>>>> Lane >>>>> >>>> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From lane at brooks.nu Tue Jan 4 19:28:31 2011 From: lane at brooks.nu (Lane Brooks) Date: Tue, 04 Jan 2011 11:28:31 -0700 Subject: [gst-devel] Video transitions using gnloperation In-Reply-To: References: <4D22BCF2.90006@brooks.nu> <1294144782.23475.5.camel@localhost> <4D235254.3040103@brooks.nu> <4D235F67.2010200@brooks.nu> Message-ID: <4D2366CF.6020009@brooks.nu> A little more information: src2 likes to have a higher priority that src1. If I remove the gnloperation completely and run only with gnlfilesources for src1 and src2, if src2 has a higher priority than src1, the pipeline hangs when src2 turns on. So if I then add the gnloperation with priority 0 and set src2 priority=1 and src1 priority=2, the pipeline hangs when the gnloperation turns on. I verify this by having src1 turn at time 0 second. Then src2 turns on at time 2 second, and it takes over as seen previously. Then I have gnloperation turn on at 3 seconds, and that is when the pipeline hangs. See the answers to questions below: On 01/04/2011 11:08 AM, Timothy Braun wrote: > What do you get when you set GST_DEBUG=2? The pipeline is most likely > freezing because of a caps filter or a pad issue. Here is the output I get when I run wiht GST_DEBUG=2. This is the same output I get if I remove the gnloperation all together, so I am not sure it is related to the issue I am having. 0:00:00.038328446 1353 0x1ed8af0 WARN gnlsource gnlsource.c:545:gnl_source_change_state: Couldn't find a valid source pad 0:00:00.054148884 1353 0x22ccd20 WARN qtdemux qtdemux_types.c:170:qtdemux_type_get: unknown QuickTime node type avc1 0:00:00.054183522 1353 0x22ccd20 WARN qtdemux qtdemux_types.c:170:qtdemux_type_get: unknown QuickTime node type avcC 0:00:01.354701757 1353 0x1ed8af0 WARN gnlsource gnlsource.c:545:gnl_source_change_state: Couldn't find a valid source pad 0:00:01.369736423 1353 0x20aad30 WARN qtdemux qtdemux_types.c:170:qtdemux_type_get: unknown QuickTime node type avc1 0:00:01.369785865 1353 0x20aad30 WARN qtdemux qtdemux_types.c:170:qtdemux_type_get: unknown QuickTime node type avcC > What is the pipeline connected to for output? The output is connected to a "queue" and an "autovideosink". The pipeline runs fine when no gnloperation is in place, and the second video takes priority. When I add the gnloperation but do not set any priorities, the pipeline runs as if the gnloperation is not there. When I set the priority on src2 to anything other than 1, then the pipeline hangs. By changing around the "start" time property on the gnloperation, the hanging seems to correspond to whenever the gnloperation starts up. Lane > > On Tue, Jan 4, 2011 at 12:56 PM, Lane Brooks > wrote: > > On 01/04/2011 10:33 AM, Timothy Braun wrote: > > Make sure the gnl input sources have different priorities and > set the gnloperations priority to 0. > > > Thanks for the idea. I tried setting the priority of the op to 0, > src1 to 1, and src2 to 2. Now when I run it the stream hangs after > 2 seconds, which is when the gnloperation and src2 are starting > up. I did some testing and the hanging only occurs if I set the > priority on src2 to anything other than 0. I can set the > priorities on the op and src1 without hanging (it still doesn't > work, however). Here is my code again with the updates: > > > comp = gst.element_factory_make("gnlcomposition") > src1 = gst.element_factory_make("gnlfilesource") > src2 = gst.element_factory_make("gnlfilesource") > > comp.add(src1) > src1.set_property("location", "/test1.mp4") > src1.set_property("start", 0 * gst.SECOND) > src1.set_property("duration", 5 * gst.SECOND) > src1.set_property("media-start", 0 * gst.SECOND) > src1.set_property("media-duration", 5 * gst.SECOND) > src1.set_property("priority", 1) > > > comp.add(src2) > src2.set_property("location", "/test2.mp4") > src2.set_property("start", 2 * gst.SECOND) > src2.set_property("duration", 5 * gst.SECOND) > src2.set_property("media-start", 0 * gst.SECOND) > src2.set_property("media-duration", 5 * gst.SECOND) > src2.set_property("priority", 2) > > > op = gst.element_factory_make("gnloperation") > tran = gst.element_factory_make("smpte") > tran.set_property("type", 234) > tran.set_property("duration", 3 * gst.SECOND) > op.add(tran) > op.set_property("start", 2 * gst.SECOND) > op.set_property("duration", 3 * gst.SECOND) > op.set_property("media-start", 0 * gst.SECOND) > op.set_property("media-duration", 3 * gst.SECOND) > op.set_property("priority", 0) > comp.add(op) > > I can get the smpte transition to work correctly on these two > streams when not using the gnonlin elements, but that is not very > interesting as I have to run the two streams in parallel. > > Any other ideas? > > Lane > > > > On Jan 4, 2011, at 12:01 PM, Lane Brooks > wrote: > > On 01/04/2011 05:39 AM, Edward Hervey wrote: > > On Mon, 2011-01-03 at 23:23 -0700, Lane Brooks wrote: > > I have a gnlcomposition containing two video > sources. I would like to > apply a smpte transition from one source to the next. > > I have tried overlapping the two gnlsources by 3 > seconds. I then put a > 'smpte' element in a 'gnloperation' and add the > 'gnloperation' to the > 'gnlcompostion'. The addition of the gnloperation > doesn't seem to do > anything. The resulting video is the same > regardless of whether I add > the gnloperation or not. The result is always that > the second video > takes over during the period of overlap. Here is > code I have (in python): > > comp = > gst.element_factory_make("gnlcomposition") > src1 = > gst.element_factory_make("gnlfilesource") > src2 = > gst.element_factory_make("gnlfilesource") > > comp.add(src1) > src1.set_property("location", "/test1.mp4") > src1.set_property("start", 0 * > gst.SECOND) > src1.set_property("duration", 5 * > gst.SECOND) > src1.set_property("media-start", 0 * > gst.SECOND) > src1.set_property("media-duration", 5 * > gst.SECOND) > > comp.add(src2) > src2.set_property("location", "/test2.mp4") > src2.set_property("start", 2 * > gst.SECOND) > src2.set_property("duration", 5 * > gst.SECOND) > src2.set_property("media-start", 0 * > gst.SECOND) > src2.set_property("media-duration", 5 * > gst.SECOND) > > op = gst.element_factory_make("gnloperation") > tran = gst.element_factory_make("smpte") > tran.set_property("type", 234) > tran.set_property("duration", 3 * gst.SECOND) > op.add(tran) > > > Set the [media-]{start|duration} propertie son op also > > > I had already tried this as well. > > op.set_property("start", 2 * gst.SECOND) > op.set_property("duration", 3 * gst.SECOND) > op.set_property("media-start", 0 * gst.SECOND) > op.set_property("media-duration", 3 * gst.SECOND) > > I am not sure what I should put the media-start at, but I > have tried > both 0 and 2 and neither works. Any other ideas? > > > > comp.add(op) > > Anyone know if using an smpte transition is > possible, and if so, what I > am doing wrong? > > Thanks, > Lane > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wmiller at sdr.com Tue Jan 4 19:52:28 2011 From: wmiller at sdr.com (Wes Miller) Date: Tue, 4 Jan 2011 10:52:28 -0800 (PST) Subject: [gst-devel] how do write a gst mixer in c In-Reply-To: <1294139524319-3173361.post@n4.nabble.com> References: <1294139524319-3173361.post@n4.nabble.com> Message-ID: <1294167148876-3174134.post@n4.nabble.com> HI, First, please note that your posting did not make it to the real gst-devel forum. You need to be a member of the forum itself to post to it. Nabble is a separate (wonderful) interface to gstreamer forums. And second, are you using gst-launch or are you writing in c/c++ or python or java? What operating system? Gstreamer on Windows is not quite as complete a solution as it is on Linux. Now, Is there some reason you can't use one of the existing mixers? ossmixer oss4mixer alsamixer pulsemixer videomixer of maybe just use the volume element to control volume? To get information out of a pipeline you can use "fakesink -v " which will tell you a LOT about your pipeline and its data. Also search this forum for a post by me (wes miller) asking about a programmatic replacement for fakesink (answer uses a callback to deep-notify). Also see this forum for questions I've posted about getting the values of element properties. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/how-do-write-a-gst-mixer-in-c-tp3173361p3174134.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From c.r.n.a at wanadoo.fr Tue Jan 4 20:02:30 2011 From: c.r.n.a at wanadoo.fr (Nicolas) Date: Tue, 04 Jan 2011 20:02:30 +0100 Subject: [gst-devel] Need help for rezising video. Message-ID: <4D236EC6.7030608@wanadoo.fr> Hi, Thanks for your reply. Ok, i do: pipeline = gst_pipeline_new ("pipeline"); playbin = gst_element_factory_make ("playbin2", "playbin"); // Filter filter = gst_element_factory_make ("capsfilter", "filter"); caps = gst_caps_new_simple ("video/x-raw-yuv", "format", GST_TYPE_FOURCC, GST_MAKE_FOURCC ('I', '4', '2', '0'), "width", G_TYPE_INT, 800, "height", G_TYPE_INT, 600, NULL); g_object_set ((GObject*) filter, "caps", caps, NULL); // My sink sink = gst_element_factory_make ("evasvideosink", "sink"); g_object_set ((GObject*) playbin, "video-sink", sink, NULL); // Tell to playbin2 to use my sink g_object_set ((GObject*) playbin, "uri", argv[1], NULL); g_object_set ((GObject*) sink, "evas-object", video, NULL); // Tell my sink to use evas image gst_bin_add_many (GST_BIN (pipeline), playbin, filter, NULL); gst_element_set_state (pipeline, GST_STATE_PLAYING); It does not change the size of the video, is this the correct way ? Thanks in advance for your reply, Nicolas. From braunsquared at gmail.com Tue Jan 4 20:18:38 2011 From: braunsquared at gmail.com (Timothy Braun) Date: Tue, 4 Jan 2011 14:18:38 -0500 Subject: [gst-devel] Need help for rezising video. In-Reply-To: <4D236EC6.7030608@wanadoo.fr> References: <4D236EC6.7030608@wanadoo.fr> Message-ID: The capsfilter isn't going to do anything for you unless something in the pipeline is able to make the needed changes to match the caps. Before the capsfilter you should add a videoscale element so it is able to match the requested caps of the filter. You may also want to take a gander at the ffmpeg_colorspace and videorate elements as well. As a side note: You can simplify the creation of the caps with the gst_caps_from_string( "video/x-raw-yuv, format=I420, width=800, height=600") call. You need to unref the caps when you're done with it as well. Just my 2c. Best, Tim On Tue, Jan 4, 2011 at 2:02 PM, Nicolas wrote: > Hi, > > Thanks for your reply. > > Ok, i do: > > pipeline = gst_pipeline_new ("pipeline"); > playbin = gst_element_factory_make ("playbin2", "playbin"); > > // Filter > filter = gst_element_factory_make ("capsfilter", "filter"); > caps = gst_caps_new_simple ("video/x-raw-yuv", "format", > GST_TYPE_FOURCC, GST_MAKE_FOURCC ('I', '4', '2', '0'), "width", > G_TYPE_INT, 800, "height", G_TYPE_INT, 600, NULL); > g_object_set ((GObject*) filter, "caps", caps, NULL); > > // My sink > sink = gst_element_factory_make ("evasvideosink", "sink"); > g_object_set ((GObject*) playbin, "video-sink", sink, NULL); // Tell to > playbin2 to use my sink > g_object_set ((GObject*) playbin, "uri", argv[1], NULL); > g_object_set ((GObject*) sink, "evas-object", video, NULL); // Tell my > sink to use evas image > > gst_bin_add_many (GST_BIN (pipeline), playbin, filter, NULL); > > gst_element_set_state (pipeline, GST_STATE_PLAYING); > > It does not change the size of the video, is this the correct way ? > > Thanks in advance for your reply, > Nicolas. > > > ------------------------------------------------------------------------------ > 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 lfarkas at lfarkas.org Tue Jan 4 19:57:24 2011 From: lfarkas at lfarkas.org (Farkas Levente) Date: Tue, 04 Jan 2011 19:57:24 +0100 Subject: [gst-devel] is gstreamer git hacked? In-Reply-To: <1294162503.23475.6.camel@localhost> References: <4D235158.5090000@lfarkas.org> <1294162503.23475.6.camel@localhost> Message-ID: <4D236D94.5010403@lfarkas.org> On 01/04/2011 06:35 PM, Edward Hervey wrote: > On Tue, 2011-01-04 at 17:56 +0100, Farkas Levente wrote: >> hi, >> i just read the docs: >> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/pwg/html/chapter-building-signals.html >> then click the link "GObject documentation" :-) >> imho it's not totally normal... > > The content of where it points to was completely out of our control. I > wouldn't call that "being hacked" :) > > Changing it to something more up-to-date. may be a link like http://www.le-hacker.org/papers/gobject/index.html never can be too authentic, something like this http://library.gnome.org/devel/gobject/unstable/signal.html should be more stable:-) -- Levente "Si vis pacem para bellum!" From shiva.varma at gmail.com Tue Jan 4 20:29:32 2011 From: shiva.varma at gmail.com (shiva varma) Date: Tue, 4 Jan 2011 11:29:32 -0800 (PST) Subject: [gst-devel] How to extract G711 u-law audio from RTP packets In-Reply-To: <1294133035.28802.58.camel@meany> References: <1294086669869-3172578.post@n4.nabble.com> <1294133035.28802.58.camel@meany> Message-ID: <1294169372219-3174203.post@n4.nabble.com> Hi Wim, Thanks for the response -Shiva -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-extract-G711-u-law-audio-from-RTP-packets-tp3172578p3174203.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From mgruenke at Tycoint.com Wed Jan 5 01:04:03 2011 From: mgruenke at Tycoint.com (Gruenke, Matt) Date: Tue, 4 Jan 2011 19:04:03 -0500 Subject: [gst-devel] GObjectClass.dispose() and GstElement Message-ID: I noticed the GObject docs (http://library.gnome.org/devel/gobject/unstable/gobject-The-Base-Object -Type.html#GObjectClass) say: Before returning, dispose should chain up to the dispose method of the parent class. This contradicts what I said about having GstElement provide a dispose hook for derived types. The descriptions of GObjectClass.dispose() and GObjectClass.finalize() make it unclear where deallocation of property values should happen, but I'd imagine it would be fine to do this from dispose(). I'm guessing that the clause "client method invocations still work" after dispose() refers mostly to GObject methods and "still work" simply means no segfaults. So, going back to my original concern, what about simply adding the following member to GstElementClass? void (*free_config) (GstElement *element); If not, then how about at least describing how to tie into GObject.dispose(), in the Plugin Writer's Guide, and maybe adding an example to the templates? Also, I noticed that section 23.1 of the Plugin Writer's Guide says: Make sure the state of an element gets reset when going to NULL. Ideally, this should set all object properties to their original state. Again, why not add a hook in GstElementClass to support this? Most people probably miss it, and it forces you to tie into state transitions - even if you're using a pre-made baseclass whose start() and stop() hooks seem to imply that doing so is unnecessary. Matt ________________________________ From: Gruenke, Matt Sent: Monday, January 03, 2011 17:57 To: gstreamer-devel at lists.sourceforge.net Subject: GObjectClass.dispose() and GstElement Based on my reading of the Plugin Writer's Guide & the GstElement reference docs, elements are supposed to complete all of their deallocation on the READY -> NULL transition. However, it's always bothered me that there's a case where you could still get a memory leak. Suppose an element's GObjectClass.set_property() gets called while the element is still in the NULL state. If this property accepts a string or some other type of heap-allocated object, the element is probably going to create a copy or add a ref. Now, suppose some problem occurs and the element is disposed before ever going to the READY state. There will be no transition back to NULL. Therefore, the only possibility of deallocating the value is by overriding GstObjetClass.dispose(). I concede that this isn't normally an issue, since the size of any such properties is likely to be very small and many applications will either exit or typically repeat this sequence a limited number of times. However, it's quite possible that some sort of automated recording application might repeat this sequence somewhat rapidly and for an indefinite period of time. Then, there's the whole philosophical issue of avoiding even small leaks, which turns into a very practical issue when trying to use tools to find slightly more serious leaks. I know you can usually work around this, through the use of filtering, but it's inconvenient and not always satisfactory... let's please try to avoid a flame war about this. Would it be so bad for GstObjectClass and GstElementClass to at least provide a dispose() hook? I'm aware that I could just save the old value & replace with my own dispose(), but something feels wrong about having the derived type dispose of its ancestors. It's also not a very user-friendly way to do it, since there are numerous ways the derived type can mess things up (i.e. not saving the old value or failing to call it last). Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: From shiva.varma at gmail.com Wed Jan 5 02:38:53 2011 From: shiva.varma at gmail.com (shiva varma) Date: Tue, 4 Jan 2011 17:38:53 -0800 (PST) Subject: [gst-devel] How to convert G722 audio from pcap to file format Message-ID: <1294191533311-3174763.post@n4.nabble.com> I have the following pipeline gst-launch filesrc location=g722.pcap ! pcapparse src-ip=10.22.120.89 src-port=27812 ! ! "application/x-rtp, payload=9, clock-rate=8000" ! rtpg722depay ! ................ After the rtpg722depay, i need an encoder which takes the audio/G722 as its sink and produces an output format which can save to file My attempt to find a suitable encoder among the existing plugins have been futile, can someone point me to the plugin I can use for this purpose Thanks Shiva -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-convert-G722-audio-from-pcap-to-file-format-tp3174763p3174763.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From rohitratri at gmail.com Wed Jan 5 05:04:44 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Wed, 5 Jan 2011 09:34:44 +0530 Subject: [gst-devel] Need help for rezising video. In-Reply-To: References: <4D236EC6.7030608@wanadoo.fr> Message-ID: @Nicolas You could use the capsfilter element like this as well - caps = gst_caps_new_simple ("video/x-raw-rgb", "bpp", G_TYPE_INT, 24, "depth", G_TYPE_INT, 24, NULL); gst_element_link_filtered(capsfilter, videosink, caps); (Considering videoscale element is already linked to capsfilter on to its left.) instead of doing a gst_object_set()... Of course the choice between the 2 depends on your convenience. Rohti On Wed, Jan 5, 2011 at 12:48 AM, Timothy Braun wrote: > The capsfilter isn't going to do anything for you unless something in the > pipeline is able to make the needed changes to match the caps. Before the > capsfilter you should add a videoscale element so it is able to match the > requested caps of the filter. You may also want to take a gander at the > ffmpeg_colorspace and videorate elements as well. > > As a side note: > You can simplify the creation of the caps with the gst_caps_from_string( > "video/x-raw-yuv, format=I420, width=800, height=600") call. > You need to unref the caps when you're done with it as well. > > Just my 2c. > > Best, > Tim > > On Tue, Jan 4, 2011 at 2:02 PM, Nicolas wrote: > >> Hi, >> >> Thanks for your reply. >> >> Ok, i do: >> >> pipeline = gst_pipeline_new ("pipeline"); >> playbin = gst_element_factory_make ("playbin2", "playbin"); >> >> // Filter >> filter = gst_element_factory_make ("capsfilter", "filter"); >> caps = gst_caps_new_simple ("video/x-raw-yuv", "format", >> GST_TYPE_FOURCC, GST_MAKE_FOURCC ('I', '4', '2', '0'), "width", >> G_TYPE_INT, 800, "height", G_TYPE_INT, 600, NULL); >> g_object_set ((GObject*) filter, "caps", caps, NULL); >> >> // My sink >> sink = gst_element_factory_make ("evasvideosink", "sink"); >> g_object_set ((GObject*) playbin, "video-sink", sink, NULL); // Tell to >> playbin2 to use my sink >> g_object_set ((GObject*) playbin, "uri", argv[1], NULL); >> g_object_set ((GObject*) sink, "evas-object", video, NULL); // Tell my >> sink to use evas image >> >> gst_bin_add_many (GST_BIN (pipeline), playbin, filter, NULL); >> >> gst_element_set_state (pipeline, GST_STATE_PLAYING); >> >> It does not change the size of the video, is this the correct way ? >> >> Thanks in advance for your reply, >> Nicolas. >> >> >> ------------------------------------------------------------------------------ >> 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 Wed Jan 5 05:05:36 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Wed, 5 Jan 2011 09:35:36 +0530 Subject: [gst-devel] Need help for rezising video. In-Reply-To: References: <4D236EC6.7030608@wanadoo.fr> Message-ID: Sorry typo - change the caps string to suit your needs. On Wed, Jan 5, 2011 at 9:34 AM, Rohit Atri wrote: > @Nicolas > > You could use the capsfilter element like this as well - > > caps = gst_caps_new_simple ("video/x-raw-rgb", > "bpp", G_TYPE_INT, 24, > "depth", G_TYPE_INT, 24, > NULL); > > gst_element_link_filtered(capsfilter, videosink, caps); (Considering > videoscale element is already linked to capsfilter on to its left.) > > instead of doing a gst_object_set()... > > Of course the choice between the 2 depends on your convenience. > > Rohti > > > > On Wed, Jan 5, 2011 at 12:48 AM, Timothy Braun wrote: > >> The capsfilter isn't going to do anything for you unless something in the >> pipeline is able to make the needed changes to match the caps. Before the >> capsfilter you should add a videoscale element so it is able to match the >> requested caps of the filter. You may also want to take a gander at the >> ffmpeg_colorspace and videorate elements as well. >> >> As a side note: >> You can simplify the creation of the caps with the gst_caps_from_string( >> "video/x-raw-yuv, format=I420, width=800, height=600") call. >> You need to unref the caps when you're done with it as well. >> >> Just my 2c. >> >> Best, >> Tim >> >> On Tue, Jan 4, 2011 at 2:02 PM, Nicolas wrote: >> >>> Hi, >>> >>> Thanks for your reply. >>> >>> Ok, i do: >>> >>> pipeline = gst_pipeline_new ("pipeline"); >>> playbin = gst_element_factory_make ("playbin2", "playbin"); >>> >>> // Filter >>> filter = gst_element_factory_make ("capsfilter", "filter"); >>> caps = gst_caps_new_simple ("video/x-raw-yuv", "format", >>> GST_TYPE_FOURCC, GST_MAKE_FOURCC ('I', '4', '2', '0'), "width", >>> G_TYPE_INT, 800, "height", G_TYPE_INT, 600, NULL); >>> g_object_set ((GObject*) filter, "caps", caps, NULL); >>> >>> // My sink >>> sink = gst_element_factory_make ("evasvideosink", "sink"); >>> g_object_set ((GObject*) playbin, "video-sink", sink, NULL); // Tell to >>> playbin2 to use my sink >>> g_object_set ((GObject*) playbin, "uri", argv[1], NULL); >>> g_object_set ((GObject*) sink, "evas-object", video, NULL); // Tell my >>> sink to use evas image >>> >>> gst_bin_add_many (GST_BIN (pipeline), playbin, filter, NULL); >>> >>> gst_element_set_state (pipeline, GST_STATE_PLAYING); >>> >>> It does not change the size of the video, is this the correct way ? >>> >>> Thanks in advance for your reply, >>> Nicolas. >>> >>> >>> ------------------------------------------------------------------------------ >>> 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 bradley.goldsmith at gmail.com Wed Jan 5 07:04:14 2011 From: bradley.goldsmith at gmail.com (Brad Goldsmith) Date: Wed, 5 Jan 2011 17:04:14 +1100 Subject: [gst-devel] Play audio from a memory buffer? In-Reply-To: <1294152909.19493.22.camel@zingle> References: <1294144535806-3173448.post@n4.nabble.com> <1294152909.19493.22.camel@zingle> Message-ID: Can appsrc be put in that pipeline, in place of filesrc, without any additional elements? Cheers, Brad On Wed, Jan 5, 2011 at 1:55 AM, Tim-Philipp M?ller wrote: > On Tue, 2011-01-04 at 04:35 -0800, bcg wrote: > > Hi, > >> This should almost be an FAQ but it's not in there so here goes: >> >> I have a pipeline thus: >> >> file-source -> ?decodebin -> audioresample -> converter -> audio_output >> >> Using location (or fd if I change file_source to a fdsrc) I can happily play >> most audio I throw at it (including mp3s) providing its in a file. What I >> would like to do is play from a memory location with the mp3 data already in >> it. >> >> I am assuming I would need to use a fakesrc with a callback to get my data >> into the pipeline - I've tried lots of variations on this and gotten >> nowhere. >> >> Can I just replace the file-source with the fakesrc, load the data and go >> (if this is possible couple someone provide an example)? Or do I have to >> worry about queues and caps? If so, can someone provide an example? Will >> decodebin suffice or will I have to be more specific? > > Use appsrc. > > 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 rohitratri at gmail.com Wed Jan 5 07:29:20 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Wed, 5 Jan 2011 11:59:20 +0530 Subject: [gst-devel] Play audio from a memory buffer? In-Reply-To: References: <1294144535806-3173448.post@n4.nabble.com> <1294152909.19493.22.camel@zingle> Message-ID: It can be put in the pipeline. But you need to pump data into it from your application. Your application would be (sort of) the 'source' and appsrc is like a cheat code to make the following elements think that there is actually a source element in there at the start of the pipeline feeding them gstbuffers. Try looking up the output format of filesrc element and package your data similarly and feed it to appsrc. Rest will be taken care by the bin/pipeline. Rohit On Wed, Jan 5, 2011 at 11:34 AM, Brad Goldsmith wrote: > Can appsrc be put in that pipeline, in place of filesrc, without any > additional elements? > > Cheers, > Brad > > On Wed, Jan 5, 2011 at 1:55 AM, Tim-Philipp M?ller > wrote: > > On Tue, 2011-01-04 at 04:35 -0800, bcg wrote: > > > > Hi, > > > >> This should almost be an FAQ but it's not in there so here goes: > >> > >> I have a pipeline thus: > >> > >> file-source -> decodebin -> audioresample -> converter -> audio_output > >> > >> Using location (or fd if I change file_source to a fdsrc) I can happily > play > >> most audio I throw at it (including mp3s) providing its in a file. What > I > >> would like to do is play from a memory location with the mp3 data > already in > >> it. > >> > >> I am assuming I would need to use a fakesrc with a callback to get my > data > >> into the pipeline - I've tried lots of variations on this and gotten > >> nowhere. > >> > >> Can I just replace the file-source with the fakesrc, load the data and > go > >> (if this is possible couple someone provide an example)? Or do I have to > >> worry about queues and caps? If so, can someone provide an example? Will > >> decodebin suffice or will I have to be more specific? > > > > Use appsrc. > > > > 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 > > > > > ------------------------------------------------------------------------------ > 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 ikt011 at gmail.com Wed Jan 5 09:07:41 2011 From: ikt011 at gmail.com (Kocsis Tibor) Date: Wed, 5 Jan 2011 09:07:41 +0100 Subject: [gst-devel] sending raw stream over rtp Message-ID: Hi, I have a basic rtsp server, and I try to use to send a raw stream to the client. On the receiver side I use a simple gstreamer pipeline like this: rtspsrc ! rtpvrawdepay ! xvimagesink. On the server there are gstreamer elements, too, some source connected to a rtpvrawpay, and to the gstrtpsink elements. The sending is started, and I can watch something live view, but it is not fluency. These messages appear on the client: 0:00:03.233100336 24189 0x7f654000bd20 WARN rtpvrawdepay gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: clipping length 540, offset 460 0:00:03.233134052 24189 0x7f654000bd20 WARN rtpvrawdepay gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: clipping length 1086, offset 278 0:00:03.233192201 24189 0x7f654000bd20 WARN rtpvrawdepay gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: clipping length 252, offset 556 0:00:03.233232346 24189 0x7f654000bd20 WARN rtpvrawdepay gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: clipping length 798, offset 374 ... What can it cause? My guess was I missed from the sdp (or built it up wrong), but the params I send over in the fmtp is the same as on the payloader's source pad (width, height, sample, even as depth, but as I see depth is not parsed on the client side). Btw my rtsp server works with h264/mpeg4 fine, so the elements are correctly linked. The full rtsp communication is here: http://pastebin.com/fC8MsJkx Any idea is welcomed Regards Tibor From bradley.goldsmith at gmail.com Wed Jan 5 09:48:19 2011 From: bradley.goldsmith at gmail.com (bcg) Date: Wed, 5 Jan 2011 00:48:19 -0800 (PST) Subject: [gst-devel] Play audio from a memory buffer? In-Reply-To: References: <1294144535806-3173448.post@n4.nabble.com> <1294152909.19493.22.camel@zingle> Message-ID: The example: appsrc-stream.c: example for using appsrc in streaming mode. Fits the bill. It loads a file as a memory map but if you shoehorn in your memory location and size at lines: 207 app->length = g_mapped_file_get_length (app->file); 208 app->data = (guint8 *) g_mapped_file_get_contents (app->file); You'll have a working example of what I was after. Cheers, Brad 1 /* GStreamer 2 * 3 * appsrc-stream.c: example for using appsrc in streaming mode. 4 * 5 * Copyright (C) 2008 Wim Taymans 6 * 7 * This library is free software; you can redistribute it and/or 8 * modify it under the terms of the GNU Library General Public 9 * License as published by the Free Software Foundation; either 10 * version 2 of the License, or (at your option) any later version. 11 * 12 * This library is distributed in the hope that it will be useful, 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of 14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 15 * Library General Public License for more details. 16 * 17 * You should have received a copy of the GNU Library General Public 18 * License along with this library; if not, write to the 19 * Free Software Foundation, Inc., 59 Temple Place - Suite 330, 20 * Boston, MA 02111-1307, USA. 21 */ 22 23 #ifdef HAVE_CONFIG_H 24 #include "config.h" 25 #endif 26 27 #include 28 29 #include 30 #include 31 #include 32 33 GST_DEBUG_CATEGORY (appsrc_playbin_debug); 34 #define GST_CAT_DEFAULT appsrc_playbin_debug 35 36 /* 37 * an example application of using appsrc in streaming push mode. We simply push 38 * buffers into appsrc. The size of the buffers we push can be any size we 39 * choose. 40 * 41 * This example is very close to how one would deal with a streaming webserver 42 * that does not support range requests or does not report the total file size. 43 * 44 * Some optimisations are done so that we don't push too much data. We connect 45 * to the need-data and enough-data signals to start/stop sending buffers. 46 * 47 * Appsrc in streaming mode (the default) does not support seeking so we don't 48 * have to handle any seek callbacks. 49 * 50 * Some formats are able to estimate the duration of the media file based on the 51 * file length (mp3, mpeg,..), others report an unknown length (ogg,..). 52 */ 53 typedef struct _App App; 54 55 struct _App 56 { 57 GstElement *playbin; 58 GstElement *appsrc; 59 60 GMainLoop *loop; 61 guint sourceid; 62 63 GMappedFile *file; 64 guint8 *data; 65 gsize length; 66 guint64 offset; 67 }; 68 69 App s_app; 70 71 #define CHUNK_SIZE 4096 72 73 /* This method is called by the idle GSource in the mainloop. We feed CHUNK_SIZE 74 * bytes into appsrc. 75 * The ide handler is added to the mainloop when appsrc requests us to start 76 * sending data (need-data signal) and is removed when appsrc has enough data 77 * (enough-data signal). 78 */ 79 static gboolean 80 read_data (App * app) 81 { 82 GstBuffer *buffer; 83 guint len; 84 GstFlowReturn ret; 85 86 buffer = gst_buffer_new (); 87 88 if (app->offset >= app->length) { 89 /* we are EOS, send end-of-stream and remove the source */ 90 g_signal_emit_by_name (app->appsrc, "end-of-stream", &ret); 91 return FALSE; 92 } 93 94 /* read the next chunk */ 95 len = CHUNK_SIZE; 96 if (app->offset + len > app->length) 97 len = app->length - app->offset; 98 99 GST_BUFFER_DATA (buffer) = app->data + app->offset; 100 GST_BUFFER_SIZE (buffer) = len; 101 102 GST_DEBUG ("feed buffer %p, offset %" G_GUINT64_FORMAT "-%u", buffer, 103 app->offset, len); 104 g_signal_emit_by_name (app->appsrc, "push-buffer", buffer, &ret); 105 if (ret != GST_FLOW_OK) { 106 /* some error, stop sending data */ 107 return FALSE; 108 } 109 110 app->offset += len; 111 112 return TRUE; 113 } 114 115 /* This signal callback is called when appsrc needs data, we add an idle handler 116 * to the mainloop to start pushing data into the appsrc */ 117 static void 118 start_feed (GstElement * playbin, guint size, App * app) 119 { 120 if (app->sourceid == 0) { 121 GST_DEBUG ("start feeding"); 122 app->sourceid = g_idle_add ((GSourceFunc) read_data, app); 123 } 124 } 125 126 /* This callback is called when appsrc has enough data and we can stop sending. 127 * We remove the idle handler from the mainloop */ 128 static void 129 stop_feed (GstElement * playbin, App * app) 130 { 131 if (app->sourceid != 0) { 132 GST_DEBUG ("stop feeding"); 133 g_source_remove (app->sourceid); 134 app->sourceid = 0; 135 } 136 } 137 138 /* this callback is called when playbin2 has constructed a source object to read 139 * from. Since we provided the appsrc:// uri to playbin2, this will be the 140 * appsrc that we must handle. We set up some signals to start and stop pushing 141 * data into appsrc */ 142 static void 143 found_source (GObject * object, GObject * orig, GParamSpec * pspec, App * app) 144 { 145 /* get a handle to the appsrc */ 146 g_object_get (orig, pspec->name, &app->appsrc, NULL); 147 148 GST_DEBUG ("got appsrc %p", app->appsrc); 149 150 /* we can set the length in appsrc. This allows some elements to estimate the 151 * total duration of the stream. It's a good idea to set the property when you 152 * can but it's not required. */ 153 g_object_set (app->appsrc, "size", app->length, NULL); 154 155 /* configure the appsrc, we will push data into the appsrc from the 156 * mainloop. */ 157 g_signal_connect (app->appsrc, "need-data", G_CALLBACK (start_feed), app); 158 g_signal_connect (app->appsrc, "enough-data", G_CALLBACK (stop_feed), app); 159 } 160 161 static gboolean 162 bus_message (GstBus * bus, GstMessage * message, App * app) 163 { 164 GST_DEBUG ("got message %s", 165 gst_message_type_get_name (GST_MESSAGE_TYPE (message))); 166 167 switch (GST_MESSAGE_TYPE (message)) { 168 case GST_MESSAGE_ERROR: 169 g_error ("received error"); 170 g_main_loop_quit (app->loop); 171 break; 172 case GST_MESSAGE_EOS: 173 g_main_loop_quit (app->loop); 174 break; 175 default: 176 break; 177 } 178 return TRUE; 179 } 180 181 int 182 main (int argc, char *argv[]) 183 { 184 App *app = &s_app; 185 GError *error = NULL; 186 GstBus *bus; 187 188 gst_init (&argc, &argv); 189 190 GST_DEBUG_CATEGORY_INIT (appsrc_playbin_debug, "appsrc-playbin", 0, 191 "appsrc playbin example"); 192 193 if (argc < 2) { 194 g_print ("usage: %s \n", argv[0]); 195 return -1; 196 } 197 198 /* try to open the file as an mmapped file */ 199 app->file = g_mapped_file_new (argv[1], FALSE, &error); 200 if (error) { 201 g_print ("failed to open file: %s\n", error->message); 202 g_error_free (error); 203 return -2; 204 } 205 /* get some vitals, this will be used to read data from the mmapped file and 206 * feed it to appsrc. */ 207 app->length = g_mapped_file_get_length (app->file); 208 app->data = (guint8 *) g_mapped_file_get_contents (app->file); 209 app->offset = 0; 210 211 /* create a mainloop to get messages and to handle the idle handler that will 212 * feed data to appsrc. */ 213 app->loop = g_main_loop_new (NULL, TRUE); 214 215 app->playbin = gst_element_factory_make ("playbin2", NULL); 216 g_assert (app->playbin); 217 218 bus = gst_pipeline_get_bus (GST_PIPELINE (app->playbin)); 219 220 /* add watch for messages */ 221 gst_bus_add_watch (bus, (GstBusFunc) bus_message, app); 222 223 /* set to read from appsrc */ 224 g_object_set (app->playbin, "uri", "appsrc://", NULL); 225 226 /* get notification when the source is created so that we get a handle to it 227 * and can configure it */ 228 g_signal_connect (app->playbin, "deep-notify::source", 229 (GCallback) found_source, app); 230 231 /* go to playing and wait in a mainloop. */ 232 gst_element_set_state (app->playbin, GST_STATE_PLAYING); 233 234 /* this mainloop is stopped when we receive an error or EOS */ 235 g_main_loop_run (app->loop); 236 237 GST_DEBUG ("stopping"); 238 239 gst_element_set_state (app->playbin, GST_STATE_NULL); 240 241 /* free the file */ 242 g_mapped_file_free (app->file); 243 244 gst_object_unref (bus); 245 g_main_loop_unref (app->loop); 246 247 return 0; 248 } On Wed, Jan 5, 2011 at 5:30 PM, rohitratri at gmail.com [via GStreamer-devel] wrote: > It can be put in the pipeline. But you need to pump data into it from your > application. Your application would be (sort of) the 'source' and appsrc is > like a cheat code to make the following elements think that there is > actually a source element in there at the start of the pipeline feeding them > gstbuffers. > Try looking up the output format of filesrc element and package your data > similarly and feed it to appsrc. Rest will be taken care by the > bin/pipeline. > Rohit > On Wed, Jan 5, 2011 at 11:34 AM, Brad Goldsmith <[hidden email]> wrote: >> >> Can appsrc be put in that pipeline, in place of filesrc, without any >> additional elements? >> >> Cheers, >> Brad >> >> On Wed, Jan 5, 2011 at 1:55 AM, Tim-Philipp M?ller <[hidden email]> wrote: >> > On Tue, 2011-01-04 at 04:35 -0800, bcg wrote: >> > >> > Hi, >> > >> >> This should almost be an FAQ but it's not in there so here goes: >> >> >> >> I have a pipeline thus: >> >> >> >> file-source -> ?decodebin -> audioresample -> converter -> audio_output >> >> >> >> Using location (or fd if I change file_source to a fdsrc) I can happily >> >> play >> >> most audio I throw at it (including mp3s) providing its in a file. What >> >> I >> >> would like to do is play from a memory location with the mp3 data >> >> already in >> >> it. >> >> >> >> I am assuming I would need to use a fakesrc with a callback to get my >> >> data >> >> into the pipeline - I've tried lots of variations on this and gotten >> >> nowhere. >> >> >> >> Can I just replace the file-source with the fakesrc, load the data and >> >> go >> >> (if this is possible couple someone provide an example)? Or do I have >> >> to >> >> worry about queues and caps? If so, can someone provide an example? >> >> Will >> >> decodebin suffice or will I have to be more specific? >> > >> > Use appsrc. >> > >> > 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 >> > [hidden email] >> > 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 >> [hidden email] >> 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 > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ________________________________ > View message @ > http://gstreamer-devel.966125.n4.nabble.com/Play-audio-from-a-memory-buffer-tp3173448p3174884.html > To unsubscribe from Play audio from a memory buffer?, click here. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Play-audio-from-a-memory-buffer-tp3173448p3174994.html Sent from the GStreamer-devel mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ikt011 at gmail.com Wed Jan 5 10:13:44 2011 From: ikt011 at gmail.com (Kocsis Tibor) Date: Wed, 5 Jan 2011 10:13:44 +0100 Subject: [gst-devel] sending raw stream over rtp In-Reply-To: References: Message-ID: The sdp I linked was wrong, my version is here: medias: media 0: media: 'video' port: '0' num_ports: '4294967295' proto: 'RTP/AVP' formats: format '96' information: '(NULL)' key: type: '(NULL)' data: '(NULL)' attributes: attribute 'rtpmap' : '96 RAW/90000' attribute 'control' : 'rtsp://10.30.0.8:1554/Live/trackID=0?channelID=98d16963-877f-4446-bb69-3aeada917523' attribute 'fmtp' : '96 width=640;height=480;sampling=YCbCr-4:2:0;depth=8' attribute 'appversion' : '5.0.0' On Wed, Jan 5, 2011 at 9:07 AM, Kocsis Tibor wrote: > Hi, > > I have a basic rtsp server, and I try to use to send a raw stream to > the client. On the receiver side I use a simple gstreamer pipeline > like this: rtspsrc ! rtpvrawdepay ! xvimagesink. On the server there > are gstreamer elements, too, some source connected to a rtpvrawpay, > and to the gstrtpsink elements. The sending is started, and I can > watch something live view, but it is not fluency. These messages > appear on the client: > > 0:00:03.233100336 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay > gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > clipping length 540, offset 460 > 0:00:03.233134052 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay > gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > clipping length 1086, offset 278 > 0:00:03.233192201 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay > gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > clipping length 252, offset 556 > 0:00:03.233232346 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay > gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > clipping length 798, offset 374 > ... > > What can it cause? > > My guess was I missed from the sdp (or built it up wrong), but the > params I send over in the fmtp is the same as on the payloader's > source pad (width, height, sample, even as depth, but as I see depth > is not parsed on the client side). Btw my rtsp server works with > h264/mpeg4 fine, so the elements are correctly linked. > > The full rtsp communication is here: http://pastebin.com/fC8MsJkx > > Any idea is welcomed > > Regards > Tibor > From vaish05_s at yahoo.com Wed Jan 5 12:17:24 2011 From: vaish05_s at yahoo.com (vaishnavi) Date: Wed, 5 Jan 2011 03:17:24 -0800 (PST) Subject: [gst-devel] OpenMax Source code Message-ID: <1294226244728-3175284.post@n4.nabble.com> Hi, Can anyone tell me where i can find the source code for openmax IL by STMicroelectronics.The code im asking for is in reference to document "Using OpenMAX Integration Layer with GStreamer" . In this document they have a flow diagram using MP3 decoder element which handles buffer management. I'm looking for this source code. Thank you -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/OpenMax-Source-code-tp3175284p3175284.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From henriette.depoulpiquet at gmail.com Wed Jan 5 12:35:13 2011 From: henriette.depoulpiquet at gmail.com (henriette de poulpiquet) Date: Wed, 5 Jan 2011 12:35:13 +0100 Subject: [gst-devel] install gst-plugins-gl-0.10.2 In-Reply-To: References: Message-ID: Hi! thanks Sudarshen, I think my env variables are ok.. other plugins installed in the same way was installed without any problems.. can anyone explain me what does the plugins blacklist mean? thanks! Henriette On 1/4/11, sudarshan bisht wrote: > No idea about Glib. But just a guess again, check LD_LIBRARY_PATH and other > environment variables. Since your Gstreamer installation is there in > /usr/local/lib not in /usr/lib. > > > On Tue, Jan 4, 2011 at 10:38 PM, henriette de poulpiquet < > henriette.depoulpiquet at gmail.com> wrote: > >> maybe my Glib version is important? >> Glib 2.26.1 >> >> .. no one ever had this problem? >> >> anyway, thanks Sudarshan for your help! >> >> Henriette >> >> On 1/4/11, henriette de poulpiquet >> wrote: >> > yes I did... >> > >> > >> > On 1/4/11, sudarshan bisht wrote: >> >> Did you do " export GST_PLUGIN_PATH=/usr/local/lib/gstreamer-0.10" >> ? >> >> >> >> On Tue, Jan 4, 2011 at 1:47 PM, henriette de poulpiquet < >> >> henriette.depoulpiquet at gmail.com> wrote: >> >> >> >>> Hi! >> >>> >> >>> I'me not sure where to post my question, hope i'me in the right place! >> >>> >> >>> so my problem is that my gst-inspect indicate that gl plugin is >> >>> blacklisted, I'me quite disappointed and can't understand why >> >>> /usr/local/lib/gstreamer-0.10/libgstopengl.so can' t be found since >> >>> libgstopengl.so do exists at this place (I checked it). >> >>> >> >>> what should I do? >> >>> >> >>> thanks for your answer! >> >>> >> >>> >> >>> Henriette >> >>> >> >>> i'me working on a >> >>> Linux: Debian 32 bits >> >>> Release: 2.6.26-2-686 >> >>> i686 >> >>> gstreamer-0.10.31 >> >>> gst-plugins-base-0.10.31 >> >>> gst-plugins-good-0.10.26 >> >>> >> >>> >> >>> >> ------------------------------------------------------------------------------ >> >>> 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 >> > > > > -- > Regards, > > Sudarshan Bisht > From rohitratri at gmail.com Wed Jan 5 12:37:35 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Wed, 5 Jan 2011 17:07:35 +0530 Subject: [gst-devel] OpenMax Source code In-Reply-To: <1294226244728-3175284.post@n4.nabble.com> References: <1294226244728-3175284.post@n4.nabble.com> Message-ID: Small correction, Openmax IL is by the khronos group, not STE. On Wed, Jan 5, 2011 at 4:47 PM, vaishnavi wrote: > > Hi, > Can anyone tell me where i can find the source code for openmax IL by > STMicroelectronics.The code im asking for is in reference to document > "Using > OpenMAX Integration Layer with GStreamer" . In this document they have a > flow diagram using MP3 decoder element which handles buffer management. I'm > looking for this source code. > > Thank you > > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/OpenMax-Source-code-tp3175284p3175284.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 ssrktce at gmail.com Wed Jan 5 12:40:40 2011 From: ssrktce at gmail.com (ssrk) Date: Wed, 5 Jan 2011 17:10:40 +0530 Subject: [gst-devel] Pipeline for converting H264 bytestream to mp4 container format? Message-ID: Hi all, Can some one give the pipeline to convert a h264 bytestream file .264 to mp4 container format? Please help. Thanks in advance. -- Regards, ssrk -------------- next part -------------- An HTML attachment was scrubbed... URL: From ikt011 at gmail.com Wed Jan 5 14:38:35 2011 From: ikt011 at gmail.com (Kocsis Tibor) Date: Wed, 5 Jan 2011 14:38:35 +0100 Subject: [gst-devel] sending raw stream over rtp In-Reply-To: References: Message-ID: I created a command line to test the problem, the result is the same, I got a lot of messages like this: 0:00:27.686127192 14222 0xf1bd50 WARN rtpvrawdepay gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: clipping length 726, offset 398 0:00:27.686175449 14222 0xf1bd50 WARN rtpvrawdepay gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: clipping length 438, offset 494 The pipelines: Sender: gst-launch -v v4l2src ! video/x-raw-yuv,width=640,height=480,framerate=10/1 ! clockoverlay ! rtpvrawpay ! udpsink host=... port=5000 sync=false Receiver: gst-launch -v udpsrc port=5000 caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)640, height=(string)480, colorimetry=(string)SMPTE240M, payload=(int)96, ssrc=(uint)2464304599, clock-base=(uint)342261484, seqnum-base=(uint)47511" ! rtpvrawdepay ! xvimagesink sync=false Caps copied from udsink.sink on the sender side. So is this warning message normal and should be info or maybe a rawdepay bug or something else? Regards, Tibor On Wed, Jan 5, 2011 at 10:13 AM, Kocsis Tibor wrote: > The sdp I linked was wrong, my version is here: > > ?medias: > ?media 0: > ? media: ? ? ? 'video' > ? port: ? ? ? ?'0' > ? num_ports: ? '4294967295' > ? proto: ? ? ? 'RTP/AVP' > ? formats: > ? ?format ?'96' > ? information: '(NULL)' > ? key: > ? ?type: ? ? ? '(NULL)' > ? ?data: ? ? ? '(NULL)' > ? attributes: > ? ?attribute 'rtpmap' : '96 RAW/90000' > ? ?attribute 'control' : > 'rtsp://10.30.0.8:1554/Live/trackID=0?channelID=98d16963-877f-4446-bb69-3aeada917523' > ? ?attribute 'fmtp' : '96 width=640;height=480;sampling=YCbCr-4:2:0;depth=8' > ? ?attribute 'appversion' : '5.0.0' > > > On Wed, Jan 5, 2011 at 9:07 AM, Kocsis Tibor wrote: >> Hi, >> >> I have a basic rtsp server, and I try to use to send a raw stream to >> the client. On the receiver side I use a simple gstreamer pipeline >> like this: rtspsrc ! rtpvrawdepay ! xvimagesink. On the server there >> are gstreamer elements, too, some source connected to a rtpvrawpay, >> and to the gstrtpsink elements. The sending is started, and I can >> watch something live view, but it is not fluency. These messages >> appear on the client: >> >> 0:00:03.233100336 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: >> clipping length 540, offset 460 >> 0:00:03.233134052 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: >> clipping length 1086, offset 278 >> 0:00:03.233192201 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: >> clipping length 252, offset 556 >> 0:00:03.233232346 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: >> clipping length 798, offset 374 >> ... >> >> What can it cause? >> >> My guess was I missed from the sdp (or built it up wrong), but the >> params I send over in the fmtp is the same as on the payloader's >> source pad (width, height, sample, even as depth, but as I see depth >> is not parsed on the client side). Btw my rtsp server works with >> h264/mpeg4 fine, so the elements are correctly linked. >> >> The full rtsp communication is here: http://pastebin.com/fC8MsJkx >> >> Any idea is welcomed >> >> Regards >> Tibor >> > From wim.taymans at gmail.com Wed Jan 5 14:48:46 2011 From: wim.taymans at gmail.com (Wim Taymans) Date: Wed, 05 Jan 2011 14:48:46 +0100 Subject: [gst-devel] sending raw stream over rtp In-Reply-To: References: Message-ID: <1294235326.28802.74.camel@meany> On Wed, 2011-01-05 at 14:38 +0100, Kocsis Tibor wrote: > I created a command line to test the problem, the result is the same, > I got a lot of messages like this: > > 0:00:27.686127192 14222 0xf1bd50 WARN rtpvrawdepay > gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > clipping length 726, offset 398 > 0:00:27.686175449 14222 0xf1bd50 WARN rtpvrawdepay > gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > clipping length 438, offset 494 Those are normal. I will change them to LOG. Wim > The pipelines: > > Sender: > > gst-launch -v v4l2src ! > video/x-raw-yuv,width=640,height=480,framerate=10/1 ! clockoverlay ! > rtpvrawpay ! udpsink host=... port=5000 sync=false > > Receiver: > > gst-launch -v udpsrc port=5000 caps="application/x-rtp, > media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, > sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)640, > height=(string)480, colorimetry=(string)SMPTE240M, payload=(int)96, > ssrc=(uint)2464304599, clock-base=(uint)342261484, > seqnum-base=(uint)47511" ! rtpvrawdepay ! xvimagesink sync=false > > Caps copied from udsink.sink on the sender side. > > So is this warning message normal and should be info or maybe a > rawdepay bug or something else? > > Regards, > Tibor > > > > On Wed, Jan 5, 2011 at 10:13 AM, Kocsis Tibor wrote: > > The sdp I linked was wrong, my version is here: > > > > medias: > > media 0: > > media: 'video' > > port: '0' > > num_ports: '4294967295' > > proto: 'RTP/AVP' > > formats: > > format '96' > > information: '(NULL)' > > key: > > type: '(NULL)' > > data: '(NULL)' > > attributes: > > attribute 'rtpmap' : '96 RAW/90000' > > attribute 'control' : > > 'rtsp://10.30.0.8:1554/Live/trackID=0?channelID=98d16963-877f-4446-bb69-3aeada917523' > > attribute 'fmtp' : '96 width=640;height=480;sampling=YCbCr-4:2:0;depth=8' > > attribute 'appversion' : '5.0.0' > > > > > > On Wed, Jan 5, 2011 at 9:07 AM, Kocsis Tibor wrote: > >> Hi, > >> > >> I have a basic rtsp server, and I try to use to send a raw stream to > >> the client. On the receiver side I use a simple gstreamer pipeline > >> like this: rtspsrc ! rtpvrawdepay ! xvimagesink. On the server there > >> are gstreamer elements, too, some source connected to a rtpvrawpay, > >> and to the gstrtpsink elements. The sending is started, and I can > >> watch something live view, but it is not fluency. These messages > >> appear on the client: > >> > >> 0:00:03.233100336 24189 0x7f654000bd20 WARN rtpvrawdepay > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > >> clipping length 540, offset 460 > >> 0:00:03.233134052 24189 0x7f654000bd20 WARN rtpvrawdepay > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > >> clipping length 1086, offset 278 > >> 0:00:03.233192201 24189 0x7f654000bd20 WARN rtpvrawdepay > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > >> clipping length 252, offset 556 > >> 0:00:03.233232346 24189 0x7f654000bd20 WARN rtpvrawdepay > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > >> clipping length 798, offset 374 > >> ... > >> > >> What can it cause? > >> > >> My guess was I missed from the sdp (or built it up wrong), but the > >> params I send over in the fmtp is the same as on the payloader's > >> source pad (width, height, sample, even as depth, but as I see depth > >> is not parsed on the client side). Btw my rtsp server works with > >> h264/mpeg4 fine, so the elements are correctly linked. > >> > >> The full rtsp communication is here: http://pastebin.com/fC8MsJkx > >> > >> Any idea is welcomed > >> > >> Regards > >> Tibor > >> > > > > ------------------------------------------------------------------------------ > 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 bisht.sudarshan at gmail.com Wed Jan 5 16:21:00 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Wed, 5 Jan 2011 17:21:00 +0200 Subject: [gst-devel] Pipeline for converting H264 bytestream to mp4 container format? In-Reply-To: References: Message-ID: Try, gst-launch filesrc location=test.264 ! h264parse ! x264enc ! mp4mux( or qtmux ) ! filesink location=test.mp4 On Wed, Jan 5, 2011 at 1:40 PM, ssrk wrote: > Hi all, > > Can some one give the pipeline to convert a h264 bytestream file .264 to > mp4 container format? > Please help. Thanks in advance. > > -- > Regards, > ssrk > > > > ------------------------------------------------------------------------------ > 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 c.r.n.a at wanadoo.fr Wed Jan 5 18:34:28 2011 From: c.r.n.a at wanadoo.fr (Nicolas) Date: Wed, 05 Jan 2011 18:34:28 +0100 Subject: [gst-devel] Need help for rezising video. Message-ID: <4D24ABA4.4020704@wanadoo.fr> Hi, Thanks everyone for your reply. Can you send me your pipeline (in C or Python), because mine does not work ? Thanks in advance, Nicolas. From bilboed at gmail.com Wed Jan 5 18:58:53 2011 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 05 Jan 2011 18:58:53 +0100 Subject: [gst-devel] Need help for rezising video. In-Reply-To: <4D24ABA4.4020704@wanadoo.fr> References: <4D24ABA4.4020704@wanadoo.fr> Message-ID: <1294250333.23475.8.camel@localhost> On Wed, 2011-01-05 at 18:34 +0100, Nicolas wrote: > Hi, > > Thanks everyone for your reply. > > Can you send me your pipeline (in C or Python), because mine does not work ? The changes of getting a response would be exponentially higher if you were to actually provide *your* pipeline. Edward > > Thanks in advance, > Nicolas. > > ------------------------------------------------------------------------------ > 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 c.r.n.a at wanadoo.fr Wed Jan 5 19:50:02 2011 From: c.r.n.a at wanadoo.fr (Nicolas) Date: Wed, 05 Jan 2011 19:50:02 +0100 Subject: [gst-devel] Need help for rezising video. Message-ID: <4D24BD5A.8000101@wanadoo.fr> Hi, In fact, i made a new pipeline just for testing and my problem is, i don't know what to link together ! In the code below nothing append, no message (gst-launch sample work perfectly). #include // Build with gcc test.c `pkg-config --cflags --libs gstreamer-0.10 glib-2.0` -o test // gst-launch filesrc location="/home/niko/jour1.avi" ! decodebin ! videoscale ! capsfilter caps=video/x-raw-yuv,width=800,height=600,framerate=30/1 ! ffmpegcolorspace ! ximagesink int main (int argc, char *argv[]) { GMainLoop *loop; GstElement *pipeline, *source, *decode, *scale, *filter, *color, *sink; GstCaps *caps; gst_init (&argc, &argv); loop = g_main_loop_new (NULL, FALSE); pipeline = gst_pipeline_new ("test_pipeline"); source = gst_element_factory_make ("filesrc", "source"); g_object_set ((GObject*) source, "location", "/home/niko/jour1.avi", NULL); decode = gst_element_factory_make ("decodebin", "decode"); scale = gst_element_factory_make ("videoscale", "scale"); filter = gst_element_factory_make ("capsfilter", "filter"); caps = gst_caps_new_simple ("video/x-raw-yuv", "format", GST_TYPE_FOURCC, GST_MAKE_FOURCC ('I', '4', '2', '0'), "width", G_TYPE_INT, 800, "height", G_TYPE_INT, 600, "framerate", GST_TYPE_FRACTION, 30, 1, NULL); g_object_set ((GObject*) filter, "caps", caps, NULL); gst_caps_unref(caps); color = gst_element_factory_make ("ffmpegcolorspace", "color"); sink = gst_element_factory_make ("ximagesink", "output"); gst_bin_add_many (GST_BIN (pipeline), source, scale, filter, color, sink, NULL); gst_element_link_many(source, scale, filter, color, sink, NULL); gst_element_set_state (pipeline, GST_STATE_PLAYING); g_main_loop_run (loop); gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (GST_OBJECT (pipeline)); return 0; } Thanks in advance for you help. Nicolas. From julien38813 at hotmail.com Wed Jan 5 20:00:20 2011 From: julien38813 at hotmail.com (Julien --) Date: Wed, 5 Jan 2011 20:00:20 +0100 Subject: [gst-devel] stop sending me your e-mail Message-ID: Hi, I erased my login gstreamer-devel and I want to stop receiving the mail sourceforge. Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From braunsquared at gmail.com Wed Jan 5 20:06:55 2011 From: braunsquared at gmail.com (Timothy Braun) Date: Wed, 5 Jan 2011 14:06:55 -0500 Subject: [gst-devel] Need help for rezising video. In-Reply-To: <4D24BD5A.8000101@wanadoo.fr> References: <4D24BD5A.8000101@wanadoo.fr> Message-ID: You always want to check the return result from any of the link calls. Also, to link the decodebin you have to listen for the pad-added (or is it new-decoded-pad, I can't recall) and link when it's available. It won't be available until the decodebin has setup the decoders and such. -Tim On Wed, Jan 5, 2011 at 1:50 PM, Nicolas wrote: > Hi, > > In fact, i made a new pipeline just for testing and my problem is, i > don't know what to link together ! > In the code below nothing append, no message (gst-launch sample work > perfectly). > > #include > > // Build with gcc test.c `pkg-config --cflags --libs gstreamer-0.10 > glib-2.0` -o test > // gst-launch filesrc location="/home/niko/jour1.avi" ! decodebin ! > videoscale ! capsfilter > caps=video/x-raw-yuv,width=800,height=600,framerate=30/1 ! > ffmpegcolorspace ! ximagesink > > int > main (int argc, > char *argv[]) > { > GMainLoop *loop; > GstElement *pipeline, *source, *decode, *scale, *filter, *color, *sink; > GstCaps *caps; > > gst_init (&argc, &argv); > loop = g_main_loop_new (NULL, FALSE); > > > pipeline = gst_pipeline_new ("test_pipeline"); > > source = gst_element_factory_make ("filesrc", "source"); > g_object_set ((GObject*) source, "location", "/home/niko/jour1.avi", > NULL); > > decode = gst_element_factory_make ("decodebin", "decode"); > scale = gst_element_factory_make ("videoscale", "scale"); > > filter = gst_element_factory_make ("capsfilter", "filter"); > caps = gst_caps_new_simple ("video/x-raw-yuv", "format", > GST_TYPE_FOURCC, GST_MAKE_FOURCC ('I', '4', '2', '0'), "width", > G_TYPE_INT, 800, "height", G_TYPE_INT, 600, "framerate", > GST_TYPE_FRACTION, 30, 1, NULL); > g_object_set ((GObject*) filter, "caps", caps, NULL); > gst_caps_unref(caps); > > color = gst_element_factory_make ("ffmpegcolorspace", "color"); > sink = gst_element_factory_make ("ximagesink", "output"); > > gst_bin_add_many (GST_BIN (pipeline), source, scale, filter, color, > sink, NULL); > > gst_element_link_many(source, scale, filter, color, sink, NULL); > > gst_element_set_state (pipeline, GST_STATE_PLAYING); > > g_main_loop_run (loop); > > gst_element_set_state (pipeline, GST_STATE_NULL); > gst_object_unref (GST_OBJECT (pipeline)); > return 0; > } > > Thanks in advance for you help. > Nicolas. > > > ------------------------------------------------------------------------------ > 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 vtorri at univ-evry.fr Wed Jan 5 20:09:04 2011 From: vtorri at univ-evry.fr (Vincent Torri) Date: Wed, 5 Jan 2011 20:09:04 +0100 (CET) Subject: [gst-devel] Need help for rezising video. In-Reply-To: <4D24BD5A.8000101@wanadoo.fr> References: <4D24BD5A.8000101@wanadoo.fr> Message-ID: On Wed, 5 Jan 2011, Nicolas wrote: > Hi, > > In fact, i made a new pipeline just for testing and my problem is, i > don't know what to link together ! > In the code below nothing append, no message (gst-launch sample work > perfectly). > > #include > > // Build with gcc test.c `pkg-config --cflags --libs gstreamer-0.10 > glib-2.0` -o test > // gst-launch filesrc location="/home/niko/jour1.avi" ! decodebin ! > videoscale ! capsfilter > caps=video/x-raw-yuv,width=800,height=600,framerate=30/1 ! > ffmpegcolorspace ! ximagesink > > int > main (int argc, > char *argv[]) > { > GMainLoop *loop; > GstElement *pipeline, *source, *decode, *scale, *filter, *color, *sink; > GstCaps *caps; > > gst_init (&argc, &argv); > loop = g_main_loop_new (NULL, FALSE); > > > pipeline = gst_pipeline_new ("test_pipeline"); > > source = gst_element_factory_make ("filesrc", "source"); > g_object_set ((GObject*) source, "location", "/home/niko/jour1.avi", > NULL); > > decode = gst_element_factory_make ("decodebin", "decode"); > scale = gst_element_factory_make ("videoscale", "scale"); > > filter = gst_element_factory_make ("capsfilter", "filter"); > caps = gst_caps_new_simple ("video/x-raw-yuv", "format", > GST_TYPE_FOURCC, GST_MAKE_FOURCC ('I', '4', '2', '0'), "width", > G_TYPE_INT, 800, "height", G_TYPE_INT, 600, "framerate", > GST_TYPE_FRACTION, 30, 1, NULL); > g_object_set ((GObject*) filter, "caps", caps, NULL); > gst_caps_unref(caps); > > color = gst_element_factory_make ("ffmpegcolorspace", "color"); > sink = gst_element_factory_make ("ximagesink", "output"); > > gst_bin_add_many (GST_BIN (pipeline), source, scale, filter, color, > sink, NULL); strangely, there is no 'decode' element here :) Vincent > > gst_element_link_many(source, scale, filter, color, sink, NULL); > > gst_element_set_state (pipeline, GST_STATE_PLAYING); > > g_main_loop_run (loop); > > gst_element_set_state (pipeline, GST_STATE_NULL); > gst_object_unref (GST_OBJECT (pipeline)); > return 0; > } > > Thanks in advance for you help. > Nicolas. > > ------------------------------------------------------------------------------ > 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 c.r.n.a at wanadoo.fr Wed Jan 5 20:38:27 2011 From: c.r.n.a at wanadoo.fr (Nicolas) Date: Wed, 05 Jan 2011 20:38:27 +0100 Subject: [gst-devel] Need help for rezising video. Message-ID: <4D24C8B3.2050204@wanadoo.fr> Hi Vincent, It does not change anything if i add it ! :) Nicolas. From suneets at gmail.com Wed Jan 5 20:50:29 2011 From: suneets at gmail.com (Suneet Shah) Date: Wed, 5 Jan 2011 11:50:29 -0800 Subject: [gst-devel] sending raw stream over rtp In-Reply-To: <1294235326.28802.74.camel@meany> References: <1294235326.28802.74.camel@meany> Message-ID: Hi Kocis-- what RTSP server are you using? I'm trying to do the opposite of what you're doing. I have a gstreamer pipeline creating RTP packets, but need to package them with RTSP and create an SDP file. Thanks, Suneet On Wed, Jan 5, 2011 at 5:48 AM, Wim Taymans wrote: > On Wed, 2011-01-05 at 14:38 +0100, Kocsis Tibor wrote: > > I created a command line to test the problem, the result is the same, > > I got a lot of messages like this: > > > > 0:00:27.686127192 14222 0xf1bd50 WARN rtpvrawdepay > > gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > > clipping length 726, offset 398 > > 0:00:27.686175449 14222 0xf1bd50 WARN rtpvrawdepay > > gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > > clipping length 438, offset 494 > > Those are normal. I will change them to LOG. > > Wim > > The pipelines: > > > > Sender: > > > > gst-launch -v v4l2src ! > > video/x-raw-yuv,width=640,height=480,framerate=10/1 ! clockoverlay ! > > rtpvrawpay ! udpsink host=... port=5000 sync=false > > > > Receiver: > > > > gst-launch -v udpsrc port=5000 caps="application/x-rtp, > > media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, > > sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)640, > > height=(string)480, colorimetry=(string)SMPTE240M, payload=(int)96, > > ssrc=(uint)2464304599, clock-base=(uint)342261484, > > seqnum-base=(uint)47511" ! rtpvrawdepay ! xvimagesink sync=false > > > > Caps copied from udsink.sink on the sender side. > > > > So is this warning message normal and should be info or maybe a > > rawdepay bug or something else? > > > > Regards, > > Tibor > > > > > > > > On Wed, Jan 5, 2011 at 10:13 AM, Kocsis Tibor wrote: > > > The sdp I linked was wrong, my version is here: > > > > > > medias: > > > media 0: > > > media: 'video' > > > port: '0' > > > num_ports: '4294967295' > > > proto: 'RTP/AVP' > > > formats: > > > format '96' > > > information: '(NULL)' > > > key: > > > type: '(NULL)' > > > data: '(NULL)' > > > attributes: > > > attribute 'rtpmap' : '96 RAW/90000' > > > attribute 'control' : > > > 'rtsp:// > 10.30.0.8:1554/Live/trackID=0?channelID=98d16963-877f-4446-bb69-3aeada917523 > ' > > > attribute 'fmtp' : '96 > width=640;height=480;sampling=YCbCr-4:2:0;depth=8' > > > attribute 'appversion' : '5.0.0' > > > > > > > > > On Wed, Jan 5, 2011 at 9:07 AM, Kocsis Tibor wrote: > > >> Hi, > > >> > > >> I have a basic rtsp server, and I try to use to send a raw stream to > > >> the client. On the receiver side I use a simple gstreamer pipeline > > >> like this: rtspsrc ! rtpvrawdepay ! xvimagesink. On the server there > > >> are gstreamer elements, too, some source connected to a rtpvrawpay, > > >> and to the gstrtpsink elements. The sending is started, and I can > > >> watch something live view, but it is not fluency. These messages > > >> appear on the client: > > >> > > >> 0:00:03.233100336 24189 0x7f654000bd20 WARN rtpvrawdepay > > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > > >> clipping length 540, offset 460 > > >> 0:00:03.233134052 24189 0x7f654000bd20 WARN rtpvrawdepay > > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > > >> clipping length 1086, offset 278 > > >> 0:00:03.233192201 24189 0x7f654000bd20 WARN rtpvrawdepay > > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > > >> clipping length 252, offset 556 > > >> 0:00:03.233232346 24189 0x7f654000bd20 WARN rtpvrawdepay > > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: > > >> clipping length 798, offset 374 > > >> ... > > >> > > >> What can it cause? > > >> > > >> My guess was I missed from the sdp (or built it up wrong), but the > > >> params I send over in the fmtp is the same as on the payloader's > > >> source pad (width, height, sample, even as depth, but as I see depth > > >> is not parsed on the client side). Btw my rtsp server works with > > >> h264/mpeg4 fine, so the elements are correctly linked. > > >> > > >> The full rtsp communication is here: http://pastebin.com/fC8MsJkx > > >> > > >> Any idea is welcomed > > >> > > >> Regards > > >> Tibor > > >> > > > > > > > > ------------------------------------------------------------------------------ > > 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 bilboed at gmail.com Wed Jan 5 20:56:20 2011 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 05 Jan 2011 20:56:20 +0100 Subject: [gst-devel] stop sending me your e-mail In-Reply-To: References: Message-ID: <1294257380.21480.0.camel@deumeu> On Wed, 2011-01-05 at 20:00 +0100, Julien -- wrote: > Hi, I erased my login gstreamer-devel and I want to stop receiving the > mail sourceforge. See the part about unsubscribing at the bottom of this page : https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > 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 vtorri at univ-evry.fr Wed Jan 5 20:56:41 2011 From: vtorri at univ-evry.fr (Vincent Torri) Date: Wed, 5 Jan 2011 20:56:41 +0100 (CET) Subject: [gst-devel] Need help for rezising video. In-Reply-To: <4D24C8B3.2050204@wanadoo.fr> References: <4D24C8B3.2050204@wanadoo.fr> Message-ID: On Wed, 5 Jan 2011, Nicolas wrote: > Hi Vincent, > > It does not change anything if i add it ! :) iirc, you set the source and decodebin, link them, then you pause the pipeline. You itrerate over the pad (that were added during the preroll). You then link your sink in the correct pad. See emotion before i implemented the pipeline with playbin2 Vincent From josel.segura at gmx.es Wed Jan 5 20:58:30 2011 From: josel.segura at gmx.es (=?ISO-8859-1?B?Sm9z6Q==?= Luis Segura Lucas) Date: Wed, 5 Jan 2011 20:58:30 +0100 Subject: [gst-devel] Pipeline paused, inside elements don't? Message-ID: <20110105205830.58bc713b@peter> Good afternoon! Some time ago I wrote a source and a sink elements using the Python bindings of Gstreamer. A few days ago, I come back to this code and, after a few tests, I see something estrange: when I set the pipeline state to "paused", my sink element continue on "playing" state. I wrote the sink element as a subclass of the "gst.BaseSink" Python class. The only thing that is supposed to do using that class is overwrite the "do_render" method. Do I need to do something more to set the element state to paused? Thanks in advance and best regards From bilboed at gmail.com Wed Jan 5 21:03:40 2011 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 05 Jan 2011 21:03:40 +0100 Subject: [gst-devel] Need help for rezising video. In-Reply-To: <4D24C8B3.2050204@wanadoo.fr> References: <4D24C8B3.2050204@wanadoo.fr> Message-ID: <1294257820.21480.2.camel@deumeu> On Wed, 2011-01-05 at 20:38 +0100, Nicolas wrote: > Hi Vincent, > > It does not change anything if i add it ! :) Could you please stop replying every time to the mail thread with a new standalone mail ? Learn how to use the reply-to-all feature of your mail client. Or switch mail client. Edward > > Nicolas. > > ------------------------------------------------------------------------------ > 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 bisht.sudarshan at gmail.com Thu Jan 6 00:06:27 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Thu, 6 Jan 2011 04:36:27 +0530 Subject: [gst-devel] Video freeze In-Reply-To: <1294028853954-3171570.post@n4.nabble.com> References: <1294028853954-3171570.post@n4.nabble.com> Message-ID: I want to suggest you something. Instead of creating elements at the time of taking screenshot , why don't you create them already, add into bin and then link them later when you want to take screen shot. And obviously by this approach you save some time in taking screenshot. And try to use bin for set of elements, for example screenshot_bin having related elements and so on. This makes your code easy to read and debug. On Mon, Jan 3, 2011 at 9:57 AM, lithium wrote: > > I've been writing a tv capture program. Basically by the end of it i'm > hoping > to have the gstreamer stuff setup somewhat like this: > > /* > /----queue----xvimagesink //screenoutput > v4l2src---------tee----- > \----queue----filesink //screenshot > \----queue----\ > ----multiplexer > ---filesink //recording > /----queue----/ > alsasrc----------tee----- > \----queue----alsasink //playback from > capture if needed > */ > > At the moment i've implemented the top video branch and am now working on > the screenshot one. > I've been using a combination of gtk and gstreamer to write the program. > > When the program starts I call the 'openVideoPipeline' function (see below) > and on clicking the screenshot button in my gtk window it runs the > 'take_screenshot' function. The problem I am having is that when I press > the > screenshot button it causes the screen output to freeze. The file is not > created by the filesink. I have found that when i click my button that > causes the tv card options menu to be created and drawn it causes the > screen > output to resume playing and the file to be written. The only functions > used > from gstreamer in the tv card menu are 'gst_tuner_list_channels', > 'gst_tuner_list_norms', and 'gst_tuner_get_norm'. It is very likely I am > going about taking screenshots the wrong way, if someone has a better idea > please suggest it. Otherwise it would be good if someone has any idea whats > causing the video to freeze. > > I have tried setting the pipeline to playing again after unlinking the > screenshot pads but it didn't change anything. > > int openVideoPipeline(videoIOStruct *videoIO) > { > //create elements > videoIO->pipeline = gst_pipeline_new("tvVideo"); > videoIO->vidSrc = gst_element_factory_make("v4l2src", "source"); > GstElement *vidSink = gst_element_factory_make("xvimagesink", > "sink"); > videoIO->tee = gst_element_factory_make("tee", "splitter"); > GstElement *vidQueue = gst_element_factory_make("queue", NULL); > > if (!videoIO->pipeline || !videoIO->vidSrc || !vidSink || > !videoIO->tee || > !vidQueue) > return 0; > > //set arguments > g_object_set(videoIO->vidSrc, "device", videoIO->devName, NULL); > > gst_bin_add_many(GST_BIN(videoIO->pipeline), videoIO->vidSrc, > videoIO->tee, > vidQueue, vidSink, NULL); > > videoIO->videoScreenPad = gst_element_get_request_pad(videoIO->tee, > "src%d"); > GstPad *staticOutputPad = gst_element_get_static_pad(vidQueue, > "sink"); > > //link stuff > gst_pad_link(videoIO->videoScreenPad, staticOutputPad); > > gst_object_unref(GST_OBJECT(videoIO->videoScreenPad)); > > gst_element_link(videoIO->vidSrc, videoIO->tee); > > gst_element_link(vidQueue, vidSink); > > if (!videoIO->pipeline) > return 0; > > return 1; > } > > void close_screenshot(GstElement *jpegEncode, videoIOStruct *videoIO) > { > GstPad *pad1 = GST_PAD(g_object_get_data(G_OBJECT(jpegEncode), > "videoScreenshotPad")); > GstPad *pad2 = GST_PAD(g_object_get_data(G_OBJECT(jpegEncode), > "staticOutputPad")); > gst_pad_unlink(pad1, pad2); > } > > void take_screenshot(GtkWidget *widget, videoIOStruct *videoIO) > { > GstElement *jpegOutput = gst_element_factory_make("filesink", > "jpegoutput"); > GstElement *jpegQueue = gst_element_factory_make("queue", > "jpgqueue"); > GstElement *jpegEncode = gst_element_factory_make("jpegenc", > "jpegencoder"); > > if (!jpegOutput || !jpegQueue || !jpegEncode || > !videoIO->jpegPipeline) > return; > > //set arguments > g_object_set(jpegOutput, "location", "test.jpg", NULL); > > //link stuff > gst_bin_add_many(GST_BIN(videoIO->pipeline), jpegEncode, jpegQueue, > jpegOutput, NULL); > > gst_element_link_many(jpegQueue, jpegEncode, jpegOutput, NULL); > > //gst_element_set_state(videoIO->jpegPipeline, GST_STATE_PLAYING); > > g_signal_connect(jpegEncode, "frame-encoded", > G_CALLBACK(close_screenshot), > videoIO); > > GstPad *videoScreenshotPad = > gst_element_get_request_pad(videoIO->tee, > "src%d"); > GstPad *staticOutputPad = gst_element_get_static_pad(jpegQueue, > "sink"); > g_object_set_data(G_OBJECT(jpegEncode), "videoScreenshotPad", > videoScreenshotPad); > g_object_set_data(G_OBJECT(jpegEncode), "staticOutputPad", > staticOutputPad); > > gst_pad_link(videoScreenshotPad, staticOutputPad); > } > > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/Video-freeze-tp3171570p3171570.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 > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From yuanqing.cai at tieto.com Thu Jan 6 02:36:43 2011 From: yuanqing.cai at tieto.com (Cai Yuanqing) Date: Thu, 6 Jan 2011 09:36:43 +0800 Subject: [gst-devel] Video freeze In-Reply-To: References: <1294028853954-3171570.post@n4.nabble.com> Message-ID: <4D251CAB.2020401@tieto.com> On 01/06/2011 07:06 AM, sudarshan bisht wrote: > I want to suggest you something. Instead of creating elements > at the time of taking screenshot , why don't you create them already, > add into bin and then link them later when you want to take screen > shot. And obviously by this approach you save some time in taking > screenshot. > > And try to use bin for set of elements, for example > screenshot_bin having related elements and so on. This makes your code > easy to read and debug. I agree with sudarshan that you'd better make the screen shot related elements into a Bin,and I guess that maybe some element can't change state to PLAYING caused your "video freezen" problem.I have no idea what happened in your case,you can enable the debug information and dig into it. BTW,taking screen shot by pipeline usually need some time at least 100ms IMHO :-) Hope it helps. > On Mon, Jan 3, 2011 at 9:57 AM, lithium > wrote: > > > I've been writing a tv capture program. Basically by the end of it > i'm hoping > to have the gstreamer stuff setup somewhat like this: > > /* > /----queue----xvimagesink > //screenoutput > v4l2src---------tee----- > \----queue----filesink //screenshot > \----queue----\ > ----multiplexer > ---filesink //recording > /----queue----/ > alsasrc----------tee----- > \----queue----alsasink //playback > from capture if needed > */ > > At the moment i've implemented the top video branch and am now > working on > the screenshot one. > I've been using a combination of gtk and gstreamer to write the > program. > > When the program starts I call the 'openVideoPipeline' function > (see below) > and on clicking the screenshot button in my gtk window it runs the > 'take_screenshot' function. The problem I am having is that when I > press the > screenshot button it causes the screen output to freeze. The file > is not > created by the filesink. I have found that when i click my button that > causes the tv card options menu to be created and drawn it causes > the screen > output to resume playing and the file to be written. The only > functions used > from gstreamer in the tv card menu are 'gst_tuner_list_channels', > 'gst_tuner_list_norms', and 'gst_tuner_get_norm'. It is very > likely I am > going about taking screenshots the wrong way, if someone has a > better idea > please suggest it. Otherwise it would be good if someone has any > idea whats > causing the video to freeze. > > I have tried setting the pipeline to playing again after unlinking the > screenshot pads but it didn't change anything. > > int openVideoPipeline(videoIOStruct *videoIO) > { > //create elements > videoIO->pipeline = gst_pipeline_new("tvVideo"); > videoIO->vidSrc = gst_element_factory_make("v4l2src", > "source"); > GstElement *vidSink = > gst_element_factory_make("xvimagesink", "sink"); > videoIO->tee = gst_element_factory_make("tee", "splitter"); > GstElement *vidQueue = gst_element_factory_make("queue", NULL); > > if (!videoIO->pipeline || !videoIO->vidSrc || !vidSink || > !videoIO->tee || > !vidQueue) > return 0; > > //set arguments > g_object_set(videoIO->vidSrc, "device", videoIO->devName, > NULL); > > gst_bin_add_many(GST_BIN(videoIO->pipeline), > videoIO->vidSrc, videoIO->tee, > vidQueue, vidSink, NULL); > > videoIO->videoScreenPad = > gst_element_get_request_pad(videoIO->tee, > "src%d"); > GstPad *staticOutputPad = > gst_element_get_static_pad(vidQueue, "sink"); > > //link stuff > gst_pad_link(videoIO->videoScreenPad, staticOutputPad); > > gst_object_unref(GST_OBJECT(videoIO->videoScreenPad)); > > gst_element_link(videoIO->vidSrc, videoIO->tee); > > gst_element_link(vidQueue, vidSink); > > if (!videoIO->pipeline) > return 0; > > return 1; > } > > void close_screenshot(GstElement *jpegEncode, videoIOStruct *videoIO) > { > GstPad *pad1 = GST_PAD(g_object_get_data(G_OBJECT(jpegEncode), > "videoScreenshotPad")); > GstPad *pad2 = GST_PAD(g_object_get_data(G_OBJECT(jpegEncode), > "staticOutputPad")); > gst_pad_unlink(pad1, pad2); > } > > void take_screenshot(GtkWidget *widget, videoIOStruct *videoIO) > { > GstElement *jpegOutput = gst_element_factory_make("filesink", > "jpegoutput"); > GstElement *jpegQueue = gst_element_factory_make("queue", > "jpgqueue"); > GstElement *jpegEncode = gst_element_factory_make("jpegenc", > "jpegencoder"); > > if (!jpegOutput || !jpegQueue || !jpegEncode || > !videoIO->jpegPipeline) > return; > > //set arguments > g_object_set(jpegOutput, "location", "test.jpg", NULL); > > //link stuff > gst_bin_add_many(GST_BIN(videoIO->pipeline), jpegEncode, > jpegQueue, > jpegOutput, NULL); > > gst_element_link_many(jpegQueue, jpegEncode, jpegOutput, NULL); > > //gst_element_set_state(videoIO->jpegPipeline, > GST_STATE_PLAYING); > > g_signal_connect(jpegEncode, "frame-encoded", > G_CALLBACK(close_screenshot), > videoIO); > > GstPad *videoScreenshotPad = > gst_element_get_request_pad(videoIO->tee, > "src%d"); > GstPad *staticOutputPad = > gst_element_get_static_pad(jpegQueue, "sink"); > g_object_set_data(G_OBJECT(jpegEncode), "videoScreenshotPad", > videoScreenshotPad); > g_object_set_data(G_OBJECT(jpegEncode), "staticOutputPad", > staticOutputPad); > > gst_pad_link(videoScreenshotPad, staticOutputPad); > } > > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/Video-freeze-tp3171570p3171570.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 > > > > > -- > Regards, > > Sudarshan Bisht -- B.R Cai Yuanqing From cmaster2 at hotmail.com Thu Jan 6 02:40:10 2011 From: cmaster2 at hotmail.com (lithium) Date: Wed, 5 Jan 2011 17:40:10 -0800 (PST) Subject: [gst-devel] Video freeze In-Reply-To: References: <1294028853954-3171570.post@n4.nabble.com> Message-ID: <1294278010574-3176765.post@n4.nabble.com> I've adjusted so its how you suggested. It does seem like a better idea, but as you can probably tell i'm completely new to gstreamer so I don't really know much about ghost pads or anything which i'm fairly sure I need to use to be able to use bins? I've had a go at it anyway. At the moment the video pauses as soon as the program starts, ie one image shows up that does not change (setup_screenshotBin is called on program starting) which occurs on this: gst_bin_add(GST_BIN(videoIO->pipeline), videoIO->screenshotBin);. I can get the video to play properly again as before, by opening the tv card menu. On taking a screenshot the release of the pad is not working. gst_element_release_request_pad(videoIO->tee, videoIO->screenshotOutPad); causes "(tvprog:6329): GStreamer-CRITICAL **: Padname '':src1 does not belong to element splitter when removing" I do get a valid jpeg created when screenshot is clicked. But as soon as the program starts an invalid (because theres no data in it) jpeg file is created in my program directory. I'm assuming this is because the pipeline is set to playing when the program starts. Is there a way I can stop it from creating this file until the screenshot button is clicked? This is what i've come up with: openVideoPipeline is the same void close_screenshot(GstElement *jpegEncode, videoIOStruct *videoIO) { gst_pad_unlink(videoIO->screenshotOutPad, videoIO->screenshotInPad); gst_element_release_request_pad(videoIO->tee, videoIO->screenshotOutPad); } void take_screenshot(GtkWidget *widget, videoIOStruct *videoIO) { videoIO->screenshotOutPad = gst_element_get_request_pad(videoIO->tee, "src%d"); gst_pad_link(videoIO->screenshotOutPad, videoIO->screenshotInPad); } int setup_screenshotBin(videoIOStruct *videoIO) { videoIO->screenshotBin = gst_bin_new("screenshot_bin"); GstElement *jpegOutput = gst_element_factory_make("filesink", "jpegoutput"); GstElement *jpegQueue = gst_element_factory_make("queue", "jpgqueue"); GstElement *jpegEncode = gst_element_factory_make("jpegenc", "jpegencoder"); if (!jpegOutput || !jpegQueue || !jpegEncode || !videoIO->screenshotBin) return 0; //set arguments gchar filename[] = "test.jpg"; g_object_set(G_OBJECT(jpegOutput), "location", filename, NULL); //link stuff gst_bin_add_many(GST_BIN(videoIO->screenshotBin), jpegQueue, jpegEncode, jpegOutput, NULL); gst_element_link_many(jpegQueue, jpegEncode, jpegOutput, NULL); g_signal_connect(jpegEncode, "frame-encoded", G_CALLBACK(close_screenshot), videoIO); GstPad *pad = gst_element_get_static_pad(jpegQueue, "sink"); gst_element_add_pad(videoIO->screenshotBin, gst_ghost_pad_new("sink", pad)); gst_object_unref(GST_OBJECT(pad)); gst_bin_add(GST_BIN(videoIO->pipeline), videoIO->screenshotBin); gst_element_sync_state_with_parent(videoIO->screenshotBin); videoIO->screenshotInPad = gst_element_get_static_pad(videoIO->screenshotBin, "sink"); return 1; } Thanks -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Video-freeze-tp3171570p3176765.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From cmaster2 at hotmail.com Thu Jan 6 02:43:47 2011 From: cmaster2 at hotmail.com (lithium) Date: Wed, 5 Jan 2011 17:43:47 -0800 (PST) Subject: [gst-devel] Video freeze In-Reply-To: <4D251CAB.2020401@tieto.com> References: <1294028853954-3171570.post@n4.nabble.com> <4D251CAB.2020401@tieto.com> Message-ID: <1294278227167-3176771.post@n4.nabble.com> I dont really have the option to take a screenshot another way. If I did another way I'd have to close the xvimagesink as the tv card that im working with's device does not allow multiple opens. I'll have a look at the debug info, thanks for mentioning that I should've realised there'd be some debug stuff in the gstreamer api. Just didn't think to check xP -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Video-freeze-tp3171570p3176771.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ssrktce at gmail.com Thu Jan 6 06:12:10 2011 From: ssrktce at gmail.com (ssrk) Date: Thu, 6 Jan 2011 10:42:10 +0530 Subject: [gst-devel] Pipeline for converting H264 bytestream to mp4 container format? In-Reply-To: References: Message-ID: Hi Sudarshan, Thanks for the reply, but a doubt even before trying that pipeline, .264 file is already an encoded AVC stream, what is the necessity to pass it to x264enc plugin? anyhow let me try the pipeline first. -- Regards, ssrk -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssrktce at gmail.com Thu Jan 6 06:18:39 2011 From: ssrktce at gmail.com (ssrk) Date: Thu, 6 Jan 2011 10:48:39 +0530 Subject: [gst-devel] Pipeline for converting H264 bytestream to mp4 container format? In-Reply-To: References: Message-ID: Hi, ERROR " h264parse x264enc " could not be linked.. If try the below pipeline : gst-launch-0.10 filesrc location=test.264 ! h264parse ! qtmux ! filesink location=test.mp4 I get this ERROR : gstqtmux.c(1347): gst_qt_mux_add_buffer (): /GstPipeline:pipeline0/GstQTMux:qtmux0: tried debuggin using the sources, looks like its because of timestamp issues. Any help on this? -- Regards, ssrk -------------- next part -------------- An HTML attachment was scrubbed... URL: From shreya.pandit25 at gmail.com Thu Jan 6 09:12:40 2011 From: shreya.pandit25 at gmail.com (Shreya Pandit) Date: Thu, 6 Jan 2011 13:42:40 +0530 Subject: [gst-devel] Query Message-ID: Hi I need to process images at 2 frames per second(input coming from my webcam). using the gstreamer interface.However, I am no t able to find a suitable command to execute the same. I would be very obliged if i get some help in this regard. Thanking You.... -------------- next part -------------- An HTML attachment was scrubbed... URL: From ikt011 at gmail.com Thu Jan 6 09:13:05 2011 From: ikt011 at gmail.com (Kocsis Tibor) Date: Thu, 6 Jan 2011 09:13:05 +0100 Subject: [gst-devel] sending raw stream over rtp In-Reply-To: References: <1294235326.28802.74.camel@meany> Message-ID: The RTSP server is written by me and my team. I'm using gstreamer elements for creating rtp packets: gstrtpbin and some udpsrc/udpsink elements, based on the gstreamer doc's example pipelines: http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-gstrtpbin.html The rtsp messages are handled through tcp connection, you need to implement the communication messages in the RTSP spec, for example if the client sends you the DESCRIBE message, you have to return the specifications of your stream in the sdp format. So if you want to handle the rtsp messages, you need to write an application. Gstreamer already has an rtsp server (gst-rtsp-server), and it is great for start, but you probably need your own in short time. Regards, Tibor On Wed, Jan 5, 2011 at 8:50 PM, Suneet Shah wrote: > Hi Kocis-- what RTSP server are you using? > I'm trying to do the opposite of what you're doing. I have a gstreamer > pipeline creating RTP packets, but need to package them with RTSP and create > an SDP file. > Thanks, > Suneet > > On Wed, Jan 5, 2011 at 5:48 AM, Wim Taymans wrote: >> >> On Wed, 2011-01-05 at 14:38 +0100, Kocsis Tibor wrote: >> > I created a command line to test the problem, the result is the same, >> > I got a lot of messages like this: >> > >> > 0:00:27.686127192 14222 ? ? ? 0xf1bd50 WARN ? ? ? ? ? ?rtpvrawdepay >> > gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: >> > clipping length 726, offset 398 >> > 0:00:27.686175449 14222 ? ? ? 0xf1bd50 WARN ? ? ? ? ? ?rtpvrawdepay >> > gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: >> > clipping length 438, offset 494 >> >> Those are normal. I will change them to LOG. >> >> Wim >> > The pipelines: >> > >> > Sender: >> > >> > gst-launch -v ?v4l2src ! >> > video/x-raw-yuv,width=640,height=480,framerate=10/1 ! clockoverlay ! >> > rtpvrawpay ! udpsink host=... port=5000 sync=false >> > >> > Receiver: >> > >> > gst-launch -v udpsrc port=5000 caps="application/x-rtp, >> > media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, >> > sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)640, >> > height=(string)480, colorimetry=(string)SMPTE240M, payload=(int)96, >> > ssrc=(uint)2464304599, clock-base=(uint)342261484, >> > seqnum-base=(uint)47511" ! rtpvrawdepay ! xvimagesink sync=false >> > >> > Caps copied from udsink.sink on the sender side. >> > >> > So is this warning message normal and should be info or maybe a >> > rawdepay bug or something else? >> > >> > Regards, >> > Tibor >> > >> > >> > >> > On Wed, Jan 5, 2011 at 10:13 AM, Kocsis Tibor wrote: >> > > The sdp I linked was wrong, my version is here: >> > > >> > > ?medias: >> > > ?media 0: >> > > ? media: ? ? ? 'video' >> > > ? port: ? ? ? ?'0' >> > > ? num_ports: ? '4294967295' >> > > ? proto: ? ? ? 'RTP/AVP' >> > > ? formats: >> > > ? ?format ?'96' >> > > ? information: '(NULL)' >> > > ? key: >> > > ? ?type: ? ? ? '(NULL)' >> > > ? ?data: ? ? ? '(NULL)' >> > > ? attributes: >> > > ? ?attribute 'rtpmap' : '96 RAW/90000' >> > > ? ?attribute 'control' : >> > > >> > > 'rtsp://10.30.0.8:1554/Live/trackID=0?channelID=98d16963-877f-4446-bb69-3aeada917523' >> > > ? ?attribute 'fmtp' : '96 >> > > width=640;height=480;sampling=YCbCr-4:2:0;depth=8' >> > > ? ?attribute 'appversion' : '5.0.0' >> > > >> > > >> > > On Wed, Jan 5, 2011 at 9:07 AM, Kocsis Tibor wrote: >> > >> Hi, >> > >> >> > >> I have a basic rtsp server, and I try to use to send a raw stream to >> > >> the client. On the receiver side I use a simple gstreamer pipeline >> > >> like this: rtspsrc ! rtpvrawdepay ! xvimagesink. On the server there >> > >> are gstreamer elements, too, some source connected to a rtpvrawpay, >> > >> and to the gstrtpsink elements. The sending is started, and I can >> > >> watch something live view, but it is not fluency. These messages >> > >> appear on the client: >> > >> >> > >> 0:00:03.233100336 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay >> > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: >> > >> clipping length 540, offset 460 >> > >> 0:00:03.233134052 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay >> > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: >> > >> clipping length 1086, offset 278 >> > >> 0:00:03.233192201 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay >> > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: >> > >> clipping length 252, offset 556 >> > >> 0:00:03.233232346 24189 0x7f654000bd20 WARN ? ? ? ? ? ?rtpvrawdepay >> > >> gstrtpvrawdepay.c:378:gst_rtp_vraw_depay_process: >> > >> clipping length 798, offset 374 >> > >> ... >> > >> >> > >> What can it cause? >> > >> >> > >> My guess was I missed from the sdp (or built it up wrong), but the >> > >> params I send over in the fmtp is the same as on the payloader's >> > >> source pad (width, height, sample, even as depth, but as I see depth >> > >> is not parsed on the client side). Btw my rtsp server works with >> > >> h264/mpeg4 fine, so the elements are correctly linked. >> > >> >> > >> The full rtsp communication is here: http://pastebin.com/fC8MsJkx >> > >> >> > >> Any idea is welcomed >> > >> >> > >> Regards >> > >> Tibor >> > >> >> > > >> > >> > >> > ------------------------------------------------------------------------------ >> > 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 lijinsyam at gmail.com Thu Jan 6 09:21:55 2011 From: lijinsyam at gmail.com (LIJIN SYAM K) Date: Thu, 6 Jan 2011 13:51:55 +0530 Subject: [gst-devel] Query In-Reply-To: References: Message-ID: Hi... What about Setting the capsfilter in the piepline ...and you can change the "framerate" property... On Thu, Jan 6, 2011 at 1:42 PM, Shreya Pandit wrote: > Hi > > I need to process images at 2 frames per second(input coming from my > webcam). using the gstreamer interface.However, I am no t able to find a > suitable command to execute the same. > > I would be very obliged if i get some help in this regard. > > > Thanking You.... > > > > ------------------------------------------------------------------------------ > 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 narendra at narendrasisodiya.com Thu Jan 6 09:31:09 2011 From: narendra at narendrasisodiya.com (Narendra Sisodiya) Date: Thu, 6 Jan 2011 14:01:09 +0530 Subject: [gst-devel] Taking snapshot from webcam using gstreamer and python Message-ID: I am looking for a script by which i can take snapshot from my webcam using gstreamer and python. even command line gst is ok, I can look for python at my own. I searched over internet but unable to find working example. When I run some commands (given in some tutorial) , I am only able to get 10% portion in image .. also, I want to take snapshot at 2 frame per second. Thanks -- ??????????????????????????? ? Narendra Sisodiya ? http://narendrasisodiya.com ??????????????????????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Thu Jan 6 12:17:44 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Thu, 6 Jan 2011 16:47:44 +0530 Subject: [gst-devel] Pipeline for converting H264 bytestream to mp4 container format? In-Reply-To: References: Message-ID: Oh yes, I am sorry . Its already an encoded file so encoder ( x264enc ) is not required in pipeline. But when you try the other pipeline what error you got, can you elaborate log ? On Thu, Jan 6, 2011 at 10:48 AM, ssrk wrote: > Hi, > > ERROR " h264parse x264enc " could not be linked.. > If try the below pipeline : > > gst-launch-0.10 filesrc location=test.264 ! h264parse ! qtmux ! filesink > location=test.mp4 > > I get this ERROR : gstqtmux.c(1347): gst_qt_mux_add_buffer (): > /GstPipeline:pipeline0/GstQTMux:qtmux0: > > tried debuggin using the sources, looks like its because of timestamp > issues. > > Any help on this? > > -- > Regards, > ssrk > > > > ------------------------------------------------------------------------------ > 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 puneeth.mv at globaledgesoft.com Thu Jan 6 12:48:21 2011 From: puneeth.mv at globaledgesoft.com (Puneeth) Date: Thu, 6 Jan 2011 03:48:21 -0800 (PST) Subject: [gst-devel] Problem in using ffdec_mp3 to decode live streaming Message-ID: <1294314501815-3177271.post@n4.nabble.com> Hi All, I m facing problem in decoding mp3 using ffdec_mp3 when i m capturing audio and listening to it, but i m able to hear when i am using mad decoder with the below command, but there are some glitches when i replace mad with ffdec_mp3. But i m able to playback the saved file using ffplay which i have captured. So i guess ffdec is not capable of decoding the live streaming data, i need some help to solve this bug. Any help will be greatly appreciated. Thanks in advance. sudo gst-launch gstrtpbin name=rtpbin alsasrc ! audioconvert ! audioresample ! lamemp3enc ! rtpmpapay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5002 host=127.0.0.1 rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=127.0.0.1 sync=false async=false udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0 sudo gst-launch gstrtpbin name=rtpbin udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)90000,encoding-name=(string)MPA,pt=(int)96" port=5002 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtpmpadepay ! mp3parse ! mad ! audioconvert ! volume volume=10 ! audioresample quality=10 ! alsasink udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=127.0.0.1 sync=false async=false -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Problem-in-using-ffdec-mp3-to-decode-live-streaming-tp3177271p3177271.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bisht.sudarshan at gmail.com Thu Jan 6 13:09:29 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Thu, 6 Jan 2011 17:39:29 +0530 Subject: [gst-devel] Pipeline for converting H264 bytestream to mp4 container format? In-Reply-To: References: Message-ID: Give one try to this pipeline; gst-launch-0.10 filesrc location=test.264 ! h264parse output-format=sample access-unit=true ! qtmux ! filesink location=test.mp4 On Thu, Jan 6, 2011 at 4:47 PM, sudarshan bisht wrote: > Oh yes, I am sorry . Its already an encoded file so encoder ( x264enc ) is > not required in pipeline. > > But when you try the other pipeline what error you got, can you elaborate > log ? > > > On Thu, Jan 6, 2011 at 10:48 AM, ssrk wrote: > >> Hi, >> >> ERROR " h264parse x264enc " could not be linked.. >> If try the below pipeline : >> >> gst-launch-0.10 filesrc location=test.264 ! h264parse ! qtmux ! filesink >> location=test.mp4 >> >> I get this ERROR : gstqtmux.c(1347): gst_qt_mux_add_buffer (): >> /GstPipeline:pipeline0/GstQTMux:qtmux0: >> >> tried debuggin using the sources, looks like its because of timestamp >> issues. >> >> Any help on this? >> >> -- >> Regards, >> ssrk >> >> >> >> ------------------------------------------------------------------------------ >> 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 > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From ikt011 at gmail.com Thu Jan 6 13:46:31 2011 From: ikt011 at gmail.com (Kocsis Tibor) Date: Thu, 6 Jan 2011 13:46:31 +0100 Subject: [gst-devel] Taking snapshot from webcam using gstreamer and python In-Reply-To: References: Message-ID: Try this: http://wiki.oz9aec.net/index.php/Gstreamer_Cheat_Sheet#Single_frame_capture On Thu, Jan 6, 2011 at 9:31 AM, Narendra Sisodiya wrote: > I am looking for a script by which i can take snapshot from? my webcam using > gstreamer and python. even command line gst is ok, I can look for python at > my own. I searched over internet but unable to find working example. > When I run some commands (given in some tutorial) , I am only able to get > 10% portion in image .. > also, I want to take snapshot at 2 frame per second. > > Thanks > > -- > ??????????????????????????? > ?? ? Narendra Sisodiya > ?? ? http://narendrasisodiya.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 ssrktce at gmail.com Thu Jan 6 14:45:44 2011 From: ssrktce at gmail.com (ssrk) Date: Thu, 6 Jan 2011 19:15:44 +0530 Subject: [gst-devel] Pipeline for converting H264 bytestream to mp4 container format? In-Reply-To: References: Message-ID: Hi sudarshan, Tried tat one also, same error :( -- Regards, ssrk -------------- next part -------------- An HTML attachment was scrubbed... URL: From lfarkas at lfarkas.org Thu Jan 6 14:50:30 2011 From: lfarkas at lfarkas.org (Farkas Levente) Date: Thu, 06 Jan 2011 14:50:30 +0100 Subject: [gst-devel] does signal has last param user_data in gstreamer? Message-ID: <4D25C8A6.2060502@lfarkas.org> hi, let see an example: both gst-inspect input-selector and http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-input-selector.html said that the signal "switch" has a last argument as gpointer user_data and the signal "block" has no such last argument in the gst-inspect output, but has in the above docs. but when i look onto the source code in gstinputselector.h none of these signal has last param! since there are many signal (almos all) which has last param user_data so my questions in general: is the last param user_data exist or not at all? or it's something glib specific thing or ...? thanks in advance. regards. -- Levente "Si vis pacem para bellum!" From gemont at igalia.com Thu Jan 6 15:29:54 2011 From: gemont at igalia.com (Guillaume Emont) Date: Thu, 06 Jan 2011 15:29:54 +0100 Subject: [gst-devel] does signal has last param user_data in gstreamer? In-Reply-To: <4D25C8A6.2060502@lfarkas.org> References: <4D25C8A6.2060502@lfarkas.org> Message-ID: <4D25D1E2.50403@igalia.com> Hi Farkas, That is indeed a gobject signal thing: when you connect [1] to a signal, you specify a "user_data" pointer to pass to the callback you're attaching, whatever the signal. This is all should be explained in the signal section of the GObject Reference Manual [2]. Guillaume [1] http://library.gnome.org/devel/gobject/2.26/gobject-Signals.html#g-signal-connect [2] http://library.gnome.org/devel/gobject/2.26/signal.html On 06/01/2011 14:50, Farkas Levente wrote: > hi, > let see an example: both gst-inspect input-selector and > http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-input-selector.html > said that the signal "switch" has a last argument as > gpointer user_data > and the signal "block" has no such last argument in the gst-inspect > output, but has in the above docs. > but when i look onto the source code in gstinputselector.h none of these > signal has last param! > > since there are many signal (almos all) which has last param user_data > so my questions in general: > is the last param user_data exist or not at all? or it's something glib > specific thing or ...? > thanks in advance. > regards. > From henriette.depoulpiquet at gmail.com Thu Jan 6 16:01:36 2011 From: henriette.depoulpiquet at gmail.com (henriette de poulpiquet) Date: Thu, 6 Jan 2011 16:01:36 +0100 Subject: [gst-devel] install gst-plugins-gl-0.10.2 In-Reply-To: References: Message-ID: Hi! I finally make it all works ! :D in fact I tried so many times to install gstreamer and somes plugins that I left some "*.so" etc every where and didn't see it.. so I resolve my problem by uninstalling every gst-thing and reinstalling them all :) thanks for your support!!! have a nice day ! regards Henriette On 1/5/11, henriette de poulpiquet wrote: > Hi! > > thanks Sudarshen, > > I think my env variables are ok.. > other plugins installed in the same way was installed without any > problems.. > can anyone explain me what does the plugins blacklist mean? > > thanks! > > Henriette > > On 1/4/11, sudarshan bisht wrote: >> No idea about Glib. But just a guess again, check LD_LIBRARY_PATH and >> other >> environment variables. Since your Gstreamer installation is there in >> /usr/local/lib not in /usr/lib. >> >> >> On Tue, Jan 4, 2011 at 10:38 PM, henriette de poulpiquet < >> henriette.depoulpiquet at gmail.com> wrote: >> >>> maybe my Glib version is important? >>> Glib 2.26.1 >>> >>> .. no one ever had this problem? >>> >>> anyway, thanks Sudarshan for your help! >>> >>> Henriette >>> >>> On 1/4/11, henriette de poulpiquet >>> wrote: >>> > yes I did... >>> > >>> > >>> > On 1/4/11, sudarshan bisht wrote: >>> >> Did you do " export GST_PLUGIN_PATH=/usr/local/lib/gstreamer-0.10" >>> ? >>> >> >>> >> On Tue, Jan 4, 2011 at 1:47 PM, henriette de poulpiquet < >>> >> henriette.depoulpiquet at gmail.com> wrote: >>> >> >>> >>> Hi! >>> >>> >>> >>> I'me not sure where to post my question, hope i'me in the right >>> >>> place! >>> >>> >>> >>> so my problem is that my gst-inspect indicate that gl plugin is >>> >>> blacklisted, I'me quite disappointed and can't understand why >>> >>> /usr/local/lib/gstreamer-0.10/libgstopengl.so can' t be found since >>> >>> libgstopengl.so do exists at this place (I checked it). >>> >>> >>> >>> what should I do? >>> >>> >>> >>> thanks for your answer! >>> >>> >>> >>> >>> >>> Henriette >>> >>> >>> >>> i'me working on a >>> >>> Linux: Debian 32 bits >>> >>> Release: 2.6.26-2-686 >>> >>> i686 >>> >>> gstreamer-0.10.31 >>> >>> gst-plugins-base-0.10.31 >>> >>> gst-plugins-good-0.10.26 >>> >>> >>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> >>> 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 >>> >> >> >> >> -- >> Regards, >> >> Sudarshan Bisht >> > From lfarkas at lfarkas.org Thu Jan 6 16:31:01 2011 From: lfarkas at lfarkas.org (Farkas Levente) Date: Thu, 06 Jan 2011 16:31:01 +0100 Subject: [gst-devel] does signal has last param user_data in gstreamer? In-Reply-To: <4D25D1E2.50403@igalia.com> References: <4D25C8A6.2060502@lfarkas.org> <4D25D1E2.50403@igalia.com> Message-ID: <4D25E035.1090506@lfarkas.org> hi, then the two signal definition in input-selector is bad? or.. http://cgit.freedesktop.org/gstreamer/gstreamer/tree/plugins/elements/gstinputselector.h since there is no last argument user_data? On 01/06/2011 03:29 PM, Guillaume Emont wrote: > Hi Farkas, > > That is indeed a gobject signal thing: when you connect [1] to a signal, > you specify a "user_data" pointer to pass to the callback you're > attaching, whatever the signal. > This is all should be explained in the signal section of the GObject > Reference Manual [2]. > > Guillaume > > [1] > http://library.gnome.org/devel/gobject/2.26/gobject-Signals.html#g-signal-connect > [2] http://library.gnome.org/devel/gobject/2.26/signal.html > > On 06/01/2011 14:50, Farkas Levente wrote: >> hi, >> let see an example: both gst-inspect input-selector and >> http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-input-selector.html >> said that the signal "switch" has a last argument as >> gpointer user_data >> and the signal "block" has no such last argument in the gst-inspect >> output, but has in the above docs. >> but when i look onto the source code in gstinputselector.h none of these >> signal has last param! >> >> since there are many signal (almos all) which has last param user_data >> so my questions in general: >> is the last param user_data exist or not at all? or it's something glib >> specific thing or ...? >> thanks in advance. >> 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 -- Levente "Si vis pacem para bellum!" From bilboed at gmail.com Thu Jan 6 16:41:47 2011 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 06 Jan 2011 16:41:47 +0100 Subject: [gst-devel] does signal has last param user_data in gstreamer? In-Reply-To: <4D25E035.1090506@lfarkas.org> References: <4D25C8A6.2060502@lfarkas.org> <4D25D1E2.50403@igalia.com> <4D25E035.1090506@lfarkas.org> Message-ID: <1294328507.23475.11.camel@localhost> On Thu, 2011-01-06 at 16:31 +0100, Farkas Levente wrote: > hi, > then the two signal definition in input-selector is bad? or.. > http://cgit.freedesktop.org/gstreamer/gstreamer/tree/plugins/elements/gstinputselector.h > since there is no last argument user_data? Those are the vmethod implementations called *on* the object (read more of the gobject docs to understand what I mean). user_data is only used in the callbacks and added by the signalling system auto-magically. Edward > > On 01/06/2011 03:29 PM, Guillaume Emont wrote: > > Hi Farkas, > > > > That is indeed a gobject signal thing: when you connect [1] to a signal, > > you specify a "user_data" pointer to pass to the callback you're > > attaching, whatever the signal. > > This is all should be explained in the signal section of the GObject > > Reference Manual [2]. > > > > Guillaume > > > > [1] > > http://library.gnome.org/devel/gobject/2.26/gobject-Signals.html#g-signal-connect > > [2] http://library.gnome.org/devel/gobject/2.26/signal.html > > > > On 06/01/2011 14:50, Farkas Levente wrote: > >> hi, > >> let see an example: both gst-inspect input-selector and > >> http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-input-selector.html > >> said that the signal "switch" has a last argument as > >> gpointer user_data > >> and the signal "block" has no such last argument in the gst-inspect > >> output, but has in the above docs. > >> but when i look onto the source code in gstinputselector.h none of these > >> signal has last param! > >> > >> since there are many signal (almos all) which has last param user_data > >> so my questions in general: > >> is the last param user_data exist or not at all? or it's something glib > >> specific thing or ...? > >> thanks in advance. > >> 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 lijinsyam at gmail.com Thu Jan 6 16:58:42 2011 From: lijinsyam at gmail.com (LIJIN SYAM K) Date: Thu, 6 Jan 2011 21:28:42 +0530 Subject: [gst-devel] Memory Issue in Gstreamer Message-ID: Hi all... I just wrote a simple program to read a mjpeg stream from axis camera and write it in to a filesink. Here i was testing the memory issue. When i run the programme for one day the memory from some anonymous location was increased to *1MB * About my programme : souphttpsrc----> multipartdemux----->queue ---->capsfilter---->jpegdec----->videorate---> ffenc_flv----->queue ----> ffmux_flv------>filesink Here to divide the recording file in to small chunks i add a data probe in filesink and in the call back when the size exceeds the predefined size it will block the src pad of the queue the one which stand before the flv_mux using the function *gst_pad_set_blocked_async* . In the block call back i change the filesink location.. Its working fine...except this memory Issue. Here is my source code .from pastebin http://pastebin.com/pTDaAyu5 Is there anything wrong in the source code ... Please Advice ... Thanks Lijin -------------- next part -------------- An HTML attachment was scrubbed... URL: From narendra at narendrasisodiya.com Thu Jan 6 18:00:32 2011 From: narendra at narendrasisodiya.com (Narendra Sisodiya) Date: Thu, 6 Jan 2011 22:30:32 +0530 Subject: [gst-devel] Taking snapshot from webcam using gstreamer and python In-Reply-To: References: Message-ID: On Thu, Jan 6, 2011 at 6:16 PM, Kocsis Tibor wrote: > Try this: > http://wiki.oz9aec.net/index.php/Gstreamer_Cheat_Sheet#Single_frame_capture > > Thanks a lot, It worked. I will write python program for it. It there any easy way to convert gst command to corresponding python command ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Thu Jan 6 18:09:09 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 06 Jan 2011 17:09:09 +0000 Subject: [gst-devel] GStreamer core/base/good/ugly/bad git modules are FROZEN FOR COMMITS Message-ID: <1294333749.17597.253.camel@zingle> Hi, GStreamer core, gst-plugins-base, gst-plugins-good, and gst-plugins-ugly and gst-plugins-bad git are now frozen for commits. Please do not commit to these modules any longer. Any patches to go in during the freeze period should be submitted by way of filing a blocker bug in bugzilla [1] (or asking on IRC for trivial fixes). A first set of pre-release tarballs should be available later today. Cheers -Tim [1] http://gstreamer.freedesktop.org/wiki/ReleasePlanning/Freezes From narendra at narendrasisodiya.com Thu Jan 6 18:09:57 2011 From: narendra at narendrasisodiya.com (Narendra Sisodiya) Date: Thu, 6 Jan 2011 22:39:57 +0530 Subject: [gst-devel] Adding TimeStamp in video recorded from GST for workshop vidoes. Message-ID: Let me explain my requirement. I am creating a "UNIQUE" way for Storing workshop or seminar content. In a proposed architecture , I am splitting content in separate streams. So Audio, Video, Slides, All will go in separate stream. So I am writing a program which will record videos , audio and play slides. Now when everything go in separate streams then how I will be able to synchronize all the streams ? so basically I am looking for a way to put some meta data in stream or outside the stream which can be used to synchronize all the streams. and how gst can help. may be you can direct me to more suitable mailing list where I can ask such question. -- ??????????????????????????? ? Narendra Sisodiya ? http://narendrasisodiya.com ??????????????????????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: From mhardt at diskohq.com Thu Jan 6 17:43:41 2011 From: mhardt at diskohq.com (Matthias Hardt) Date: Thu, 6 Jan 2011 17:43:41 +0100 Subject: [gst-devel] Linking elements in running state Message-ID: <20110106164341.GA8567@miniriot> Hello, I'm trying to link elements to streams in an already running pipeline. Now the problem is, that these elements don't have the segments start/stop... times as they didn't receive the NEW_SEGMENT event, which was sent earlier. Is it possible to have this event sent again? For now I tried querying the segment from the pads parent element I'm linking to, but unfortunately I don't get the correct times. And the format is GST_FORMAT_BYTES although I requested GST_FORMAT_TIME. Maybe I'm completely wrong and it can be done another way. Any help would be appreciated. Here is the code I am using: /* get segment times from stream pads parent element */ GstElement *el = gst_pad_get_parent_element(pad); GstQuery* query = gst_query_new_segment(GST_FORMAT_TIME); gboolean ret = gst_element_query(el, query); if(ret) { GstFormat format; gdouble rate; gint64 start, stop; gst_query_parse_segment(query, &rate, &format, &start, &stop); gst_query_unref(query); gint64 position; if(!gst_element_query_position(el, &format, &position)) position = 0; gst_pad_push_event (pad, gst_event_new_new_segment (FALSE, rate, format, start, stop, position)); printf("configured NEWSEGMENT rate %lf, format %d, %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT ", pos %" GST_TIME_FORMAT "\n", rate, format, GST_TIME_ARGS(start), GST_TIME_ARGS(stop), GST_TIME_ARGS(position)); } gst_object_unref(el); Best regards, Matthias From rohitratri at gmail.com Fri Jan 7 04:37:34 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Fri, 7 Jan 2011 09:07:34 +0530 Subject: [gst-devel] Problem in using ffdec_mp3 to decode live streaming In-Reply-To: <1294314501815-3177271.post@n4.nabble.com> References: <1294314501815-3177271.post@n4.nabble.com> Message-ID: Have you looked at the debug logs? (You could start with level 3) Did you find anything out of place there? May be a warning or something? Also, I suggest you take logs for both the decoders and compare them. You should be able to spot the problem. Rohit On Thu, Jan 6, 2011 at 5:18 PM, Puneeth wrote: > > Hi All, > > I m facing problem in decoding mp3 using ffdec_mp3 when i m capturing > audio and listening to it, but i m able to hear when i am using mad > decoder > with the below command, but there are some glitches when i replace mad > with > ffdec_mp3. > But i m able to playback the saved file using ffplay which i have captured. > So i guess ffdec is not capable of decoding > the live streaming data, i need some help to solve this bug. Any help will > be greatly appreciated. Thanks in advance. > > sudo gst-launch gstrtpbin name=rtpbin alsasrc ! audioconvert ! > audioresample > ! lamemp3enc ! rtpmpapay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! > udpsink port=5002 host=127.0.0.1 rtpbin.send_rtcp_src_0 ! udpsink port=5003 > host=127.0.0.1 sync=false async=false udpsrc port=5007 ! > rtpbin.recv_rtcp_sink_0 > > sudo gst-launch gstrtpbin name=rtpbin udpsrc > > caps="application/x-rtp,media=(string)audio,clock-rate=(int)90000,encoding-name=(string)MPA,pt=(int)96" > port=5002 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtpmpadepay ! mp3parse ! mad ! > audioconvert ! volume volume=10 ! audioresample quality=10 ! alsasink > udpsrc > port=5003 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink > port=5007 host=127.0.0.1 sync=false async=false > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/Problem-in-using-ffdec-mp3-to-decode-live-streaming-tp3177271p3177271.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 pierre.crepieux at orange-ftgroup.com Fri Jan 7 11:36:46 2011 From: pierre.crepieux at orange-ftgroup.com (pierre.crepieux at orange-ftgroup.com) Date: Fri, 07 Jan 2011 11:36:46 +0100 Subject: [gst-devel] sending audio/video on RTP with C API Message-ID: <4D26ECBE.4020800@orange-ftgroup.com> An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: orange_logo.gif Type: image/gif Size: 1264 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: ampersand.gif Type: image/gif Size: 1081 bytes Desc: not available URL: From bilboed at gmail.com Fri Jan 7 11:46:37 2011 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 07 Jan 2011 11:46:37 +0100 Subject: [gst-devel] sending audio/video on RTP with C API In-Reply-To: <4D26ECBE.4020800@orange-ftgroup.com> References: <4D26ECBE.4020800@orange-ftgroup.com> Message-ID: <1294397198.23475.13.camel@localhost> On Fri, 2011-01-07 at 11:36 +0100, pierre.crepieux at orange-ftgroup.com wrote: > Hi, > > I'm trying to write a simple C program that sends webcam/microphone > streams on rtp with gstrtpbin. > I gave a look at the provided server-alsasrc-PCMA.c test program and > mimics it to add h264 stream to the rtpbin (just like > server-v4l2-H264-alsasrc-PCMA.sh does). > Sadly, using VLC to receive the streams, give me a fixed image and a > pretty bad sound (delay, sometime stop/restart,...). > Are there any tricks to be aware of for doing this translation of > gst-launch pipe definition to C code ? Does gstrtpbin require subtle > manipulations (that gst-launch would do behind the scene ... ??? ) to > operate with 2 rtp stream ? You might want to use gst-rtsp-server instead. It does all the magic for you. Edward > > Pierre > -- > > CREPIEUX Pierre RD-SIRP-LAN > LAN/SIRP/ASF > Tel : +33 296053490 > Email: pierre.crepieux at orange-ftgroup.com > R&D Lannion 2, avenue Pierre Marzin 22307 LANNION Cedex - France > > > Mailer ThunderBird sous YourDev Linux > Installez simplement votre poste de travail Linux avec > http://web2000.rd.francetelecom.fr > > > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 Jan 7 11:49:38 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Fri, 7 Jan 2011 12:49:38 +0200 Subject: [gst-devel] How to convert G722 audio from pcap to file format In-Reply-To: <1294191533311-3174763.post@n4.nabble.com> References: <1294191533311-3174763.post@n4.nabble.com> Message-ID: Hi, On Wed, Jan 5, 2011 at 3:38 AM, shiva varma wrote: > > I have the following pipeline > > gst-launch filesrc location=g722.pcap ! pcapparse src-ip=10.22.120.89 > src-port=27812 ! ! "application/x-rtp, payload=9, clock-rate=8000" ! > rtpg722depay ! ................ > > After the rtpg722depay, i need an encoder which takes the audio/G722 as its > sink and produces an output format which can save to file > > My attempt to find a suitable encoder among the existing plugins have been > futile, can someone point me to the plugin I can use for this purpose > afaik it's not easy to find a standard muxer capable to handle g722 or any other speech codec. You can anyway try with gdppay / gdpdepay. Regards > Thanks > Shiva > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-convert-G722-audio-from-pcap-to-file-format-tp3174763p3174763.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 Jan 7 12:21:24 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Fri, 7 Jan 2011 13:21:24 +0200 Subject: [gst-devel] Getting converted frame from playbin2 element In-Reply-To: <1294053159060-3171830.post@n4.nabble.com> References: <1294053159060-3171830.post@n4.nabble.com> Message-ID: Hi, On Mon, Jan 3, 2011 at 1:12 PM, vaisaari wrote: > > Hi all, > > I'm trying to implement a function that returns the current frame of > playbin2 element as a QImage (Qt data type). When I call the function (for > the same video), sometimes it works perfectly, sometimes parts of the frame > (bottom rows) are missing. What am I doing wrong? Should I somehow be able > to wait for the "convert-frame" operation to finish, or what? You can find > my example code below. Thank you for your help! imo this should not happen, as the signal handler (delegated to playsink) internally calls gst_play_sink_get_last_frame, which locks playsink. Can you post to pastebin the output you get after setting GST_DEBUG=playsink:5 ? Which video decoder are you using? Besides, if you're trying to render a video clip in Qt, you may consider using QtGStreamer.. Regards > > > QImage Video::getFrame() > { > > GstBuffer *buffer; > GstCaps *caps; > > caps = gst_caps_new_simple("video/x-raw-rgb","bpp", G_TYPE_INT, 24,"depth", > G_TYPE_INT, 24,NULL); > > g_signal_emit_by_name(player, "convert-frame", caps, &buffer); > > unsigned char* data=(unsigned char *) GST_BUFFER_DATA (buffer); > QImage img(data, width, height, QImage::Format_RGB888); > > return img; > } > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Getting-converted-frame-from-playbin2-element-tp3171830p3171830.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 miguel at mtelleria.com Fri Jan 7 12:05:30 2011 From: miguel at mtelleria.com (Miguel Telleria de Esteban) Date: Fri, 7 Jan 2011 12:05:30 +0100 Subject: [gst-devel] sending audio/video on RTP with C API In-Reply-To: <4D26ECBE.4020800@orange-ftgroup.com> References: <4D26ECBE.4020800@orange-ftgroup.com> Message-ID: <20110107120530.2f8713ab@mtelleria.com> Hello, Like you are doing now, I followed the same goal of using gstrtpbin and found many small nuances. You can find here an implementation that I did to send ogg (vorbis and theora) audio through RTP using gstrtpbin. http://www.ctr.unican.es/asignaturas/dec/Doc/ogg_audio_video_rtp_sender_receiver.tar.gz Hope it helps. And yes, I plan to submit it as a bugzilla addition sometime in the future :). Miguel On Fri, 07 Jan 2011 11:36:46 +0100 pierre.crepieux wrote: > Hi, > > I'm trying to write a simple C program that sends webcam/microphone > streams on rtp with gstrtpbin. I gave a look at the provided > server-alsasrc-PCMA.c test program and mimics it to add h264 stream > to the rtpbin (just like server-v4l2-H264-alsasrc-PCMA.sh does). > Sadly, using VLC to receive the streams, give me a fixed image and a > pretty bad sound (delay, sometime stop/restart,...). Are there any > tricks to be aware of for doing this translation of gst-launch pipe > definition to C code ? Does gstrtpbin require subtle manipulations > (that gst-launch would do behind the scene ... ??? ) to operate with > 2 rtp stream ? > > Pierre > -- > > CREPIEUX Pierre RD-SIRP-LAN > LAN/SIRP/ASF > Tel : +33 296053490 > Email: pierre.crepieux at orange-ftgroup.com > R&D Lannion 2, avenue Pierre Marzin 22307 LANNION Cedex - France > > > Mailer ThunderBird sous YourDev Linux > Installez simplement votre poste de travail Linux avec > http://web2000.rd.francetelecom.fr > > > -- (O-O) ---oOO-(_)-OOo----------------------------------------------------- Miguel TELLERIA DE ESTEBAN http://www.mtelleria.com Email: miguel at mtelleria.com Tel GSM: +34 650 801098 Tel Fix: +34 942 280174 Miembro de http://www.linuca.org Membre du http://www.bxlug.be ?Usuario captivo o libre? http://www.obtengalinux.org/windows/ Free or captive user? http://www.getgnulinux.org/windows/ ------------------------------------------------------------------- -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: not available URL: From ceyusa at gmail.com Fri Jan 7 12:35:48 2011 From: ceyusa at gmail.com (=?ISO-8859-1?Q?Victor_Manuel_J=E1quez_Leal?=) Date: Fri, 7 Jan 2011 12:35:48 +0100 Subject: [gst-devel] OpenMax Source code In-Reply-To: <1294226244728-3175284.post@n4.nabble.com> References: <1294226244728-3175284.post@n4.nabble.com> Message-ID: On Wed, Jan 5, 2011 at 12:17 PM, vaishnavi wrote: > > Hi, > Can anyone tell me where i can find the source code for openmax IL by > STMicroelectronics.The code im asking for is in reference to document "Using > OpenMAX Integration Layer with GStreamer" . In this document they have a > flow diagram using MP3 decoder element which handles buffer management. I'm > looking for this source code. I guess you're asking for this: http://omxil.sourceforge.net/ vmjl > > Thank you > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/OpenMax-Source-code-tp3175284p3175284.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 vaish05_s at yahoo.com Fri Jan 7 14:22:01 2011 From: vaish05_s at yahoo.com (vaishnavi) Date: Fri, 7 Jan 2011 05:22:01 -0800 (PST) Subject: [gst-devel] Buffer copying Message-ID: <1294406521793-3179110.post@n4.nabble.com> We are developing a plug-in which has interface with Hardware driver for transcoding. To avoid copying, in the driver, we use get_user_pages to map user space memory to the kernel space. When we use filesrc to read from the file and pass on to this plug-in, we are currently copying the buffer passed from filesrc to another memory which is allocated using posix_memalign in the plug-in. Is there a way where we can avoid this copying by asking filesrc to do page alignment when it allocates a buffer for reading from the file? Thank you -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Buffer-copying-tp3179110p3179110.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From rob at ti.com Fri Jan 7 16:54:31 2011 From: rob at ti.com (Clark, Rob) Date: Fri, 7 Jan 2011 09:54:31 -0600 Subject: [gst-devel] Buffer copying In-Reply-To: <1294406521793-3179110.post@n4.nabble.com> References: <1294406521793-3179110.post@n4.nabble.com> Message-ID: it may be a bit heavy handed to align *all* buffers to page boundary, but you could do that by using the --with-buffer-alignment argument to gstreamer core configure script BR, -R On Fri, Jan 7, 2011 at 7:22 AM, vaishnavi wrote: > > We are developing a plug-in which has interface with Hardware driver for > transcoding. > > To avoid copying, in the driver, we use get_user_pages to map user space > memory to the > > kernel space. > > > > When we use filesrc to read from the file and pass on to this plug-in, we > are currently > > copying the buffer passed from filesrc to another memory which is allocated > using > > posix_memalign in the plug-in. > > > > Is there a way where we can avoid this copying by asking filesrc to do page > alignment > > when it allocates a buffer for reading from the file? > > > Thank you > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/Buffer-copying-tp3179110p3179110.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any > company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 wmiller at sdr.com Fri Jan 7 21:09:52 2011 From: wmiller at sdr.com (Wes Miller) Date: Fri, 7 Jan 2011 12:09:52 -0800 (PST) Subject: [gst-devel] Volume High/Low Pass Filters, AGC filter Message-ID: <1294430992866-3193445.post@n4.nabble.com> I looked through the list returned by gst-inspect and in the list of all plugins on the website and I don't see quite what I'm after. First, Is there a stage to do volume pass filtering? Idea is to sniff audio streams looking for low volume passages and replace them with silence or whitenoise af a fixed volume. Secondly, is there an audio AGC (automatic gain control) filter? Following a 300 to 3000 Hz band pass filter and an element that can do the volume pass operation, this element should clean audio (speech over noise mostly) to control highly variable background noise and maximize the speech quality. Think of a microphone in use in an outdoor industrial setting where machines or heavy trucks start and stop continually. First select only the usual voice band, flatten static, wind noise, motor/engine rumble, etc. then modify the overall volume to a pleasant level. Yes, of course this would make more sense as a hardware process, but I though I might be able to use it to assist existing hardware or to augment a hardware solution. Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Volume-High-Low-Pass-Filters-AGC-filter-tp3193445p3193445.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From mgruenke at Tycoint.com Fri Jan 7 21:57:37 2011 From: mgruenke at Tycoint.com (Gruenke, Matt) Date: Fri, 7 Jan 2011 15:57:37 -0500 Subject: [gst-devel] Volume High/Low Pass Filters, AGC filter In-Reply-To: <1294430992866-3193445.post@n4.nabble.com> References: <1294430992866-3193445.post@n4.nabble.com> Message-ID: I don't know much about the audio side of GStreamer, but I think I've heard of what you refer to as "volume pass filtering" as squelching or squelch filtering, in the context of two-way radios. Also, RTP has a concept of "silence suppression", "talkspurts", and "comfort noise". Perhaps this information might aid you in your search for available plugins, code snippets, etc. Matt -----Original Message----- From: Wes Miller [mailto:wmiller at sdr.com] Sent: Friday, January 07, 2011 15:10 To: gstreamer-devel at lists.sourceforge.net Subject: [gst-devel] Volume High/Low Pass Filters, AGC filter I looked through the list returned by gst-inspect and in the list of all plugins on the website and I don't see quite what I'm after. First, Is there a stage to do volume pass filtering? Idea is to sniff audio streams looking for low volume passages and replace them with silence or whitenoise af a fixed volume. Secondly, is there an audio AGC (automatic gain control) filter? Following a 300 to 3000 Hz band pass filter and an element that can do the volume pass operation, this element should clean audio (speech over noise mostly) to control highly variable background noise and maximize the speech quality. Think of a microphone in use in an outdoor industrial setting where machines or heavy trucks start and stop continually. First select only the usual voice band, flatten static, wind noise, motor/engine rumble, etc. then modify the overall volume to a pleasant level. Yes, of course this would make more sense as a hardware process, but I though I might be able to use it to assist existing hardware or to augment a hardware solution. Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Volume-High-Low-Pass-Filters -AGC-filter-tp3193445p3193445.html Sent from the GStreamer-devel mailing list archive at Nabble.com. ------------------------------------------------------------------------ ------ Gaining the trust of online customers is vital for the success of any company that requires sensitive data to be transmitted over the Web. Learn how to best implement a security strategy that keeps consumers' information secure and instills the confidence they need to proceed with transactions. 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 lane at brooks.nu Fri Jan 7 22:55:46 2011 From: lane at brooks.nu (Lane Brooks) Date: Fri, 07 Jan 2011 14:55:46 -0700 Subject: [gst-devel] gnonlin and ffmpeg encoder problems Message-ID: <4D278BE2.2040607@brooks.nu> I have a gnonlin frontend that works fine when feeding an autovideosink and the x264enc encoder. However, when I switch the encoder to a ffenc_mpeg4, it quits encoding after the first source and starting spewing out "Invalid timestamp" errors. I have a simple self-contained test that shows the problem that I posted to http://pastebin.com/D8UgAHqt The script creates two gnlsources that encapsulate two videotestsrc with different patterns that are arranged to play serially. The encapsulating gnlcomposition feeds a ffenc_mpeg element. Only the first gnlsource gets encoded in the resulting test.mp4 file. If you switch the script to use the x264enc or the autovideosink, however, it works fine. The problem also occurs with the mpeg2enc encoder and all the other ffmpeg encoders that I have tried. The script is completely self contained and I would be curious if others can download it and see the same behavior. Am I doing something wrong or is there an incompatibility between gnonlin and ffmpeg encoders. Thanks, Lane From bilboed at gmail.com Sat Jan 8 10:54:41 2011 From: bilboed at gmail.com (Edward Hervey) Date: Sat, 08 Jan 2011 10:54:41 +0100 Subject: [gst-devel] gnonlin and ffmpeg encoder problems In-Reply-To: <4D278BE2.2040607@brooks.nu> References: <4D278BE2.2040607@brooks.nu> Message-ID: <1294480481.2470.7.camel@deumeu> On Fri, 2011-01-07 at 14:55 -0700, Lane Brooks wrote: > I have a gnonlin frontend that works fine when feeding an autovideosink > and the x264enc encoder. However, when I switch the encoder to a > ffenc_mpeg4, it quits encoding after the first source and starting > spewing out "Invalid timestamp" errors. > > I have a simple self-contained test that shows the problem that I posted > to http://pastebin.com/D8UgAHqt > > The script creates two gnlsources that encapsulate two videotestsrc with > different patterns that are arranged to play serially. The encapsulating > gnlcomposition feeds a ffenc_mpeg element. Only the first gnlsource gets > encoded in the resulting test.mp4 file. If you switch the script to use > the x264enc or the autovideosink, however, it works fine. The problem > also occurs with the mpeg2enc encoder and all the other ffmpeg encoders > that I have tried. > > The script is completely self contained and I would be curious if others > can download it and see the same behavior. Am I doing something wrong or > is there an incompatibility between gnonlin and ffmpeg encoders. The problem is related to elements that don't properly take into account segments. GNonLin makes extensive usage of playback segments (see the documentation on new-segment event and the design docs for more info) to do time shifting. That way it doesn't need to do a costly re-timestamping of all buffers but instead just tweak the outgoing segments. Very simplified, this means that if you have X sources, you will end up with gnlcomposition outputting at least X segments NEWSEGMENT buffers from first source NEWSEGMENT buffers from second source .... Elements operating on time should normally use/accumulate those newsegments (using GstSegment) to convert the buffer timestamps to running time timestamps (basesink does this for example, which is why you don't see any problem when using a videosink with gnonlin). Since a lot of elements don't do that... you need to make sure they receive one continuous stream (with intermediary newsegment events swallowed and buffer timestamps modified). Identity has got a property you can activate (single-segment) to do exactly that. So basic rule of thumb : if you're using gnlcomposition, put a "single-segment identity=True" after it (or at least before your encoders). EncodeBin supports that internally. Edward > > Thanks, > Lane > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 josel.segura at gmx.es Sat Jan 8 11:25:03 2011 From: josel.segura at gmx.es (=?ISO-8859-1?B?Sm9z6Q==?= Luis Segura Lucas) Date: Sat, 8 Jan 2011 11:25:03 +0100 Subject: [gst-devel] Pipeline paused, inside elements don't? In-Reply-To: <20110105205830.58bc713b@peter> References: <20110105205830.58bc713b@peter> Message-ID: <20110108112503.7230f4bd@peter> I have discover the mistake: I was changing the state using gobject functions (as gobject.idle_add). Changing the state with "set_state" pased to gobject.idle_add, the set_state is called again and again because it alwys return something different to "False". I wrote a little wrapper that, in the case that the state change is succesful, it returns True and gobject.idle_add stops to call the set_state func. El Wed, 5 Jan 2011 20:58:30 +0100 Jos? Luis Segura Lucas escribi?: > Good afternoon! > > Some time ago I wrote a source and a sink elements using the Python > bindings of Gstreamer. > > A few days ago, I come back to this code and, after a few tests, I see > something estrange: when I set the pipeline state to "paused", my sink > element continue on "playing" state. > > I wrote the sink element as a subclass of the "gst.BaseSink" Python > class. The only thing that is supposed to do using that class is > overwrite the "do_render" method. > > Do I need to do something more to set the element state to paused? > > Thanks in advance and best regards From ensonic at hora-obscura.de Sat Jan 8 12:09:20 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 08 Jan 2011 13:09:20 +0200 Subject: [gst-devel] What is the purpose of gst audioconvert plugin In-Reply-To: <1293630257660-3167118.post@n4.nabble.com> References: <1293447146580-3164871.post@n4.nabble.com> <1293630257660-3167118.post@n4.nabble.com> Message-ID: <4D2845E0.5000405@hora-obscura.de> Am 29.12.2010 15:44, schrieb Wes Miller: > > 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. Elements are docuemnted as part of the code. Each element (should) have a doc-blob at the start of the source file. The comment starts like this: /** * SECTION:element-audioconvert * * ... */ The documentation there is extracted and can be read on the gstreamer webpage or e.g. locally in devhelp. We happily accept patches to improve such documentation. Stefan > > 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 > > From ensonic at hora-obscura.de Sat Jan 8 12:16:35 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 08 Jan 2011 13:16:35 +0200 Subject: [gst-devel] HTTP video source serves individual JPEG images In-Reply-To: <4D21A711.1060504@ingelabs.com> References: <4D21A711.1060504@ingelabs.com> Message-ID: <4D284793.4030605@hora-obscura.de> Am 03.01.2011 12:38, schrieb Guillermo Rodriguez Garcia: > Hello all, > > I am trying to use gstreamer to stream video (sort of) from a HTTP source > that serves individual JPEG images. This is not a MJPEG stream -- instead, > a new HTTP request is required for each frame that is to be retrieved. > > This pipeline works for MJPEG: > > gst-launch souphttpsrc location=http://user:pass at hostname/path ! jpegdec ! xvimagesink > > If I try to use the same pipeline for an HTTP source that serves individual > JPEG images, only one image is shown (since the HTTP response actually > contains just one single image). gst-launch souphttpsrc location=http://user:pass at hostname/path ! multipartdemux ! jpegdec ! xvimagesink Stefan > > Is there any way to have souphttpsrc make additional requests periodically > from the specified location? > > Is there any other way to achieve this? > > Thanks, > > GRodriguez > > > ------------------------------------------------------------------------------ > 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 ensonic at hora-obscura.de Sat Jan 8 12:20:28 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 08 Jan 2011 13:20:28 +0200 Subject: [gst-devel] How to add a complicate property to an element In-Reply-To: <1A42CE6F5F474C41B63392A5F80372B231D2AEB2@shsmsx501.ccr.corp.intel.com> References: <1A42CE6F5F474C41B63392A5F80372B231D2AEB2@shsmsx501.ccr.corp.intel.com> Message-ID: <4D28487C.5010206@hora-obscura.de> Am 01.01.2011 04:52, schrieb Hu, Gang A: > Hi, > I want to add a property to a camera source element. This property contains five > field. How can create this property with one property id and extract it? > For example, to add the ae_windows property to the v4l2src, the user can use the > command > gst-launch v4l2src ae_windows=?width=49, height=60, centerx=20, centery=30, > weight=3? ! xvimagesink. > to specify the ae_windows. While you could define your own gparamspec type, i would sugest you to simply add 4 properties instead. Please use a descriptive name so that people won't have to guess what "ae" stands for. Stefan > 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 ensonic at hora-obscura.de Sat Jan 8 12:24:39 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 08 Jan 2011 13:24:39 +0200 Subject: [gst-devel] Need help for rezising video. In-Reply-To: References: <4D221C80.10601@wanadoo.fr> Message-ID: <4D284977.2010204@hora-obscura.de> Am 03.01.2011 22:03, schrieb liJin: > Hi.. > > If you are looking for a plugin to resize the image ..i think capsfilter will > help you.. capsfilter could be used to constrain caps. As nicolas noticed videoscale whould be the right choise. Nicals, just put videoscale infront of your videosink. Stefan > > Reagrds > Lijin > > > On Tue, Jan 4, 2011 at 12:29 AM, Nicolas > wrote: > > Hello, > > I have created a plugin that displays a video in an evas image. > > Here is my problem: > When I resize the picture I am calling "evas_object_image_fill_set", it > works correctly, however the consumption of CPU is just horrible, more > than 120% ! > If I leave it in normal size, the consumption of CPU is normal. > > I want to know if there is a way to resize the video (via gstreamer) in > my plugin. > Have you any idea? > May be I could create a function "buffer alloc" > (gstbase_sink_class->buffer_alloc = mybufferalloc) and modify the buffer > in it ? > > I know that there is "videoscale" but I do not know how to use it inside > the plugin. > > Thank you in advance for your reply, > Sincerely, > Nicolas. > > Ps: sorry for bad english. > > ------------------------------------------------------------------------------ > 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 ensonic at hora-obscura.de Sat Jan 8 12:28:10 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 08 Jan 2011 13:28:10 +0200 Subject: [gst-devel] Need help for rezising video. In-Reply-To: <4D223A73.9050802@wanadoo.fr> References: <4D223A73.9050802@wanadoo.fr> Message-ID: <4D284A4A.6010503@hora-obscura.de> Am 03.01.2011 23:06, schrieb Nicolas: > Hi Lijin, > > Thanks for your reply. > > But how capsfilter can help me ? > > I only see in the doc: > capsfilter ? Pass data without modification, limiting formats Nicolas. forget about the capsfilter. The videodecoder is ouputing a certain size and you want it to be scaled to the size of your evas widget. Just add a videoscale between the decoder and the evassink and when the window is resized set update your caps. Have a look at ximagesink (this won't resize itself as well and relies on videoscale). Stefan > > Regards, > Nicolas. > > ------------------------------------------------------------------------------ > 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 gang.a.hu at intel.com Sat Jan 8 13:04:48 2011 From: gang.a.hu at intel.com (Hu, Gang A) Date: Sat, 8 Jan 2011 20:04:48 +0800 Subject: [gst-devel] How to add a complicate property to an element In-Reply-To: <4D28487C.5010206@hora-obscura.de> References: <1A42CE6F5F474C41B63392A5F80372B231D2AEB2@shsmsx501.ccr.corp.intel.com> <4D28487C.5010206@hora-obscura.de> Message-ID: <1A42CE6F5F474C41B63392A5F80372B231DCED6C@shsmsx501.ccr.corp.intel.com> Stefan. Thanks for your suggestion. I would like to keep the property list small. As composited property is not common in GStreamer, adding 5 property for the Windows settings maybe a good choice right now. Thanks. -----Original Message----- From: Stefan Kost [mailto:ensonic at hora-obscura.de] Sent: Saturday, January 08, 2011 7:20 PM To: Discussion of the development of GStreamer Cc: Hu, Gang A Subject: Re: [gst-devel] How to add a complicate property to an element Am 01.01.2011 04:52, schrieb Hu, Gang A: > Hi, > I want to add a property to a camera source element. This property contains five > field. How can create this property with one property id and extract it? > For example, to add the ae_windows property to the v4l2src, the user can use the > command > gst-launch v4l2src ae_windows="width=49, height=60, centerx=20, centery=30, > weight=3" ! xvimagesink. > to specify the ae_windows. While you could define your own gparamspec type, i would sugest you to simply add 4 properties instead. Please use a descriptive name so that people won't have to guess what "ae" stands for. Stefan > 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 t.i.m at zen.co.uk Sat Jan 8 13:15:12 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sat, 08 Jan 2011 12:15:12 +0000 Subject: [gst-devel] pre-releases! GStreamer core, -base, -good, -ugly, -bad, -python Message-ID: <1294488912.3488.9.camel@zingle> Hi, pre-releases for GStreamer core, gst-plugins-base, gst-plugins-good, gst-plugins-ugly, gst-plugins-bad, and gst-python are now available at the following locations: 5abbe163ccea45aea9e2172054ac2aa1 gstreamer-0.10.31.2.tar.gz 161b75f9e47202a2d1d88b31c77a40dc gstreamer-0.10.31.2.tar.bz2 http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.31.2.tar.gz http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.31.2.tar.bz2 00e8901b8fc5ebfb41f66a4fcbb0c251 gst-plugins-base-0.10.31.2.tar.gz 1f0b5f59b3eef8255d404953376e2a6a gst-plugins-base-0.10.31.2.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.31.2.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.31.2.tar.bz2 e7ae5155a823ae7c8552e9c486cd485f gst-plugins-good-0.10.26.2.tar.gz b00af005a868ea8daa651b15ecd66f93 gst-plugins-good-0.10.26.2.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-good/pre/gst-plugins-good-0.10.26.2.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-good/pre/gst-plugins-good-0.10.26.2.tar.bz2 1b51c37bcf13509ca58ce846b64476f7 gst-plugins-ugly-0.10.16.2.tar.gz 310a4b039e85c2a2c8535c7145ebad1b gst-plugins-ugly-0.10.16.2.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-ugly/pre/gst-plugins-ugly-0.10.16.2.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-ugly/pre/gst-plugins-ugly-0.10.16.2.tar.bz2 2b302fa5c8943460da5243075f64647f gst-plugins-bad-0.10.20.2.tar.gz c103781cd0cdfed2ff3fbbd897300247 gst-plugins-bad-0.10.20.2.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.20.2.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.20.2.tar.bz2 6232946f5a33591e014214765a735fde gst-python-0.10.20.2.tar.gz c57820367f819ac69ccaa10052f313c9 gst-python-0.10.20.2.tar.bz2 http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.20.2.tar.gz http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.20.2.tar.bz2 There might also be a gnonlin pre-release to follow soon. Ubuntu users might also be able to test these pre-releases by using the GStreamer PPA (as soon as they get uploaded there). Please test them thoroughly with all your favourite apps, and let us know of any regressions or other important issues by filing blocker bugs in bugzilla: http://gstreamer.freedesktop.org/bugs/ Further pre-releases will follow as needed. If all goes well, releases should happen later this month, as per http://gstreamer.freedesktop.org/wiki/ReleasePlanning2011 A list of possible blocker bugs can be found at: http://bit.ly/h3MzgU Happy testing! Cheers -Tim From lane at brooks.nu Sat Jan 8 19:25:24 2011 From: lane at brooks.nu (Lane Brooks) Date: Sat, 08 Jan 2011 11:25:24 -0700 Subject: [gst-devel] gnonlin and ffmpeg encoder problems In-Reply-To: <1294480481.2470.7.camel@deumeu> References: <4D278BE2.2040607@brooks.nu> <1294480481.2470.7.camel@deumeu> Message-ID: <4D28AC14.3080000@brooks.nu> On 01/08/2011 02:54 AM, Edward Hervey wrote: > On Fri, 2011-01-07 at 14:55 -0700, Lane Brooks wrote: >> I have a gnonlin frontend that works fine when feeding an autovideosink >> and the x264enc encoder. However, when I switch the encoder to a >> ffenc_mpeg4, it quits encoding after the first source and starting >> spewing out "Invalid timestamp" errors. >> >> I have a simple self-contained test that shows the problem that I posted >> to http://pastebin.com/D8UgAHqt >> >> The script creates two gnlsources that encapsulate two videotestsrc with >> different patterns that are arranged to play serially. The encapsulating >> gnlcomposition feeds a ffenc_mpeg element. Only the first gnlsource gets >> encoded in the resulting test.mp4 file. If you switch the script to use >> the x264enc or the autovideosink, however, it works fine. The problem >> also occurs with the mpeg2enc encoder and all the other ffmpeg encoders >> that I have tried. >> >> The script is completely self contained and I would be curious if others >> can download it and see the same behavior. Am I doing something wrong or >> is there an incompatibility between gnonlin and ffmpeg encoders. > > The problem is related to elements that don't properly take into > account segments. GNonLin makes extensive usage of playback segments > (see the documentation on new-segment event and the design docs for more > info) to do time shifting. That way it doesn't need to do a costly > re-timestamping of all buffers but instead just tweak the outgoing > segments. > Very simplified, this means that if you have X sources, you will end > up with gnlcomposition outputting at least X segments > NEWSEGMENT > buffers from first source > NEWSEGMENT > buffers from second source > .... > > Elements operating on time should normally use/accumulate those > newsegments (using GstSegment) to convert the buffer timestamps to > running time timestamps (basesink does this for example, which is why > you don't see any problem when using a videosink with gnonlin). > > Since a lot of elements don't do that... you need to make sure they > receive one continuous stream (with intermediary newsegment events > swallowed and buffer timestamps modified). > Identity has got a property you can activate (single-segment) to do > exactly that. > > So basic rule of thumb : if you're using gnlcomposition, put a > "single-segment identity=True" after it (or at least before your > encoders). > > EncodeBin supports that internally. > > Edward Thanks for the detailed feedback. The identity element does indeed solve the problem. Lane From ensonic at hora-obscura.de Sat Jan 8 21:25:30 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 08 Jan 2011 22:25:30 +0200 Subject: [gst-devel] Volume High/Low Pass Filters, AGC filter In-Reply-To: <1294430992866-3193445.post@n4.nabble.com> References: <1294430992866-3193445.post@n4.nabble.com> Message-ID: <4D28C839.9010504@hora-obscura.de> Am 07.01.2011 22:09, schrieb Wes Miller: > > I looked through the list returned by gst-inspect and in the list of all > plugins on the website and I don't see quite what I'm after. > > First, Is there a stage to do volume pass filtering? Idea is to sniff audio > streams looking for low volume passages and replace them with silence or > whitenoise af a fixed volume. Replacing it with silence would be a noise-gate and I am sure there is one in the form of a ladspa plugin. But then I am wondering, if you are actually looking for a comfort noise generator? > > Secondly, is there an audio AGC (automatic gain control) filter? Following > a 300 to 3000 Hz band pass filter and an element that can do the volume > pass operation, this element should clean audio (speech over noise mostly) > to control highly variable background noise and maximize the speech quality. > Think of a microphone in use in an outdoor industrial setting where machines > or heavy trucks start and stop continually. First select only the usual > voice band, flatten static, wind noise, motor/engine rumble, etc. then > modify the overall volume to a pleasant level. I don't think we have that one. Wanna try writing one? We have filters and e.g. the level element. Mix well together and add whats missing :) Stefan > > Yes, of course this would make more sense as a hardware process, but I > though I might be able to use it to assist existing hardware or to augment a > hardware solution. > > Wes From lfarkas at lfarkas.org Sat Jan 8 22:12:53 2011 From: lfarkas at lfarkas.org (Farkas Levente) Date: Sat, 08 Jan 2011 22:12:53 +0100 Subject: [gst-devel] does signal has last param user_data in gstreamer? In-Reply-To: <1294328507.23475.11.camel@localhost> References: <4D25C8A6.2060502@lfarkas.org> <4D25D1E2.50403@igalia.com> <4D25E035.1090506@lfarkas.org> <1294328507.23475.11.camel@localhost> Message-ID: <4D28D355.3030900@lfarkas.org> On 01/06/2011 04:41 PM, Edward Hervey wrote: > On Thu, 2011-01-06 at 16:31 +0100, Farkas Levente wrote: >> hi, >> then the two signal definition in input-selector is bad? or.. >> http://cgit.freedesktop.org/gstreamer/gstreamer/tree/plugins/elements/gstinputselector.h >> since there is no last argument user_data? > > Those are the vmethod implementations called *on* the object (read > more of the gobject docs to understand what I mean). > user_data is only used in the callbacks and added by the signalling > system auto-magically. > > Edward but it'd be nice to use consistent docs/help ie. either all gst-inspect signal contain it as last param or none (see for example gst-inspect input-selector). >> >> On 01/06/2011 03:29 PM, Guillaume Emont wrote: >>> Hi Farkas, >>> >>> That is indeed a gobject signal thing: when you connect [1] to a signal, >>> you specify a "user_data" pointer to pass to the callback you're >>> attaching, whatever the signal. >>> This is all should be explained in the signal section of the GObject >>> Reference Manual [2]. >>> >>> Guillaume >>> >>> [1] >>> http://library.gnome.org/devel/gobject/2.26/gobject-Signals.html#g-signal-connect >>> [2] http://library.gnome.org/devel/gobject/2.26/signal.html >>> >>> On 06/01/2011 14:50, Farkas Levente wrote: >>>> hi, >>>> let see an example: both gst-inspect input-selector and >>>> http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-input-selector.html >>>> said that the signal "switch" has a last argument as >>>> gpointer user_data >>>> and the signal "block" has no such last argument in the gst-inspect >>>> output, but has in the above docs. >>>> but when i look onto the source code in gstinputselector.h none of these >>>> signal has last param! >>>> >>>> since there are many signal (almos all) which has last param user_data >>>> so my questions in general: >>>> is the last param user_data exist or not at all? or it's something glib >>>> specific thing or ...? >>>> thanks in advance. >>>> 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 >> >> > > > > ------------------------------------------------------------------------------ > 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 -- Levente "Si vis pacem para bellum!" From t.i.m at zen.co.uk Sun Jan 9 00:44:14 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sat, 08 Jan 2011 23:44:14 +0000 Subject: [gst-devel] does signal has last param user_data in gstreamer? In-Reply-To: <4D28D355.3030900@lfarkas.org> References: <4D25C8A6.2060502@lfarkas.org> <4D25D1E2.50403@igalia.com> <4D25E035.1090506@lfarkas.org> <1294328507.23475.11.camel@localhost> <4D28D355.3030900@lfarkas.org> Message-ID: <1294530254.6105.3.camel@zingle> On Sat, 2011-01-08 at 22:12 +0100, Farkas Levente wrote: > but it'd be nice to use consistent docs/help ie. either all gst-inspect > signal contain it as last param or none (see for example gst-inspect > input-selector). For action signals (and vfuncs for such action signals) no user_data argument should be displayed, since that doesn't make sense. gst-inspect doesn't show a user_data argument here, but gtk-doc (html docs) does, which looks like a gtk-doc bug to me. For normal signals, there should be a user data argument, because what you're displaying is basically the signature your callback function should have. Two different things really, and hence two different conventions. Cheers -Tim From osmoma at gmail.com Sun Jan 9 14:36:41 2011 From: osmoma at gmail.com (Osmo Antero Maatta) Date: Sun, 09 Jan 2011 14:36:41 +0100 Subject: [gst-devel] Getting default audio source and sink (devices) in the GNOME-desktop. Message-ID: <4D29B9E9.4090407@gmail.com> Hello, How can I read the default values for audio-sink and audio-source in Linux/GNOME-desktop. These values (device names) are set in the GNOME's gnome-volume-control program. Start $ gnome-volume-control Picture: http://bildr.no/view/796255 showing both [Input] and [Output] pages. The values are stored in user's $HOME/.pulse/ directory. $ ls ~/.pulse/ b3b6877727df4f4ae34dba2b00000009-default-sink b3b6877727df4f4ae34dba2b00000009-default-source ... $ cat ~/.pulse/b3b6877727df4f4ae34dba2b00000009-default-source alsa_output.pci-0000_04_02.0.analog-stereo $ cat ~/.pulse/b3b6877727df4f4ae34dba2b00000009-default-sink alsa_input.pci-0000_00_1b.0.analog-stereo Is there a function call in GStreamer or Pulseaudio to get these values, or should I simply read the files. Kindly Moma Antero From malloblenne at gmail.com Sun Jan 9 15:56:39 2011 From: malloblenne at gmail.com (Mauro Brenna) Date: Sun, 9 Jan 2011 15:56:39 +0100 Subject: [gst-devel] Obtain a JPEG sequence from a RTP MJPEG stream (RFC 2435) Message-ID: Hello, I am developing a video streamer of jpeg images MJPEG through RTP. I would like to make it compliant to RFC2435, and I am testing it using, as a receiver, a gstreamer pipeline. By now, I can saw the video in a window, with an error in the decoding process, using the following pipelines on MS Windows: C:\Program Files\OSSBuild\GStreamer\v0.10.6\bin>gst-launch.exe -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 ! autovideosink udpsrc port=9997 ! rtpbin. recv_rtcp_sink_1 rtpbin.send_rtcp_src_1 ! multiudpsink clients="127.0.0.1:10001" sync=false async=false or simply: gst-launch.exe -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 ! autovideosink In order to investigate the problems in my implementation, I would like to change the sink and have as output the file jpeg as they are decoded. I mean, I don't want the jpeg decoded and re-encoded another time, which could lose their original header or change the compression, quantization tables or other parameters. My idea is to compare the jpeg I sent with the image I retrieve by gstreamer. Is there a simple way to do it? I can also run the program under GNU/Linux but MS Windows is preferable. Thanks, Mauro From lane at brooks.nu Sun Jan 9 20:14:58 2011 From: lane at brooks.nu (Lane Brooks) Date: Sun, 09 Jan 2011 12:14:58 -0700 Subject: [gst-devel] Arbitrary color background image generation Message-ID: I need a solid color background image, and I can't find a plugin that will let me generate an arbitrary color. I see that videotestsrc will let me generate a white, black, red, green, or blue image. But is there a source plugin that lets me generate an arbitrary color? Lane From ds at entropywave.com Sun Jan 9 23:33:00 2011 From: ds at entropywave.com (David Schleef) Date: Sun, 9 Jan 2011 17:33:00 -0500 Subject: [gst-devel] Arbitrary color background image generation In-Reply-To: References: Message-ID: <20110109223300.GA15404@cooker.entropywave.com> On Sun, Jan 09, 2011 at 12:14:58PM -0700, Lane Brooks wrote: > I need a solid color background image, and I can't find a plugin that will let me generate an arbitrary color. I see that videotestsrc will let me generate a white, black, red, green, or blue image. But is there a source plugin that lets me generate an arbitrary color? gst-launch videotestsrc pattern=white foreground-color=0x00ffaa77 ! xvimagesink David From vaish05_s at yahoo.com Mon Jan 10 05:19:10 2011 From: vaish05_s at yahoo.com (vaishnavi) Date: Sun, 9 Jan 2011 20:19:10 -0800 (PST) Subject: [gst-devel] Buffer copying In-Reply-To: References: <1294406521793-3179110.post@n4.nabble.com> Message-ID: <1294633150573-3206599.post@n4.nabble.com> Thank you.. Il try that... -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Buffer-copying-tp3179110p3206599.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From yuanqing.cai at tieto.com Mon Jan 10 07:16:50 2011 From: yuanqing.cai at tieto.com (Cai Yuanqing) Date: Mon, 10 Jan 2011 14:16:50 +0800 Subject: [gst-devel] Getting default audio source and sink (devices) in the GNOME-desktop. In-Reply-To: <4D29B9E9.4090407@gmail.com> References: <4D29B9E9.4090407@gmail.com> Message-ID: <4D2AA452.1030108@tieto.com> Hi Osmo: I don't quite clear where do you want to use this functions call and here are some informations I can give to: 1.You can get default sink and source in pulseaudio by functions: > pa_sink* pa_namereg_get_default_sink(pa_core *c) > pa_source* pa_namereg_get_default_source(pa_core *c) These 2 functions will return you the pointer of default sink or source,thus you can get name by sink.name or source.name. You can also check this informations in pacmd: $ pacmd <<< list-sinks <<< list-sources the first one is the default device in generally. 2.In the Gstreamer said I don't know very well,seems that pulsesink and pulsesrc have no default device,maybe you need to set them first and the get these arguments as others by: > g_object_get(G_OBJECT(pulsesink),"device-name",dev_name,NULL); Hope this helps :-) On 01/09/2011 09:36 PM, Osmo Antero Maatta wrote: > Hello, > > How can I read the default values for audio-sink and audio-source in > Linux/GNOME-desktop. > These values (device names) are set in the GNOME's gnome-volume-control > program. Start > $ gnome-volume-control > Picture: http://bildr.no/view/796255 showing both [Input] and [Output] > pages. > > The values are stored in user's $HOME/.pulse/ directory. > > $ ls ~/.pulse/ > b3b6877727df4f4ae34dba2b00000009-default-sink > b3b6877727df4f4ae34dba2b00000009-default-source > ... > > $ cat ~/.pulse/b3b6877727df4f4ae34dba2b00000009-default-source > alsa_output.pci-0000_04_02.0.analog-stereo > > $ cat ~/.pulse/b3b6877727df4f4ae34dba2b00000009-default-sink > alsa_input.pci-0000_00_1b.0.analog-stereo > > Is there a function call in GStreamer or Pulseaudio to get these values, > or should I simply read the files. > > Kindly > Moma Antero > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 cmaster2 at hotmail.com Mon Jan 10 08:41:34 2011 From: cmaster2 at hotmail.com (lithium) Date: Sun, 9 Jan 2011 23:41:34 -0800 (PST) Subject: [gst-devel] Video freeze In-Reply-To: References: <1294028853954-3171570.post@n4.nabble.com> Message-ID: <1294645294937-3206707.post@n4.nabble.com> Well I've been working on debugging this. I've created debug logs of my program run with --gst-debug-level 5 if i use gst_pad_set_blocked: the first run of take screenshot causes a 0byte jpeg to be created. The second time i click screenshot I get the image that was there when screenshot was clicked the first time written to the jpeg, the video freezes and the log shows this: 0:00:03.492790081  6199 0x1703f30 LOG   GST_SCHEDULING gstpad.c:1036:gst_pad_set_blocked_async_full: blocking pad 0:00:03.492806842  6199 0x1703f30 LOG   GST_SCHEDULING gstpad.c:1048:gst_pad_set_blocked_async_full: waiting for block after which point nothing more shows up if i remove the gst_pad_set_blocked lines: the first run of take screenshot causes a 0byte jpeg to be created. The second time i click screenshot I get the image that was there when screenshot was clicked the first time written to the jpeg, the video does not freeze, and the log continues having information about everything im doing written to it. Further clicks cause the image that was there when the screenshot was clicked the first time to be re-written to the jpeg file. In the bus sync handler I check for state paused because for reasons unknown to me the message i construct telling it to set it to GST_STATE_NULL does not set it to GST_STATE_NULL but GST_STATE_PAUSED heres the current functions: void close_screenshot(GstElement *jpegEncode, videoIOStruct *videoIO) { GstPad *sinkpad = gst_element_get_pad(videoIO->screenshotBin, "sink"); GstPad *srcpad = gst_pad_get_peer(sinkpad); if (!srcpad) return; //if we dont have a src pad its already been released need to check because another jpeg may be encoded before we close the screenshot gst_pad_set_blocked(srcpad, TRUE); gst_pad_unlink(srcpad, sinkpad); gst_pad_set_blocked(srcpad, FALSE); gst_element_release_request_pad(videoIO->tee, srcpad); gst_object_unref(srcpad); gst_object_unref(sinkpad); //send a state request back to main thread GstMessage *setState = gst_message_new_request_state(GST_OBJECT(videoIO->screenshotBin), GST_STATE_NULL); GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE(videoIO->pipeline)); gst_bus_post(bus, setState); gst_object_unref(bus); } void take_screenshot(GtkWidget *widget, videoIOStruct *videoIO) { gst_element_set_state(videoIO->screenshotBin, GST_STATE_PAUSED); gst_bin_add(GST_BIN(videoIO->pipeline), videoIO->screenshotBin); GstPad *sinkpad = gst_element_get_pad(videoIO->screenshotBin, "sink"); GstPad *srcpad = gst_element_get_request_pad(videoIO->tee, "src%d"); g_print("srcpadref: %d\n", GST_OBJECT_REFCOUNT(GST_OBJECT(srcpad))); gst_pad_set_blocked(srcpad, TRUE); gst_pad_link(srcpad, sinkpad); gst_pad_set_blocked(srcpad, FALSE); gst_object_unref(sinkpad); gst_element_set_state(videoIO->pipeline, GST_STATE_PLAYING); } int setup_screenshotBin(videoIOStruct *videoIO) { videoIO->screenshotBin = gst_bin_new("screenshot_bin"); GstElement *jpegOutput = gst_element_factory_make("filesink", "jpegoutput"); GstElement *jpegQueue = gst_element_factory_make("queue", "jpgqueue"); GstElement *jpegEncode = gst_element_factory_make("jpegenc", "jpegencoder"); if (!jpegOutput || !jpegQueue || !jpegEncode || !videoIO->screenshotBin) return 0; //set arguments // g_object_set(G_OBJECT(jpegOutput), "location", g_strdup("test.jpg"), NULL); //FIXME: why cant i use set_data??? //set arguments gchar filename[] = "test.jpg"; g_object_set(G_OBJECT(jpegOutput), "location", filename, NULL); //link stuff gst_bin_add_many(GST_BIN(videoIO->screenshotBin), jpegQueue, jpegEncode, jpegOutput, NULL); gst_element_link_many(jpegQueue, jpegEncode, jpegOutput, NULL); g_signal_connect(jpegEncode, "frame-encoded", G_CALLBACK(close_screenshot), videoIO); GstPad *sinkpad = gst_element_get_static_pad(jpegQueue, "sink"); gst_element_add_pad(videoIO->screenshotBin, gst_ghost_pad_new("sink", sinkpad)); gst_object_unref(GST_OBJECT(sinkpad)); return 1; } int openVideoPipeline(videoIOStruct *videoIO) { /* /----queue----xvimagesink //screenoutput v4l2src-----tee----- \----queue----filesink //screenshot \----queue----\ ----multiplexer ---filesink //recording /----queue----/ alsasrc-----tee----- \----queue----alsasink //playback from capture if needed */ //create elements videoIO->pipeline = gst_pipeline_new("tvVideo"); videoIO->vidSrc = gst_element_factory_make("v4l2src", "v4lsource"); GstElement *vidSink = gst_element_factory_make("xvimagesink", "xvsink"); videoIO->tee = gst_element_factory_make("tee", "splitter"); GstElement *vidQueue = gst_element_factory_make("queue", "screenQueue"); if (!videoIO->pipeline || !videoIO->vidSrc || !vidSink || !videoIO->tee || !vidQueue) return 0; //set arguments g_object_set(videoIO->vidSrc, "device", videoIO->devName, NULL); gst_bin_add_many(GST_BIN(videoIO->pipeline), videoIO->vidSrc, videoIO->tee, vidQueue, vidSink, NULL); GstPad *teeSrc = gst_element_get_request_pad(videoIO->tee, "src%d"); //FIXME: release on program exit GstPad *screenQueue = gst_element_get_static_pad(vidQueue, "sink"); //link stuff gst_pad_link(teeSrc, screenQueue); gst_object_unref(screenQueue); gst_element_link(videoIO->vidSrc, videoIO->tee); gst_element_link(vidQueue, vidSink); if (!videoIO->pipeline) return 0; return 1; } GstBusSyncReply bus_sync_handler(GstBus *bus, GstMessage *message, videoIOStruct *videoIO) { if (GST_MESSAGE_TYPE(message) == GST_MESSAGE_ASYNC_DONE) { GstState state; GstState pending; gst_element_get_state(videoIO->screenshotBin, &state, &pending, NULL); if (state == GST_STATE_PAUSED) { GstObject *parent = gst_element_get_parent(videoIO->screenshotBin); if (parent) { gst_bin_remove(GST_BIN(videoIO->pipeline), videoIO->screenshotBin); gst_object_unref(parent); } } } if (GST_MESSAGE_TYPE(message) != GST_MESSAGE_ELEMENT) return GST_BUS_PASS; if (!gst_structure_has_name(message->structure, "prepare-xwindow-id")) return GST_BUS_PASS; if (*videoIO->videoWindowXId != 0) { GstXOverlay *xOverlay = GST_X_OVERLAY(GST_MESSAGE_SRC(message)); gst_x_overlay_set_xwindow_id(xOverlay, *videoIO->videoWindowXId); } else g_warning("Should have obtained video_window_xid by now!"); gst_message_unref(message); return GST_BUS_DROP; } any ideas? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Video-freeze-tp3171570p3206707.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From grodriguez at ingelabs.com Mon Jan 10 10:23:15 2011 From: grodriguez at ingelabs.com (grodriguez at ingelabs.com) Date: Mon, 10 Jan 2011 10:23:15 +0100 Subject: [gst-devel] HTTP video source serves individual JPEG images In-Reply-To: <4D284793.4030605@hora-obscura.de> References: <4D21A711.1060504@ingelabs.com> <4D284793.4030605@hora-obscura.de> Message-ID: <4D2AD003.6040700@ingelabs.com> Hello Stefan, El 08/01/2011 12:16, Stefan Kost escribi?: > Am 03.01.2011 12:38, schrieb Guillermo Rodriguez Garcia: >> Hello all, >> >> I am trying to use gstreamer to stream video (sort of) from a HTTP source >> that serves individual JPEG images. This is not a MJPEG stream -- instead, >> a new HTTP request is required for each frame that is to be retrieved. >> >> This pipeline works for MJPEG: >> >> gst-launch souphttpsrc location=http://user:pass at hostname/path ! jpegdec ! xvimagesink >> >> If I try to use the same pipeline for an HTTP source that serves individual >> JPEG images, only one image is shown (since the HTTP response actually >> contains just one single image). > > gst-launch souphttpsrc location=http://user:pass at hostname/path ! multipartdemux > ! jpegdec ! xvimagesink Thanks for your answer. This pipeline does work for MJPEG (just like the one I provided above), but does not solve the problem. I don't have an MJPEG stream. What I have is a HTTP source that serves individual JPEG images (one single JPEG image per HTTP request). What I actually need is a way to have souphttpsrc (or another HTTP src) make additional HTTP requests periodically from the same URL. Is this possible? If not, can anyone suggest how this could be implemented? Thanks, G Rodriguez From jorney_dong at asus.com Mon Jan 10 10:39:14 2011 From: jorney_dong at asus.com (jorney) Date: Mon, 10 Jan 2011 01:39:14 -0800 (PST) Subject: [gst-devel] amr-wb seek problem Message-ID: <1294652354379-3206831.post@n4.nabble.com> Hi I am writing amr-wb player with gstreamer on Marvell pxa310. When I seek to a position by time format, there is a long time blocking (10 sec or even to 1 min), and there is no voice during the block, then an EOS message was emitted, but the position I seeked is far away to the end of the file. Bytes format has the same problem! here is my seek code: breslt = gst_element_seek(pipeline, 1.0, GST_FORMAT_BYTES, GST_SEEK_FLAG_FLUSH,//|GST_SEEK_FLAG_KEY_UNIT|GST_SEEK_FLAG_SEGMENT, GST_SEEK_TYPE_SET, bytePos,//GST_SECOND*nForw*count, GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE ); I use gstreamer-0.10.29 and gst-plugins-ugly-0.10.15's amrwbdec, gst-plugins-bad-0.10.13's amrparse. like this: gst-launch filesrc location=abc.amr ! amrparse ! amrwbdec ! alsasink And the codec is opencore-amr-0.1.2 Could any body help? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/amr-wb-seek-problem-tp3206831p3206831.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ikt011 at gmail.com Mon Jan 10 11:15:07 2011 From: ikt011 at gmail.com (Kocsis Tibor) Date: Mon, 10 Jan 2011 11:15:07 +0100 Subject: [gst-devel] HTTP video source serves individual JPEG images In-Reply-To: <4D2AD003.6040700@ingelabs.com> References: <4D21A711.1060504@ingelabs.com> <4D284793.4030605@hora-obscura.de> <4D2AD003.6040700@ingelabs.com> Message-ID: Maybe you should request the frames manually and use appsrc to push jpeg images as buffers into a decoding pipeline. On Mon, Jan 10, 2011 at 10:23 AM, wrote: > Hello Stefan, > > El 08/01/2011 12:16, Stefan Kost escribi?: >> Am 03.01.2011 12:38, schrieb Guillermo Rodriguez Garcia: >>> Hello all, >>> >>> I am trying to use gstreamer to stream video (sort of) from a HTTP source >>> that serves individual JPEG images. This is not a MJPEG stream -- instead, >>> a new HTTP request is required for each frame that is to be retrieved. >>> >>> This pipeline works for MJPEG: >>> >>> gst-launch souphttpsrc location=http://user:pass at hostname/path ! jpegdec ! xvimagesink >>> >>> If I try to use the same pipeline for an HTTP source that serves individual >>> JPEG images, only one image is shown (since the HTTP response actually >>> contains just one single image). >> >> gst-launch souphttpsrc location=http://user:pass at hostname/path ! multipartdemux >> ! jpegdec ! xvimagesink > > Thanks for your answer. This pipeline does work for MJPEG (just like > the one I provided above), but does not solve the problem. > > I don't have an MJPEG stream. What I have is a HTTP source that serves > individual JPEG images (one single JPEG image per HTTP request). What > I actually need is a way to have souphttpsrc (or another HTTP src) make > additional HTTP requests periodically from the same URL. Is this > possible? If not, can anyone suggest how this could be implemented? > > Thanks, > > G Rodriguez > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. ? Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 remy.perrier at endirectv.com Mon Jan 10 12:58:23 2011 From: remy.perrier at endirectv.com (=?ISO-8859-1?Q?R=E9my_PERRIER?=) Date: Mon, 10 Jan 2011 12:58:23 +0100 Subject: [gst-devel] Plungin GL installation with Mingw Message-ID: <4D2AF45F.1080201@endirectv.com> Hi all, I am trying to install Gstreamer GL plugin and the mingw-get autoconf package doesnt seem to match. I need to compile & install autoconf 2.6.7 but I have to face some difficulities. Here is what I get : bob at bob /c/autoconf-2.67 $ make .... autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: autom4te: make[3]: *** [m4sugar.m4f] Error 1 make[3]: Leaving directory `/c/autoconf-2.67/lib/m4sugar' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/c/autoconf-2.67/lib' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/c/autoconf-2.67' make: *** [all] Error 2 My configuration : gst plugin GL : 0.10.2 gstreamer : 0.10. 31 I am using migw -get 0.1 alpha 5. I have tried different version of m4 : 1.4.15 , 1.4.4 now I am using 1.4.9 perl : v5.6.1 gcc : 4.5.0 Is somebody already have this problem ? What do you think about that ? regards, remy -------------- next part -------------- An HTML attachment was scrubbed... URL: From remy.perrier at endirectv.com Mon Jan 10 13:08:59 2011 From: remy.perrier at endirectv.com (=?ISO-8859-1?Q?R=E9my_PERRIER?=) Date: Mon, 10 Jan 2011 13:08:59 +0100 Subject: [gst-devel] Plungin GL installation with Mingw In-Reply-To: <4D2AF45F.1080201@endirectv.com> References: <4D2AF45F.1080201@endirectv.com> Message-ID: <4D2AF6DB.8090306@endirectv.com> I forget to display the plugin GL compilation result : bob at bob /c/gst-plugins-gl-0.10.2 $ make cat: ./win32/MANIFEST: No such file or directory make all-recursive cat: ./win32/MANIFEST: No such file or directory make[1]: Entering directory `/c/gst-plugins-gl-0.10.2' Making all in gst-libs make[2]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs' Making all in gst make[3]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' Making all in gl make[4]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' CC libgstgl_0.10_la-gstgldisplay.lo libtool: Version mismatch error. This is libtool 2.4, but the libtool: definition of this LT_INIT comes from libtool 2.2.6b. libtool: You should recreate aclocal.m4 with macros from libtool 2.4 libtool: and run autoconf again. make[4]: *** [libgstgl_0.10_la-gstgldisplay.lo] Error 63 make[4]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/c/gst-plugins-gl-0.10.2' make: *** [all] Error 2 Le 10/01/2011 12:58, R?my PERRIER a ?crit : > Hi all, > > > I am trying to install Gstreamer GL plugin and the mingw-get autoconf > package doesnt seem to match. > I need to compile & install autoconf 2.6.7 but I have to face some > difficulities. > Here is what I get : > > > bob at bob /c/autoconf-2.67 > $ make > > .... > > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > make[3]: *** [m4sugar.m4f] Error 1 > make[3]: Leaving directory `/c/autoconf-2.67/lib/m4sugar' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/c/autoconf-2.67/lib' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory `/c/autoconf-2.67' > make: *** [all] Error 2 > > > My configuration : > > gst plugin GL : 0.10.2 > gstreamer : 0.10. 31 > > I am using migw -get 0.1 alpha 5. > I have tried different version of m4 : 1.4.15 , 1.4.4 now I am using > 1.4.9 > perl : v5.6.1 > gcc : 4.5.0 > > Is somebody already have this problem ? > > What do you think about that ? > > > > regards, > > > remy > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 Mon Jan 10 14:00:44 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Mon, 10 Jan 2011 15:00:44 +0200 Subject: [gst-devel] amr-wb seek problem In-Reply-To: <1294652354379-3206831.post@n4.nabble.com> References: <1294652354379-3206831.post@n4.nabble.com> Message-ID: It seems amrwbdec does not handle src pad event (seek ) and thats why there is a blocking. There is a lack of implementation of src event handling thats why kept in gst-plugins-ugly. On Mon, Jan 10, 2011 at 11:39 AM, jorney wrote: > > Hi > I am writing amr-wb player with gstreamer on Marvell pxa310. > When I seek to a position by time format, there is a long time blocking > (10 sec or even to 1 min), and there is no voice during the block, then > an EOS message was emitted, > but the position I seeked is far away to the end of the file. > > Bytes format has the same problem! here is my seek code: > > breslt = gst_element_seek(pipeline, > 1.0, > GST_FORMAT_BYTES, > GST_SEEK_FLAG_FLUSH,//|GST_SEEK_FLAG_KEY_UNIT|GST_SEEK_FLAG_SEGMENT, > GST_SEEK_TYPE_SET, > bytePos,//GST_SECOND*nForw*count, > GST_SEEK_TYPE_NONE, > GST_CLOCK_TIME_NONE > ); > > I use gstreamer-0.10.29 and gst-plugins-ugly-0.10.15's amrwbdec, > gst-plugins-bad-0.10.13's amrparse. like this: > > gst-launch filesrc location=abc.amr ! amrparse ! amrwbdec ! alsasink > > > And the codec is opencore-amr-0.1.2 > Could any body help? > > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/amr-wb-seek-problem-tp3206831p3206831.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any > company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 bisht.sudarshan at gmail.com Mon Jan 10 14:09:41 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Mon, 10 Jan 2011 15:09:41 +0200 Subject: [gst-devel] amr-wb seek problem In-Reply-To: References: <1294652354379-3206831.post@n4.nabble.com> Message-ID: check the return value of gst_element_seek too . On Mon, Jan 10, 2011 at 3:00 PM, sudarshan bisht wrote: > > It seems amrwbdec does not handle src pad event (seek ) and thats why > there is a blocking. There is a lack of implementation of src event > handling thats why kept in gst-plugins-ugly. > > > > On Mon, Jan 10, 2011 at 11:39 AM, jorney wrote: > >> >> Hi >> I am writing amr-wb player with gstreamer on Marvell pxa310. >> When I seek to a position by time format, there is a long time blocking >> (10 sec or even to 1 min), and there is no voice during the block, then >> an EOS message was emitted, >> but the position I seeked is far away to the end of the file. >> >> Bytes format has the same problem! here is my seek code: >> >> breslt = gst_element_seek(pipeline, >> 1.0, >> GST_FORMAT_BYTES, >> GST_SEEK_FLAG_FLUSH,//|GST_SEEK_FLAG_KEY_UNIT|GST_SEEK_FLAG_SEGMENT, >> GST_SEEK_TYPE_SET, >> bytePos,//GST_SECOND*nForw*count, >> GST_SEEK_TYPE_NONE, >> GST_CLOCK_TIME_NONE >> ); >> >> I use gstreamer-0.10.29 and gst-plugins-ugly-0.10.15's amrwbdec, >> gst-plugins-bad-0.10.13's amrparse. like this: >> >> gst-launch filesrc location=abc.amr ! amrparse ! amrwbdec ! alsasink >> >> >> And the codec is opencore-amr-0.1.2 >> Could any body help? >> >> -- >> View this message in context: >> http://gstreamer-devel.966125.n4.nabble.com/amr-wb-seek-problem-tp3206831p3206831.html >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> >> >> ------------------------------------------------------------------------------ >> Gaining the trust of online customers is vital for the success of any >> company >> that requires sensitive data to be transmitted over the Web. Learn how >> to >> best implement a security strategy that keeps consumers' information >> secure >> and instills the confidence they need to proceed with transactions. >> 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 > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From puneeth.mv at globaledgesoft.com Mon Jan 10 15:10:19 2011 From: puneeth.mv at globaledgesoft.com (Puneeth) Date: Mon, 10 Jan 2011 06:10:19 -0800 (PST) Subject: [gst-devel] Problem in using ffdec_mp3 to decode live streaming In-Reply-To: <1294314501815-3177271.post@n4.nabble.com> References: <1294314501815-3177271.post@n4.nabble.com> Message-ID: <1294668619597-3207224.post@n4.nabble.com> Hi All I could not resolve this problem, still i m facing same problem using ffdec_mp3 to decode and playback the output of lame encoder.This is working fine with mad decoder, is there any LGPL mp3 decoders which works fine for the belpw command. Please let me know if there r any, any help will be greatly appreciated. Thanks in advance. sudo gst-launch gstrtpbin name=rtpbin alsasrc ! audioconvert ! audioresample ! lamemp3enc ! rtpmpapay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5002 host=127.0.0.1 rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=127.0.0.1 sync=false async=false udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0 sudo gst-launch gstrtpbin name=rtpbin udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)90000,encoding-name=(string)MPA,pt=(int)96" port=5002 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtpmpadepay ! mp3parse ! mad ! audioconvert ! volume volume=10 ! audioresample quality=10 ! alsasink udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=127.0.0.1 sync=false async=false The above command works fine, but when i replace with ffdec_mp3 there is lot of noice, how could i solve this or is there any other LGPL decoders which would do the needful. Please help. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Problem-in-using-ffdec-mp3-to-decode-live-streaming-tp3177271p3207224.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From vaisaari at gmail.com Mon Jan 10 15:15:53 2011 From: vaisaari at gmail.com (vaisaari) Date: Mon, 10 Jan 2011 06:15:53 -0800 (PST) Subject: [gst-devel] Getting converted frame from playbin2 element In-Reply-To: References: <1294053159060-3171830.post@n4.nabble.com> Message-ID: <1294668953244-3207234.post@n4.nabble.com> > imo this should not happen, as the signal handler (delegated to > playsink) internally calls gst_play_sink_get_last_frame, which locks > playsink. Can you post to pastebin the output you get after setting > GST_DEBUG=playsink:5 ? You can find the output from http://pastebin.com/raw.php?i=HgD1S8Uz here . The output looks the same whether or not the conversion is successful. > Which video decoder are you using? I have tried it with both uncompressed and MPEG4 encoded videos. > Besides, if you're trying to render a video clip in Qt, you may > consider using QtGStreamer.. It looks very promising. At some point, I most certainly will. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Getting-converted-frame-from-playbin2-element-tp3171830p3207234.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From malloblenne at gmail.com Mon Jan 10 15:22:29 2011 From: malloblenne at gmail.com (Mauro Brenna) Date: Mon, 10 Jan 2011 15:22:29 +0100 Subject: [gst-devel] Obtain a JPEG sequence from a RTP MJPEG stream (RFC 2435) In-Reply-To: References: Message-ID: I solved it using a pipeline like: gst-launch.exe -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 ! multifilesink location="test%05d.jpg" Mauro From remy.perrier at endirectv.com Mon Jan 10 15:26:36 2011 From: remy.perrier at endirectv.com (=?ISO-8859-1?Q?R=E9my_PERRIER?=) Date: Mon, 10 Jan 2011 15:26:36 +0100 Subject: [gst-devel] Plungin GL installation with Mingw In-Reply-To: <4D2AF6DB.8090306@endirectv.com> References: <4D2AF45F.1080201@endirectv.com> <4D2AF6DB.8090306@endirectv.com> Message-ID: <4D2B171C.7060505@endirectv.com> ok my mingw installation was a mess, I have cleaned up autoconf & m4. I have installed autoconf 2.68 and m4 1.4.14 package thanks to mingw-get , run ./autogen.sh and installation is ok However gl plugins dont appear in the gst-inspect command. They are not black listed and of course I cannot run the GL examples. The "make check" in the gst-plugin gl repertory works good Where does it may come from ? kind regards, remy Le 10/01/2011 13:08, R?my PERRIER a ?crit : > I forget to display the plugin GL compilation result : > > > bob at bob /c/gst-plugins-gl-0.10.2 > $ make > cat: ./win32/MANIFEST: No such file or directory > make all-recursive > cat: ./win32/MANIFEST: No such file or directory > make[1]: Entering directory `/c/gst-plugins-gl-0.10.2' > Making all in gst-libs > make[2]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs' > Making all in gst > make[3]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' > Making all in gl > make[4]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' > CC libgstgl_0.10_la-gstgldisplay.lo > libtool: Version mismatch error. This is libtool 2.4, but the > libtool: definition of this LT_INIT comes from libtool 2.2.6b. > libtool: You should recreate aclocal.m4 with macros from libtool 2.4 > libtool: and run autoconf again. > make[4]: *** [libgstgl_0.10_la-gstgldisplay.lo] Error 63 > make[4]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory `/c/gst-plugins-gl-0.10.2' > make: *** [all] Error 2 > > > Le 10/01/2011 12:58, R?my PERRIER a ?crit : >> Hi all, >> >> >> I am trying to install Gstreamer GL plugin and the mingw-get >> autoconf package doesnt seem to match. >> I need to compile & install autoconf 2.6.7 but I have to face some >> difficulities. >> Here is what I get : >> >> >> bob at bob /c/autoconf-2.67 >> $ make >> >> .... >> >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> autom4te: >> make[3]: *** [m4sugar.m4f] Error 1 >> make[3]: Leaving directory `/c/autoconf-2.67/lib/m4sugar' >> make[2]: *** [all-recursive] Error 1 >> make[2]: Leaving directory `/c/autoconf-2.67/lib' >> make[1]: *** [all-recursive] Error 1 >> make[1]: Leaving directory `/c/autoconf-2.67' >> make: *** [all] Error 2 >> >> >> My configuration : >> >> gst plugin GL : 0.10.2 >> gstreamer : 0.10. 31 >> >> I am using migw -get 0.1 alpha 5. >> I have tried different version of m4 : 1.4.15 , 1.4.4 now I am using >> 1.4.9 >> perl : v5.6.1 >> gcc : 4.5.0 >> >> Is somebody already have this problem ? >> >> What do you think about that ? >> >> >> >> regards, >> >> >> remy >> >> >> ------------------------------------------------------------------------------ >> Gaining the trust of online customers is vital for the success of any company >> that requires sensitive data to be transmitted over the Web. Learn how to >> best implement a security strategy that keeps consumers' information secure >> and instills the confidence they need to proceed with transactions. >> 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 > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 bilboed at gmail.com Mon Jan 10 16:02:43 2011 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 10 Jan 2011 16:02:43 +0100 Subject: [gst-devel] pre-release of GStreamer Editing Services 0.10.0.2 Message-ID: <1294671763.23475.24.camel@localhost> Hi all, The first pre-release for the GStreamer Editing Services is available at: http://gstreamer.freedesktop.org/data/src/gst-editing-services/pre/ Documentation can be found online at: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-editing-services/html/ Ubuntu users might also be able to test this pre-release by using the GStreamer PPA (as soon as they get uploaded there). Please test it thoroughly, and let us know of any regressions or other important issues by filing blocker bugs in bugzilla: http://gstreamer.freedesktop.org/bugs/ (Component: Further pre-releases will follow as needed. If all goes well, 0.10.1 release should happen later this month when GStreamer core/-base 0.10.32 are out, as per http://gstreamer.freedesktop.org/wiki/ReleasePlanning2011 Edward From malloblenne at gmail.com Mon Jan 10 16:23:18 2011 From: malloblenne at gmail.com (Mauro Brenna) Date: Mon, 10 Jan 2011 16:23:18 +0100 Subject: [gst-devel] Streaming RTP jpeg images (RFC2435) with restart marker header. Is it unsupported? Why? Message-ID: Hello, I would like to stream jpeg images using RTP (RFC2435). The images which are coming from my camera include the restart marker (0xFF 0xDD). I tested to send an image via RTP with the following pipelines, under windows: #STREAM gst-launch.exe -v gstrtpbin name =rtpbin multifilesrc location="camera.jpg" caps="image/jpeg,framerate=1/1" ! rt pjpegpay ! 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" s ync=false async=false udpsrc port=10000 #RECEIVE gst-launch.exe -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 ! multifilesink location="test%05d.jpg" I notice that only a little part of the image is visible in the resulting jpeg. In particular, the only difference between the original image (streaming)and the received one is the missing of the line that specifies the restart interval between MCUs: e.g.: ff dd 00 04 00 32 before the SOS: ff c0. I looked in the source files for a possible explanation (gstrtpjpegpay.c and gstrtpjpegdepay.c) and I read this comment in gstrtpjpegpay.c, line 223: /* FIXME: restart marker header currently unsupported */ I read the code and I don't understand when the DRI part is not included (in the creation of RTP packed or in its reception). I would like to know what are the problems in implementing that part. In my case, I only need to prepend those 6 byte before the start of scan (by hand) and I managed to see the correct picture. Thanks, Mauro From ensonic at hora-obscura.de Mon Jan 10 17:03:03 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 10 Jan 2011 18:03:03 +0200 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: <4D2B2DB7.8070001@hora-obscura.de> On 21.12.2010 20:53, 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? > You can use a uri query: gst_query_new_uri(). Stefan > > > -----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 > > > ------------------------------------------------------------------------------ > 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 remy.perrier at endirectv.com Mon Jan 10 17:09:50 2011 From: remy.perrier at endirectv.com (=?ISO-8859-1?Q?R=E9my_PERRIER?=) Date: Mon, 10 Jan 2011 17:09:50 +0100 Subject: [gst-devel] Plungin GL installation with Mingw In-Reply-To: <4D2B171C.7060505@endirectv.com> References: <4D2AF45F.1080201@endirectv.com> <4D2AF6DB.8090306@endirectv.com> <4D2B171C.7060505@endirectv.com> Message-ID: <4D2B2F4E.7000008@endirectv.com> Hi, It s me again. I have just realized that there is no dll object with libgstopengl and libgstgl in /usr/local/lib and /usr/local/lib/gstreamer-0.10 ( contrary to the other lib ). It may be the cause of my problem. How can I generate those missing ( I guess) files ? Thanks for your help. regards, Le 10/01/2011 15:26, R?my PERRIER a ?crit : > ok my mingw installation was a mess, I have cleaned up autoconf & m4. > I have installed autoconf 2.68 and m4 1.4.14 package thanks to > mingw-get , run ./autogen.sh and installation is ok > > However gl plugins dont appear in the gst-inspect command. > They are not black listed and of course I cannot run the GL examples. > > The "make check" in the gst-plugin gl repertory works good > > Where does it may come from ? > > > kind regards, > > remy > > > Le 10/01/2011 13:08, R?my PERRIER a ?crit : >> I forget to display the plugin GL compilation result : >> >> >> bob at bob /c/gst-plugins-gl-0.10.2 >> $ make >> cat: ./win32/MANIFEST: No such file or directory >> make all-recursive >> cat: ./win32/MANIFEST: No such file or directory >> make[1]: Entering directory `/c/gst-plugins-gl-0.10.2' >> Making all in gst-libs >> make[2]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs' >> Making all in gst >> make[3]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' >> Making all in gl >> make[4]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' >> CC libgstgl_0.10_la-gstgldisplay.lo >> libtool: Version mismatch error. This is libtool 2.4, but the >> libtool: definition of this LT_INIT comes from libtool 2.2.6b. >> libtool: You should recreate aclocal.m4 with macros from libtool 2.4 >> libtool: and run autoconf again. >> make[4]: *** [libgstgl_0.10_la-gstgldisplay.lo] Error 63 >> make[4]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' >> make[3]: *** [all-recursive] Error 1 >> make[3]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' >> make[2]: *** [all-recursive] Error 1 >> make[2]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs' >> make[1]: *** [all-recursive] Error 1 >> make[1]: Leaving directory `/c/gst-plugins-gl-0.10.2' >> make: *** [all] Error 2 >> >> >> Le 10/01/2011 12:58, R?my PERRIER a ?crit : >>> Hi all, >>> >>> >>> I am trying to install Gstreamer GL plugin and the mingw-get >>> autoconf package doesnt seem to match. >>> I need to compile & install autoconf 2.6.7 but I have to face some >>> difficulities. >>> Here is what I get : >>> >>> >>> bob at bob /c/autoconf-2.67 >>> $ make >>> >>> .... >>> >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> autom4te: >>> make[3]: *** [m4sugar.m4f] Error 1 >>> make[3]: Leaving directory `/c/autoconf-2.67/lib/m4sugar' >>> make[2]: *** [all-recursive] Error 1 >>> make[2]: Leaving directory `/c/autoconf-2.67/lib' >>> make[1]: *** [all-recursive] Error 1 >>> make[1]: Leaving directory `/c/autoconf-2.67' >>> make: *** [all] Error 2 >>> >>> >>> My configuration : >>> >>> gst plugin GL : 0.10.2 >>> gstreamer : 0.10. 31 >>> >>> I am using migw -get 0.1 alpha 5. >>> I have tried different version of m4 : 1.4.15 , 1.4.4 now I am >>> using 1.4.9 >>> perl : v5.6.1 >>> gcc : 4.5.0 >>> >>> Is somebody already have this problem ? >>> >>> What do you think about that ? >>> >>> >>> >>> regards, >>> >>> >>> remy >>> >>> >>> ------------------------------------------------------------------------------ >>> Gaining the trust of online customers is vital for the success of any company >>> that requires sensitive data to be transmitted over the Web. Learn how to >>> best implement a security strategy that keeps consumers' information secure >>> and instills the confidence they need to proceed with transactions. >>> 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 >> >> >> ------------------------------------------------------------------------------ >> Gaining the trust of online customers is vital for the success of any company >> that requires sensitive data to be transmitted over the Web. Learn how to >> best implement a security strategy that keeps consumers' information secure >> and instills the confidence they need to proceed with transactions. >> 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 > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 david.henningsson at canonical.com Mon Jan 10 17:24:53 2011 From: david.henningsson at canonical.com (David Henningsson) Date: Mon, 10 Jan 2011 10:24:53 -0600 Subject: [gst-devel] [pulseaudio-discuss] [PATCH 0/3] Fighting rewinds In-Reply-To: <4D059A7F.1000007@canonical.com> References: <4D00DFAF.3000807@canonical.com> <4D0231DF.7040008@canonical.com> <4D059A7F.1000007@canonical.com> Message-ID: <4D2B32D5.1050902@canonical.com> On 2010-12-12 22:01, David Henningsson wrote: > 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). > Hi Pierre and rest of gstreamer-devel, Given the discussion above, did you have any time to do more thinking about it? I'm attaching a simple patch. Some testing of that patch hasn't reported any regressions, and it does start to send larger packages (which in turn saves PA from additional packet overhead). I'm attaching a patch for Ubuntu, but given that I give you a proper git header etc to that patch, would the folks here apply it? // David -------------- next part -------------- A non-text attachment was scrubbed... Name: default-buffer-size.patch Type: text/x-patch Size: 2024 bytes Desc: not available URL: From ensonic at hora-obscura.de Mon Jan 10 17:29:58 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 10 Jan 2011 18:29:58 +0200 Subject: [gst-devel] does signal has last param user_data in gstreamer? In-Reply-To: <1294530254.6105.3.camel@zingle> References: <4D25C8A6.2060502@lfarkas.org> <4D25D1E2.50403@igalia.com> <4D25E035.1090506@lfarkas.org> <1294328507.23475.11.camel@localhost> <4D28D355.3030900@lfarkas.org> <1294530254.6105.3.camel@zingle> Message-ID: <4D2B3406.9000902@hora-obscura.de> On 09.01.2011 01:44, Tim-Philipp M?ller wrote: > On Sat, 2011-01-08 at 22:12 +0100, Farkas Levente wrote: > >> but it'd be nice to use consistent docs/help ie. either all gst-inspect >> signal contain it as last param or none (see for example gst-inspect >> input-selector). > For action signals (and vfuncs for such action signals) no user_data > argument should be displayed, since that doesn't make sense. gst-inspect > doesn't show a user_data argument here, but gtk-doc (html docs) does, > which looks like a gtk-doc bug to me. Hmm, one can still conect to an action signal and the callback is called when the signal is emitted. I agree that it would be quite a strange thing to do. On the gtk-doc side no one else has complained about it so far. Filed as enhancement request: https://bugzilla.gnome.org/show_bug.cgi?id=639155 Stefan > For normal signals, there should be a user data argument, because what > you're displaying is basically the signature your callback function > should have. > > Two different things really, and hence two different conventions. > > Cheers > -Tim > > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 Mon Jan 10 17:57:01 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Mon, 10 Jan 2011 22:27:01 +0530 Subject: [gst-devel] iFrame identification in encoded stream Message-ID: Hi Guys, Is there a way to tell if an encoded frame returned by a demuxer(say qtdemux) is an inter-Frame (iFrame) or not? Right now, I've written a routine to do this by parsing the initial few bits but that becomes difficult with increase in number of formats to support. Is there a GstBuffer property I can check to see if its an iFrame? Thanks Rohit -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Mon Jan 10 18:10:05 2011 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 10 Jan 2011 18:10:05 +0100 Subject: [gst-devel] iFrame identification in encoded stream In-Reply-To: References: Message-ID: <1294679405.10348.2.camel@localhost> On Mon, 2011-01-10 at 22:27 +0530, Rohit Atri wrote: > Hi Guys, > > > Is there a way to tell if an encoded frame returned by a demuxer(say > qtdemux) is an inter-Frame (iFrame) or not? > Right now, I've written a routine to do this by parsing the initial > few bits but that becomes difficult with increase in number of formats > to support. Is there a GstBuffer property I can check to see if its an > iFrame? > Check if the GST_BUFFER_FLAG_DELTA_UNIT flag is set or not. If it's not set... it's a keyframe. http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstBuffer.html#GstBufferFlag > > Thanks > Rohit > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 lane at brooks.nu Mon Jan 10 21:51:17 2011 From: lane at brooks.nu (Lane Brooks) Date: Mon, 10 Jan 2011 13:51:17 -0700 Subject: [gst-devel] kenburns plugin Message-ID: <4D2B7145.7010503@brooks.nu> I have written a kenburns plugin (http://en.wikipedia.org/wiki/Ken_burns_effect) that does smooth zooming/panning simultaneously. I have it committed in a git repo at https://github.com/dirjud/kenburns There are some gst-launch samples in the src/gstkenburns.c file. Currently I have only implemented nearest neighbor sampling, so it is fast but can show aliasing artifacts. Is this something that the developers would be interested in seeing integrated into gstreamer? Any feedback would be appreciated. Thanks, Lane From el.cameleon.1 at gmail.com Mon Jan 10 22:19:09 2011 From: el.cameleon.1 at gmail.com (Vincent) Date: Mon, 10 Jan 2011 22:19:09 +0100 Subject: [gst-devel] MP4 video does not play with totem but no problem with VLC Message-ID: Hi gents, I met an issue with some MP4 video recorded during holidays: Shotwell cannot display a cover for these movies and totem cannot play it ("impossible to determine the type of flux")... However, VLC read it without any problems... Can anyone which has some knowledge in MP4 decoding can have a look to one of these video to try to explain why I meet this issue? Shotwell developers tell me to ask on this list which seems to be more appropriate for this kind of problems. I have uploaded the video here: http://ubuntuone.com/p/XHX/ Thanks in advance for your help! Best regards -- Vincent -------------- next part -------------- An HTML attachment was scrubbed... URL: From giorgio9 at libero.it Tue Jan 11 00:44:08 2011 From: giorgio9 at libero.it (giorgino) Date: Mon, 10 Jan 2011 15:44:08 -0800 (PST) Subject: [gst-devel] sip videophone or java videoconference Message-ID: <1294703048910-3208226.post@n4.nabble.com> Hi all, I need your support in order to help me to solve an issue. I need to integrate in my java swing desktop application a videoconference tool. Firstly I integrate into my application a SIP client that works well for audio, but I have not idea how I can add video (captured from a usb webcam, encoded with h264, encapsulte with rtp and sent to internet). I think that a starting point can be the server-v4l2-H264-alsasrc-PCMA.sh script but I would like to know your opinion on that. You can give me some suggestions? Have a nice day G. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/sip-videophone-or-java-videoconference-tp3208226p3208226.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From mgruenke at Tycoint.com Tue Jan 11 02:01:17 2011 From: mgruenke at Tycoint.com (Gruenke, Matt) Date: Mon, 10 Jan 2011 20:01:17 -0500 Subject: [gst-devel] kenburns plugin References: <4D2B7145.7010503@brooks.nu> Message-ID: Speaking for myself, I'm interested in an efficient, high-quality element that zooms, shifts, and crops. At least one of the sampling options should be a low-pass filter that can be used to do any interpolation, decimation, and phase-shifting in one pass. The number of taps should be selectable and either use an optimal filter or at least support some common window functions. IMO, this is the hard part. Once you have it, then the Ken Burns Effect can be implemented via Dynamic Parameters (see Plugin Writer's Guide: Ch. 15). However, the benefit of making it a generic element is that you can also just use it as a fast, high-quality way to statically zoom and/or crop. Matt ________________________________ From: Lane Brooks [mailto:lane at brooks.nu] Sent: Mon 1/10/2011 3:51 PM To: Discussion of the development of GStreamer Subject: [gst-devel] kenburns plugin I have written a kenburns plugin (http://en.wikipedia.org/wiki/Ken_burns_effect) that does smooth zooming/panning simultaneously. I have it committed in a git repo at https://github.com/dirjud/kenburns There are some gst-launch samples in the src/gstkenburns.c file. Currently I have only implemented nearest neighbor sampling, so it is fast but can show aliasing artifacts. Is this something that the developers would be interested in seeing integrated into gstreamer? Any feedback would be appreciated. Thanks, Lane ------------------------------------------------------------------------------ Gaining the trust of online customers is vital for the success of any company that requires sensitive data to be transmitted over the Web. Learn how to best implement a security strategy that keeps consumers' information secure and instills the confidence they need to proceed with transactions. 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 -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 5112 bytes Desc: not available URL: From lane at brooks.nu Tue Jan 11 07:31:33 2011 From: lane at brooks.nu (Lane Brooks) Date: Mon, 10 Jan 2011 23:31:33 -0700 Subject: [gst-devel] AYUV text overlay Message-ID: <4D2BF945.5000500@brooks.nu> Sebastian, Looking through the git log, it looks like you added support for the AYUV format in the pango textoverlay element. It does not quite behave how I was hoping it would, so I wanted to discuss it with you to understand the current implementation. If I create a completely transparent background image, I was hoping that textoverlay would still overlay the requested text appropriately, but the current implementation leaves the resulting image (including the text overlay) completely transparent (so the text is invisible). You can see this by running this pipeline: gst-launch videotestsrc pattern=white ! video/x-raw-yuv,format=\(fourcc\)AYUV ! alpha alpha=0.0 ! textoverlay text=Testing auto-resize=False font-desc=60px ! videomixer ! ffmpegcolorspace ! autovideosink Attached is a patch that changes the alpha compositing behavior to use a "A OVER B" scheme (http://en.wikipedia.org/wiki/Alpha_compositing). Can you comment on this approach? With this patch, the above pipeline works like I want it to in that the text overlays and is visible regardless of alpha of the background image. I am not sure, however, the reasons for the current implementation and what advantages it has. In case there is confusion, if you run the above pipeline with and without this patch, you will instantly see the difference. Without this patch, you get a completely transparent image and the text overlay is invisible. With the patch you get a transparent image everywhere except for the text in the overlay. If the behavior from this patch is considered the appropriate behavior, I can also fix the the RGBA/ABGR path to do the same thing. Thanks, Lane -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: patch_over.txt URL: From t.i.m at zen.co.uk Tue Jan 11 10:13:47 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 11 Jan 2011 09:13:47 +0000 Subject: [gst-devel] AYUV text overlay In-Reply-To: <4D2BF945.5000500@brooks.nu> References: <4D2BF945.5000500@brooks.nu> Message-ID: <1294737227.19538.1.camel@zingle> On Mon, 2011-01-10 at 23:31 -0700, Lane Brooks wrote: Hi Lane, > [textoverlay] does not quite behave how I was hoping it would (...) > > If I create a completely transparent background image, I was hoping that > textoverlay would still overlay the requested text appropriately, but > the current implementation leaves the resulting image (including the > text overlay) completely transparent (so the text is invisible). (...) > > Attached is a patch that changes the alpha compositing behavior (...) Please could you open a bug for this in bugzilla and attach your patch there, so it doesn't get overlooked? Thanks! http://gstreamer.freedesktop.org/bugs/ Cheers -Tim From pierre.crepieux at orange-ftgroup.com Tue Jan 11 11:51:58 2011 From: pierre.crepieux at orange-ftgroup.com (pierre.crepieux at orange-ftgroup.com) Date: Tue, 11 Jan 2011 11:51:58 +0100 Subject: [gst-devel] sending audio/video on RTP with C API In-Reply-To: <20110107120530.2f8713ab@mtelleria.com> References: <4D26ECBE.4020800@orange-ftgroup.com> <20110107120530.2f8713ab@mtelleria.com> Message-ID: <4D2C364E.6040604@orange-ftgroup.com> Thanks, I gave a look at your code, and found it instructive. It didn't really highlight the problem I had (i'm using dshow and directsound src), but anyway, I got my sample working :-) There is still one thing i don't get: my sample works if I use h263 encoding but doesn't with h264 ... I still have many thing to learn regarding gstreamer ! Miguel Telleria de Esteban a ?crit : > Hello, > > Like you are doing now, I followed the same goal of using gstrtpbin and > found many small nuances. > > You can find here an implementation that I did to send ogg (vorbis and > theora) audio through RTP using gstrtpbin. > > http://www.ctr.unican.es/asignaturas/dec/Doc/ogg_audio_video_rtp_sender_receiver.tar.gz > > Hope it helps. > > And yes, I plan to submit it as a bugzilla addition sometime in the > future :). > > Miguel > > > On Fri, 07 Jan 2011 11:36:46 +0100 pierre.crepieux > wrote: > > >> Hi, >> >> I'm trying to write a simple C program that sends webcam/microphone >> streams on rtp with gstrtpbin. I gave a look at the provided >> server-alsasrc-PCMA.c test program and mimics it to add h264 stream >> to the rtpbin (just like server-v4l2-H264-alsasrc-PCMA.sh does). >> Sadly, using VLC to receive the streams, give me a fixed image and a >> pretty bad sound (delay, sometime stop/restart,...). Are there any >> tricks to be aware of for doing this translation of gst-launch pipe >> definition to C code ? Does gstrtpbin require subtle manipulations >> (that gst-launch would do behind the scene ... ??? ) to operate with >> 2 rtp stream ? >> >> Pierre >> -- >> >> > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 hadess at hadess.net Tue Jan 11 11:31:29 2011 From: hadess at hadess.net (Bastien Nocera) Date: Tue, 11 Jan 2011 10:31:29 +0000 Subject: [gst-devel] MP4 video does not play with totem but no problem with VLC In-Reply-To: References: Message-ID: <1294741892.2600.16.camel@novo.hadess.net> On Mon, 2011-01-10 at 22:19 +0100, Vincent wrote: > Hi gents, > > I met an issue with some MP4 video recorded during holidays: Shotwell > cannot display a cover for these movies and totem cannot play it > ("impossible to determine the type of flux")... However, VLC read it > without any problems... Can anyone which has some knowledge in MP4 > decoding can have a look to one of these video to try to explain why I > meet this issue? Shotwell developers tell me to ask on this list which > seems to be more appropriate for this kind of problems. > > I have uploaded the video here: http://ubuntuone.com/p/XHX/ Your best bet is to try and reproduce the problem with gst-launch, and see if the problem occurs there. See: http://projects.gnome.org/totem/#bugs Then file a bug against GStreamer if you can reproduce the problem, or against Totem if you cannot. FWIW, I could not play the file: $ file shotwell-broken-file.mp4 shotwell-broken-file.mp4: gzip compressed data, max compression $ mv shotwell-broken-file.mp4 shotwell-broken-file.mp4.gz $ gunzip shotwell-broken-file.mp4.gz gzip: shotwell-broken-file.mp4.gz: invalid compressed data--format violated Cheers From t.i.m at zen.co.uk Tue Jan 11 12:22:16 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 11 Jan 2011 11:22:16 +0000 Subject: [gst-devel] MP4 video does not play with totem but no problem with VLC In-Reply-To: References: Message-ID: <1294744936.19538.19.camel@zingle> On Mon, 2011-01-10 at 22:19 +0100, Vincent wrote: Hi Vincent, > I met an issue with some MP4 video recorded during holidays: Shotwell > cannot display a cover for these movies and totem cannot play it > ("impossible to determine the type of flux")... However, VLC read it > without any problems... Can anyone which has some knowledge in MP4 > decoding can have a look to one of these video to try to explain why I > meet this issue? Shotwell developers tell me to ask on this list which > seems to be more appropriate for this kind of problems. > > I have uploaded the video here: http://ubuntuone.com/p/XHX/ > > Thanks in advance for your help! This works fine for me both with git and older releases. You don't say what distro/version and/or version of the various GStreamer modules you are using, but it sounds like maybe you might need to update. Cheers -Tim From el.cameleon.1 at gmail.com Tue Jan 11 13:26:32 2011 From: el.cameleon.1 at gmail.com (Vincent) Date: Tue, 11 Jan 2011 13:26:32 +0100 Subject: [gst-devel] MP4 video does not play with totem but no problem with VLC In-Reply-To: <1294744936.19538.19.camel@zingle> References: <1294744936.19538.19.camel@zingle> Message-ID: On Tue, Jan 11, 2011 at 12:22 PM, Tim-Philipp M?ller wrote: > You don't say what distro/version and/or version of the various > GStreamer modules you are using, but it sounds like maybe you might need > to update. > I use Ubuntu Maverick, but I do not know which release of GSTreamer... Could you explain me a little bit more how can I update? -------------- next part -------------- An HTML attachment was scrubbed... URL: From lane at brooks.nu Tue Jan 11 16:05:32 2011 From: lane at brooks.nu (Lane Brooks) Date: Tue, 11 Jan 2011 08:05:32 -0700 Subject: [gst-devel] AYUV text overlay In-Reply-To: <1294737227.19538.1.camel@zingle> References: <4D2BF945.5000500@brooks.nu> <1294737227.19538.1.camel@zingle> Message-ID: <4D2C71BC.9000106@brooks.nu> On 01/11/2011 02:13 AM, Tim-Philipp M?ller wrote: > On Mon, 2011-01-10 at 23:31 -0700, Lane Brooks wrote: > > Hi Lane, > >> [textoverlay] does not quite behave how I was hoping it would (...) >> >> If I create a completely transparent background image, I was hoping that >> textoverlay would still overlay the requested text appropriately, but >> the current implementation leaves the resulting image (including the >> text overlay) completely transparent (so the text is invisible). (...) >> >> Attached is a patch that changes the alpha compositing behavior (...) > > Please could you open a bug for this in bugzilla and attach your patch > there, so it doesn't get overlooked? Thanks! > I am happy to file a bug with this patch. Is bugzilla where you prefer the discussion take place on what the desired overlay behavior should be? In other words, I am not sure the reasons and advantages of the current implementation, so I wanted to first understand that prior to seeking the application of my patch. Lane From rafael.lmsousa at gmail.com Tue Jan 11 16:10:57 2011 From: rafael.lmsousa at gmail.com (Rafael Sousa) Date: Tue, 11 Jan 2011 11:10:57 -0400 Subject: [gst-devel] Changing the x264 encoding properties in real-time In-Reply-To: References: Message-ID: Hi all, I'd like to change the properties of x264 encoding in execution time. The propertie that I'd change is the I-frame frequency. The criteria of changing is the information about jitter and packet loss obtained by RTP. Theses information I already have, but I'm having difficulties in changing the I-frame frequency on the encoder. Any tips? I'm implementing in C. thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From pierre.crepieux at orange-ftgroup.com Fri Jan 7 12:12:05 2011 From: pierre.crepieux at orange-ftgroup.com (pierre.crepieux at orange-ftgroup.com) Date: Fri, 07 Jan 2011 12:12:05 +0100 Subject: [gst-devel] sending audio/video on RTP with C API In-Reply-To: <1294397198.23475.13.camel@localhost> References: <4D26ECBE.4020800@orange-ftgroup.com> <1294397198.23475.13.camel@localhost> Message-ID: <4D26F505.6010600@orange-ftgroup.com> An HTML attachment was scrubbed... URL: From kaija at ms2.hinet.net Sat Jan 1 03:57:29 2011 From: kaija at ms2.hinet.net (kaija) Date: Fri, 31 Dec 2010 18:57:29 -0800 (PST) Subject: [gst-devel] dynamically remove custom branch from tee and videomixer In-Reply-To: <1293287707152-3163782.post@n4.nabble.com> References: <1293287707152-3163782.post@n4.nabble.com> Message-ID: <1293850649741-3169949.post@n4.nabble.com> anyone help? kaija wrote: > > 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-tp3163782p3169949.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From sholzer at my-gekko.com Tue Jan 4 12:12:04 2011 From: sholzer at my-gekko.com (gekko) Date: Tue, 4 Jan 2011 03:12:04 -0800 (PST) Subject: [gst-devel] how do write a gst mixer in c Message-ID: <1294139524319-3173361.post@n4.nabble.com> Hello! does anybody know how to write a gsreamer audio mixer in c? i want to control the volume and also i want to get some informations like time or rate... -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/how-do-write-a-gst-mixer-in-c-tp3173361p3173361.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From holzer.st at akfree.it Tue Jan 4 12:29:15 2011 From: holzer.st at akfree.it (bonkers) Date: Tue, 4 Jan 2011 03:29:15 -0800 (PST) Subject: [gst-devel] input_channel_t not found Message-ID: <1294140555103-3173384.post@n4.nabble.com> when i compile my c source i get the error: mixer.c:55: error: ?input_channel_t? undeclared (first use in this function) can anyone help me? i already included - gst/includes/mixer.h - gst.h -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/input-channel-t-not-found-tp3173384p3173384.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From pcd7 at btinternet.com Tue Jan 4 17:57:26 2011 From: pcd7 at btinternet.com (pcd) Date: Tue, 4 Jan 2011 08:57:26 -0800 (PST) Subject: [gst-devel] Play videos with playbin2 - gapless method ? Message-ID: <1294160246840-3173911.post@n4.nabble.com> Hi, I want to play a number of video .ts files with playbin2 - without breaks. I have heard of the gapless feature - but cannot find any usage example. Can someone please describe (or give links) how to do this ? I am familiar with pipelines and also python scripts. Thanks in advance. Regards, pcd. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Play-videos-with-playbin2-gapless-method-tp3173911p3173911.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From brad at blacksheepsoftware.com.au Wed Jan 5 06:31:04 2011 From: brad at blacksheepsoftware.com.au (Brad Goldsmith) Date: Wed, 5 Jan 2011 16:31:04 +1100 Subject: [gst-devel] Play audio from a memory buffer? In-Reply-To: <1294152909.19493.22.camel@zingle> References: <1294144535806-3173448.post@n4.nabble.com> <1294152909.19493.22.camel@zingle> Message-ID: Can appsrc be put in that pipeline in place of filesrc without any additional elements? Cheers, Brad On 5 January 2011 01:55, Tim-Philipp M?ller wrote: > On Tue, 2011-01-04 at 04:35 -0800, bcg wrote: > > Hi, > > > This should almost be an FAQ but it's not in there so here goes: > > > > I have a pipeline thus: > > > > file-source -> decodebin -> audioresample -> converter -> audio_output > > > > Using location (or fd if I change file_source to a fdsrc) I can happily > play > > most audio I throw at it (including mp3s) providing its in a file. What I > > would like to do is play from a memory location with the mp3 data already > in > > it. > > > > I am assuming I would need to use a fakesrc with a callback to get my > data > > into the pipeline - I've tried lots of variations on this and gotten > > nowhere. > > > > Can I just replace the file-source with the fakesrc, load the data and go > > (if this is possible couple someone provide an example)? Or do I have to > > worry about queues and caps? If so, can someone provide an example? Will > > decodebin suffice or will I have to be more specific? > > Use appsrc. > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From krkrams27 at gmail.com Wed Jan 5 07:52:53 2011 From: krkrams27 at gmail.com (rams k) Date: Wed, 5 Jan 2011 12:22:53 +0530 Subject: [gst-devel] Playbin multiple sink suport 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-0.10.25/gst/ playback/ files and want to use the playbin for playing multiple videos MVC file rgds, rams -------------- next part -------------- An HTML attachment was scrubbed... URL: From sandrewartha14 at gmail.com Thu Jan 6 10:19:50 2011 From: sandrewartha14 at gmail.com (sandrewarthac76J8) Date: Thu, 6 Jan 2011 01:19:50 -0800 (PST) Subject: [gst-devel] Alsasink problem? In-Reply-To: References: Message-ID: <1294305590140-3177087.post@n4.nabble.com> I am having a similar problem, did you find an answer of solution? My sound card is working and the driver and ALSA drivrs. I can test the sound and get audio thru both speakers. However something has a hold on my sound output. Does someone know how to detect what has it locked from being accessable? How to breack that connection to free it? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Alsasink-problem-tp3166307p3177087.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From yfangjin at gmail.com Mon Jan 10 14:58:11 2011 From: yfangjin at gmail.com (golden_hz) Date: Mon, 10 Jan 2011 05:58:11 -0800 (PST) Subject: [gst-devel] Camera used by GStreamer Message-ID: <1294667891877-3207204.post@n4.nabble.com> Dear all: I am developing a test camera for meeGo device, I have research the GStreamer API and V4L2 API, it seems the we should use the GStreamer API, because the V4L2 API, we should controle the video encoder. I have some questions: 1, Can we use the QT frameowork and GStreamer API for camera, because the camera viewfinder should sue the surface, QT can provide the surface to GStreamer? I wonder. 2, Some GStreamer sampe also use the camerabin, What is the different with CameraBin and GStreamer? And what is the different between CameraBin and CameraBin2? 3, If you have some demo for this, please share it with me. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Camera-used-by-GStreamer-tp3207204p3207204.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From radivojejovanovic at gmail.com Tue Jan 11 17:19:22 2011 From: radivojejovanovic at gmail.com (Radivoje Jovanovic) Date: Tue, 11 Jan 2011 08:19:22 -0800 Subject: [gst-devel] Alsasink problem? In-Reply-To: <1294305590140-3177087.post@n4.nabble.com> References: <1294305590140-3177087.post@n4.nabble.com> Message-ID: The problem I was facing was in the unusual rate from the source song. I have defined the virtual device in my .asoundrc file and I am using this device to play all the content. It seems to me that the alsa-lib is controlling virtual device so if the content is not match to hardware specs it will resample the data to match hardware profile. Cheers, Ogi On Thu, Jan 6, 2011 at 1:19 AM, sandrewarthac76J8 wrote: > > I am having a similar problem, did you find an answer of solution? My > sound > card is working and the driver and ALSA drivrs. I can test the sound and > get audio thru both speakers. However something has a hold on my sound > output. Does someone know how to detect what has it locked from being > accessable? How to breack that connection to free it? > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/Alsasink-problem-tp3166307p3177087.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any > company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 overkill.shining at tiscali.it Tue Jan 11 17:25:15 2011 From: overkill.shining at tiscali.it (Overkill) Date: Tue, 11 Jan 2011 17:25:15 +0100 Subject: [gst-devel] decoding H.264 using an hardware codec Message-ID: > 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. You have to write a plugin that at least properly setup the hardware, load the H264 data on the HW decoder and extract the decoded stream from the HW. Then the plugin has to be insert in a pipeline. For example gst-launch filesrc location=myfile.avi ! avidemux ! my_hw_decoder_plugin ! videosink To write your plugin you must know how to use the hardware... If you're lucky you may find someone that is working on the same component. Ivan From ivan.z at tiscali.it Tue Jan 11 17:31:54 2011 From: ivan.z at tiscali.it (Ivan Zoli) Date: Tue, 11 Jan 2011 17:31:54 +0100 Subject: [gst-devel] decoding H.264 using an hardware codec In-Reply-To: References: Message-ID: > 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. You have to write a plugin that at least properly setup the hardware, load the H264 data on the HW decoder and extract the decoded stream from the HW. Then the plugin has to be insert in a pipeline. For example gst-launch filesrc location=myfile.avi ! avidemux ! my_hw_decoder_plugin ! videosink To write your plugin you must know how to use the hardware... If you're lucky you may find someone that is working on the same component. Ivan From olivier.crete at collabora.co.uk Tue Jan 11 18:10:58 2011 From: olivier.crete at collabora.co.uk (Olivier =?ISO-8859-1?Q?Cr=EAte?=) Date: Tue, 11 Jan 2011 12:10:58 -0500 Subject: [gst-devel] Single Pipelne for rtp audio and dtmf (rfc 2833) In-Reply-To: <1293312398219-3163961.post@n4.nabble.com> References: <1293312398219-3163961.post@n4.nabble.com> Message-ID: <1294765858.19057.4062.camel@TesterTop4> On Sat, 2010-12-25 at 13:26 -0800, qwerty-1 wrote: > 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? You can't, you must use gstrtpptdemux to do that, then listen for the pad-added signals and connect the depayloaders/decoders there. You may be interested in farsight2 which does that (and a lot more) for you. -- Olivier Cr?te olivier.crete at collabora.co.uk -------------- 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 bisht.sudarshan at gmail.com Tue Jan 11 19:13:32 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 11 Jan 2011 18:13:32 +0000 Subject: [gst-devel] how to encode to jpeg plz help In-Reply-To: <1292972273013-3159848.post@n4.nabble.com> References: <1292972273013-3159848.post@n4.nabble.com> Message-ID: Instead of first encoding yuv to jpeg and again decoding back it to yuv, its better to encode it just before pushing to udpsink. And this would maintain your final goal also. -queue -- > jpecgenc->udpsink src ---> caps filter -> tee -queue --- > ffmpegcolorspace ! xvimagesink gst-launch v4l2src ! 'video/x-raw-yuv,width=320,height=240,format=(fourcc)I420,framerate=(fraction)20/1' ! tee name=t ! queue ! jpegenc ! udpsink host=127.0.0.1 port=5000 t. ! queue ! ffmpegcolorspace ! xvimagesink On Wed, Dec 22, 2010 at 4:27 AM, sfaizanh wrote: > > 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. > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any > company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 bilboed at gmail.com Tue Jan 11 20:45:02 2011 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 11 Jan 2011 20:45:02 +0100 Subject: [gst-devel] second pre-release of GStreamer Editing Services 0.10.0.3 In-Reply-To: <1294671763.23475.24.camel@localhost> References: <1294671763.23475.24.camel@localhost> Message-ID: <1294775102.5430.3.camel@deumeu> Hi all, The second pre-release for the GStreamer Editing Services is available at: http://gstreamer.freedesktop.org/data/src/gst-editing-services/pre/ Documentation can be found online at: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-editing-services/html/ This second pre-release contains: * many fixes for various build systems and distros * ges-launch renamed to ges-launch-0.10 to be in sync with other gst tools naming * header cleanup * symbol cleanup * memleak fix Ubuntu users might also be able to test this pre-release by using the GStreamer PPA (as soon as they get uploaded there). Please test it thoroughly, and let us know of any important issues by filing blocker bugs in bugzilla: http://gstreamer.freedesktop.org/bugs/ (Component: gst-editing-services) Further pre-releases will follow as needed. If all goes well, 0.10.1 release should happen later this month when GStreamer core/-base 0.10.32 are out, as per http://gstreamer.freedesktop.org/wiki/ReleasePlanning2011 Edward From nathanael at gnat.ca Tue Jan 11 20:54:52 2011 From: nathanael at gnat.ca (Nathanael D. Noblet) Date: Tue, 11 Jan 2011 12:54:52 -0700 Subject: [gst-devel] dynamically remove custom branch from tee and videomixer In-Reply-To: <1293287707152-3163782.post@n4.nabble.com> References: <1293287707152-3163782.post@n4.nabble.com> Message-ID: <4D2CB58C.6080209@gnat.ca> On 12/25/2010 07:35 AM, kaija wrote: > > 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? http://gstreamer-devel.966125.n4.nabble.com/SOLVED-Dynamically-Recording-From-a-Live-Stream-amp-EOS-Handling-td3057813.html Here is what I had to do to remove a branch from a running pipeline. Not exactly for the same purpose, however my guess is that you are running into the same issue. *IF* you don't have both audio and video streams then perhaps you can do the simpler version detailed here: http://gstreamer-devel.966125.n4.nabble.com/Handling-EOS-in-a-branched-pipeline-td2965563.html I'm a newbie to this so can't profess expertise in solving this for you - good luck From giorgio9 at libero.it Tue Jan 11 20:59:23 2011 From: giorgio9 at libero.it (giorgino) Date: Tue, 11 Jan 2011 19:59:23 +0000 (UTC) Subject: [gst-devel] sending audio/video on RTP with C API References: <4D26ECBE.4020800@orange-ftgroup.com> <20110107120530.2f8713ab@mtelleria.com> <4D2C364E.6040604@orange-ftgroup.com> Message-ID: orange-ftgroup.com> writes: Dear Pierre could you give me some suggestion about your working example with h.263 altough like you I'm interested to work with h.264. G. From bisht.sudarshan at gmail.com Tue Jan 11 22:34:45 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Wed, 12 Jan 2011 03:04:45 +0530 Subject: [gst-devel] video frame type identification In-Reply-To: <1292428891718-3089291.post@n4.nabble.com> References: <1292428891718-3089291.post@n4.nabble.com> Message-ID: I think this has been already asked by you in previous mail, and you got the answer also :) > Hi Guys, > > > Is there a way to tell if an encoded frame returned by a demuxer(say > qtdemux) is an inter-Frame (iFrame) or not? > Right now, I've written a routine to do this by parsing the initial > few bits but that becomes difficult with increase in number of formats > to support. Is there a GstBuffer property I can check to see if its an > iFrame? > Check if the GST_BUFFER_FLAG_DELTA_UNIT flag is set or not. If it's not set... it's a keyframe. http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstBuffer.html#GstBufferFlag On Wed, Dec 15, 2010 at 9:31 PM, rohitratri at gmail.com wrote: > > 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. > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any > company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 bisht.sudarshan at gmail.com Tue Jan 11 23:00:03 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Wed, 12 Jan 2011 03:30:03 +0530 Subject: [gst-devel] Live Streaming In-Reply-To: <1292341654412-3087339.post@n4.nabble.com> References: <1292341654412-3087339.post@n4.nabble.com> Message-ID: Why video is not smooth is because you are not forming rtp packets of encoded data and that's why some of the packets are getting lost or not reaching in proper order while transferring data from sender to receiver. Its would be better if you use rtp packetizer and depacketizer plugins at respective side. So pipeline should be something like this, Sender: gst-launch v4l2src ! video/x-raw-yuv,width=320,height=240,framerate=\(fraction\)30/1 ! ffenc_mpeg2video ! rtpmp2tpay ! udpsink host="10.0.0.217" port=1234 Faruk Receiver: gst-launch udpsrc buffer-size=16777216 port=1234 ! rtpmp2tdepay ! ffdec_mpeg2video ! autovideosink On Tue, Dec 14, 2010 at 9:17 PM, frknml wrote: > 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: Live Streaming > Sent from the GStreamer-devel mailing list archiveat Nabble.com. > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any > company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 t.i.m at zen.co.uk Wed Jan 12 01:02:19 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 12 Jan 2011 00:02:19 +0000 Subject: [gst-devel] more pre-releases! GStreamer core, -base, -good, -ugly, -bad, -python In-Reply-To: <1294488912.3488.9.camel@zingle> References: <1294488912.3488.9.camel@zingle> Message-ID: <1294790539.27422.19.camel@zingle> On Sat, 2011-01-08 at 12:15 +0000, Tim-Philipp M?ller wrote: Hello A second round of pre-releases for GStreamer core, gst-plugins-base, gst-plugins-good, gst-plugins-ugly, gst-plugins-bad, and gst-python is now available at the following locations: 980d8f2f7027cbdee85ddbe33283583e gstreamer-0.10.31.3.tar.gz 5c04f0fbb2d092f377fdffbdd5ae64d9 gstreamer-0.10.31.3.tar.bz2 http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.31.3.tar.gz http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.31.3.tar.bz2 cba4f15911c3a67358e1cb54fb547717 gst-plugins-base-0.10.31.3.tar.gz 45c258f8316a7a5a883ace5fc4eb83e9 gst-plugins-base-0.10.31.3.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.31.3.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.31.3.tar.bz2 b154c394dd8a3175ed6a47fc7569f2f4 gst-plugins-good-0.10.26.3.tar.gz 371a130c1fa98f42ac52d8041a1b02c7 gst-plugins-good-0.10.26.3.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-good/pre/gst-plugins-good-0.10.26.3.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-good/pre/gst-plugins-good-0.10.26.3.tar.bz2 743524d647329d2f98a57d64b0347ac3 gst-plugins-ugly-0.10.16.3.tar.gz 3a6b124ea887aa6e047c3f357079fc40 gst-plugins-ugly-0.10.16.3.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-ugly/pre/gst-plugins-ugly-0.10.16.3.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-ugly/pre/gst-plugins-ugly-0.10.16.3.tar.bz2 cd04ca6bbe5b8a86d6bc12ab7e0155bb gst-plugins-bad-0.10.20.3.tar.gz 7ae5ff7468a0e3e1ee29037e4fbfb53b gst-plugins-bad-0.10.20.3.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.20.3.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.20.3.tar.bz2 b977c2f73c1dd12d81e979377183ea9d gst-python-0.10.20.3.tar.gz c72da4f1aa081d35c47eea5ca072d673 gst-python-0.10.20.3.tar.bz2 http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.20.3.tar.gz http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.20.3.tar.bz2 ----- Ubuntu users will also be able to test these pre-releases by using the GStreamer PPA where they will be available shortly. ----- Changes from last pre-releases: Core: - fix a few bash-isms - fix registry issue (don't replace valid existing plugins by blacklisted ones) - basesrc: don't hang when shutting down live sources - output-selector: improve getcaps behaviour - misc. gobject-introspection build fixes gst-plugins-base: - fix examples compilation with latest gtk+ - misc. gobject-introspection build fixes gst-plugins-good: - pulsesink: Make corking during pause synchronous gst-plugins-ugly: - detect more DRM GUIDs and error out properly on those gst-plugins-bad: - mpegtsdemux: fix re-syncing on invalid data after seek ----- PACKAGERS: note that plugins were moved in this release cycle: the selector and valve plugins from -bad were merged into corelements in core, and the jack plugin was moved to gst-plugins-good. ----- Please test them thoroughly with all your favourite apps, and let us know of any regressions or other important issues by filing blocker bugs in bugzilla: http://gstreamer.freedesktop.org/bugs/ Further pre-releases will follow as needed. If all goes well, releases should happen later this month, as per http://gstreamer.freedesktop.org/wiki/ReleasePlanning2011 A list of possible blocker bugs can be found at: http://bit.ly/h3MzgU Happy testing! Cheers -Tim From kiagiadakis.george at gmail.com Wed Jan 12 03:14:37 2011 From: kiagiadakis.george at gmail.com (George Kiagiadakis) Date: Wed, 12 Jan 2011 04:14:37 +0200 Subject: [gst-devel] pre-release QtGStreamer 0.10.0.2 available Message-ID: Hello, I am happy to announce that the first pre-release of QtGStreamer (0.10.0.2) is now available at: http://gstreamer.freedesktop.org/src/qt-gstreamer/pre/ API documentation will soon be uploaded too. For anyone interested, please check it out and report any issues at: https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=qt-gstreamer If everything goes well, the release will happen before the end of the week. Best regards, George From rohitratri at gmail.com Wed Jan 12 06:27:34 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Wed, 12 Jan 2011 10:57:34 +0530 Subject: [gst-devel] video frame type identification In-Reply-To: <1292428891718-3089291.post@n4.nabble.com> References: <1292428891718-3089291.post@n4.nabble.com> Message-ID: Yes this was an old mail. Didn't get a response so reposted :) Edward was kind enough to offer help. Sorry for spamming :) Thanks On Wed, Dec 15, 2010 at 9:31 PM, rohitratri at gmail.com wrote: > > 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. > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any > company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 dstaples at friendmts.co.uk Wed Jan 12 09:25:11 2011 From: dstaples at friendmts.co.uk (Darren Staples) Date: Wed, 12 Jan 2011 08:25:11 +0000 Subject: [gst-devel] Decoded video frame timings Message-ID: Hi, I'm decoding video frames (various file formats and codecs) and converting them into YUV420 format so that I can perform image processing on them. And I have the commercial Fluendo codec pack installed. The pipeline looks like this: uridecodebin -> queue -> ffmpegcolorspace -> tee -> queue -> appsink YUV420 conversion is done in the ffmpegcolorspace component, and the other source pad of the tee is for a future feature. This is built up dynamically in C, not on a command line. My problem is that I need to extract the frame timings for each of the video frames being processed, and I'm seeing a large variance in this. For example, one of my video files is a H.264 MP4, 82s long, 2057 frames and plays at 25 fps, so the average frame interval is 40ms. When I recover a frame from the appsink component I use the gst_element_query_position() function to get the timing for that frame. And what I'm seeing is that the timing is highly variable, even from one run to the next. e.g. the time between frames for the first 32 frames for run 1: 0: 1020226us 1: 0us 2: 0us 3: 0us 4: 0us 5: 0us 6: 0us 7: 0us 8: 0us 9: 0us 10: 0us 11: 0us 12: 0us 13: 0us 14: 0us 15: 0us 16: 0us 17: 0us 18: 0us 19: 0us 20: 0us 21: 0us 22: 0us 23: 0us 24: 0us 25: 0us 26: 19774us 27: 40000us 28: 40000us 29: 1022947us 30: 46440us 31: 23220us the first 32 frames for run 2: 0: 53151us 1: 0us 2: 26849us 3: 40000us 4: 40000us 5: 40000us 6: 40000us 7: 40000us 8: 40000us 9: 40000us 10: 40000us 11: 40000us 12: 40000us 13: 40000us 14: 40000us 15: 40000us 16: 40000us 17: 40000us 18: 40000us 19: 40000us 20: 1011428us 21: 46440us 22: 46440us 23: 23220us 24: 46440us 25: 46440us 26: 46440us 27: 23219us 28: 46440us 29: 46440us 30: 46440us 31: 23220us Why do so many of the frames in the first run above have zero time delta between them ? Is this a consequence of the file being in H.264/mp4 format ? And if, so is there an easy way to 'smooth' out the frame timings ? Or would you suggest I do this 'manually' by approximating/averaging the timing based on the duration and no. of frames ? Thanks ! Darren -------------- next part -------------- An HTML attachment was scrubbed... URL: From jesuas at gmail.com Wed Jan 12 10:19:05 2011 From: jesuas at gmail.com (Jesu Anuroop Suresh) Date: Wed, 12 Jan 2011 01:19:05 -0800 (PST) Subject: [gst-devel] audioresample Message-ID: <1294823945910-3213586.post@n4.nabble.com> Hi All, Does anyone tried the pipeline for audioresample using C. gst-launch -vvv filesrc location=01_Pepercut.mp3 ! id3demux ! mad ! audioconvert ! audioresample ! audio/x-raw-int,width=16,rate=22050,channels=1,depth=16,signed=true,endianness=1234 ! alsasink Thanks and regards Jesu Anuroop Suresh -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/audioresample-tp3213586p3213586.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From remy.perrier at endirectv.com Wed Jan 12 11:25:31 2011 From: remy.perrier at endirectv.com (=?ISO-8859-1?Q?R=E9my_PERRIER?=) Date: Wed, 12 Jan 2011 11:25:31 +0100 Subject: [gst-devel] Plungin GL installation with Mingw In-Reply-To: <4D2B2F4E.7000008@endirectv.com> References: <4D2AF45F.1080201@endirectv.com> <4D2AF6DB.8090306@endirectv.com> <4D2B171C.7060505@endirectv.com> <4D2B2F4E.7000008@endirectv.com> Message-ID: <4D2D819B.7030404@endirectv.com> Hi, In the plugin gl lib documentation, they talk about using cmake to build libgstopen.dl for mingw32. However there is no cmakelist.txt in gst-plugin gl repertory... How could I generate this file ? Thanks, Le 10/01/2011 17:09, R?my PERRIER a ?crit : > Hi, > > It s me again. > I have just realized that there is no dll object with libgstopengl and > libgstgl in /usr/local/lib and /usr/local/lib/gstreamer-0.10 ( > contrary to the other lib ). > It may be the cause of my problem. How can I generate those missing ( > I guess) files ? > Thanks for your help. > > regards, > > > Le 10/01/2011 15:26, R?my PERRIER a ?crit : >> ok my mingw installation was a mess, I have cleaned up autoconf & m4. >> I have installed autoconf 2.68 and m4 1.4.14 package thanks to >> mingw-get , run ./autogen.sh and installation is ok >> >> However gl plugins dont appear in the gst-inspect command. >> They are not black listed and of course I cannot run the GL examples. >> >> The "make check" in the gst-plugin gl repertory works good >> >> Where does it may come from ? >> >> >> kind regards, >> >> remy >> >> >> Le 10/01/2011 13:08, R?my PERRIER a ?crit : >>> I forget to display the plugin GL compilation result : >>> >>> >>> bob at bob /c/gst-plugins-gl-0.10.2 >>> $ make >>> cat: ./win32/MANIFEST: No such file or directory >>> make all-recursive >>> cat: ./win32/MANIFEST: No such file or directory >>> make[1]: Entering directory `/c/gst-plugins-gl-0.10.2' >>> Making all in gst-libs >>> make[2]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs' >>> Making all in gst >>> make[3]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' >>> Making all in gl >>> make[4]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' >>> CC libgstgl_0.10_la-gstgldisplay.lo >>> libtool: Version mismatch error. This is libtool 2.4, but the >>> libtool: definition of this LT_INIT comes from libtool 2.2.6b. >>> libtool: You should recreate aclocal.m4 with macros from libtool 2.4 >>> libtool: and run autoconf again. >>> make[4]: *** [libgstgl_0.10_la-gstgldisplay.lo] Error 63 >>> make[4]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' >>> make[3]: *** [all-recursive] Error 1 >>> make[3]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' >>> make[2]: *** [all-recursive] Error 1 >>> make[2]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs' >>> make[1]: *** [all-recursive] Error 1 >>> make[1]: Leaving directory `/c/gst-plugins-gl-0.10.2' >>> make: *** [all] Error 2 >>> >>> >>> Le 10/01/2011 12:58, R?my PERRIER a ?crit : >>>> Hi all, >>>> >>>> >>>> I am trying to install Gstreamer GL plugin and the mingw-get >>>> autoconf package doesnt seem to match. >>>> I need to compile & install autoconf 2.6.7 but I have to face some >>>> difficulities. >>>> Here is what I get : >>>> >>>> >>>> bob at bob /c/autoconf-2.67 >>>> $ make >>>> >>>> .... >>>> >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> autom4te: >>>> make[3]: *** [m4sugar.m4f] Error 1 >>>> make[3]: Leaving directory `/c/autoconf-2.67/lib/m4sugar' >>>> make[2]: *** [all-recursive] Error 1 >>>> make[2]: Leaving directory `/c/autoconf-2.67/lib' >>>> make[1]: *** [all-recursive] Error 1 >>>> make[1]: Leaving directory `/c/autoconf-2.67' >>>> make: *** [all] Error 2 >>>> >>>> >>>> My configuration : >>>> >>>> gst plugin GL : 0.10.2 >>>> gstreamer : 0.10. 31 >>>> >>>> I am using migw -get 0.1 alpha 5. >>>> I have tried different version of m4 : 1.4.15 , 1.4.4 now I am >>>> using 1.4.9 >>>> perl : v5.6.1 >>>> gcc : 4.5.0 >>>> >>>> Is somebody already have this problem ? >>>> >>>> What do you think about that ? >>>> >>>> >>>> >>>> regards, >>>> >>>> >>>> remy >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Gaining the trust of online customers is vital for the success of any company >>>> that requires sensitive data to be transmitted over the Web. Learn how to >>>> best implement a security strategy that keeps consumers' information secure >>>> and instills the confidence they need to proceed with transactions. >>>> 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 >>> >>> >>> ------------------------------------------------------------------------------ >>> Gaining the trust of online customers is vital for the success of any company >>> that requires sensitive data to be transmitted over the Web. Learn how to >>> best implement a security strategy that keeps consumers' information secure >>> and instills the confidence they need to proceed with transactions. >>> 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 >> >> >> ------------------------------------------------------------------------------ >> Gaining the trust of online customers is vital for the success of any company >> that requires sensitive data to be transmitted over the Web. Learn how to >> best implement a security strategy that keeps consumers' information secure >> and instills the confidence they need to proceed with transactions. >> 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 > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 smcnam at gmail.com Wed Jan 12 11:59:53 2011 From: smcnam at gmail.com (Sean McNamara) Date: Wed, 12 Jan 2011 05:59:53 -0500 Subject: [gst-devel] audioresample In-Reply-To: <1294823945910-3213586.post@n4.nabble.com> References: <1294823945910-3213586.post@n4.nabble.com> Message-ID: Hi, I don't entirely understand your question. Are you trying to convert a pipeline into an equivalent C application? Doing so is just based on understanding of C and the gstreamer C API... it is certainly possible, insofar as your pipeline is valid. That said, look at gst_parse_* functions for a way to enter a pipeline spec in the gst-launch syntax and get either a GstBin or a GstPipeline out of it. Sean On Jan 12, 2011 4:20 AM, "Jesu Anuroop Suresh" wrote: > > Hi All, > > Does anyone tried the pipeline for audioresample using C. > > gst-launch -vvv filesrc location=01_Pepercut.mp3 ! id3demux ! mad ! > audioconvert ! audioresample ! > audio/x-raw-int,width=16,rate=22050,channels=1,depth=16,signed=true,endianness=1234 > ! alsasink > > Thanks and regards > Jesu Anuroop Suresh > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/audioresample-tp3213586p3213586.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 julien.isorce at gmail.com Wed Jan 12 12:07:19 2011 From: julien.isorce at gmail.com (Julien Isorce) Date: Wed, 12 Jan 2011 12:07:19 +0100 Subject: [gst-devel] Plungin GL installation with Mingw In-Reply-To: <4D2D819B.7030404@endirectv.com> References: <4D2AF45F.1080201@endirectv.com> <4D2AF6DB.8090306@endirectv.com> <4D2B171C.7060505@endirectv.com> <4D2B2F4E.7000008@endirectv.com> <4D2D819B.7030404@endirectv.com> Message-ID: 2011/1/12 R?my PERRIER > Hi, > > In the plugin gl lib documentation, they talk about using cmake to build > libgstopen.dl for mingw32. > However there is no cmakelist.txt in gst-plugin gl repertory... > How could I generate this file ? > Hi, http://cgit.freedesktop.org/gstreamer/gst-plugins-gl/tree/CMakeLists.txt Sincerely Julien > > Thanks, > > > > > Le 10/01/2011 17:09, R?my PERRIER a ?crit : > > Hi, > > It s me again. > I have just realized that there is no dll object with libgstopengl and > libgstgl in /usr/local/lib and /usr/local/lib/gstreamer-0.10 ( contrary to > the other lib ). > It may be the cause of my problem. How can I generate those missing ( I > guess) files ? > Thanks for your help. > > regards, > > > Le 10/01/2011 15:26, R?my PERRIER a ?crit : > > ok my mingw installation was a mess, I have cleaned up autoconf & m4. > I have installed autoconf 2.68 and m4 1.4.14 package thanks to mingw-get , > run ./autogen.sh and installation is ok > > However gl plugins dont appear in the gst-inspect command. > They are not black listed and of course I cannot run the GL examples. > > The "make check" in the gst-plugin gl repertory works good > > Where does it may come from ? > > > kind regards, > > remy > > > Le 10/01/2011 13:08, R?my PERRIER a ?crit : > > I forget to display the plugin GL compilation result : > > > bob at bob /c/gst-plugins-gl-0.10.2 > $ make > cat: ./win32/MANIFEST: No such file or directory > make all-recursive > cat: ./win32/MANIFEST: No such file or directory > make[1]: Entering directory `/c/gst-plugins-gl-0.10.2' > Making all in gst-libs > make[2]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs' > Making all in gst > make[3]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' > Making all in gl > make[4]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' > CC libgstgl_0.10_la-gstgldisplay.lo > libtool: Version mismatch error. This is libtool 2.4, but the > libtool: definition of this LT_INIT comes from libtool 2.2.6b. > libtool: You should recreate aclocal.m4 with macros from libtool 2.4 > libtool: and run autoconf again. > make[4]: *** [libgstgl_0.10_la-gstgldisplay.lo] Error 63 > make[4]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory `/c/gst-plugins-gl-0.10.2' > make: *** [all] Error 2 > > > Le 10/01/2011 12:58, R?my PERRIER a ?crit : > > Hi all, > > > I am trying to install Gstreamer GL plugin and the mingw-get autoconf > package doesnt seem to match. > I need to compile & install autoconf 2.6.7 but I have to face some > difficulities. > Here is what I get : > > > bob at bob /c/autoconf-2.67 > $ make > > .... > > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > autom4te: > make[3]: *** [m4sugar.m4f] Error 1 > make[3]: Leaving directory `/c/autoconf-2.67/lib/m4sugar' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/c/autoconf-2.67/lib' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory `/c/autoconf-2.67' > make: *** [all] Error 2 > > > My configuration : > > gst plugin GL : 0.10.2 > gstreamer : 0.10. 31 > > I am using migw -get 0.1 alpha 5. > I have tried different version of m4 : 1.4.15 , 1.4.4 now I am using 1.4.9 > perl : v5.6.1 > gcc : 4.5.0 > > Is somebody already have this problem ? > > What do you think about that ? > > > > regards, > > > remy > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions.http://p.sf.net/sfu/oracle-sfdevnl > > > _______________________________________________ > gstreamer-devel mailing listgstreamer-devel at lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions.http://p.sf.net/sfu/oracle-sfdevnl > > > _______________________________________________ > gstreamer-devel mailing listgstreamer-devel at lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions.http://p.sf.net/sfu/oracle-sfdevnl > > > _______________________________________________ > gstreamer-devel mailing listgstreamer-devel at lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions.http://p.sf.net/sfu/oracle-sfdevnl > > > _______________________________________________ > gstreamer-devel mailing listgstreamer-devel at lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 sss0420 at gmail.com Wed Jan 12 01:45:43 2011 From: sss0420 at gmail.com (yanping) Date: Tue, 11 Jan 2011 16:45:43 -0800 (PST) Subject: [gst-devel] i want to convert to MPEG2 TS from RTSPsrc. Message-ID: <1294793143896-3209967.post@n4.nabble.com> Hi,all: I want to convert to MPEG2 TS and save to *.ts file. Source is element stream of RTSP/RTP. Video is H.264 and Audio is MPEG4 general. this is my command: $ gst-launch-0.10 rtspsrc location=rtsp://....mp4 ! rtph264depay ! mpegtsmux ! filesink location=cap.ts This command was able to convert to MPEG2 TS. Of course audio data didn't contain MPEG2 TS. This composition is my image. rtph264depay rtsp -< >--- mpegtsmux ---- filesink---> *.ts rtpmp4gdepay So I want to contain audio and video to MPEG2 TS. What i have to do?..... -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/i-want-to-convert-to-MPEG2-TS-from-RTSPsrc-tp3209967p3209967.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From remy.perrier at endirectv.com Wed Jan 12 12:53:23 2011 From: remy.perrier at endirectv.com (=?ISO-8859-1?Q?R=E9my_PERRIER?=) Date: Wed, 12 Jan 2011 12:53:23 +0100 Subject: [gst-devel] Plungin GL installation with Mingw In-Reply-To: References: <4D2AF45F.1080201@endirectv.com> <4D2AF6DB.8090306@endirectv.com> <4D2B171C.7060505@endirectv.com> <4D2B2F4E.7000008@endirectv.com> <4D2D819B.7030404@endirectv.com> Message-ID: <4D2D9633.2060008@endirectv.com> Thank you. Le 12/01/2011 12:07, Julien Isorce a ?crit : > > > 2011/1/12 R?my PERRIER > > > Hi, > > In the plugin gl lib documentation, they talk about using cmake to > build libgstopen.dl for mingw32. > However there is no cmakelist.txt in gst-plugin gl repertory... > How could I generate this file ? > > > Hi, > > http://cgit.freedesktop.org/gstreamer/gst-plugins-gl/tree/CMakeLists.txt > > Sincerely > Julien > > > Thanks, > > > > > Le 10/01/2011 17:09, R?my PERRIER a ?crit : >> Hi, >> >> It s me again. >> I have just realized that there is no dll object with >> libgstopengl and libgstgl in /usr/local/lib and >> /usr/local/lib/gstreamer-0.10 ( contrary to the other lib ). >> It may be the cause of my problem. How can I generate those >> missing ( I guess) files ? >> Thanks for your help. >> >> regards, >> >> >> Le 10/01/2011 15:26, R?my PERRIER a ?crit : >>> ok my mingw installation was a mess, I have cleaned up autoconf >>> & m4. >>> I have installed autoconf 2.68 and m4 1.4.14 package thanks to >>> mingw-get , run ./autogen.sh and installation is ok >>> >>> However gl plugins dont appear in the gst-inspect command. >>> They are not black listed and of course I cannot run the GL >>> examples. >>> >>> The "make check" in the gst-plugin gl repertory works good >>> >>> Where does it may come from ? >>> >>> >>> kind regards, >>> >>> remy >>> >>> >>> Le 10/01/2011 13:08, R?my PERRIER a ?crit : >>>> I forget to display the plugin GL compilation result : >>>> >>>> >>>> bob at bob /c/gst-plugins-gl-0.10.2 >>>> $ make >>>> cat: ./win32/MANIFEST: No such file or directory >>>> make all-recursive >>>> cat: ./win32/MANIFEST: No such file or directory >>>> make[1]: Entering directory `/c/gst-plugins-gl-0.10.2' >>>> Making all in gst-libs >>>> make[2]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs' >>>> Making all in gst >>>> make[3]: Entering directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' >>>> Making all in gl >>>> make[4]: Entering directory >>>> `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' >>>> CC libgstgl_0.10_la-gstgldisplay.lo >>>> libtool: Version mismatch error. This is libtool 2.4, but the >>>> libtool: definition of this LT_INIT comes from libtool 2.2.6b. >>>> libtool: You should recreate aclocal.m4 with macros from >>>> libtool 2.4 >>>> libtool: and run autoconf again. >>>> make[4]: *** [libgstgl_0.10_la-gstgldisplay.lo] Error 63 >>>> make[4]: Leaving directory >>>> `/c/gst-plugins-gl-0.10.2/gst-libs/gst/gl' >>>> make[3]: *** [all-recursive] Error 1 >>>> make[3]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs/gst' >>>> make[2]: *** [all-recursive] Error 1 >>>> make[2]: Leaving directory `/c/gst-plugins-gl-0.10.2/gst-libs' >>>> make[1]: *** [all-recursive] Error 1 >>>> make[1]: Leaving directory `/c/gst-plugins-gl-0.10.2' >>>> make: *** [all] Error 2 >>>> >>>> >>>> Le 10/01/2011 12:58, R?my PERRIER a ?crit : >>>>> Hi all, >>>>> >>>>> >>>>> I am trying to install Gstreamer GL plugin and the mingw-get >>>>> autoconf package doesnt seem to match. >>>>> I need to compile & install autoconf 2.6.7 but I have to face >>>>> some difficulities. >>>>> Here is what I get : >>>>> >>>>> >>>>> bob at bob /c/autoconf-2.67 >>>>> $ make >>>>> >>>>> .... >>>>> >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> autom4te: >>>>> make[3]: *** [m4sugar.m4f] Error 1 >>>>> make[3]: Leaving directory `/c/autoconf-2.67/lib/m4sugar' >>>>> make[2]: *** [all-recursive] Error 1 >>>>> make[2]: Leaving directory `/c/autoconf-2.67/lib' >>>>> make[1]: *** [all-recursive] Error 1 >>>>> make[1]: Leaving directory `/c/autoconf-2.67' >>>>> make: *** [all] Error 2 >>>>> >>>>> >>>>> My configuration : >>>>> >>>>> gst plugin GL : 0.10.2 >>>>> gstreamer : 0.10. 31 >>>>> >>>>> I am using migw -get 0.1 alpha 5. >>>>> I have tried different version of m4 : 1.4.15 , 1.4.4 now I >>>>> am using 1.4.9 >>>>> perl : v5.6.1 >>>>> gcc : 4.5.0 >>>>> >>>>> Is somebody already have this problem ? >>>>> >>>>> What do you think about that ? >>>>> >>>>> >>>>> >>>>> regards, >>>>> >>>>> >>>>> remy >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Gaining the trust of online customers is vital for the success of any company >>>>> that requires sensitive data to be transmitted over the Web. Learn how to >>>>> best implement a security strategy that keeps consumers' information secure >>>>> and instills the confidence they need to proceed with transactions. >>>>> 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 >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Gaining the trust of online customers is vital for the success of any company >>>> that requires sensitive data to be transmitted over the Web. Learn how to >>>> best implement a security strategy that keeps consumers' information secure >>>> and instills the confidence they need to proceed with transactions. >>>> 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 >>> >>> >>> ------------------------------------------------------------------------------ >>> Gaining the trust of online customers is vital for the success of any company >>> that requires sensitive data to be transmitted over the Web. Learn how to >>> best implement a security strategy that keeps consumers' information secure >>> and instills the confidence they need to proceed with transactions. >>> 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 >> >> >> ------------------------------------------------------------------------------ >> Gaining the trust of online customers is vital for the success of any company >> that requires sensitive data to be transmitted over the Web. Learn how to >> best implement a security strategy that keeps consumers' information secure >> and instills the confidence they need to proceed with transactions. >> 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 > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and > how you > can protect your company and customers by using code signing. > 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 > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 jesuas at gmail.com Wed Jan 12 13:41:21 2011 From: jesuas at gmail.com (Jesu Anuroop Suresh) Date: Wed, 12 Jan 2011 04:41:21 -0800 (PST) Subject: [gst-devel] audioresample In-Reply-To: References: <1294823945910-3213586.post@n4.nabble.com> Message-ID: Hi Sean, Yes, what I was trying is to resample the decoded mp3 data to the fixed (22KHZ S16LE) formate, no matter what is the input rate using a C application. Thanks for your response. Here is the piece of the code for the same but it does not work with audioresample with the caps filter 'resmux'. This code does work without the caps filter 'resmux'. GstElement *source, *demuxer, *decoder, *conv, *sink, *resample, *resmux; GstCaps *caps; gst_init(NULL, NULL); /* Create gstreamer elements */ musicPlayer.playPipeline = gst_pipeline_new ("audio-player"); source = gst_element_factory_make ("filesrc", "file-source"); sink = gst_element_factory_make ("alsasink", "audio-output"); resample = gst_element_factory_make ("audioresample", "audio-resample"); conv = gst_element_factory_make ("audioconvert", "converter1"); caps = gst_caps_new_simple ("audio/x-raw-int", "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, "rate", G_TYPE_INT, 22050, "channels",G_TYPE_INT, 2, NULL ); if (!musicPlayer.playPipeline || !source || !sink || !resample || !resmux || !caps || !conv) { g_print ("NO MEM Exiting.\n"); return 1; } /* we set the input filename to the source element */ g_object_set (G_OBJECT (source), "location", filePath, NULL); demuxer = gst_element_factory_make ("id3demux", "id3-demuxer"); decoder = gst_element_factory_make ("mad", "mp3-decoder"); if (!demuxer || !decoder || !conv1) { g_print ("NO MEM Exiting.\n"); return 1; } g_object_set (G_OBJECT (resmux), "caps", caps, NULL); gst_caps_unref (caps); /* file-source -> demuxer -> decoder -> alsa-output */ gst_bin_add_many (GST_BIN (musicPlayer.playPipeline), source, demuxer, decoder, conv, resample, resmux,sink, NULL); gst_element_link (source, demuxer); gst_element_link_many (decoder, conv, resample,resmux,sink, NULL); g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), decoder); GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(musicPlayer.playPipeline)); gst_bus_add_watch(bus, bus_call, NULL); gst_object_unref(bus); gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), GST_STATE_PLAYING); musicPlayer.playLoop = g_main_loop_new(NULL, FALSE); g_main_loop_run(musicPlayer.playLoop); gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), GST_STATE_NULL); gst_object_unref(GST_OBJECT(musicPlayer.playPipeline)); With Warm Regards Jesu Anuroop Suresh "Any intelligent fool can make things bigger, more complex, and more violent. It takes a touch of genius -- and a lot of courage -- to move in the opposite direction." "Anyone who has never made a mistake has never tried anything new." On Wed, Jan 12, 2011 at 4:31 PM, Sean McNamara-4 [via GStreamer-devel] < ml-node+3213735-1942695576-203210 at n4.nabble.com > wrote: > Hi, > > I don't entirely understand your question. Are you trying to convert a > pipeline into an equivalent C application? Doing so is just based on > understanding of C and the gstreamer C API... it is certainly possible, > insofar as your pipeline is valid. > > That said, look at gst_parse_* functions for a way to enter a pipeline spec > in the gst-launch syntax and get either a GstBin or a GstPipeline out of it. > > Sean > On Jan 12, 2011 4:20 AM, "Jesu Anuroop Suresh" <[hidden email]> > wrote: > > > > Hi All, > > > > Does anyone tried the pipeline for audioresample using C. > > > > gst-launch -vvv filesrc location=01_Pepercut.mp3 ! id3demux ! mad ! > > audioconvert ! audioresample ! > > > audio/x-raw-int,width=16,rate=22050,channels=1,depth=16,signed=true,endianness=1234 > > ! alsasink > > > > Thanks and regards > > Jesu Anuroop Suresh > > > > -- > > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/audioresample-tp3213586p3213586.html > > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > > > ------------------------------------------------------------------------------ > > Protect Your Site and Customers from Malware Attacks > > Learn about various malware tactics and how to avoid them. Understand > > malware threats, the impact they can have on your business, and how you > > can protect your company and customers by using code signing. > > http://p.sf.net/sfu/oracle-sfdevnl > > _______________________________________________ > > gstreamer-devel mailing list > > [hidden email] > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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/audioresample-tp3213586p3213735.html > To unsubscribe from audioresample, click here. > > -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/audioresample-tp3213586p3213856.html Sent from the GStreamer-devel mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: From emmanuel at gnome.org Wed Jan 12 13:30:58 2011 From: emmanuel at gnome.org (Emmanuel Pacaud) Date: Wed, 12 Jan 2011 13:30:58 +0100 Subject: [gst-devel] Release of Aravis 0.1.3 Message-ID: <1294835458.4279.159.camel@lappc-p348> Hi, I've just released a new unstable version of Aravis, a LGPLv2+ gobject based library for the acquisition of video streams from genicam based digital camera. http://ftp.gnome.org/pub/GNOME/sources/aravis/0.1/ http://blogs.gnome.org/emmanuel/category/aravis/ It includes: * Support for gigabit ethernet cameras * Support for a large subset of the Genicam interface * A simple API for easy camera control * A simple gstreamer source element * A simple video viewer * Gobject-introspection support * An ethernet camera simulator * A work-in-progress documentation The changes since the last release are: * Simple viewer based on ArvCamera API (requires gtk and gstreamer) * Add a "new-buffer" signal to ArvStream * Fix stream IP address setting for JAI Pulnix cameras (Tom Cobb) * Fix use or Aravis from a C++ application (Tom Cobb) * Fix division of integers in ArvEvaluator when asked for float result (Tom Cobb) * Add an API for retrieving the genicam data (Tom Cobb) * Fix minimum offset of ROI (Tom Cobb) * Fake camera can now simulate gain and exposure setting Wiki: http://live.gnome.org/Aravis Bug reports: https://bugzilla.gnome.org/browse.cgi?product=aravis Blog: http://blogs.gnome.org/emmanuel/category/aravis/ For the next release I will try to find a place for a mailing list dedicated to aravis. Emmanuel. From katcipis at inf.ufsc.br Wed Jan 12 14:05:54 2011 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Wed, 12 Jan 2011 11:05:54 -0200 Subject: [gst-devel] sending audio/video on RTP with C API In-Reply-To: References: <4D26ECBE.4020800@orange-ftgroup.com> <20110107120530.2f8713ab@mtelleria.com> <4D2C364E.6040604@orange-ftgroup.com> Message-ID: On Tue, Jan 11, 2011 at 5:59 PM, giorgino wrote: > orange-ftgroup.com> writes: > > Dear Pierre could you give me some suggestion about your working example > with > h.263 altough like you I'm interested to work with h.264. > I have this small application that sends audio(G711)/video(H263): http://gitorious.org/ufscwork/works/blobs/master/c/gstreamer/video_conf_rtp/simple.c i tested it making a video conference linux->linux and it worked ok. The only thing missing is a gstrtpbin, there is no jitter buffer etc. hope it helps. best regards, Katcipis > > G. > > > > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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://www.getgnulinux.org/windows -------------- next part -------------- An HTML attachment was scrubbed... URL: From marete at toshnix.com Wed Jan 12 15:09:34 2011 From: marete at toshnix.com (Brian Gitonga Marete) Date: Wed, 12 Jan 2011 17:09:34 +0300 Subject: [gst-devel] sending audio/video on RTP with C API In-Reply-To: <1294397198.23475.13.camel@localhost> References: <4D26ECBE.4020800@orange-ftgroup.com> <1294397198.23475.13.camel@localhost> Message-ID: On Fri, Jan 7, 2011 at 1:46 PM, Edward Hervey wrote: > On Fri, 2011-01-07 at 11:36 +0100, pierre.crepieux at orange-ftgroup.com > wrote: >> Hi, >> >> I'm trying to write a simple C program that sends webcam/microphone >> streams on rtp with gstrtpbin. >> I gave a look at the provided server-alsasrc-PCMA.c test program and >> mimics it to add h264 stream to the rtpbin (just like >> server-v4l2-H264-alsasrc-PCMA.sh does). >> Sadly, using VLC to receive the streams, give me a fixed image and a >> pretty bad sound (delay, sometime stop/restart,...). >> Are there any tricks to be aware of for doing this translation of >> gst-launch pipe definition to C code ? Does gstrtpbin require subtle >> manipulations (that gst-launch would do behind the scene ... ??? ) to >> operate with 2 rtp stream ? > > ?You might want to use gst-rtsp-server instead. It does all the magic > for you. > By this do you mean the transfer of SDP data or something more? Thanks. BGM. -- Brian Gitonga Marete Toshnix Systems Tel: +254722151590 From ensonic at hora-obscura.de Wed Jan 12 16:24:50 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 12 Jan 2011 17:24:50 +0200 Subject: [gst-devel] kenburns plugin In-Reply-To: <4D2B7145.7010503@brooks.nu> References: <4D2B7145.7010503@brooks.nu> Message-ID: <4D2DC7C2.1000807@hora-obscura.de> On 10.01.2011 22:51, Lane Brooks wrote: > I have written a kenburns plugin > (http://en.wikipedia.org/wiki/Ken_burns_effect) that does smooth > zooming/panning simultaneously. > > I have it committed in a git repo at https://github.com/dirjud/kenburns > > There are some gst-launch samples in the src/gstkenburns.c file. Such a plugin would be welcome. Make a patch against gst-plugins-bad and put it to bugzilla. Add yourself to the copyright comment. > Currently I have only implemented nearest neighbor sampling, so it is > fast but can show aliasing artifacts. Yes, n-tap filtering would be nice to have, but thats no show-stopper for getting it into bad. Do you know about GstController? I would prefer to only have "zoom", "xcenter", "ycenter" properties, make then controllable and leave the animations to GstController. With that you get envelops and LFOs right now. Stefan > Is this something that the developers would be interested in seeing > integrated into gstreamer? > > Any feedback would be appreciated. > > Thanks, > Lane > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 ensonic at hora-obscura.de Wed Jan 12 16:28:34 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 12 Jan 2011 17:28:34 +0200 Subject: [gst-devel] playbin multiple sink In-Reply-To: References: Message-ID: <4D2DC8A2.2060308@hora-obscura.de> On 30.12.2010 16:43, rk wrote: > Hi, > Im trying to play MVC file through PLAYBIN with the mvc plugin. 3d video is not supported offically now. We need now caps for that. Streaming them as independent streams is not trivial modification in playbin2. You should leave playbin (1) untouched it is legacy. Stefan > > 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: playbin multiple sink > > Sent from the GStreamer-devel mailing list archive > at Nabble.com. > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 dstaples at friendmts.co.uk Wed Jan 12 16:50:38 2011 From: dstaples at friendmts.co.uk (Darren Staples) Date: Wed, 12 Jan 2011 15:50:38 +0000 Subject: [gst-devel] Video frame timings (revisited) Message-ID: Hi, I'm hoping someone can help with this please, as I'm struggling... When doing audio and video decoding of a H.264 MP4 file I find that the timing of the video frames seems to be affected by the processing of the audio frames. My pipeline looks like this: ?? ? ? ? ? ? ? ? ?? / ffmpegcolorspace - queue - appsink uridecodebin \ audioconvert - queue - appsink I have setup callbacks on both the audio and video appsinks. In the video callback I get the buffer from the appsink and then call gst_element_query_position() to get the frame time. In the audio callback I'm just getting the buffer and discarding it (for now). The frame times I see in the video callback seem to be affected by the audio decoding i.e. I don't see the regular frame times that I expected. They're quite variable and there seem to be extended periods (1-3s) between some frames. If I remove the audio branch of the pipeline I see perfectly regular (exactly 40ms or 25fps) video frame times. Am I missing some vital component from my piepline ? Can anyone suggest, please, where I might be going wrong ? Thanks in advance, Darren From msmith at xiph.org Wed Jan 12 17:40:14 2011 From: msmith at xiph.org (Michael Smith) Date: Wed, 12 Jan 2011 08:40:14 -0800 Subject: [gst-devel] Video frame timings (revisited) In-Reply-To: References: Message-ID: On Wed, Jan 12, 2011 at 7:50 AM, Darren Staples wrote: > Hi, I'm hoping someone can help with this please, as I'm struggling... > > When doing audio and video decoding of a H.264 MP4 file I find that > the timing of the video frames seems to be affected by the processing > of the audio frames. > My pipeline looks like this: > > ?? ? ? ? ? ? ? ? ?? / ffmpegcolorspace - queue - appsink > uridecodebin > ? ? ? ? ? ? ? ? ? ?\ audioconvert - queue - appsink > > I have setup callbacks on both the audio and video appsinks. > In the video callback I get the buffer from the appsink and then call > gst_element_query_position() to get the frame time. > In the audio callback I'm just getting the buffer and discarding it (for now). Don't use gst_element_query_position, then. It's not doing what you want. Instead, use the buffer timestamps directly (GST_BUFFER_TIMESTAMP(buffer)). Mike From andre.dieb at gmail.com Wed Jan 12 18:07:16 2011 From: andre.dieb at gmail.com (=?UTF-8?Q?Andr=C3=A9_Dieb?=) Date: Wed, 12 Jan 2011 15:07:16 -0200 Subject: [gst-devel] How can I add RTSP media mapping to the gst-rtsp-server in Python? In-Reply-To: References: <1284454972893-2538670.post@n4.nabble.com> Message-ID: Hello, The patch was commited (thanks devs!) today, so, if you're still interested, you'll find the methods and sample code on test.py, included on the patch at https://bugzilla.gnome.org/show_bug.cgi?id=622296. On Tue, Sep 14, 2010 at 8:19 AM, Andr? Dieb wrote: > Hello, > > Currently gst-rtsp-server (master) can't add media mappings. I'm currently > in the process of cleaning up my python-gst-rtsp-server patches, which > include MediaMapping and MediaFactory bindings. It's likely to be submitted > to bugzilla this week (at least an initial version). If you're interested, I > can send you the dirty patches :) passing a launch string to the factory > already works. > > On Tue, Sep 14, 2010 at 6:02 AM, wl2776 wrote: > >> >> I've created a server instance, but cannot add media-mapping to it. >> Are there any examples of using gst-rtsp-server in Python? >> >> import gobject >> gobject.threads_init() >> >> import sys >> import pygst >> pygst.require('0.10') >> >> import os >> >> from gst import rtspserver >> >> class myrtspserver: >> def __init__(self): >> self.server=rtspserver.Server() >> self.server.props.media_mapping.set_property('/test', >> '( videotestsrc is-live=1 ! x264enc ! >> rtph264pay name=pay0 pt=96 )') >> >> >> The last line of the code above gives the error >> TypeError: object of type `GstRTSPMediaMapping' does not have property >> `/test' >> >> If I replace set_property with set_data, it doesn't give errors, but logs >> show that the server cannot find the media mapping when processing >> connection. >> Tried also set_data >> -- >> View this message in context: >> http://gstreamer-devel.966125.n4.nabble.com/How-can-I-add-RTSP-media-mapping-to-the-gst-rtsp-server-in-Python-tp2538670p2538670.html >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> >> >> ------------------------------------------------------------------------------ >> Start uncovering the many advantages of virtual appliances >> and start using them to simplify application deployment and >> accelerate your shift to cloud computing. >> http://p.sf.net/sfu/novell-sfdev2dev >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > Andr? Dieb Martins > -- Andr? Dieb Martins -------------- next part -------------- An HTML attachment was scrubbed... URL: From felipe.contreras at gmail.com Wed Jan 12 21:53:06 2011 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Wed, 12 Jan 2011 22:53:06 +0200 Subject: [gst-devel] Problem in using ffdec_mp3 to decode live streaming In-Reply-To: <1294668619597-3207224.post@n4.nabble.com> References: <1294314501815-3177271.post@n4.nabble.com> <1294668619597-3207224.post@n4.nabble.com> Message-ID: On Mon, Jan 10, 2011 at 4:10 PM, Puneeth wrote: > The above command works fine, but when i replace with ffdec_mp3 there is lot > of noice, how could i solve this or is there any other ?LGPL decoders which > would do the needful. Please help. Maybe you could try gst-av[1], it can use FFmpeg (libavcodec) just like gst-ffmpeg, but it's much simpler. It's definitely not production-ready, but being only 455 lines of code I think you might be able to find the issues ;) [1] https://github.com/felipec/gst-av -- Felipe Contreras From tbrendt at googlemail.com Wed Jan 12 22:21:46 2011 From: tbrendt at googlemail.com (Thorsten Brendt) Date: Wed, 12 Jan 2011 22:21:46 +0100 Subject: [gst-devel] RTP audio stream to filesink, issue with packet loss Message-ID: Hi! I'm using Gstreamer to save RTP audio streams to wav files. This works in general, however, I need to deal with packet loss and in this case the pipeline I'm currently using fails: filesrc location=foobar.pcap ! pcapparse ! "application/x-rtp, payload=0, clock-rate=8000" ! gstrtpjitterbuffer ! rtppcmudepay ! mulawdec ! audioconvert ! audioresample ! wavenc ! filesink location=foobar.wav This produces audio files that are shorter than the original RTP stream, e.g. for a certain amount of consecutively lost packets a second of audio is missing from the wav file. For a real audio sink this doesn't happen as it plays audio when something is available, the filesink however -- I assume -- writes data sequentially and only when data is actually available. By the way: the effect is the same for udpsrc as it is for filesrc+pcapparse. I tried the above pipeline with filesink's sync property set to true ([1]), this didn't make any difference. I read somewhere in the Gstreamer docs that the global clock is [usually] dependent on the sink element. Do I need to reverse this and make the sink's clock dependent on the source's clock? 1. < http://www.gstreamer.net/data/doc/gstreamer/head/gstreamer-libs/html/GstBaseSink.html#gst-base-sink-set-sync > Maybe I'm totally off in this regard and there is no implicit way to achieve what I need. Ultimately some element in the pipeline needs to produce "silence/empty/zero data" if its input buffer is empty. Is there an explicit way (helper element/attribute) to do this? Thorsten -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Thu Jan 13 00:16:37 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 12 Jan 2011 23:16:37 +0000 Subject: [gst-devel] RTP audio stream to filesink, issue with packet loss In-Reply-To: References: Message-ID: <1294874197.25646.0.camel@zingle> On Wed, 2011-01-12 at 22:21 +0100, Thorsten Brendt wrote: Hi, > (snip) > > Ultimately some element in the pipeline needs to produce > "silence/empty/zero data" if its input buffer is empty. Is there an > explicit way (helper element/attribute) to do this? Tried adding an audiorate element? Cheers -Tim From lane at brooks.nu Thu Jan 13 00:48:45 2011 From: lane at brooks.nu (Lane Brooks) Date: Wed, 12 Jan 2011 16:48:45 -0700 Subject: [gst-devel] kenburns plugin In-Reply-To: <4D2DC7C2.1000807@hora-obscura.de> References: <4D2B7145.7010503@brooks.nu> <4D2DC7C2.1000807@hora-obscura.de> Message-ID: <4D2E3DDD.8030101@brooks.nu> On 01/12/2011 08:24 AM, Stefan Kost wrote: > On 10.01.2011 22:51, Lane Brooks wrote: >> I have written a kenburns plugin >> (http://en.wikipedia.org/wiki/Ken_burns_effect) that does smooth >> zooming/panning simultaneously. >> >> I have it committed in a git repo at https://github.com/dirjud/kenburns >> >> There are some gst-launch samples in the src/gstkenburns.c file. > > Such a plugin would be welcome. Make a patch against gst-plugins-bad and > put it to bugzilla. Add yourself to the copyright comment. >> Currently I have only implemented nearest neighbor sampling, so it is >> fast but can show aliasing artifacts. > Yes, n-tap filtering would be nice to have, but thats no show-stopper > for getting it into bad. > > Do you know about GstController? I would prefer to only have "zoom", > "xcenter", "ycenter" properties, make then controllable and leave the > animations to GstController. With that you get envelops and LFOs right now. I do know about controllers and opted not to use them because of the complication I thought it would put on the user for anything other than the linear case. For example, I implemented an internal controller that ramps up the velocity linearly until it reaches a paramaterized time. It then holds the velocity constant until it reaches the parameter mirror at the end where it ramps the velocity back down. This gives a nice looking effect. It requires taking some time derivatives (converting velocity to position) and what not. I wanted to have nice movement options that did not require the user to do a lot of math. That said, not enabling external controllers does limit the functionality. For example, my current implementation only allows for one pan/zoom on a linear path. Using controllers, you could do curved paths or whatever. Let me play a little with the controllers to see if I can get similarly nice movement with them. Controllers would make it more generic, but I don't want to loose the ease of use. Actually, I just had another thought. Currently I have a few different user selectable internal controllers like the velocity ramping one described above. Would it work to add another option that lets the user select to use an external controller instead? Then for the simple use case, you just set "zoom1", "xcenter1", "ycenter1", "zoom2", "xcenter2", "ycenter2", and "duration". For the more advanced usage, you instead put a controller on "zoom1", "xcenter1", and "ycenter1" and select to disable the internal controller. Lane From shiva.varma at gmail.com Thu Jan 13 00:50:05 2011 From: shiva.varma at gmail.com (shiva varma) Date: Wed, 12 Jan 2011 15:50:05 -0800 (PST) Subject: [gst-devel] RTP audio stream to filesink, issue with packet loss In-Reply-To: <1294874197.25646.0.camel@zingle> References: <1294874197.25646.0.camel@zingle> Message-ID: <1294876205704-3215009.post@n4.nabble.com> You can take the output of the rtppcmudepay directly to wavenc... The below pipeline should work... filesrc location=foobar.pcap ! pcapparse ! "application/x-rtp, payload=0, clock-rate=8000" ! gstrtpjitterbuffer ! rtppcmudepay ! wavenc ! filesink location=foobar.wav -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/RTP-audio-stream-to-filesink-issue-with-packet-loss-tp3214825p3215009.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From yuanqing.cai at tieto.com Thu Jan 13 02:48:21 2011 From: yuanqing.cai at tieto.com (Cai Yuanqing) Date: Thu, 13 Jan 2011 09:48:21 +0800 Subject: [gst-devel] audioresample In-Reply-To: References: <1294823945910-3213586.post@n4.nabble.com> Message-ID: <4D2E59E5.6020905@tieto.com> Hi Suresh: Your application have a little problem. :-) On 01/12/2011 08:41 PM, Jesu Anuroop Suresh wrote: > Hi Sean, > > Yes, what I was trying is to resample the decoded mp3 data to the > fixed (22KHZ S16LE) formate, > > no matter what is the input rate using a C application. > > Thanks for your response. > > Here is the piece of the code for the same but it does not work with > audioresample with the caps filter 'resmux'. This code does work > without the caps filter 'resmux'. > > GstElement *source, *demuxer, *decoder, *conv, *sink, > *resample, *resmux; > GstCaps *caps; > > gst_init(NULL, NULL); > > /* Create gstreamer elements */ > musicPlayer.playPipeline = gst_pipeline_new ("audio-player"); > source = gst_element_factory_make ("filesrc", "file-source"); > sink = gst_element_factory_make ("alsasink", "audio-output"); > resample = gst_element_factory_make ("audioresample", > "audio-resample"); > conv = gst_element_factory_make ("audioconvert", > "converter1"); > > caps = gst_caps_new_simple ("audio/x-raw-int", > "width", G_TYPE_INT, 16, > "depth", G_TYPE_INT, 16, > "rate", G_TYPE_INT, 22050, > "channels",G_TYPE_INT, 2, NULL > ); > > if (!musicPlayer.playPipeline || !source || !sink || > !resample || !resmux || !caps || !conv) > { > g_print ("NO MEM Exiting.\n"); > return 1; > } resmux is not initialized yet,here maybe some random value,you'd better remove it from check list. > > /* we set the input filename to the source element */ > g_object_set (G_OBJECT (source), "location", filePath, NULL); > > demuxer = gst_element_factory_make ("id3demux", "id3-demuxer"); > decoder = gst_element_factory_make ("mad", "mp3-decoder"); > > if (!demuxer || !decoder || !conv1) conv1 ? dose it should be conv? > { > g_print ("NO MEM Exiting.\n"); > return 1; > } > > g_object_set (G_OBJECT (resmux), "caps", caps, NULL); > gst_caps_unref (caps); > as I said before,resmux haven't initialized ,that's not quite right. and I suggest you to remove these two lines. > /* file-source -> demuxer -> decoder -> alsa-output */ > gst_bin_add_many (GST_BIN (musicPlayer.playPipeline), > source, demuxer, decoder, conv, resample, > resmux,sink, NULL); > > gst_element_link (source, demuxer); > gst_element_link_many (decoder, conv, resample,resmux,sink, NULL); You can use gst_element_link_filtered to link resample and sink with caps instead of this way. something like: gst_element_link (source, demuxer); gst_element_link_many (decoder, conv, resample, NULL); if ( !gst_element_link_filtered(resample,sink,caps) ){ g_printerr("Failed to link elements resample and alsa-sink"); } > g_signal_connect (demuxer, "pad-added", G_CALLBACK > (on_pad_added), decoder); > > GstBus *bus = > gst_pipeline_get_bus(GST_PIPELINE(musicPlayer.playPipeline)); > gst_bus_add_watch(bus, bus_call, NULL); > gst_object_unref(bus); > > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), > GST_STATE_PLAYING); > > musicPlayer.playLoop = g_main_loop_new(NULL, FALSE); > > g_main_loop_run(musicPlayer.playLoop); > > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), > GST_STATE_NULL); > gst_object_unref(GST_OBJECT(musicPlayer.playPipeline)); > > > > > With Warm Regards > Jesu Anuroop Suresh > > "Any intelligent fool can make things bigger, more complex, and more > violent. It takes a touch of genius -- and a lot of courage -- to move > in the opposite direction." > "Anyone who has never made a mistake has never tried anything new." > > I attached my modified source code ,you can try it. Hope it helps. Thanks. -- B.R Cai Yuanqing -------------- next part -------------- A non-text attachment was scrubbed... Name: test.c Type: text/x-csrc Size: 6087 bytes Desc: not available URL: From tbrendt at googlemail.com Thu Jan 13 03:10:41 2011 From: tbrendt at googlemail.com (Thorsten Brendt) Date: Thu, 13 Jan 2011 03:10:41 +0100 Subject: [gst-devel] RTP audio stream to filesink, issue with packet loss In-Reply-To: <1294874197.25646.0.camel@zingle> References: <1294874197.25646.0.camel@zingle> Message-ID: On Thu, Jan 13, 2011 at 12:16 AM, Tim-Philipp M?ller wrote: > On Wed, 2011-01-12 at 22:21 +0100, Thorsten Brendt wrote: > > Hi, > >> (snip) >> >> Ultimately some element in the pipeline needs to produce >> ?"silence/empty/zero data" if its input buffer is empty. ?Is there an >> ? explicit way (helper element/attribute) to do this? > > Tried adding an audiorate element? Do you mean a caps-filter element like this?: 'audio/x-raw-int, width=16, depth=16, rate=8000, endianness=1234, signed=(boolean)true, channels=1' The right place to put this seems to be before the encoder element (wavenc in my case), however in that case the pipeline can't be linked together. It works before an autoaudiosink though. Anyhow, I assume that the audio's sampling rate is known to the encoder already, as it both given explicitly as a clock-rate attribute in a previous caps-filter and should be implicitly known by the mulaw-decoder (plus the audio file "sounds right" in general). From tbrendt at googlemail.com Thu Jan 13 03:13:52 2011 From: tbrendt at googlemail.com (Thorsten Brendt) Date: Thu, 13 Jan 2011 03:13:52 +0100 Subject: [gst-devel] RTP audio stream to filesink, issue with packet loss In-Reply-To: <1294876205704-3215009.post@n4.nabble.com> References: <1294874197.25646.0.camel@zingle> <1294876205704-3215009.post@n4.nabble.com> Message-ID: On Thu, Jan 13, 2011 at 12:50 AM, shiva varma wrote: > > You can take the output of the rtppcmudepay directly to wavenc... > > The below pipeline should work... > > filesrc location=foobar.pcap ! pcapparse > ? ! "application/x-rtp, payload=0, clock-rate=8000" > ? ! gstrtpjitterbuffer ! rtppcmudepay ! wavenc ! filesink > location=foobar.wav Thanks for the suggestion, but I need to have uncompressed PCM audio in the wav file. Should have stated that more clearly, "a wav file" is really kind of a noop. From yiliangb at gmail.com Thu Jan 13 03:20:14 2011 From: yiliangb at gmail.com (yiliang) Date: Wed, 12 Jan 2011 18:20:14 -0800 (PST) Subject: [gst-devel] x264enc not working with tee + video output sink In-Reply-To: References: Message-ID: <1294885214052-3215129.post@n4.nabble.com> Hi, I am seeing the same problem on Ubuntu 10.10. Has there any fix for this already? Thanks! Yiliang -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/x264enc-not-working-with-tee-video-output-sink-tp2327474p3215129.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From jesuas at gmail.com Thu Jan 13 05:21:12 2011 From: jesuas at gmail.com (Jesu Anuroop Suresh) Date: Wed, 12 Jan 2011 20:21:12 -0800 (PST) Subject: [gst-devel] audioresample In-Reply-To: <4D2E59E5.6020905@tieto.com> References: <1294823945910-3213586.post@n4.nabble.com> <4D2E59E5.6020905@tieto.com> Message-ID: Hi Cai, Thanks for you response, I Will try out your suggestion of using the filtered link. Sorry there was some typoerror in my code what I shared. I did initialized the 'resmux' as capasity filter and used the conv not conv1. The cocde works for me for mp3 playback in its original settings. resample = gst_element_factory_make ("audioresample", "audio-resample"); conv = gst_element_factory_make ("audioconvert", "converter1"); resmux = gst_element_factory_make ("capsfilter", "filter"); caps = gst_caps_new_simple ("audio/x-raw-int", "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, "rate", G_TYPE_INT, 22050, "channels",G_TYPE_INT, 2, NULL ); if (!musicPlayer.playPipeline || !source || !sink || !resample || !resmux || !caps || !conv) { g_print ("NO MEM Exiting.\n"); return 1; } /* we set the input filename to the source element */ g_object_set (G_OBJECT (source), "location", filePath, NULL); demuxer = gst_element_factory_make ("id3demux", "id3-demuxer"); decoder = gst_element_factory_make ("mad", "mp3-decoder"); if (!demuxer || !decoder || !conv) { g_print ("NO MEM Exiting.\n"); return 1; } With Warm Regards Jesu Anuroop Suresh "Any intelligent fool can make things bigger, more complex, and more violent. It takes a touch of genius -- and a lot of courage -- to move in the opposite direction." "Anyone who has never made a mistake has never tried anything new." On Thu, Jan 13, 2011 at 7:16 AM, Cai Yuanqing [via GStreamer-devel] < ml-node+3215098-821959213-203210 at n4.nabble.com > wrote: > Hi Suresh: > Your application have a little problem. :-) > > > On 01/12/2011 08:41 PM, Jesu Anuroop Suresh wrote: > > > Hi Sean, > > > > Yes, what I was trying is to resample the decoded mp3 data to the > > fixed (22KHZ S16LE) formate, > > > > no matter what is the input rate using a C application. > > > > Thanks for your response. > > > > Here is the piece of the code for the same but it does not work with > > audioresample with the caps filter 'resmux'. This code does work > > without the caps filter 'resmux'. > > > > GstElement *source, *demuxer, *decoder, *conv, *sink, > > *resample, *resmux; > > GstCaps *caps; > > > > gst_init(NULL, NULL); > > > > /* Create gstreamer elements */ > > musicPlayer.playPipeline = gst_pipeline_new ("audio-player"); > > source = gst_element_factory_make ("filesrc", "file-source"); > > sink = gst_element_factory_make ("alsasink", "audio-output"); > > > resample = gst_element_factory_make ("audioresample", > > "audio-resample"); > > conv = gst_element_factory_make ("audioconvert", > > "converter1"); > > > > caps = gst_caps_new_simple ("audio/x-raw-int", > > "width", G_TYPE_INT, 16, > > "depth", G_TYPE_INT, 16, > > "rate", G_TYPE_INT, 22050, > > "channels",G_TYPE_INT, 2, NULL > > ); > > > > if (!musicPlayer.playPipeline || !source || !sink || > > !resample || !resmux || !caps || !conv) > > { > > g_print ("NO MEM Exiting.\n"); > > return 1; > > } > resmux is not initialized yet,here maybe some random value,you'd better > remove it from check list. > > > > > /* we set the input filename to the source element */ > > g_object_set (G_OBJECT (source), "location", filePath, NULL); > > > > demuxer = gst_element_factory_make ("id3demux", "id3-demuxer"); > > decoder = gst_element_factory_make ("mad", "mp3-decoder"); > > > > if (!demuxer || !decoder || !conv1) > conv1 ? dose it should be conv? > > > { > > g_print ("NO MEM Exiting.\n"); > > return 1; > > } > > > > g_object_set (G_OBJECT (resmux), "caps", caps, NULL); > > gst_caps_unref (caps); > > > as I said before,resmux haven't initialized ,that's not quite right. > and I suggest you to remove these two lines. > > > /* file-source -> demuxer -> decoder -> alsa-output */ > > gst_bin_add_many (GST_BIN (musicPlayer.playPipeline), > > source, demuxer, decoder, conv, resample, > > resmux,sink, NULL); > > > > gst_element_link (source, demuxer); > > gst_element_link_many (decoder, conv, resample,resmux,sink, > NULL); > You can use gst_element_link_filtered to link resample and sink with > caps instead of this way. > something like: > gst_element_link (source, demuxer); > gst_element_link_many (decoder, conv, resample, NULL); > if ( !gst_element_link_filtered(resample,sink,caps) ){ > g_printerr("Failed to link elements resample and alsa-sink"); > } > > > > g_signal_connect (demuxer, "pad-added", G_CALLBACK > > (on_pad_added), decoder); > > > > GstBus *bus = > > gst_pipeline_get_bus(GST_PIPELINE(musicPlayer.playPipeline)); > > gst_bus_add_watch(bus, bus_call, NULL); > > gst_object_unref(bus); > > > > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), > > GST_STATE_PLAYING); > > > > musicPlayer.playLoop = g_main_loop_new(NULL, FALSE); > > > > g_main_loop_run(musicPlayer.playLoop); > > > > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), > > GST_STATE_NULL); > > gst_object_unref(GST_OBJECT(musicPlayer.playPipeline)); > > > > > > > > > > With Warm Regards > > Jesu Anuroop Suresh > > > > "Any intelligent fool can make things bigger, more complex, and more > > violent. It takes a touch of genius -- and a lot of courage -- to move > > in the opposite direction." > > "Anyone who has never made a mistake has never tried anything new." > > > > > I attached my modified source code ,you can try it. > Hope it helps. > > Thanks. > > > -- > B.R > > Cai Yuanqing > > > ------------------------------------------------------------------------------ > > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > http://p.sf.net/sfu/oracle-sfdevnl > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > *test.c* (6K) Download Attachment > > > ------------------------------ > View message @ > http://gstreamer-devel.966125.n4.nabble.com/audioresample-tp3213586p3215098.html > > To unsubscribe from audioresample, click here. > > -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/audioresample-tp3213586p3215225.html Sent from the GStreamer-devel mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jesuas at gmail.com Thu Jan 13 06:39:32 2011 From: jesuas at gmail.com (Anuroop Jesu) Date: Thu, 13 Jan 2011 11:09:32 +0530 Subject: [gst-devel] audioresample In-Reply-To: References: <1294823945910-3213586.post@n4.nabble.com> <4D2E59E5.6020905@tieto.com> Message-ID: Hi , I Tried the suggestion provided for the by Cai. Thanks for the code Cai. It works something like this It only allows to playback the 22KHz S16LE audio. What I was trying is to convert the any input format into a 22KHz S16LE so I can mux it with other stream of the same property and mux multiple streams using alsasink plug:dmix. With Warm Regards Jesu Anuroop Suresh "Any intelligent fool can make things bigger, more complex, and more violent. It takes a touch of genius -- and a lot of courage -- to move in the opposite direction." "Anyone who has never made a mistake has never tried anything new." On Thu, Jan 13, 2011 at 9:51 AM, Jesu Anuroop Suresh wrote: > Hi Cai, > > > Thanks for you response, I Will try out your suggestion of using the > filtered link. > > Sorry there was some typoerror in my code what I shared. > > I did initialized the 'resmux' as capasity filter and used the conv not > conv1. > > The cocde works for me for mp3 playback in its original settings. > > > resample = gst_element_factory_make ("audioresample", > "audio-resample"); > conv = gst_element_factory_make ("audioconvert", > "converter1"); > resmux = gst_element_factory_make ("capsfilter", "filter"); > > caps = gst_caps_new_simple ("audio/x-raw-int", > "width", G_TYPE_INT, 16, > "depth", G_TYPE_INT, 16, > "rate", G_TYPE_INT, 22050, > "channels",G_TYPE_INT, 2, NULL > ); > > if (!musicPlayer.playPipeline || !source || !sink || > !resample || !resmux || !caps || !conv) > { > g_print ("NO MEM Exiting.\n"); > return 1; > } > > /* we set the input filename to the source element */ > g_object_set (G_OBJECT (source), "location", filePath, NULL); > > demuxer = gst_element_factory_make ("id3demux", "id3-demuxer"); > decoder = gst_element_factory_make ("mad", "mp3-decoder"); > > if (!demuxer || !decoder || !conv) > { > g_print ("NO MEM Exiting.\n"); > return 1; > } > > With Warm Regards > Jesu Anuroop Suresh > > "Any intelligent fool can make things bigger, more complex, and more > violent. It takes a touch of genius -- and a lot of courage -- to move in > the opposite direction." > "Anyone who has never made a mistake has never tried anything new." > > On Thu, Jan 13, 2011 at 7:16 AM, Cai Yuanqing [via GStreamer-devel] <[hidden > email] > wrote: > >> Hi Suresh: >> Your application have a little problem. :-) >> >> >> On 01/12/2011 08:41 PM, Jesu Anuroop Suresh wrote: >> >> > Hi Sean, >> > >> > Yes, what I was trying is to resample the decoded mp3 data to the >> > fixed (22KHZ S16LE) formate, >> > >> > no matter what is the input rate using a C application. >> > >> > Thanks for your response. >> > >> > Here is the piece of the code for the same but it does not work with >> > audioresample with the caps filter 'resmux'. This code does work >> > without the caps filter 'resmux'. >> > >> > GstElement *source, *demuxer, *decoder, *conv, *sink, >> > *resample, *resmux; >> > GstCaps *caps; >> > >> > gst_init(NULL, NULL); >> > >> > /* Create gstreamer elements */ >> > musicPlayer.playPipeline = gst_pipeline_new ("audio-player"); >> > source = gst_element_factory_make ("filesrc", "file-source"); >> > sink = gst_element_factory_make ("alsasink", >> "audio-output"); >> > resample = gst_element_factory_make ("audioresample", >> > "audio-resample"); >> > conv = gst_element_factory_make ("audioconvert", >> > "converter1"); >> > >> > caps = gst_caps_new_simple ("audio/x-raw-int", >> > "width", G_TYPE_INT, 16, >> > "depth", G_TYPE_INT, 16, >> > "rate", G_TYPE_INT, 22050, >> > "channels",G_TYPE_INT, 2, NULL >> > ); >> > >> > if (!musicPlayer.playPipeline || !source || !sink || >> > !resample || !resmux || !caps || !conv) >> > { >> > g_print ("NO MEM Exiting.\n"); >> > return 1; >> > } >> resmux is not initialized yet,here maybe some random value,you'd better >> remove it from check list. >> >> > >> > /* we set the input filename to the source element */ >> > g_object_set (G_OBJECT (source), "location", filePath, NULL); >> > >> > demuxer = gst_element_factory_make ("id3demux", "id3-demuxer"); >> >> > decoder = gst_element_factory_make ("mad", "mp3-decoder"); >> > >> > if (!demuxer || !decoder || !conv1) >> conv1 ? dose it should be conv? >> >> > { >> > g_print ("NO MEM Exiting.\n"); >> > return 1; >> > } >> > >> > g_object_set (G_OBJECT (resmux), "caps", caps, NULL); >> > gst_caps_unref (caps); >> > >> as I said before,resmux haven't initialized ,that's not quite right. >> and I suggest you to remove these two lines. >> >> > /* file-source -> demuxer -> decoder -> alsa-output */ >> > gst_bin_add_many (GST_BIN (musicPlayer.playPipeline), >> > source, demuxer, decoder, conv, resample, >> > resmux,sink, NULL); >> > >> > gst_element_link (source, demuxer); >> > gst_element_link_many (decoder, conv, resample,resmux,sink, >> NULL); >> You can use gst_element_link_filtered to link resample and sink with >> caps instead of this way. >> something like: >> gst_element_link (source, demuxer); >> gst_element_link_many (decoder, conv, resample, NULL); >> if ( !gst_element_link_filtered(resample,sink,caps) ){ >> g_printerr("Failed to link elements resample and alsa-sink"); >> } >> >> >> > g_signal_connect (demuxer, "pad-added", G_CALLBACK >> > (on_pad_added), decoder); >> > >> > GstBus *bus = >> > gst_pipeline_get_bus(GST_PIPELINE(musicPlayer.playPipeline)); >> > gst_bus_add_watch(bus, bus_call, NULL); >> > gst_object_unref(bus); >> > >> > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), >> > GST_STATE_PLAYING); >> > >> > musicPlayer.playLoop = g_main_loop_new(NULL, FALSE); >> > >> > g_main_loop_run(musicPlayer.playLoop); >> > >> > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), >> > GST_STATE_NULL); >> > gst_object_unref(GST_OBJECT(musicPlayer.playPipeline)); >> > >> > >> > >> > >> > With Warm Regards >> > Jesu Anuroop Suresh >> > >> > "Any intelligent fool can make things bigger, more complex, and more >> > violent. It takes a touch of genius -- and a lot of courage -- to move >> > in the opposite direction." >> > "Anyone who has never made a mistake has never tried anything new." >> > >> > >> I attached my modified source code ,you can try it. >> Hope it helps. >> >> Thanks. >> >> >> -- >> B.R >> >> Cai Yuanqing >> >> >> ------------------------------------------------------------------------------ >> >> Protect Your Site and Customers from Malware Attacks >> Learn about various malware tactics and how to avoid them. Understand >> malware threats, the impact they can have on your business, and how you >> can protect your company and customers by using code signing. >> http://p.sf.net/sfu/oracle-sfdevnl >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> *test.c* (6K) Download Attachment >> >> >> ------------------------------ >> View message @ >> http://gstreamer-devel.966125.n4.nabble.com/audioresample-tp3213586p3215098.html >> >> To unsubscribe from audioresample, click here. >> >> > > > ------------------------------ > View this message in context: Re: audioresample > > Sent from the GStreamer-devel mailing list archiveat Nabble.com. > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 mgruenke at Tycoint.com Thu Jan 13 07:27:53 2011 From: mgruenke at Tycoint.com (Gruenke, Matt) Date: Thu, 13 Jan 2011 01:27:53 -0500 Subject: [gst-devel] kenburns plugin In-Reply-To: <4D2E3DDD.8030101@brooks.nu> References: <4D2B7145.7010503@brooks.nu> <4D2DC7C2.1000807@hora-obscura.de> <4D2E3DDD.8030101@brooks.nu> Message-ID: I'd recommend making a custom bin to encapsulate the full controls with a set of simplified parameters. You could add a controller interface to kenburns, and then make a custom bin called kenburns_easy, for instance, that provides a simplified interface by using an internal controller. One example of this is rtspsrc, which is a bin. It encapsulates a lot of other elements that one can also use in a stand-alone fashion. Matt -----Original Message----- From: Lane Brooks [mailto:lane at brooks.nu] Sent: Wednesday, January 12, 2011 18:49 To: Discussion of the development of GStreamer Subject: Re: [gst-devel] kenburns plugin On 01/12/2011 08:24 AM, Stefan Kost wrote: > On 10.01.2011 22:51, Lane Brooks wrote: Actually, I just had another thought. Currently I have a few different user selectable internal controllers like the velocity ramping one described above. Would it work to add another option that lets the user select to use an external controller instead? Then for the simple use case, you just set "zoom1", "xcenter1", "ycenter1", "zoom2", "xcenter2", "ycenter2", and "duration". For the more advanced usage, you instead put a controller on "zoom1", "xcenter1", and "ycenter1" and select to disable the internal controller. Lane From sramali2 at cisco.com Thu Jan 13 07:47:55 2011 From: sramali2 at cisco.com (senthil) Date: Wed, 12 Jan 2011 22:47:55 -0800 (PST) Subject: [gst-devel] x264enc not working with tee + video output sink In-Reply-To: References: Message-ID: <1294901275424-3215317.post@n4.nabble.com> Hello, The queue will block the supplier when there is no more space in it. Please refer http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-queue.html http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-queue.html Since x264enc may take more time to consume, setting leaky property (as suggested in the doc) on the queue before xvimagesink might help. Senthil -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/x264enc-not-working-with-tee-video-output-sink-tp2327474p3215317.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From yiliangb at gmail.com Thu Jan 13 08:12:59 2011 From: yiliangb at gmail.com (yiliang) Date: Wed, 12 Jan 2011 23:12:59 -0800 (PST) Subject: [gst-devel] x264enc not working with tee + video output sink In-Reply-To: <1294901275424-3215317.post@n4.nabble.com> References: <1294901275424-3215317.post@n4.nabble.com> Message-ID: <1294902779367-3215331.post@n4.nabble.com> Hi Senthil, Thanks for the reply. I have done more tests after the post. After some simplification (remove tee and queue, etc), here is the pipeline that gives me the problem. It will be stuck a couple of seconds after start, then it will exit abnormally. gst-launch -e v4l2src device=/dev/video0 ! x264enc byte-stream=true ! qtmux ! filesink location=a.mov If I change byte-stream=false, the pipeline will go on, but the movie file recorded can not be played. If I change qtmux to ffmux_mp4, the recording can go on no matter I set byte-stream to true or false. The movie can be played, but the movie file has lots of artifacts. Lastly, if I use avimux, the pipeline works fine, and the movie file recorded can be played smoothly. So after all the test, it seems to be more a problem related how the qtmux (or ffmux_mp4) interacts with other elements. Also it seems to be a different problem from that reported in the original post, since I can use avimux with x264enc. I may post a message with a different subject. Thanks! Yiliang -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/x264enc-not-working-with-tee-video-output-sink-tp2327474p3215331.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wl2776 at gmail.com Thu Jan 13 09:24:54 2011 From: wl2776 at gmail.com (wl2776) Date: Thu, 13 Jan 2011 00:24:54 -0800 (PST) Subject: [gst-devel] What is 'streaming thread'? Message-ID: <1294907094576-3215407.post@n4.nabble.com> This terms is met in many places of the GStreamer documentation, but I can't find its strict definition. So, what is it, exactly? Or, please, point to the definition. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-is-streaming-thread-tp3215407p3215407.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Thu Jan 13 10:41:38 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 13 Jan 2011 09:41:38 +0000 Subject: [gst-devel] x264enc not working with tee + video output sink In-Reply-To: <1294902779367-3215331.post@n4.nabble.com> References: <1294901275424-3215317.post@n4.nabble.com> <1294902779367-3215331.post@n4.nabble.com> Message-ID: <1294911698.28313.3.camel@zingle> > I have done more tests after the post. After some simplification (remove tee > and queue, etc), here is the pipeline that gives me the problem. It will be > stuck a couple of seconds after start, then it will exit abnormally. Exit abnormally? How? What's the error message? (presumably a not-negotiated error, which is qtmux rejecting H.264 in bytestream format) > gst-launch -e v4l2src device=/dev/video0 ! x264enc byte-stream=true ! > qtmux ! filesink location=a.mov > > If I change byte-stream=false, the pipeline will go on, but the movie file > recorded can not be played. Works fine for me with git/latest pre-releases. Is gst-launch finishing up the file properly? (ie. sending the EOS and waiting for the EOS) Cheers -Tim From t.i.m at zen.co.uk Thu Jan 13 10:44:35 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 13 Jan 2011 09:44:35 +0000 Subject: [gst-devel] What is 'streaming thread'? In-Reply-To: <1294907094576-3215407.post@n4.nabble.com> References: <1294907094576-3215407.post@n4.nabble.com> Message-ID: <1294911875.28313.6.camel@zingle> On Thu, 2011-01-13 at 00:24 -0800, wl2776 wrote: > This terms is met in many places of the GStreamer documentation, but I can't > find its strict definition. > So, what is it, exactly? Or, please, point to the definition. Any thread not created by the application, but by GStreamer. Often used as 'streaming thread' (in which dataflow happens) vs. 'application thread' (in which things like property setting/getting is usually done, and state changes). There are usually multiple streaming threads in a typical GStreamer pipeline. For application writers it rarely matters which one it is exactly though, just that it's a different thread than the main application's thread. Cheers -Tim From t.i.m at zen.co.uk Thu Jan 13 10:46:28 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 13 Jan 2011 09:46:28 +0000 Subject: [gst-devel] RTP audio stream to filesink, issue with packet loss In-Reply-To: References: <1294874197.25646.0.camel@zingle> Message-ID: <1294911988.28313.8.camel@zingle> On Thu, 2011-01-13 at 03:10 +0100, Thorsten Brendt wrote: > > Tried adding an audiorate element? > > Do you mean a caps-filter element like this?: > > 'audio/x-raw-int, width=16, depth=16, rate=8000, endianness=1234, > signed=(boolean)true, channels=1' No, I meant an "audiorate" element, which is part of gst-plugins-base. Factory Details: Long name: Audio rate adjuster Class: Filter/Effect/Audio Description: Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream Cheers -Tim From parveen.jain at one97.net Thu Jan 13 10:56:28 2011 From: parveen.jain at one97.net (Parveen Kumar Jain) Date: Thu, 13 Jan 2011 15:26:28 +0530 Subject: [gst-devel] What is 'streaming thread'? In-Reply-To: <1294911875.28313.6.camel@zingle> References: <1294907094576-3215407.post@n4.nabble.com> <1294911875.28313.6.camel@zingle> Message-ID: Hi Tim, Just a small cross question on this: How one can control these no. of threads creation on a single pipeline ? or if I put it in another way, how these threads get created in gstreamer ? e.g. does it depend on the no. of "elements" in a single pipeline or some other factor matters. Regards, Parveen Jain On Thu, Jan 13, 2011 at 3:14 PM, Tim-Philipp M?ller wrote: > On Thu, 2011-01-13 at 00:24 -0800, wl2776 wrote: > >> This terms is met in many places of the GStreamer documentation, but I can't >> find its strict definition. >> So, what is it, exactly? Or, please, point to the definition. > > Any thread not created by the application, but by GStreamer. Often used > as 'streaming thread' (in which dataflow happens) vs. 'application > thread' (in which things like property setting/getting is usually done, > and state changes). There are usually multiple streaming threads in a > typical GStreamer pipeline. For application writers it rarely matters > which one it is exactly though, just that it's a different thread than > the main application's thread. > > Cheers > ?-Tim > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 > -- Parveen Jain Technical Lead ? Network Engineering One97 Communications (P) Ltd B121, Sector 5, Noida, UP 201301 P:? + 91 120 4770770? ? ? Extn:352 M: + 91?? 9212708203 W: www.one97world.com From bilboed at gmail.com Thu Jan 13 11:39:57 2011 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 13 Jan 2011 11:39:57 +0100 Subject: [gst-devel] GNonLin pre-release 0.10.16.2 Message-ID: <1294915197.27670.10.camel@localhost> Better late than never. The first pre-release of the GNonLin non-linear editing plugins is available at: http://gstreamer.freedesktop.org/data/src/gnonlin/pre/ Ubuntu users might also be able to test this pre-release by using the GStreamer PPA (as soon as they get uploaded there). Please test it thoroughly, and let us know of any regressions or other important issues by filing blocker bugs in bugzilla: http://gstreamer.freedesktop.org/bugs/ (Component: gnonlin) Further pre-releases will follow as needed. If all goes well, 0.10.17 release should happen later this month when GStreamer core/-base 0.10.32 are out, as per http://gstreamer.freedesktop.org/wiki/ReleasePlanning2011 Edward From wl2776 at gmail.com Thu Jan 13 11:54:49 2011 From: wl2776 at gmail.com (wl2776) Date: Thu, 13 Jan 2011 02:54:49 -0800 (PST) Subject: [gst-devel] What is 'streaming thread'? In-Reply-To: References: <1294907094576-3215407.post@n4.nabble.com> <1294911875.28313.6.camel@zingle> Message-ID: <1294916089384-3215629.post@n4.nabble.com> Parveen Kumar Jain wrote: > > How one can control these no. of threads creation on a single > pipeline ? or if I put it in another way, how these threads get > created in gstreamer ? > This is covered in the docs. http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/chapter-threads.html http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/part-messages.txt - GST_MESSAGE_STREAM_STATUS -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-is-streaming-thread-tp3215407p3215629.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wl2776 at gmail.com Thu Jan 13 12:02:01 2011 From: wl2776 at gmail.com (wl2776) Date: Thu, 13 Jan 2011 03:02:01 -0800 (PST) Subject: [gst-devel] What is 'streaming thread'? In-Reply-To: <1294911875.28313.6.camel@zingle> References: <1294907094576-3215407.post@n4.nabble.com> <1294911875.28313.6.camel@zingle> Message-ID: <1294916521208-3215636.post@n4.nabble.com> Thanks. Also, do the terms 'streaming thread' and 'streaming task' have the same meaning? Tim-Philipp M?ller-2 wrote: > >> This term is met in many places of the GStreamer documentation, but I >> can't >> find its strict definition. >> So, what is it, exactly? Or, please, point to the definition. > > Any thread not created by the application, but by GStreamer. Often used > as 'streaming thread' (in which dataflow happens) vs. 'application > thread' (in which things like property setting/getting is usually done, > and state changes). > -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-is-streaming-thread-tp3215407p3215636.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ikt011 at gmail.com Thu Jan 13 12:29:41 2011 From: ikt011 at gmail.com (Kocsis Tibor) Date: Thu, 13 Jan 2011 12:29:41 +0100 Subject: [gst-devel] GNonLin pre-release 0.10.16.2 In-Reply-To: <1294915197.27670.10.camel@localhost> References: <1294915197.27670.10.camel@localhost> Message-ID: Hi, is there any documentation for GNonLin? I googled the internet but found only broken links and two blog entries. Regards, Tibor On Thu, Jan 13, 2011 at 11:39 AM, Edward Hervey wrote: > Better late than never. > > ?The first pre-release of the GNonLin non-linear editing plugins is > available at: > ?http://gstreamer.freedesktop.org/data/src/gnonlin/pre/ > > Ubuntu users might also be able to test this pre-release by using the > GStreamer PPA (as soon as they get uploaded there). > > Please test it thoroughly, and let us know of any regressions or other > important issues by filing blocker bugs in bugzilla: > http://gstreamer.freedesktop.org/bugs/ (Component: gnonlin) > > Further pre-releases will follow as needed. If all goes well, 0.10.17 > release should happen later this month when GStreamer core/-base 0.10.32 > are out, as per > http://gstreamer.freedesktop.org/wiki/ReleasePlanning2011 > > ?Edward > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 t.i.m at zen.co.uk Thu Jan 13 12:36:16 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 13 Jan 2011 11:36:16 +0000 Subject: [gst-devel] What is 'streaming thread'? In-Reply-To: <1294916521208-3215636.post@n4.nabble.com> References: <1294907094576-3215407.post@n4.nabble.com> <1294911875.28313.6.camel@zingle> <1294916521208-3215636.post@n4.nabble.com> Message-ID: <1294918576.28313.10.camel@zingle> On Thu, 2011-01-13 at 03:02 -0800, wl2776 wrote: > Also, do the terms 'streaming thread' and 'streaming task' have the same > meaning? Yes (see the GstTask API which is convenience API for starting/stopping threads in a GStreamer context). Cheers -tim From pierre.crepieux at orange-ftgroup.com Thu Jan 13 12:41:57 2011 From: pierre.crepieux at orange-ftgroup.com (pierre.crepieux at orange-ftgroup.com) Date: Thu, 13 Jan 2011 12:41:57 +0100 Subject: [gst-devel] sending audio/video on RTP with C API In-Reply-To: References: <4D26ECBE.4020800@orange-ftgroup.com> <20110107120530.2f8713ab@mtelleria.com> <4D2C364E.6040604@orange-ftgroup.com> Message-ID: <4D2EE505.6020600@orange-ftgroup.com> giorgino a ?crit : > orange-ftgroup.com> writes: > > Dear Pierre could you give me some suggestion about your working example with > h.263 altough like you I'm interested to work with h.264. > > G. > Here it is. It's really a simple copy-paste of the sample provided with the source distribution of gstreamer. I only mimic the first RTP sink to add the second (also added a preview). I didn't tried again with h264. Maybe there are a few more constraints this simple code doesn't handle ... I'll tell you if I succeed. > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 -------------- A non-text attachment was scrubbed... Name: rtpbin_sample.c Type: text/x-csrc Size: 10766 bytes Desc: not available URL: From bilboed at gmail.com Thu Jan 13 12:46:52 2011 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 13 Jan 2011 12:46:52 +0100 Subject: [gst-devel] GNonLin pre-release 0.10.16.2 In-Reply-To: References: <1294915197.27670.10.camel@localhost> Message-ID: <1294919213.27670.12.camel@localhost> On Thu, 2011-01-13 at 12:29 +0100, Kocsis Tibor wrote: > Hi, > > is there any documentation for GNonLin? I googled the internet but > found only broken links and two blog entries. Apart from the plugin doc, no: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gnonlin/html/ > > Regards, > Tibor > > On Thu, Jan 13, 2011 at 11:39 AM, Edward Hervey wrote: > > Better late than never. > > > > The first pre-release of the GNonLin non-linear editing plugins is > > available at: > > http://gstreamer.freedesktop.org/data/src/gnonlin/pre/ > > > > Ubuntu users might also be able to test this pre-release by using the > > GStreamer PPA (as soon as they get uploaded there). > > > > Please test it thoroughly, and let us know of any regressions or other > > important issues by filing blocker bugs in bugzilla: > > http://gstreamer.freedesktop.org/bugs/ (Component: gnonlin) > > > > Further pre-releases will follow as needed. If all goes well, 0.10.17 > > release should happen later this month when GStreamer core/-base 0.10.32 > > are out, as per > > http://gstreamer.freedesktop.org/wiki/ReleasePlanning2011 > > > > Edward > > > > > > ------------------------------------------------------------------------------ > > Protect Your Site and Customers from Malware Attacks > > Learn about various malware tactics and how to avoid them. Understand > > malware threats, the impact they can have on your business, and how you > > can protect your company and customers by using code signing. > > 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 > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 timothejahan at gmail.com Thu Jan 13 12:55:18 2011 From: timothejahan at gmail.com (timothe jahan) Date: Thu, 13 Jan 2011 12:55:18 +0100 Subject: [gst-devel] What is 'streaming thread'? In-Reply-To: <1294916089384-3215629.post@n4.nabble.com> References: <1294907094576-3215407.post@n4.nabble.com> <1294911875.28313.6.camel@zingle> <1294916089384-3215629.post@n4.nabble.com> Message-ID: Hi, After reading the doc on the thread (few weeks ago) I still had a question on multithreading for which I did not found the answer in the doc. Does some plugin have their own multithreading capability ? My interest is for the x264 encoder. Multithreading of the h264 encoding process is a mandatory thing as soon as you do live encoding. If anybody knows about this... Regards, Timothe On Thu, Jan 13, 2011 at 11:54 AM, wl2776 wrote: > > > Parveen Kumar Jain wrote: > > > > How one can control these no. of threads creation on a single > > pipeline ? or if I put it in another way, how these threads get > > created in gstreamer ? > > > > This is covered in the docs. > > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/chapter-threads.html > > http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/part-messages.txt > - GST_MESSAGE_STREAM_STATUS > > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/What-is-streaming-thread-tp3215407p3215629.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 shamun.toha at gmail.com Thu Jan 13 13:11:48 2011 From: shamun.toha at gmail.com (Shamun toha md) Date: Thu, 13 Jan 2011 13:11:48 +0100 Subject: [gst-devel] Mac - not yet ? Message-ID: Hello, How can i test in mac osx ? Linux and Windows was working, but is there any specific package for Mac ? Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From tbrendt at googlemail.com Thu Jan 13 13:23:10 2011 From: tbrendt at googlemail.com (Thorsten Brendt) Date: Thu, 13 Jan 2011 13:23:10 +0100 Subject: [gst-devel] RTP audio stream to filesink, issue with packet loss In-Reply-To: <1294911988.28313.8.camel@zingle> References: <1294874197.25646.0.camel@zingle> <1294911988.28313.8.camel@zingle> Message-ID: On Thu, Jan 13, 2011 at 10:46 AM, Tim-Philipp M?ller wrote: > On Thu, 2011-01-13 at 03:10 +0100, Thorsten Brendt wrote: > >> > Tried adding an audiorate element? >> >> Do you mean a caps-filter element like this?: >> >> 'audio/x-raw-int, width=16, depth=16, rate=8000, endianness=1234, >> signed=(boolean)true, channels=1' > > No, I meant an "audiorate" element, which is part of gst-plugins-base. > > Factory Details: > ?Long name: ? ?Audio rate adjuster > ?Class: ? ? ? ?Filter/Effect/Audio > ?Description: ?Drops/duplicates/adjusts timestamps on audio samples to > make a perfect stream Ah, I should've looked at the gstreamer sources instead of searching for "gstreamer audiorate"... lazy me! Anyway, this was a great pointer! Just adding audiorate to the pipeline didn't help. However, I could trace down the issue to the jitter buffer: I needed to set do-lost=true to ask the jitter buffer element to send packet-lost events downstream. Now this combination works just like I expected. For the sake of completeness here's the pipeline I used: filesrc location=foobar.pcap ! pcapparse ! "application/x-rtp, payload=0, clock-rate=8000" ! gstrtpjitterbuffer do-lost=true ! rtppcmudepay ! mulawdec ! audiorate ! wavenc ! filesink location=foobar.wav Tim, thank you very much for your help! Cheers, Thorsten From wl2776 at gmail.com Thu Jan 13 14:51:26 2011 From: wl2776 at gmail.com (wl2776) Date: Thu, 13 Jan 2011 05:51:26 -0800 (PST) Subject: [gst-devel] What is 'streaming thread'? In-Reply-To: References: <1294907094576-3215407.post@n4.nabble.com> <1294911875.28313.6.camel@zingle> <1294916089384-3215629.post@n4.nabble.com> Message-ID: <1294926686375-3215875.post@n4.nabble.com> timothe jahan wrote: > > After reading the doc on the thread (few weeks ago) I still had a question > on multithreading for which I did not found the answer in the doc. > Does some plugin have their own multithreading capability ? > It depends on the plugin. For example, mpeg2 PS demuxer has. timothe jahan wrote: > > My interest is for the x264 encoder. Multithreading of the h264 encoding > process is a mandatory thing as soon as you do live encoding. > You could search the source code for GstTask or GThread. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-is-streaming-thread-tp3215407p3215875.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Thu Jan 13 15:14:19 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 13 Jan 2011 14:14:19 +0000 Subject: [gst-devel] What is 'streaming thread'? In-Reply-To: References: <1294907094576-3215407.post@n4.nabble.com> <1294911875.28313.6.camel@zingle> <1294916089384-3215629.post@n4.nabble.com> Message-ID: <1294928059.28313.13.camel@zingle> On Thu, 2011-01-13 at 12:55 +0100, timothe jahan wrote: > Does some plugin have their own multithreading capability ? > My interest is for the x264 encoder. Multithreading of the h264 > encoding process is a mandatory thing as soon as you do live encoding. This is done at the x264 library level and hidden from both GStreamer and the application (but configurable of course, see gst-inspect x264enc). Cheers -Tim From timothejahan at gmail.com Thu Jan 13 15:37:26 2011 From: timothejahan at gmail.com (timothe jahan) Date: Thu, 13 Jan 2011 15:37:26 +0100 Subject: [gst-devel] What is 'streaming thread'? In-Reply-To: <1294928059.28313.13.camel@zingle> References: <1294907094576-3215407.post@n4.nabble.com> <1294911875.28313.6.camel@zingle> <1294916089384-3215629.post@n4.nabble.com> <1294928059.28313.13.camel@zingle> Message-ID: Thank you both for your fast and accurate answers Regards, Timothe On Thu, Jan 13, 2011 at 3:14 PM, Tim-Philipp M?ller wrote: > On Thu, 2011-01-13 at 12:55 +0100, timothe jahan wrote: > > > Does some plugin have their own multithreading capability ? > > My interest is for the x264 encoder. Multithreading of the h264 > > encoding process is a mandatory thing as soon as you do live encoding. > > This is done at the x264 library level and hidden from both GStreamer > and the application (but configurable of course, see gst-inspect > x264enc). > > Cheers > -Tim > > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 nvineeth at gmail.com Thu Jan 13 16:14:36 2011 From: nvineeth at gmail.com (vineeth) Date: Thu, 13 Jan 2011 20:44:36 +0530 Subject: [gst-devel] reg gst_check_run_suite Message-ID: Hi, Though it is a very minor problem since the function gst_check_run_suite runs only once per testcase, I think there is a memory leak ingst_check_run_suite (gstreamer\libs\gst\check\gstcheck.c) for the local variable xmlfilename, which needs to g_free()'d. Should I raise a bug request? --vineeth -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Thu Jan 13 16:44:42 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 13 Jan 2011 15:44:42 +0000 Subject: [gst-devel] reg gst_check_run_suite In-Reply-To: References: Message-ID: <1294933482.28313.20.camel@zingle> On Thu, 2011-01-13 at 20:44 +0530, vineeth wrote: > Though it is a very minor problem since the > function gst_check_run_suite runs only once per testcase, I think > there is a memory leak in gst_check_run_suite (gstreamer\libs\gst > \check\gstcheck.c) for the local variable xmlfilename, which needs to > g_free()'d. > Should I raise a bug request? Usually it's best to just file a bug. I've already fixed this locally now though, and will push it after the release. Cheers -Tim From malloblenne at gmail.com Thu Jan 13 16:49:24 2011 From: malloblenne at gmail.com (Mauro Brenna) Date: Thu, 13 Jan 2011 16:49:24 +0100 Subject: [gst-devel] Is RTSPU implemented in gstreamer? Can I use RTSP over UDP? Message-ID: Hello, I am implementing a server which streams video using RTP. To read the stream from a client I use gstreamer. For the session protocol I would like to use RTSP and I am reading the RFC 2326. I read in a passage that RTSP should be sent using TCP while you should call RTSPU for UDP. I'll cite from the RFC: " The scheme rtsp requires that commands are issued via a reliable protocol (within the Internet, TCP), while the scheme rtspu identifies an unreliable protocol (within the Internet, UDP)." I would like to know if the implementation of the RTSP provided by gstreamer uses TCP or UDP. Is it possible to use RTSPU with gstreamer? Anybody knows if RTPS over UDP or RTPSU is supported in other media player like vlc? Thanks, Mauro From wim.taymans at gmail.com Thu Jan 13 16:58:37 2011 From: wim.taymans at gmail.com (Wim Taymans) Date: Thu, 13 Jan 2011 16:58:37 +0100 Subject: [gst-devel] Is RTSPU implemented in gstreamer? Can I use RTSP over UDP? In-Reply-To: References: Message-ID: <1294934317.1875.48.camel@meany> On Thu, 2011-01-13 at 16:49 +0100, Mauro Brenna wrote: > Hello, > > I am implementing a server which streams video using RTP. To read the > stream from a client I use gstreamer. > For the session protocol I would like to use RTSP and I am reading the RFC 2326. > > I read in a passage that RTSP should be sent using TCP while you > should call RTSPU for UDP. > I'll cite from the RFC: > > " The scheme rtsp requires that commands are issued via a reliable > protocol (within the Internet, TCP), while the scheme rtspu identifies > an unreliable protocol (within the Internet, UDP)." > > I would like to know if the implementation of the RTSP provided by > gstreamer uses TCP or UDP. Is it possible to use RTSPU with > gstreamer? Anybody knows if RTPS over UDP or RTPSU is supported in > other media player like vlc? rtspsrc supports data transport over TCP and UDP and RTSP over tcp. I am not aware of any player doing the RTSP requests over UDP nor is it clear from the spec how this is supposed to work, or how a server should support it. I don't know why that part is in the spec or what it means. rtspsrc supportes rtspu:// protocols but interpretes it that the RTP transport is only done over UDP unicast or multicast. VLC doesn't seem to understand the rtspu protocol. Wim > > Thanks, > > Mauro > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 gang.a.hu at intel.com Thu Jan 13 17:09:40 2011 From: gang.a.hu at intel.com (Hu, Gang A) Date: Fri, 14 Jan 2011 00:09:40 +0800 Subject: [gst-devel] Pipeline fail to change to playing when use two pads. Message-ID: <1A42CE6F5F474C41B63392A5F80372B231E76826@shsmsx501.ccr.corp.intel.com> Hi, I am developing a three pads camera source element. I created this element based on GstBaseSrc, and add another two pads to GstBaseSrc and created a new basesrc in my camera source. My currently design is the vfsrc pad always has data flow. Another copy of data flow will be image or video depends on the mode setting to the camera source. This pipeline is working now. gst-launch -ev mfldv4l2camsrc num-buffers=2 .vfsrc ! video/x-raw-yuv,width=640,height=480 ! sdlvideosink But if I want use the imgpad, gst-launch -ev mfldv4l2camsrc num-buffers=2 name=aa .imgsrc ! video/x-raw-yuv,width=640,height=480 ! fakesink aa.vfsrc ! video/x-raw-yuv,width=640,height=480 ! sdlvideosink the pipeline will stop at waiting for playing. gst_base_sink_wait_preroll: waiting in preroll for flush or PLAYING cat What is the potential issue? This is my source code repository: http://meego.gitorious.org/maemo-multimedia/mfldv4l2camsrc/commits/camerabin2 This is the some of the log of the basesrc anb basesink. gst_cam_base_src_init: creating vfsrc pad gst_cam_base_src_init: creating imgsrc pad gst_cam_base_src_init: creating vidsrc pad gst_cam_base_src_init: setting functions on src pad gst_cam_base_src_init: adding three src pads gst_cam_base_src_init: init done gst_cam_base_src_set_property: Setting pad mode: 1 gst_cam_base_src_set_pad_mode: Set pad mode to 1 gst_base_sink_change_state: READY to PAUSED gst_base_sink_change_state: doing async state change gst_base_sink_pad_activate: Trying pull mode first gst_base_sink_pad_activate: pull mode disabled gst_base_sink_pad_activate: Falling back to push mode gst_base_sink_pad_activate: Success activating push mode gst_base_sink_change_state: READY to PAUSED gst_base_sink_change_state: doing async state change gst_base_sink_pad_activate: Trying pull mode first gst_base_sink_pad_activate: pull mode disabled gst_base_sink_pad_activate: Falling back to push mode gst_base_sink_pad_activate: Success activating push mode gst_cam_base_src_activate_push: vidsrc activating in push mode gst_cam_base_src_start: starting source of pad vidsrc gst_cam_base_src_start: format: time, have size: 1, size: 18446744073709551615, duration: -1 gst_cam_base_src_start: is seekable: 0 gst_cam_base_src_start: is random_access: 0 gst_cam_base_src_negotiate: negotiate of pad vidsrc gst_cam_base_src_activate_push: imgsrc activating in push mode gst_cam_base_src_start: starting source of pad imgsrc gst_cam_base_src_start: format: time, have size: 1, size: 18446744073709551615, duration: -1 gst_cam_base_src_start: is seekable: 0 gst_cam_base_src_start: is random_access: 0 gst_cam_base_src_negotiate: negotiate of pad imgsrc gst_cam_base_src_activate_push: vfsrc activating in push mode gst_cam_base_src_start: starting source of pad vfsrc gst_cam_base_src_start: format: time, have size: 1, size: 18446744073709551615, duration: -1 gst_cam_base_src_start: is seekable: 0 gst_cam_base_src_start: is random_access: 0 gst_cam_base_src_negotiate: negotiate of pad vfsrc gst_cam_base_src_perform_seek: doing seek: (NULL) gst_cam_base_src_perform_seek: seek with seqnum 19 gst_cam_base_src_perform_seek: segment configured from 0 to -1, position 0 gst_cam_base_src_perform_seek: Sending newsegment from 0 to -1 gst_cam_base_src_loop: next_ts 99:99:99.999999999 size 4096 gst_cam_base_src_wait_playing: live source waiting for running state gst_base_sink_query_latency: we are not yet ready for LATENCY query gst_base_sink_query_latency: latency query failed and we are live gst_base_sink_query: query latency returns 0 gst_base_sink_query_latency: we are not yet ready for LATENCY query gst_base_sink_query_latency: latency query failed but we are not live gst_base_sink_query_latency: latency query: live: 0, have_latency 0, upstream: 0, min 0:00:00.000000000, max 99:99:99.999999999 gst_base_sink_query: query latency returns 1 gst_cam_base_src_change_state: PAUSED->PLAYING gst_cam_base_src_set_playing: unschedule clock gst_cam_base_src_set_playing: live running 1 gst_cam_base_src_set_playing: unlock stop gst_cam_base_src_set_playing: signal gst_cam_base_src_wait_playing: live source unlocked gst_cam_base_src_get_range: calling create offset 18446744073709551615 length 4096, time 0 gst_cam_base_src_do_sync: no latency needed, live 1, sync 0 gst_cam_base_src_do_sync: startup timestamp: 0:00:00.299494749, running_time 0:00:00.332892825 gst_cam_base_src_do_sync: no timestamp offset needed gst_cam_base_src_do_sync: no sync needed gst_cam_base_src_get_range: buffer ok gst_base_sink_event: received event 0x8f49f68 newsegment event from 'vfsrc' at time 99:99:99.999999999: GstEventNewsegment, update=(boolean)false, rate=(double)1, applied-rate=(double)1, format=(GstFormat)GST_FORMAT_TIME, start=(gint64)0, stop=(gint64)-1, position=(gint64)0; gst_base_sink_event: newsegment 0x8f49f68 gst_base_sink_configure_segment: configured NEWSEGMENT update 0, rate 1.000000, applied rate 1.000000, format GST_FORMAT_TIME, 0:00:00.000000000 -- 99:99:99.999999999, time 0:00:00.000000000, accum 0:00:00.000000000 gst_base_sink_queue_object_unlocked: now 0 prerolled items gst_base_sink_queue_object_unlocked: need more preroll data 0 <= 0 gst_base_sink_chain_unlocked: got times start: 0:00:00.299494749, end: 0:00:00.332828082 gst_base_sink_queue_object_unlocked: now 1 prerolled items gst_base_sink_preroll_object: prerolling object 0x8ddf300 gst_base_sink_preroll_object: preroll buffer 0:00:00.299494749 st_base_sink_set_last_buffer_unlocked: setting last buffer to 0x8ddf300 gst_base_sink_commit_state: commiting state to PAUSED gst_base_sink_commit_state: posting PAUSED state change message gst_base_sink_commit_state: posting async-done message gst_base_sink_queue_object_unlocked: rendering queued object 0x8f49f68 gst_base_sink_do_sync: non syncable object 0x8f49f68 gst_base_sink_render_object: rendering event 0x8f49f68, type newsegment gst_base_sink_render_object: Got seqnum #19 gst_base_sink_configure_segment: configured NEWSEGMENT update 0, rate 1.000000, applied rate 1.000000, format GST_FORMAT_TIME, 0:00:00.000000000 -- 99:99:99.999999999, time 0:00:00.000000000, accum 0:00:00.000000000 gst_base_sink_render_object: object unref after render 0x8f49f68 gst_base_sink_get_sync_times: got times start: 0:00:00.299494749, stop: 0:00:00.332828082, do_sync 1 gst_base_sink_do_preroll: prerolling object 0x8ddf300 gst_base_sink_preroll_object: prerolling object 0x8ddf300 gst_base_sink_wait_preroll: waiting in preroll for flush or PLAYING 3:gst_cam_base_src_event_handler: Hanlde event of pad vfsrc 1:gst_cam_base_src_default_event: handle event navigation event from 'src' at time 99:99:99.999999999: application/x-gst-navigation, event=(string)mouse-move, button=(int)0, pointer_x=(double)639, pointer_y=(double)179; gst_cam_base_src_event_handler: Hanlde event of pad vfsrc gst_cam_base_src_change_state: PLAYING->PAUSED gst_cam_base_src_set_playing: unlock gst_cam_base_src_set_playing: unschedule clock gst_cam_base_src_set_playing: live running 0 gst_base_sink_set_flushing: flushing out data thread, need preroll to TRUE gst_base_sink_preroll_queue_flush: flushing queue 0x8e8c0a0 gst_base_sink_wait_preroll: preroll interrupted because of flush gst_base_sink_do_preroll: preroll failed -2 gst_base_sink_do_sync: preroll failed gst_base_sink_render_object: do_sync returned wrong-state gst_base_sink_render_object: object unref after render 0x8ddf300 gst_cam_base_src_loop: pausing after gst_pad_push() to vfpad = wrong-state gst_cam_base_src_loop: pausing task, reason wrong-state st_base_sink_set_last_buffer_unlocked: setting last buffer to (nil) gst_base_sink_change_state: PAUSED to READY, don't need_preroll gst_base_sink_set_flushing: flushing out data thread, need preroll to TRUE gst_base_sink_preroll_queue_flush: flushing queue 0x8e86288 gst_base_sink_change_state: PAUSED to READY, posting async-done gst_cam_base_src_activate_push: Deactivating in push mode gst_cam_base_src_activate_push: Deactivating in push mode gst_cam_base_src_activate_push: Deactivating in push mode gst_cam_base_src_stop: stopping source gst_cam_base_src_change_state: Sending EOS event Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Thu Jan 13 17:17:03 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Thu, 13 Jan 2011 18:17:03 +0200 Subject: [gst-devel] kenburns plugin In-Reply-To: References: <4D2B7145.7010503@brooks.nu> <4D2DC7C2.1000807@hora-obscura.de> <4D2E3DDD.8030101@brooks.nu> Message-ID: <4D2F257F.6070305@hora-obscura.de> Hi matt & Lane, On 13.01.2011 08:27, Gruenke, Matt wrote: > I'd recommend making a custom bin to encapsulate the full controls with > a set of simplified parameters. You could add a controller interface to > kenburns, and then make a custom bin called kenburns_easy, for instance, > that provides a simplified interface by using an internal controller. > > One example of this is rtspsrc, which is a bin. It encapsulates a lot > of other elements that one can also use in a stand-alone fashion. > Not necessarily a bin, but yes. What about having a kenburns plugins with several elements, all sharing the same codebase. One can provide single controlable zoom, x/y-center props and the other has what you have now. Stefan > Matt > > > -----Original Message----- > From: Lane Brooks [mailto:lane at brooks.nu] > Sent: Wednesday, January 12, 2011 18:49 > To: Discussion of the development of GStreamer > Subject: Re: [gst-devel] kenburns plugin > > On 01/12/2011 08:24 AM, Stefan Kost wrote: >> On 10.01.2011 22:51, Lane Brooks wrote: > > > Actually, I just had another thought. Currently I have a few different > user selectable internal controllers like the velocity ramping one > described above. Would it work to add another option that lets the user > select to use an external controller instead? Then for the simple use > case, you just set "zoom1", "xcenter1", "ycenter1", "zoom2", "xcenter2", > > "ycenter2", and "duration". For the more advanced usage, you instead put > > a controller on "zoom1", "xcenter1", and "ycenter1" and select to > disable the internal controller. > > Lane > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 Victor.Johns at technicolor.com Thu Jan 13 18:13:50 2011 From: Victor.Johns at technicolor.com (Johns Victor) Date: Thu, 13 Jan 2011 12:13:50 -0500 Subject: [gst-devel] is it possible to set properties on request pads ( outside of using the caps functionality) Message-ID: <9A2E6A78FABA774FB6CD8907DA5D118A325E4FE62B@INDYSMAILMB04.am.thmulti.com> All, We have an application that would require to have a set of request pads. The properties of said pads need to be specified explicitly ( I believe in finer granularity than is possible using the gst_pad_set_caps()/ pad template functionality. As gstpads derive from Gobject, one thought is to extend the GstPad object w/in the element that is being developed. (say GstExtendedPad, which derives from GstPad) 1) Is this the correct way to approach solving this problem 2) If so, I believe from a c/c++ programming standpoint, the process of setting the properties on the request pads should be fairly straighthforward, my concern is how this manifests when using gst-launch i.e. is it possible to : a) request a pad of type GstExtendedPad and alsoset the properties of this pad using gst-launch b) are there any implications to the above approach to be aware of Regards Victor -------------- next part -------------- An HTML attachment was scrubbed... URL: From praveen.pande1 at gmail.com Thu Jan 13 19:05:17 2011 From: praveen.pande1 at gmail.com (praveen pandey) Date: Thu, 13 Jan 2011 23:35:17 +0530 Subject: [gst-devel] Pipeline fail to change to playing when use two pads. In-Reply-To: <1A42CE6F5F474C41B63392A5F80372B231E76826@shsmsx501.ccr.corp.intel.com> References: <1A42CE6F5F474C41B63392A5F80372B231E76826@shsmsx501.ccr.corp.intel.com> Message-ID: connect 2 'queue' elements after camera element. BR, Praveen Pandey On Thu, Jan 13, 2011 at 9:39 PM, Hu, Gang A wrote: > Hi, > I am developing a three pads camera source element. I created this element > based on GstBaseSrc, and add another two pads to GstBaseSrc and created a > new basesrc in my camera source. > My currently design is the vfsrc pad always has data flow. Another copy of > data flow will be image or video depends on the mode setting to the camera > source. > > This pipeline is working now. > gst-launch -ev mfldv4l2camsrc num-buffers=2 .vfsrc ! > video/x-raw-yuv,width=640,height=480 ! sdlvideosink > > But if I want use the imgpad, > gst-launch -ev mfldv4l2camsrc num-buffers=2 name=aa .imgsrc ! > video/x-raw-yuv,width=640,height=480 ! fakesink aa.vfsrc ! > video/x-raw-yuv,width=640,height=480 ! sdlvideosink > > the pipeline will stop at waiting for playing. > gst_base_sink_wait_preroll: waiting in preroll for flush or > PLAYING > cat > What is the potential issue? > > This is my source code repository: > * > http://meego.gitorious.org/maemo-multimedia/mfldv4l2camsrc/commits/camerabin2 > * > This is the some of the log of the basesrc anb basesink. > > gst_cam_base_src_init: creating vfsrc pad > gst_cam_base_src_init: creating imgsrc pad > gst_cam_base_src_init: creating vidsrc pad > gst_cam_base_src_init: setting functions on src pad > gst_cam_base_src_init: adding three src pads > gst_cam_base_src_init: init done > gst_cam_base_src_set_property: Setting pad mode: 1 > gst_cam_base_src_set_pad_mode: Set pad mode to 1 > gst_base_sink_change_state: READY to PAUSED > gst_base_sink_change_state: doing async state change > gst_base_sink_pad_activate: Trying pull mode first > gst_base_sink_pad_activate: pull mode disabled > gst_base_sink_pad_activate: Falling back to push mode > gst_base_sink_pad_activate: Success activating push mode > gst_base_sink_change_state: READY to PAUSED > gst_base_sink_change_state: doing async state change > gst_base_sink_pad_activate: Trying pull mode first > gst_base_sink_pad_activate: pull mode disabled > gst_base_sink_pad_activate: Falling back to push mode > gst_base_sink_pad_activate: Success activating push mode > gst_cam_base_src_activate_push: vidsrc activating in push > mode > gst_cam_base_src_start: starting source of pad vidsrc > gst_cam_base_src_start: format: time, have size: 1, size: > 18446744073709551615, duration: -1 > gst_cam_base_src_start: is seekable: 0 > gst_cam_base_src_start: is random_access: 0 > gst_cam_base_src_negotiate: negotiate of pad vidsrc > gst_cam_base_src_activate_push: imgsrc activating in push > mode > gst_cam_base_src_start: starting source of pad imgsrc > gst_cam_base_src_start: format: time, have size: 1, size: > 18446744073709551615, duration: -1 > gst_cam_base_src_start: is seekable: 0 > gst_cam_base_src_start: is random_access: 0 > gst_cam_base_src_negotiate: negotiate of pad imgsrc > gst_cam_base_src_activate_push: vfsrc activating in push > mode > gst_cam_base_src_start: starting source of pad vfsrc > gst_cam_base_src_start: format: time, have size: 1, size: > 18446744073709551615, duration: -1 > gst_cam_base_src_start: is seekable: 0 > gst_cam_base_src_start: is random_access: 0 > gst_cam_base_src_negotiate: negotiate of pad vfsrc > > gst_cam_base_src_perform_seek: doing seek: (NULL) > gst_cam_base_src_perform_seek: seek with seqnum 19 > gst_cam_base_src_perform_seek: segment configured from 0 > to -1, position 0 > gst_cam_base_src_perform_seek: Sending newsegment from 0 > to -1 > gst_cam_base_src_loop: next_ts 99:99:99.999999999 size > 4096 > gst_cam_base_src_wait_playing: live source waiting for > running state > gst_base_sink_query_latency: we are not yet ready for > LATENCY query > gst_base_sink_query_latency: latency query failed and we are > live > gst_base_sink_query: query latency returns 0 > gst_base_sink_query_latency: we are not yet ready for LATENCY > query > gst_base_sink_query_latency: latency query failed but we are not > live > gst_base_sink_query_latency: latency query: live: 0, > have_latency 0, upstream: 0, min 0:00:00.000000000, max 99:99:99.999999999 > gst_base_sink_query: query latency returns 1 > gst_cam_base_src_change_state: PAUSED->PLAYING > gst_cam_base_src_set_playing: unschedule clock > gst_cam_base_src_set_playing: live running 1 > gst_cam_base_src_set_playing: unlock stop > gst_cam_base_src_set_playing: signal > gst_cam_base_src_wait_playing: live source unlocked > gst_cam_base_src_get_range: calling create offset > 18446744073709551615 length 4096, time 0 > gst_cam_base_src_do_sync: no latency needed, live 1, sync > 0 > gst_cam_base_src_do_sync: startup timestamp: > 0:00:00.299494749, running_time 0:00:00.332892825 > gst_cam_base_src_do_sync: no timestamp offset needed > gst_cam_base_src_do_sync: no sync needed > gst_cam_base_src_get_range: buffer ok > gst_base_sink_event: received event 0x8f49f68 newsegment > event from 'vfsrc' at time 99:99:99.999999999: GstEventNewsegment, > update=(boolean)false, rate=(double)1, applied-rate=(double)1, > format=(GstFormat)GST_FORMAT_TIME, start=(gint64)0, stop=(gint64)-1, > position=(gint64)0; > gst_base_sink_event: newsegment 0x8f49f68 > gst_base_sink_configure_segment: configured NEWSEGMENT > update 0, rate 1.000000, applied rate 1.000000, format GST_FORMAT_TIME, > 0:00:00.000000000 -- 99:99:99.999999999, time 0:00:00.000000000, accum > 0:00:00.000000000 > gst_base_sink_queue_object_unlocked: now 0 prerolled items > gst_base_sink_queue_object_unlocked: need more preroll data > 0 <= 0 > gst_base_sink_chain_unlocked: got times start: > 0:00:00.299494749, end: 0:00:00.332828082 > gst_base_sink_queue_object_unlocked: now 1 prerolled items > gst_base_sink_preroll_object: prerolling object 0x8ddf300 > gst_base_sink_preroll_object: preroll buffer > 0:00:00.299494749 > st_base_sink_set_last_buffer_unlocked: setting last buffer > to 0x8ddf300 > gst_base_sink_commit_state: commiting state to PAUSED > gst_base_sink_commit_state: posting PAUSED state change > message > gst_base_sink_commit_state: posting async-done message > gst_base_sink_queue_object_unlocked: rendering queued object > 0x8f49f68 > gst_base_sink_do_sync: non syncable object 0x8f49f68 > gst_base_sink_render_object: rendering event 0x8f49f68, type > newsegment > gst_base_sink_render_object: Got seqnum #19 > gst_base_sink_configure_segment: configured NEWSEGMENT > update 0, rate 1.000000, applied rate 1.000000, format GST_FORMAT_TIME, > 0:00:00.000000000 -- 99:99:99.999999999, time 0:00:00.000000000, accum > 0:00:00.000000000 > gst_base_sink_render_object: object unref after render > 0x8f49f68 > gst_base_sink_get_sync_times: got times start: > 0:00:00.299494749, stop: 0:00:00.332828082, do_sync 1 > gst_base_sink_do_preroll: prerolling object 0x8ddf300 > gst_base_sink_preroll_object: prerolling object 0x8ddf300 > gst_base_sink_wait_preroll: waiting in preroll for flush or > PLAYING > 3:gst_cam_base_src_event_handler: Hanlde event of pad > vfsrc > 1:gst_cam_base_src_default_event: handle event navigation > event from 'src' at time 99:99:99.999999999: application/x-gst-navigation, > event=(string)mouse-move, button=(int)0, pointer_x=(double)639, > pointer_y=(double)179; > gst_cam_base_src_event_handler: Hanlde event of pad vfsrc > > gst_cam_base_src_change_state: PLAYING->PAUSED > gst_cam_base_src_set_playing: unlock > gst_cam_base_src_set_playing: unschedule clock > gst_cam_base_src_set_playing: live running 0 > gst_base_sink_set_flushing: flushing out data thread, need > preroll to TRUE > gst_base_sink_preroll_queue_flush: flushing queue 0x8e8c0a0 > *gst_base_sink_wait_preroll: preroll interrupted because of > flush* > *gst_base_sink_do_preroll: preroll failed -2* > *gst_base_sink_do_sync: preroll failed* > gst_base_sink_render_object: do_sync returned wrong-state > gst_base_sink_render_object: object unref after render > 0x8ddf300 > gst_cam_base_src_loop: pausing after gst_pad_push() to > vfpad = wrong-state > gst_cam_base_src_loop: pausing task, reason wrong-state > st_base_sink_set_last_buffer_unlocked: setting last buffer > to (nil) > gst_base_sink_change_state: PAUSED to READY, don't > need_preroll > gst_base_sink_set_flushing: flushing out data thread, need > preroll to TRUE > gst_base_sink_preroll_queue_flush: flushing queue 0x8e86288 > gst_base_sink_change_state: PAUSED to READY, posting async-done > gst_cam_base_src_activate_push: Deactivating in push mode > gst_cam_base_src_activate_push: Deactivating in push mode > gst_cam_base_src_activate_push: Deactivating in push mode > gst_cam_base_src_stop: stopping source > gst_cam_base_src_change_state: Sending EOS event > > Thanks. > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 Jan 13 19:14:08 2011 From: praveen.pande1 at gmail.com (praveen pandey) Date: Thu, 13 Jan 2011 23:44:08 +0530 Subject: [gst-devel] Decoded video frame timings In-Reply-To: References: Message-ID: There could be two possible reasons for this. 1) your stream might be a corrupted one 2) Any element in the pipeline is multifunctional (parser and/or decoder) you can verify this using qtdemux and ffdec_h264 BR, Praveen Pandey On Wed, Jan 12, 2011 at 1:55 PM, Darren Staples wrote: > Hi, > > I'm decoding video frames (various file formats and codecs) and converting > them into YUV420 format so that I can perform image processing on them. > And I have the commercial Fluendo codec pack installed. > > The pipeline looks like this: uridecodebin -> queue -> ffmpegcolorspace -> > tee -> queue -> appsink > YUV420 conversion is done in the ffmpegcolorspace component, and the other > source pad of the tee is for a future feature. > This is built up dynamically in C, not on a command line. > > My problem is that I need to extract the frame timings for each of the > video frames being processed, and I'm seeing a large variance in this. > For example, one of my video files is a H.264 MP4, 82s long, 2057 frames > and plays at 25 fps, so the average frame interval is 40ms. > When I recover a frame from the appsink component I use the > gst_element_query_position() function to get the timing for that frame. > And what I'm seeing is that the timing is highly variable, even from one > run to the next. > > e.g. the time between frames for the first 32 frames for run 1: > 0: 1020226us > 1: 0us > 2: 0us > 3: 0us > 4: 0us > 5: 0us > 6: 0us > 7: 0us > 8: 0us > 9: 0us > 10: 0us > 11: 0us > 12: 0us > 13: 0us > 14: 0us > 15: 0us > 16: 0us > 17: 0us > 18: 0us > 19: 0us > 20: 0us > 21: 0us > 22: 0us > 23: 0us > 24: 0us > 25: 0us > 26: 19774us > 27: 40000us > 28: 40000us > 29: 1022947us > 30: 46440us > 31: 23220us > > the first 32 frames for run 2: > 0: 53151us > 1: 0us > 2: 26849us > 3: 40000us > 4: 40000us > 5: 40000us > 6: 40000us > 7: 40000us > 8: 40000us > 9: 40000us > 10: 40000us > 11: 40000us > 12: 40000us > 13: 40000us > 14: 40000us > 15: 40000us > 16: 40000us > 17: 40000us > 18: 40000us > 19: 40000us > 20: 1011428us > 21: 46440us > 22: 46440us > 23: 23220us > 24: 46440us > 25: 46440us > 26: 46440us > 27: 23219us > 28: 46440us > 29: 46440us > 30: 46440us > 31: 23220us > > Why do so many of the frames in the first run above have zero time delta > between them ? > Is this a consequence of the file being in H.264/mp4 format ? And if, so is > there an easy way to 'smooth' out the frame timings ? > Or would you suggest I do this 'manually' by approximating/averaging the > timing based on the duration and no. of frames ? > > Thanks ! > Darren > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 Jan 13 19:18:17 2011 From: praveen.pande1 at gmail.com (praveen pandey) Date: Thu, 13 Jan 2011 23:48:17 +0530 Subject: [gst-devel] matroska In-Reply-To: <1293462920665-3165093.post@n4.nabble.com> References: <1293462920665-3165093.post@n4.nabble.com> Message-ID: use queue element before audio and video decoders BR, Praveen Pandey On Mon, Dec 27, 2010 at 8:45 PM, stojnovsky wrote: > > 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. > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any > company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 lazyt at gmx.net Thu Jan 13 20:31:03 2011 From: lazyt at gmx.net (Thomas Loewe) Date: Thu, 13 Jan 2011 11:31:03 -0800 (PST) Subject: [gst-devel] Mac - not yet ? In-Reply-To: References: Message-ID: <1294947063074-3216495.post@n4.nabble.com> You can build GStreamer via http://www.macports.org/ macports , but it takes some hours... -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Mac-not-yet-tp3215721p3216495.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From braunsquared at gmail.com Thu Jan 13 20:37:43 2011 From: braunsquared at gmail.com (Timothy Braun) Date: Thu, 13 Jan 2011 14:37:43 -0500 Subject: [gst-devel] Mac - not yet ? In-Reply-To: <1294947063074-3216495.post@n4.nabble.com> References: <1294947063074-3216495.post@n4.nabble.com> Message-ID: This is another option: http://archive.itee.uq.edu.au/~davel/gstreamer/ Prebuilt installer package. Also has instructions for building the latest release on OSX. -Tim On Jan 13, 2011, at 2:31 PM, Thomas Loewe wrote: > > You can build GStreamer via http://www.macports.org/ macports , but it takes > some hours... > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Mac-not-yet-tp3215721p3216495.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 t.i.m at zen.co.uk Thu Jan 13 21:00:11 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 13 Jan 2011 20:00:11 +0000 Subject: [gst-devel] matroska In-Reply-To: <1293462920665-3165093.post@n4.nabble.com> References: <1293462920665-3165093.post@n4.nabble.com> Message-ID: <1294948811.28313.22.camel@zingle> On Mon, 2010-12-27 at 07:15 -0800, stojnovsky wrote: > gst-launch-0.10 -v filesrc location=bilqna.webm ! matroskademux name=d > d.audio_00 ! vorbisdec ! audioconvert ! pulsesink d.video_00 ! vp8dec ! > xvimagesink Try: gst-launch-0.10 playbin2 uri=file:///path/to/bilqna.webm (the problem with your pipeline are the missing queues after the demuxer, as already pointed out by another poster) Cheers -Tim From t.i.m at zen.co.uk Thu Jan 13 21:04:54 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 13 Jan 2011 20:04:54 +0000 Subject: [gst-devel] Decoded video frame timings In-Reply-To: References: Message-ID: <1294949094.28313.25.camel@zingle> On Wed, 2011-01-12 at 08:25 +0000, Darren Staples wrote: Hi, (is this a double-post?) > I'm decoding video frames (various file formats and codecs) and > converting them into YUV420 format so that I can perform image > processing on them. (..) > My problem is that I need to extract the frame timings (..), > and I'm seeing a large variance in this. (..) > When I recover a frame from the appsink component I use the > gst_element_query_position() function to get the timing for that > frame. Try looking at the GST_BUFFER_TIMESTAMP (buffer) instead. Cheers -Tim From antoine at nagafix.co.uk Thu Jan 13 22:21:43 2011 From: antoine at nagafix.co.uk (Antoine Martin) Date: Fri, 14 Jan 2011 04:21:43 +0700 Subject: [gst-devel] Mac - not yet ? In-Reply-To: <1294947063074-3216495.post@n4.nabble.com> References: <1294947063074-3216495.post@n4.nabble.com> Message-ID: <4D2F6CE7.3010302@nagafix.co.uk> On 01/14/2011 02:31 AM, Thomas Loewe wrote: > > You can build GStreamer via http://www.macports.org/ macports , but it takes > some hours... You can also use jhbuild (from gtk-osx), it does take quite a while... Antoine From Jorney_Dong at asus.com Fri Jan 14 02:26:28 2011 From: Jorney_Dong at asus.com (Jorney_Dong at asus.com) Date: Fri, 14 Jan 2011 09:26:28 +0800 Subject: [gst-devel] =?gb2312?b?tPC4tDogIGFtci13YiBzZWVrIHByb2JsZW0=?= References: <1294652354379-3206831.post@n4.nabble.com> Message-ID: Thanks for your reply! There is not a blocking in every seek event. And the gst_element_seek's return value is TRUE when there is a blocking. I didn't find a src pad event handler in amrwbdec in ugly(15 or 16). but I's not sure how to handle src pad event in AMRWBDEC, could you give me a sample? Thanks! -----Original Message----- From: sudarshan bisht [mailto:bisht.sudarshan at gmail.com] Sent: 2011-1-10 (???) 21:09 To: Discussion of the development of GStreamer Subject: Re: [gst-devel] amr-wb seek problem check the return value of gst_element_seek too . On Mon, Jan 10, 2011 at 3:00 PM, sudarshan bisht wrote: > > It seems amrwbdec does not handle src pad event (seek ) and thats why > there is a blocking. There is a lack of implementation of src event > handling thats why kept in gst-plugins-ugly. > > > > On Mon, Jan 10, 2011 at 11:39 AM, jorney wrote: > >> >> Hi >> I am writing amr-wb player with gstreamer on Marvell pxa310. >> When I seek to a position by time format, there is a long time blocking >> (10 sec or even to 1 min), and there is no voice during the block, then >> an EOS message was emitted, >> but the position I seeked is far away to the end of the file. >> >> Bytes format has the same problem! here is my seek code: >> >> breslt = gst_element_seek(pipeline, >> 1.0, >> GST_FORMAT_BYTES, >> GST_SEEK_FLAG_FLUSH,//|GST_SEEK_FLAG_KEY_UNIT|GST_SEEK_FLAG_SEGMENT, >> GST_SEEK_TYPE_SET, >> bytePos,//GST_SECOND*nForw*count, >> GST_SEEK_TYPE_NONE, >> GST_CLOCK_TIME_NONE >> ); >> >> I use gstreamer-0.10.29 and gst-plugins-ugly-0.10.15's amrwbdec, >> gst-plugins-bad-0.10.13's amrparse. like this: >> >> gst-launch filesrc location=abc.amr ! amrparse ! amrwbdec ! alsasink >> >> >> And the codec is opencore-amr-0.1.2 >> Could any body help? >> >> -- >> View this message in context: >> http://gstreamer-devel.966125.n4.nabble.com/amr-wb-seek-problem-tp3206831p3206831.html >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> >> >> ------------------------------------------------------------------------------ >> Gaining the trust of online customers is vital for the success of any >> company >> that requires sensitive data to be transmitted over the Web. Learn how >> to >> best implement a security strategy that keeps consumers' information >> secure >> and instills the confidence they need to proceed with transactions. >> 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 > -- Regards, Sudarshan Bisht ===================================================================================================================================== This email and any attachments to it contain confidential information and are intended solely for the use of the individual to whom it is addressed.If you are not the intended recipient or receive it accidentally, please immediately notify the sender by e-mail and delete the message and any attachments from your computer system, and destroy all hard copies. 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URL: From Jorney_Dong at asus.com Fri Jan 14 04:24:40 2011 From: Jorney_Dong at asus.com (Jorney_Dong at asus.com) Date: Fri, 14 Jan 2011 11:24:40 +0800 Subject: [gst-devel] amr-wb seek problem References: <1294652354379-3206831.post@n4.nabble.com> Message-ID: I use --gst-debug=amrwbdec:5 to debug , the normal seeking with a "mode 8, block 61" output log. but the block seeking with: "mode 13, block 0" or "mode 11, block 0" -----Original Message----- From: Jorney Dong(??????_?A?T?K??) Sent: 2011-1-14 (??????) 9:26 To: Discussion of the development of GStreamer; Discussion of the development of GStreamer Subject: ????: [gst-devel] amr-wb seek problem Thanks for your reply! There is not a blocking in every seek event. And the gst_element_seek's return value is TRUE when there is a blocking. I didn't find a src pad event handler in amrwbdec in ugly(15 or 16). but I's not sure how to handle src pad event in AMRWBDEC, could you give me a sample? Thanks! -----Original Message----- From: sudarshan bisht [mailto:bisht.sudarshan at gmail.com] Sent: 2011-1-10 (??????) 21:09 To: Discussion of the development of GStreamer Subject: Re: [gst-devel] amr-wb seek problem check the return value of gst_element_seek too . On Mon, Jan 10, 2011 at 3:00 PM, sudarshan bisht wrote: > > It seems amrwbdec does not handle src pad event (seek ) and thats why > there is a blocking. There is a lack of implementation of src event > handling thats why kept in gst-plugins-ugly. > > > > On Mon, Jan 10, 2011 at 11:39 AM, jorney wrote: > >> >> Hi >> I am writing amr-wb player with gstreamer on Marvell pxa310. >> When I seek to a position by time format, there is a long time blocking >> (10 sec or even to 1 min), and there is no voice during the block, then >> an EOS message was emitted, >> but the position I seeked is far away to the end of the file. >> >> Bytes format has the same problem! here is my seek code: >> >> breslt = gst_element_seek(pipeline, >> 1.0, >> GST_FORMAT_BYTES, >> GST_SEEK_FLAG_FLUSH,//|GST_SEEK_FLAG_KEY_UNIT|GST_SEEK_FLAG_SEGMENT, >> GST_SEEK_TYPE_SET, >> bytePos,//GST_SECOND*nForw*count, >> GST_SEEK_TYPE_NONE, >> GST_CLOCK_TIME_NONE >> ); >> >> I use gstreamer-0.10.29 and gst-plugins-ugly-0.10.15's amrwbdec, >> gst-plugins-bad-0.10.13's amrparse. like this: >> >> gst-launch filesrc location=abc.amr ! amrparse ! amrwbdec ! alsasink >> >> >> And the codec is opencore-amr-0.1.2 >> Could any body help? >> >> -- >> View this message in context: >> http://gstreamer-devel.966125.n4.nabble.com/amr-wb-seek-problem-tp3206831p3206831.html >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> >> >> ------------------------------------------------------------------------------ >> Gaining the trust of online customers is vital for the success of any >> company >> that requires sensitive data to be transmitted over the Web. Learn how >> to >> best implement a security strategy that keeps consumers' information >> secure >> and instills the confidence they need to proceed with transactions. >> 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 > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... 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Thank you for your cooperation. ===================================================================================================================================== From umakantgoyal1 at gmail.com Fri Jan 14 08:47:27 2011 From: umakantgoyal1 at gmail.com (Umakant Goyal) Date: Fri, 14 Jan 2011 13:17:27 +0530 Subject: [gst-devel] Streaming of Image File for Specific duration Message-ID: Hi All, Can i stream image (.jpg, .png) file on network after transcoding in pcma format for some specific duration using gstreamer? Thanks in Advance From arun.raghavan at collabora.co.uk Fri Jan 14 09:06:33 2011 From: arun.raghavan at collabora.co.uk (Arun Raghavan) Date: Fri, 14 Jan 2011 13:36:33 +0530 Subject: [gst-devel] [pulseaudio-discuss] [PATCH 0/3] Fighting rewinds In-Reply-To: <4D2B32D5.1050902@canonical.com> References: <4D00DFAF.3000807@canonical.com> <4D0231DF.7040008@canonical.com> <4D059A7F.1000007@canonical.com> <4D2B32D5.1050902@canonical.com> Message-ID: <1294992393.25127.8.camel@snowflake> On Mon, 2011-01-10 at 10:24 -0600, David Henningsson wrote: > On 2010-12-12 22:01, David Henningsson wrote: [...] > > 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). > > > > Hi Pierre and rest of gstreamer-devel, > > Given the discussion above, did you have any time to do more thinking > about it? I'm attaching a simple patch. Some testing of that patch > hasn't reported any regressions, and it does start to send larger > packages (which in turn saves PA from additional packet overhead). > I'm attaching a patch for Ubuntu, but given that I give you a proper git > header etc to that patch, would the folks here apply it? Ultimately, the size of the write also depends on the number of samples passed to the sink from the decoder, and there is currently no way to negotiate this (it's something being looked at for GStreamer 1.0, though). That said, IMO this patch makes sense - I see no reason to not push as much data as possible on each iteration. -- Arun From wl2776 at gmail.com Fri Jan 14 09:52:11 2011 From: wl2776 at gmail.com (wl2776) Date: Fri, 14 Jan 2011 00:52:11 -0800 (PST) Subject: [gst-devel] How can I add RTSP media mapping to the gst-rtsp-server in Python? In-Reply-To: References: <1284454972893-2538670.post@n4.nabble.com> Message-ID: <1294995131927-3217303.post@n4.nabble.com> Thanks, it's not very actual for me right now, I'll look into it as time permits. Andr? Dieb wrote: > > Hello, > The patch was commited (thanks devs!) today, so, if you're still > interested, > you'll find the methods and sample code on test.py, included on the patch > at > https://bugzilla.gnome.org/show_bug.cgi?id=622296. > > On Tue, Sep 14, 2010 at 8:19 AM, Andr? Dieb wrote: > >> Hello, >> >> Currently gst-rtsp-server (master) can't add media mappings. I'm >> currently >> in the process of cleaning up my python-gst-rtsp-server patches, which > >>> I've created a server instance, but cannot add media-mapping to it. >>> Are there any examples of using gst-rtsp-server in Python? >>> > -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-can-I-add-RTSP-media-mapping-to-the-gst-rtsp-server-in-Python-tp2538670p3217303.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From yogeshm.007 at gmail.com Fri Jan 14 12:31:05 2011 From: yogeshm.007 at gmail.com (Yogesh Marwaha) Date: Fri, 14 Jan 2011 17:01:05 +0530 Subject: [gst-devel] How to connect to cdparanoiasrc's transport-error signal? Message-ID: Hi, I need to connect to cdparanoiasrc's transport-error signal. The "transport-error" signal: void user_function (GstCdParanoiaSrc *cdparanoia, sector, gpointer user_data) This signal is emitted whenever an error occurs while reading. CdParanoia will attempt to recover the data. The problem is that I am unable to find a header file where GstCdParanoiaSrc is defined. Regards, -- Yogesh M http://sparklemedia.sourceforge.net/ http://mazedaar.wordpress.com/ http://snakeeyes.wordpress.com/ From t.i.m at zen.co.uk Fri Jan 14 12:55:17 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 14 Jan 2011 11:55:17 +0000 Subject: [gst-devel] How to connect to cdparanoiasrc's transport-error signal? In-Reply-To: References: Message-ID: <1295006117.13745.0.camel@zingle> On Fri, 2011-01-14 at 17:01 +0530, Yogesh Marwaha wrote: Hi, > I need to connect to cdparanoiasrc's transport-error signal. > > The "transport-error" signal: > void user_function (GstCdParanoiaSrc *cdparanoia, sector, gpointer user_data) > This signal is emitted whenever an error occurs while reading. > CdParanoia will attempt to recover the data. > > The problem is that I am unable to find a header file where > GstCdParanoiaSrc is defined. There are no header files. You don't need to know how GstCdParanoiaSrc is defined. Just g_signal_connect (element, "transport-error", G_CALLBACK (my_transport_error_callback), NULL); Cheers -Tim From yogeshm.007 at gmail.com Fri Jan 14 16:10:26 2011 From: yogeshm.007 at gmail.com (Yogesh Marwaha) Date: Fri, 14 Jan 2011 20:40:26 +0530 Subject: [gst-devel] How to connect to cdparanoiasrc's transport-error signal? In-Reply-To: <1295006117.13745.0.camel@zingle> References: <1295006117.13745.0.camel@zingle> Message-ID: <201101142040.26993.yogeshm.007@gmail.com> Sorry if I am missing something trivial, but I need to declare and define a callback function like this:- void my_transport_error_callback(GstCddaBaseSrc *cdparanoia, gint sector, gpointer userData); Thanks for replying, On Friday 14 Jan 2011 5:25:17 pm Tim-Philipp M?ller wrote: > On Fri, 2011-01-14 at 17:01 +0530, Yogesh Marwaha wrote: > > Hi, > > > I need to connect to cdparanoiasrc's transport-error signal. > > > > The "transport-error" signal: > > void user_function (GstCdParanoiaSrc *cdparanoia, sector, gpointer > > user_data) This signal is emitted whenever an error occurs while > > reading. > > CdParanoia will attempt to recover the data. > > > > The problem is that I am unable to find a header file where > > GstCdParanoiaSrc is defined. > > There are no header files. You don't need to know how GstCdParanoiaSrc > is defined. > > Just > > g_signal_connect (element, "transport-error", > G_CALLBACK (my_transport_error_callback), NULL); > > Cheers > -Tim > > > > > --------------------------------------------------------------------------- > --- Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 t.i.m at zen.co.uk Fri Jan 14 16:54:49 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 14 Jan 2011 15:54:49 +0000 Subject: [gst-devel] How to connect to cdparanoiasrc's transport-error signal? In-Reply-To: <201101142040.26993.yogeshm.007@gmail.com> References: <1295006117.13745.0.camel@zingle> <201101142040.26993.yogeshm.007@gmail.com> Message-ID: <1295020489.23277.1.camel@zingle> On Fri, 2011-01-14 at 20:40 +0530, Yogesh Marwaha wrote: > Sorry if I am missing something trivial, but I need to declare and define a > callback function like this:- > > void my_transport_error_callback(GstCddaBaseSrc *cdparanoia, gint sector, > gpointer userData); Not sure where you got the BaseSrc bit from, but you can just do: void my_transport_error_callback (GstElement * cdparanoia, gint sector, gpointer user_data) { ... } Cheers -Tim From bertd at tplogic.com Fri Jan 14 19:20:07 2011 From: bertd at tplogic.com (Bert Douglas) Date: Fri, 14 Jan 2011 10:20:07 -0800 Subject: [gst-devel] aravis and color Message-ID: Greetings ! I have Aravis working with Baser camera model acA1300-30gc But only in 8-bit gray mode. When using Pylon viewer, this camera shows color. But when using Aravis library and gstreamer I cannot get color. Have tried every format that the camera can produce. Aravis is sending a genicam message to the camera to set it into mono-8 mode. Thanks Much, Bert Douglas -------------- next part -------------- An HTML attachment was scrubbed... URL: From bertd at tplogic.com Fri Jan 14 20:02:29 2011 From: bertd at tplogic.com (Bert Douglas) Date: Fri, 14 Jan 2011 11:02:29 -0800 Subject: [gst-devel] aravis and color Message-ID: I think I have answered my own question. I found this code in aravis/gst/gstaravis.c gs = gst_structure_empty_new ("video"); gst_structure_set_name (gs, "video/x-raw-gray"); gst_structure_set (gs, "bpp", G_TYPE_INT, 8, "depth", G_TYPE_INT, 8, "width", G_TYPE_INT, width, "height", G_TYPE_INT, height, NULL); I looked elsewhere for subsequent modifications, but did not find any. >From this I concluded that aravis only supports 8-bit gray. Thanks, Bert Douglas -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.isorce at gmail.com Fri Jan 14 20:08:31 2011 From: julien.isorce at gmail.com (Julien Isorce) Date: Fri, 14 Jan 2011 20:08:31 +0100 Subject: [gst-devel] playing live mpegts stream over udp Message-ID: Hi, I am trying to play nicely a usual mpegts stream from network. (720p h264/ac3 mpegts over udp multicast) The pipeline is: gst-launch uridecodebin uri=239.102.101.3:1013 ! then audio and video renderers. My network interface is connected in 1GO. With default properties, the video has a lot of macro block. If I record the network packets in a pcap file then I can perfeclty play the stream using pcapparse. So I think the problem comes from some buffering pbs. When configuring with some heuristics use-buffering=1, buffer-duration and buffer-size properties of uridecodebin, the video contains less macro blocks but it still contains enough block to make me fill it's not normal. With VLC the stream is played almost perfectly. Am I doing something wrong ? Should I need a something like a jitterbuffer (no rtp here) ? Any help would be appreciated. Sincerely Julien Isorce -------------- next part -------------- An HTML attachment was scrubbed... URL: From emmanuel at gnome.org Fri Jan 14 20:55:18 2011 From: emmanuel at gnome.org (Emmanuel Pacaud) Date: Fri, 14 Jan 2011 20:55:18 +0100 Subject: [gst-devel] aravis and color In-Reply-To: References: Message-ID: <1295034918.2330.19.camel@lappc-p348> Hi, Aravis has a mailing list now, please use aravis at freelists.org instead of gstreamer-devel. Le vendredi 14 janvier 2011 ? 19:20 +0100, Bert Douglas a ?crit : > I have Aravis working with Baser camera model acA1300-30gc > But only in 8-bit gray mode. > > When using Pylon viewer, this camera shows color. > But when using Aravis library and gstreamer I cannot get color. > Have tried every format that the camera can produce. > > Aravis is sending a genicam message to the camera to set it into > mono-8 mode. Yes. The gstreamer plugin has very limited capabilities, and only support 8bit grayscale for now. In Aravis 0.1.3, I've implemented a simple video viewer which should be able to handle more pixel format, including 24bits rgb. Which formats does support your camera ? Could you try arv-viewer ? The gstreamer plugin needs work in order to make it more useful. Emmanuel. From keevon at gmail.com Sat Jan 15 04:20:09 2011 From: keevon at gmail.com (KEVIN WEST) Date: Fri, 14 Jan 2011 19:20:09 -0800 Subject: [gst-devel] Directshow sync issues Message-ID: <4802BCAE-5510-4E27-B787-62F00EFEFD58@gmail.com> I am running into a strange issue with a few videos and dshowdecwrapper. Basically I will transcode the video, but the output is all wrong. If you open the video, it will show the first frame while playing the entire audio stream, and finally it will play the video stream once it hits the end of the audio. I thought there may be some synchronization issues since dshow may not be using the same clock as gstreamer, so I implemented an IReferenceClock that calls into the gstreamer clock, however this doesn't seem to be fixing the issue. For reference, here is my current pipeline: gst-launch.exe -v --seek=0 uridecodebin name=dec uri="test.wmv" dec. ! videoscale ! videorate ! video/x-raw-yuv, width=320, height=240, framerate=24000/1001 ! ffmpegcolorspace ! x264enc ! queue2 max-size-bytes=0 max-size-time=0 max-size-buffers=0 ! mux. dec. ! audioconvert ! audioresample ! audio/x-raw-int, rate=44100, channels=2 ! faac bitrate=128000 ! queue2 max-size-bytes=0 max-size-time=0 max-size-buffers=0 ! mux. flvmux name=mux ! filesink location="output.flv" Any ideas what might be going on? From praveen.pande1 at gmail.com Sat Jan 15 11:41:14 2011 From: praveen.pande1 at gmail.com (praveen pandey) Date: Sat, 15 Jan 2011 16:11:14 +0530 Subject: [gst-devel] Directshow sync issues In-Reply-To: <4802BCAE-5510-4E27-B787-62F00EFEFD58@gmail.com> References: <4802BCAE-5510-4E27-B787-62F00EFEFD58@gmail.com> Message-ID: are you able to play video alone? BR, Praveen Pandey On Sat, Jan 15, 2011 at 8:50 AM, KEVIN WEST wrote: > I am running into a strange issue with a few videos and dshowdecwrapper. > Basically I will transcode the video, but the output is all wrong. If you > open the video, it will show the first frame while playing the entire audio > stream, and finally it will play the video stream once it hits the end of > the audio. I thought there may be some synchronization issues since dshow > may not be using the same clock as gstreamer, so I implemented an > IReferenceClock that calls into the gstreamer clock, however this doesn't > seem to be fixing the issue. > > For reference, here is my current pipeline: > > gst-launch.exe -v --seek=0 uridecodebin name=dec uri="test.wmv" dec. ! > videoscale ! videorate ! video/x-raw-yuv, width=320, height=240, > framerate=24000/1001 ! ffmpegcolorspace ! x264enc ! queue2 max-size-bytes=0 > max-size-time=0 max-size-buffers=0 ! mux. dec. ! audioconvert ! > audioresample ! audio/x-raw-int, rate=44100, channels=2 ! faac > bitrate=128000 ! queue2 max-size-bytes=0 max-size-time=0 max-size-buffers=0 > ! mux. flvmux name=mux ! filesink location="output.flv" > > Any ideas what might be going on? > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 t.i.m at zen.co.uk Sat Jan 15 13:01:45 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sat, 15 Jan 2011 12:01:45 +0000 Subject: [gst-devel] Directshow sync issues In-Reply-To: <4802BCAE-5510-4E27-B787-62F00EFEFD58@gmail.com> References: <4802BCAE-5510-4E27-B787-62F00EFEFD58@gmail.com> Message-ID: <1295092905.12462.4.camel@zingle> On Fri, 2011-01-14 at 19:20 -0800, KEVIN WEST wrote: > I am running into a strange issue with a few videos and > dshowdecwrapper. Basically I will transcode the video, but the output > is all wrong. If you open the video, it will show the first frame > while playing the entire audio stream, and finally it will play the > video stream once it hits the end of the audio. I thought there may > be some synchronization issues since dshow may not be using the same > clock as gstreamer, so I implemented an IReferenceClock that calls > into the gstreamer clock, however this doesn't seem to be fixing the > issue. Clocks don't matter much in this kind of pipeline - there's neither a source nor a sink syncing against the clock. Timestamps on the buffers (and newsegment events preceding the data flow) carry the timing information. > For reference, here is my current pipeline: > > gst-launch.exe -v --seek=0 uridecodebin name=dec uri="test.wmv" dec. ! > videoscale ! videorate ! video/x-raw-yuv, width=320, height=240, > framerate=24000/1001 ! ffmpegcolorspace ! x264enc ! queue2 > max-size-bytes=0 max-size-time=0 max-size-buffers=0 ! mux. dec. ! > audioconvert ! audioresample ! audio/x-raw-int, rate=44100, channels=2 > ! faac bitrate=128000 ! queue2 max-size-bytes=0 max-size-time=0 > max-size-buffers=0 ! mux. flvmux name=mux ! filesink > location="output.flv" Have you tried other muxers? Have you tried with other decoders? (e.g. gst-ffmpeg based) Have you tried other encoders? Does playback alone work? Where does the --seek option come from? That's not in upstream gst-launch. Try without it. (Also: use a normal queue instead of queue2 here). Cheers -Tim From shamun.toha at gmail.com Sat Jan 15 15:30:12 2011 From: shamun.toha at gmail.com (Shamun toha md) Date: Sat, 15 Jan 2011 15:30:12 +0100 Subject: [gst-devel] Gstreamer vs Vlc ???? Message-ID: Hello, I am getting confused with VLC and Gstreamer. Please have a look ex: http://images.videolan.org/images/documentation/global-diagram.jpg - http://wiki.videolan.org/Codec - it works in Windows/Mac/Linux/Xyz... - What is the benefit using gstreamer compared to vlc ? - Also gstreamer need to be ported in mac manually, we dont have officially 1 that is prepared to just give someone Can some one kindly share some thoughts. Best regards Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: From smcnam at gmail.com Sat Jan 15 17:01:07 2011 From: smcnam at gmail.com (Sean McNamara) Date: Sat, 15 Jan 2011 11:01:07 -0500 Subject: [gst-devel] Gstreamer vs Vlc ???? In-Reply-To: References: Message-ID: On Sat, Jan 15, 2011 at 9:30 AM, Shamun toha md wrote: > Hello, > > I am getting confused with VLC and Gstreamer. > > Please have a look ex: > http://images.videolan.org/images/documentation/global-diagram.jpg > - http://wiki.videolan.org/Codec > - it works in Windows/Mac/Linux/Xyz... > - What is the benefit using gstreamer compared to vlc ? > - Also gstreamer need to be ported in mac manually, we dont have officially > 1 that is prepared to just give someone This could turn into a debate like vi vs emacs, where people are strongly opinionated and there is never really any end to the debate. It might therefore be wise to do a cost-benefit analysis on your own and make up your mind based on your own particular requirements or preferences. Here are some of the topics of debate that would come up, which you did not yet mention in your message: 1. Legality of the codec implementations. For each product, which codecs are available? Do they comply with the letter of the law in all jurisdictions where you and anyone to whom you distribute your software plans to use it? For VLC, I am unaware of any MPEG-LA codec implementation that will run on VLC that is legally licensed in the U.S. or any country respecting U.S. patent law. To use MPEG-4, AAC, WMA, H.264, or any number of other patent-encumbered codecs, you can either break the law; live in a country where it isn't breaking the law; or use legally licensed codecs. Since VLC has none, that leaves only Gstreamer and the codecs that you can buy from Fluendo: http://www.fluendo.com 2. Available plugins. I'm aware that VLC is based on a similar filter graph infrastructure as Gstreamer, but I think that the number and variety of plugins available for Gstreamer makes it more useful for writing applications. Gstreamer is often accessible even to new users because of its console commands, gst-launch and gst-inspect, which help people to understand how the plugins fit together before they develop an application. Again, Gstreamer's focus is not to deliver an end-user product, but to provide an intuitive API and wide variety of generally-useful plugins for application development. By contrast, VLC *is* the application, so the development focus is on using the existing VLC graphical interface to perform typical user tasks, like video playback, streaming, and conversion. Note that "available plugins" doesn't necessarily translate to "available codecs": there are a lot of Gstreamer plugins that are not a codec, and these are equally important as the codecs. 3. Available codecs. If you just install the legally licensed "free as in freedom" codecs in Gstreamer, from gst-plugins-{base,good,bad}, it's clear that VLC wins the codec variety battle. But you can also stack a great number of additional codecs into Gstreamer by adding in gst-plugins-{gl,ugly,ffmpeg}, with legal strings attached, or gst-plugins-fluendo (the paid Fluendo codecs) without legal strings attached, but with a price tag attached :-). In the end, a fully loaded gstreamer (legal or otherwise) can more or less compete with VLC on codecs, but you have to either consider your legal situation very carefully, or disregard it entirely, which I can't recommend (but many people seem to do it anyway). 4. Platform integration. It's hard for VLC to keep up here, because VLC is its own world. It doesn't have any familial resemblance to any of the popular Linux desktop environments. Gstreamer, on the other hand, is a well-behaving GLib-based library, and so it integrates quite easily into GNOME applications. So if you are thinking of using GLib, GTK+, etc. in your application, you have a huge incentive to also use Gstreamer over any other media framework, because the programming paradigms it uses will fit quite nicely into your overall app design. VLC's preferred interface uses Qt4, which should look good on KDE, and decent on a properly-configured GNOME desktop, but you still don't get that fully platform integrated feel. 5. Performance. If you're doing a lot of data processing, or just any kind of processing on small hardware (e.g. embedded), you could actually care about performance. I won't make any claims about the relative performance of Gstreamer and VLC, because I don't know how it goes; but I do know that Gstreamer has been successfully deployed on mobile devices with minimal modifications. Usually it is good enough to just, for example, use one of the fastest settings for resampling, and work with integers rather than floats across the pipeline. These little tweaks can be made by tapping element properties before you play your pipeline, and make a big difference in the performance of the pipeline. I don't know to what extent VLC has similar options because I am less familiar with it. 6. Bindings. What bindings does VLC have for developing software against the VLC core using non-C languages? Well, it has Python bindings, sure (so does Gstreamer); but what else? If you have a particular programming language you care about, make sure that VLC supports bindings for that language. If Gstreamer has bindings that VLC doesn't for your desired language, that might be a strong reason to use Gstreamer (or, alternatively, contribute new bindings to the VLC project!) 7. Community. This is a hard one to quantify, or even qualify, but the friendliness and helpfulness of the community makes a big difference when I'm doing software development. I haven't worked too much with the VLC community, but I know that the Gstreamer community is one of the best in terms of being civil, providing sound advice, and being available over a wide array of media (IRC, email, etc.). Not to mention that you can get professional Gstreamer support from some companies, I understand ;-). Anyway, you may or may not agree with these assessments, but the _topics_ of each of those items are certainly relevant in making a choice between these two open source projects. This topic has the potential for strong words and disagreements, so I will refrain from replying further to this thread. Just keep in mind that this kind of choice is difficult to make in the general case (i.e. for all users); the most useful evaluation is one coming from your *own* perspective, focusing on your *own* requirements and preferences. My claims in this message are just coming from my perspective, which may be completely irrelevant to yours. Take it or leave it. HTH, Sean > > > Can some one kindly share some thoughts. > > > Best regards > Thank you > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 Sat Jan 15 20:42:12 2011 From: umakantgoyal1 at gmail.com (Umakant Goyal) Date: Sun, 16 Jan 2011 01:12:12 +0530 Subject: [gst-devel] Duration of Pipeline Message-ID: Hi All, Can i make a pipeline to run for specific period of time? Thanks From dhoyt at hoytsoft.org Sat Jan 15 20:36:50 2011 From: dhoyt at hoytsoft.org (David Hoyt) Date: Sat, 15 Jan 2011 11:36:50 -0800 Subject: [gst-devel] Gstreamer vs Vlc ???? In-Reply-To: References: Message-ID: <00c601cbb4eb$8e32cfc0$aa986f40$@org> > 1. Legality of the codec implementations. For each product, which > codecs are available? Do they comply with the letter of the law in all > jurisdictions where you and anyone to whom you distribute your > software plans to use it? For VLC, I am unaware of any MPEG-LA codec > implementation that will run on VLC that is legally licensed in the > U.S. or any country respecting U.S. patent law. To use MPEG-4, AAC, > WMA, H.264, or any number of other patent-encumbered codecs, you can > either break the law; live in a country where it isn't breaking the > law; or use legally licensed codecs. Since VLC has none, that leaves > only Gstreamer and the codecs that you can buy from Fluendo: > http://www.fluendo.com This is important -- the vlc devs themselves recently asked apple to pull a vlc build from the app store due to licensing (GPL) issues. It's my understanding, however, that you can license x264 and use a vlc+x264 build legally. I've never done it before and so don't know the ramifications, but it seems possible. > 6. Bindings. What bindings does VLC have for developing software > against the VLC core using non-C languages? Well, it has Python > bindings, sure (so does Gstreamer); but what else? If you have a > particular programming language you care about, make sure that VLC > supports bindings for that language. If Gstreamer has bindings that > VLC doesn't for your desired language, that might be a strong reason > to use Gstreamer (or, alternatively, contribute new bindings to the > VLC project!) vlc has java bindings at least. > 7. Community. This is a hard one to quantify, or even qualify, but the > friendliness and helpfulness of the community makes a big difference > when I'm doing software development. I haven't worked too much with > the VLC community, but I know that the Gstreamer community is one of > the best in terms of being civil, providing sound advice, and being > available over a wide array of media (IRC, email, etc.). Not to > mention that you can get professional Gstreamer support from some > companies, I understand ;-). This is what really did it for me at first because I was evaluating vlc vs gstreamer vs mplayer vs helix. The gstreamer community (especially on IRC for me) has been exceptional compared to vlc. With vlc I was basically told I was stupid for even asking, even though I was a complete beginner, wanted to help, but didn't know how to go about doing what I wanted (being unfamiliar with the api). A few times of that and I was done with them. From dhoyt at hoytsoft.org Sat Jan 15 20:40:10 2011 From: dhoyt at hoytsoft.org (David Hoyt) Date: Sat, 15 Jan 2011 11:40:10 -0800 Subject: [gst-devel] Directshow sync issues In-Reply-To: <4802BCAE-5510-4E27-B787-62F00EFEFD58@gmail.com> References: <4802BCAE-5510-4E27-B787-62F00EFEFD58@gmail.com> Message-ID: <00c701cbb4ec$05af09b0$110d1d10$@org> On Fri, 2011-01-14 at 19:20 -0800, KEVIN WEST wrote: > I am running into a strange issue with a few videos and > dshowdecwrapper. Basically I will transcode the video, but the output > is all wrong. If you open the video, it will show the first frame > while playing the entire audio stream, and finally it will play the > video stream once it hits the end of the audio. I thought there may > be some synchronization issues since dshow may not be using the same > clock as gstreamer, so I implemented an IReferenceClock that calls > into the gstreamer clock, however this doesn't seem to be fixing the > issue. Where did you get the windows build? Was it your own or via ossbuild (gstreamer winbuilds) or cygwin ports? And what version? From thiago.sousa.santos at collabora.co.uk Sat Jan 15 21:44:09 2011 From: thiago.sousa.santos at collabora.co.uk (Thiago Sousa Santos) Date: Sat, 15 Jan 2011 17:44:09 -0300 Subject: [gst-devel] Duration of Pipeline In-Reply-To: References: Message-ID: <1295124249.4478.32.camel@blacksheep> On Sun, 2011-01-16 at 01:12 +0530, Umakant Goyal wrote: > Hi All, > > Can i make a pipeline to run for specific period of time? Yes. You can send a seek event to the pipeline with a start and a stop time. Check http://www.gstreamer.net/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-seek -- Thiago > > Thanks > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 keevon at gmail.com Sun Jan 16 04:43:05 2011 From: keevon at gmail.com (KEVIN WEST) Date: Sat, 15 Jan 2011 19:43:05 -0800 Subject: [gst-devel] Directshow sync issues In-Reply-To: <00c701cbb4ec$05af09b0$110d1d10$@org> References: <4802BCAE-5510-4E27-B787-62F00EFEFD58@gmail.com> <00c701cbb4ec$05af09b0$110d1d10$@org> Message-ID: <9311E98B-4CBA-47E2-A562-04314E035FF6@gmail.com> > Message: 5 > Date: Sat, 15 Jan 2011 12:01:45 +0000 > From: Tim-Philipp M?ller > Subject: Re: [gst-devel] Directshow sync issues > To: gstreamer-devel at lists.sourceforge.net > Message-ID: <1295092905.12462.4.camel at zingle> > Content-Type: text/plain; charset="UTF-8" > > On Fri, 2011-01-14 at 19:20 -0800, KEVIN WEST wrote: > >> I am running into a strange issue with a few videos and >> dshowdecwrapper. Basically I will transcode the video, but the output >> is all wrong. If you open the video, it will show the first frame >> while playing the entire audio stream, and finally it will play the >> video stream once it hits the end of the audio. I thought there may >> be some synchronization issues since dshow may not be using the same >> clock as gstreamer, so I implemented an IReferenceClock that calls >> into the gstreamer clock, however this doesn't seem to be fixing the >> issue. > > Clocks don't matter much in this kind of pipeline - there's neither a > source nor a sink syncing against the clock. Timestamps on the buffers > (and newsegment events preceding the data flow) carry the timing > information. Thanks for the info here Tim. I wasn't sure it would fix the problem and it's good to know it shouldn't be necessary. > >> For reference, here is my current pipeline: >> >> gst-launch.exe -v --seek=0 uridecodebin name=dec uri="test.wmv" dec. ! >> videoscale ! videorate ! video/x-raw-yuv, width=320, height=240, >> framerate=24000/1001 ! ffmpegcolorspace ! x264enc ! queue2 >> max-size-bytes=0 max-size-time=0 max-size-buffers=0 ! mux. dec. ! >> audioconvert ! audioresample ! audio/x-raw-int, rate=44100, channels=2 >> ! faac bitrate=128000 ! queue2 max-size-bytes=0 max-size-time=0 >> max-size-buffers=0 ! mux. flvmux name=mux ! filesink >> location="output.flv" > > Have you tried other muxers? I have tried avimux, flvmux, and mpegtsmux. With flvmux I get the playback as I described above playing in Adobe Media Player. With avi and mpg I get either only sound or no playback in Windows Media Player. > > Have you tried with other decoders? (e.g. gst-ffmpeg based) Are you suggesting using something other than dshowdecwrapper for decoding? I have one mov file with this synchronization issue (h264 video) when using the dshowdecwrapper, and then when using a separate plugin for decoding it works just fine. It seems to be related to the wrapper. > > Have you tried other encoders? I have not, just x264. This only happens for some videos, so I don't suspect the encoder, but I will give it a try and report back. > > Does playback alone work? Yes, playback in Windows Media Player of the original wmv plays fine. > > Where does the --seek option come from? That's not in upstream > gst-launch. Try without it. Sorry, that's from custom code I have added to gst-launch. Using the straight ossbuild version produces the same sync issue. > > (Also: use a normal queue instead of queue2 here). Haven't tried this but I'm curious about why this may work. Shot in the dark or is there more reasoning behind it? I'll give it a try either way. Thanks! > > Cheers > -Tim > On Jan 15, 2011, at 11:40 AM, David Hoyt wrote: > On Fri, 2011-01-14 at 19:20 -0800, KEVIN WEST wrote: > >> I am running into a strange issue with a few videos and >> dshowdecwrapper. Basically I will transcode the video, but the output >> is all wrong. If you open the video, it will show the first frame >> while playing the entire audio stream, and finally it will play the >> video stream once it hits the end of the audio. I thought there may >> be some synchronization issues since dshow may not be using the same >> clock as gstreamer, so I implemented an IReferenceClock that calls >> into the gstreamer clock, however this doesn't seem to be fixing the >> issue. > > Where did you get the windows build? Was it your own or via ossbuild > (gstreamer winbuilds) or cygwin ports? And what version? > > As far as the windows build we are using, it's based off of the ossbuild. We have made some modifications, but I always go back and test using the original dlls as well to make sure it's not something we caused with our changes. I believe the version is > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 bug-track at fisher-privat.net Sun Jan 16 10:37:50 2011 From: bug-track at fisher-privat.net (Alexey Fisher) Date: Sun, 16 Jan 2011 10:37:50 +0100 Subject: [gst-devel] [camerabin] buggy webcams with no autoexposure Message-ID: <1295170670.6629.29.camel@zwerg> Hi, i thing camerabin is the right place to discuss this problem, some build in webcams on some laptops provide only static framereate without autoexposure. With 30fps i get goot frame rate but bad image, with 10fps i get good picture, but camerabin will take 30fps by default. For example the cam on fujisu lifebook a530 is more tricky. It support only 30fps on 640x480 frame size and about 9fps on 1280x1024. So to get good picture for video chat i should do fallowing workaround: gst-launch-0.10 v4l2src ! ffmpegcolorspace ! video/x-raw-yuv,width=1280,height=1024,framerate=9/1 ! videoscale ! video/x-raw-yuv,width=320,height=240 ! fpsdisplaysink I have fallowing questions: How can we find if the cam is buggy? Add v4l2_ctrl_quirk? Or if cam do nor have exposure controls this mean it use static exposure time? For video chat we need not more than 15fps, so if we have only choice between 30fps and 10fps, best decision will be probably 10fps. Less fps -> less videotmpnoise -> better image quality -> less load for encoder -> less traffic. -- Regards, Alexey From yogeshm.007 at gmail.com Sun Jan 16 15:31:11 2011 From: yogeshm.007 at gmail.com (Yogesh Marwaha) Date: Sun, 16 Jan 2011 20:01:11 +0530 Subject: [gst-devel] How to connect to cdparanoiasrc's transport-error signal? In-Reply-To: <1295020489.23277.1.camel@zingle> References: <201101142040.26993.yogeshm.007@gmail.com> <1295020489.23277.1.camel@zingle> Message-ID: <201101162001.12232.yogeshm.007@gmail.com> Thank you very much :) On Friday 14 Jan 2011 9:24:49 pm Tim-Philipp M?ller wrote: > On Fri, 2011-01-14 at 20:40 +0530, Yogesh Marwaha wrote: > > Sorry if I am missing something trivial, but I need to declare and define > > a callback function like this:- > > > > void my_transport_error_callback(GstCddaBaseSrc *cdparanoia, gint sector, > > gpointer userData); > > Not sure where you got the BaseSrc bit from, but you can just do: > > void > my_transport_error_callback (GstElement * cdparanoia, > gint sector, gpointer user_data) > { > ... > } > > Cheers > -Tim > > > --------------------------------------------------------------------------- > --- Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 Sun Jan 16 20:34:15 2011 From: umakantgoyal1 at gmail.com (Umakant Goyal) Date: Mon, 17 Jan 2011 01:04:15 +0530 Subject: [gst-devel] Duration of Pipeline In-Reply-To: <1295124249.4478.32.camel@blacksheep> References: <1295124249.4478.32.camel@blacksheep> Message-ID: Hi Thiago, I tried to run pipeline for specified period of time as suggested by you. But it did not work for me. I want to run pipeline for 15 seconds for that i have done bool flag = gst_element_seek (pipeline, 1.0, GST_FORMAT_TIME, (GstSeekFlags)(GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE), GST_SEEK_TYPE_NONE, 0, GST_SEEK_TYPE_SET, 15*GST_SECOND); i have also tried following line of code bool falg = gst_element_seek (vBin, 1.0, GST_FORMAT_TIME, (GstSeekFlags)(GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE), GST_SEEK_TYPE_SET, 0 * 15*GST_SECOND, GST_SEEK_TYPE_SET, 15*GST_SECOND); i have run above line of code in once in READY state and once in PAUSED state. I have also tried GST_MSECOND instead of GST_SECOND. But nothing works for me. Please help by telling where i am doing wrong while calling above line of code. Thanks On Sun, Jan 16, 2011 at 2:14 AM, Thiago Sousa Santos < thiago.sousa.santos at collabora.co.uk> wrote: > On Sun, 2011-01-16 at 01:12 +0530, Umakant Goyal wrote: > > Hi All, > > > > Can i make a pipeline to run for specific period of time? > > Yes. You can send a seek event to the pipeline with a start and a stop > time. Check > > http://www.gstreamer.net/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-seek > > -- > Thiago > > > > > Thanks > > > > > ------------------------------------------------------------------------------ > > Protect Your Site and Customers from Malware Attacks > > Learn about various malware tactics and how to avoid them. Understand > > malware threats, the impact they can have on your business, and how you > > can protect your company and customers by using code signing. > > 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 > > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 bertd at tplogic.com Mon Jan 17 00:10:20 2011 From: bertd at tplogic.com (Bert Douglas) Date: Sun, 16 Jan 2011 15:10:20 -0800 Subject: [gst-devel] question about changing videorate element to produce blue screen Message-ID: Hi All, I have a pipeline with a videorate element. The video source is multicast rtp. When the rtp stream stops, the videorate element duplicates the last frame received forever. I have an idea to modify videorate to produce a blue screen after a timeout. Is this a dumb idea? Thanks, Bert Douglas -------------- next part -------------- An HTML attachment was scrubbed... URL: From yuanqing.cai at tieto.com Mon Jan 17 03:26:19 2011 From: yuanqing.cai at tieto.com (Cai Yuanqing) Date: Mon, 17 Jan 2011 10:26:19 +0800 Subject: [gst-devel] audioresample In-Reply-To: References: <1294823945910-3213586.post@n4.nabble.com> <4D2E59E5.6020905@tieto.com> Message-ID: <4D33A8CB.3080302@tieto.com> Hi, On 01/13/2011 01:39 PM, Anuroop Jesu wrote: > Hi , > > I Tried the suggestion provided for the by Cai. Thanks for the code Cai. > > It works something like this It only allows to playback the 22KHz > S16LE audio. > > What I was trying is to convert the any input format into a 22KHz > S16LE so I can mux it with other stream of the same property and mux > multiple streams using alsasink plug:dmix. I see what you mean :-) I tried pipeline like this: gst-launch-0.10 filesrc location=yellow.mp3 ! id3demux ! mad ! audioconvert ! audioresample ! 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true' ! alsasink It works well to first decode any type of mp3 files into PCM,and then re-sample them into 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true. So you can playback this stream ,or you can replace 'alsasink' to other elements: $ file yellow.mp3 yellow.mp3: Audio file with ID3 version 2.3.0, contains: MPEG ADTS, layer III, v1, 128 kbps, 44.1 kHz, JntStereo $ gst-launch-0.10 filesrc location=yellow.mp3 ! id3demux ! mad ! audioconvert ! audioresample ! 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true' ! lame ! filesink location=haha.mp3 $ file haha.mp3 haha.mp3: MPEG ADTS, layer III, v2, 128 kbps, 22.05 kHz, JntStereo The pipeline above turn your stream to encode int a mp3 file with property as 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true'. Just add elements behind audioresample and caps in the C code. Hope it helps :-) Thanks. > > With Warm Regards > Jesu Anuroop Suresh > > "Any intelligent fool can make things bigger, more complex, and more > violent. It takes a touch of genius -- and a lot of courage -- to move > in the opposite direction." > "Anyone who has never made a mistake has never tried anything new." > > > > > > > On Thu, Jan 13, 2011 at 9:51 AM, Jesu Anuroop Suresh > wrote: > > Hi Cai, > > > Thanks for you response, I Will try out your suggestion of using > the filtered link. > > Sorry there was some typoerror in my code what I shared. > > I did initialized the 'resmux' as capasity filter and used the > conv not conv1. > > The cocde works for me for mp3 playback in its original settings. > > > resample = gst_element_factory_make ("audioresample", > "audio-resample"); > conv = gst_element_factory_make ("audioconvert", > "converter1"); > resmux = gst_element_factory_make ("capsfilter", "filter"); > > caps = gst_caps_new_simple ("audio/x-raw-int", > "width", G_TYPE_INT, 16, > "depth", G_TYPE_INT, 16, > "rate", G_TYPE_INT, 22050, > "channels",G_TYPE_INT, 2, NULL > ); > > if (!musicPlayer.playPipeline || !source || !sink || > !resample || !resmux || !caps || !conv) > { > g_print ("NO MEM Exiting.\n"); > return 1; > } > > /* we set the input filename to the source element */ > g_object_set (G_OBJECT (source), "location", filePath, NULL); > > demuxer = gst_element_factory_make ("id3demux", > "id3-demuxer"); > decoder = gst_element_factory_make ("mad", "mp3-decoder"); > > if (!demuxer || !decoder || !conv) > { > g_print ("NO MEM Exiting.\n"); > return 1; > } > > With Warm Regards > Jesu Anuroop Suresh > > "Any intelligent fool can make things bigger, more complex, and > more violent. It takes a touch of genius -- and a lot of courage > -- to move in the opposite direction." > "Anyone who has never made a mistake has never tried anything new." > > On Thu, Jan 13, 2011 at 7:16 AM, Cai Yuanqing [via > GStreamer-devel] <[hidden email] > > wrote: > > Hi Suresh: > Your application have a little problem. :-) > > > On 01/12/2011 08:41 PM, Jesu Anuroop Suresh wrote: > > > Hi Sean, > > > > Yes, what I was trying is to resample the decoded mp3 data > to the > > fixed (22KHZ S16LE) formate, > > > > no matter what is the input rate using a C application. > > > > Thanks for your response. > > > > Here is the piece of the code for the same but it does not > work with > > audioresample with the caps filter 'resmux'. This code does > work > > without the caps filter 'resmux'. > > > > GstElement *source, *demuxer, *decoder, *conv, *sink, > > *resample, *resmux; > > GstCaps *caps; > > > > gst_init(NULL, NULL); > > > > /* Create gstreamer elements */ > > musicPlayer.playPipeline = gst_pipeline_new > ("audio-player"); > > source = gst_element_factory_make ("filesrc", > "file-source"); > > sink = gst_element_factory_make ("alsasink", > "audio-output"); > > resample = gst_element_factory_make ("audioresample", > > "audio-resample"); > > conv = gst_element_factory_make ("audioconvert", > > "converter1"); > > > > caps = gst_caps_new_simple ("audio/x-raw-int", > > "width", G_TYPE_INT, 16, > > "depth", G_TYPE_INT, 16, > > "rate", G_TYPE_INT, > 22050, > > "channels",G_TYPE_INT, > 2, NULL > > ); > > > > if (!musicPlayer.playPipeline || !source || !sink || > > !resample || !resmux || !caps || !conv) > > { > > g_print ("NO MEM Exiting.\n"); > > return 1; > > } > resmux is not initialized yet,here maybe some random > value,you'd better > remove it from check list. > > > > > /* we set the input filename to the source element */ > > g_object_set (G_OBJECT (source), "location", > filePath, NULL); > > > > demuxer = gst_element_factory_make ("id3demux", > "id3-demuxer"); > > decoder = gst_element_factory_make ("mad", > "mp3-decoder"); > > > > if (!demuxer || !decoder || !conv1) > conv1 ? dose it should be conv? > > > { > > g_print ("NO MEM Exiting.\n"); > > return 1; > > } > > > > g_object_set (G_OBJECT (resmux), "caps", caps, NULL); > > gst_caps_unref (caps); > > > as I said before,resmux haven't initialized ,that's not quite > right. > and I suggest you to remove these two lines. > > > /* file-source -> demuxer -> decoder -> > alsa-output */ > > gst_bin_add_many (GST_BIN (musicPlayer.playPipeline), > > source, demuxer, decoder, conv, > resample, > > resmux,sink, NULL); > > > > gst_element_link (source, demuxer); > > gst_element_link_many (decoder, conv, > resample,resmux,sink, NULL); > You can use gst_element_link_filtered to link resample and > sink with > caps instead of this way. > something like: > gst_element_link (source, demuxer); > gst_element_link_many (decoder, conv, resample, NULL); > if ( !gst_element_link_filtered(resample,sink,caps) ){ > g_printerr("Failed to link elements resample and > alsa-sink"); > } > > > > g_signal_connect (demuxer, "pad-added", G_CALLBACK > > (on_pad_added), decoder); > > > > GstBus *bus = > > gst_pipeline_get_bus(GST_PIPELINE(musicPlayer.playPipeline)); > > gst_bus_add_watch(bus, bus_call, NULL); > > gst_object_unref(bus); > > > > > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), > > GST_STATE_PLAYING); > > > > musicPlayer.playLoop = g_main_loop_new(NULL, FALSE); > > > > g_main_loop_run(musicPlayer.playLoop); > > > > > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), > > GST_STATE_NULL); > > gst_object_unref(GST_OBJECT(musicPlayer.playPipeline)); > > > > > > > > > > With Warm Regards > > Jesu Anuroop Suresh > > > > "Any intelligent fool can make things bigger, more complex, > and more > > violent. It takes a touch of genius -- and a lot of courage > -- to move > > in the opposite direction." > > "Anyone who has never made a mistake has never tried > anything new." > > > > > I attached my modified source code ,you can try it. > Hope it helps. > > Thanks. > > > -- > B.R > > Cai Yuanqing > > > ------------------------------------------------------------------------------ > > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. > Understand > malware threats, the impact they can have on your business, > and how you > can protect your company and customers by using code signing. > 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/audioresample-tp3213586p3215098.html > > > > To unsubscribe from audioresample, click here > . > > > > > ------------------------------------------------------------------------ > View this message in context: Re: audioresample > > > > Sent from the GStreamer-devel mailing list archive > at Nabble.com. > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and > how you > can protect your company and customers by using code signing. > 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 rohitratri at gmail.com Mon Jan 17 05:42:21 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Sun, 16 Jan 2011 20:42:21 -0800 Subject: [gst-devel] get_*_static_pad () on GstBin throws a seg fault Message-ID: Hi, I've a pipeline like this - gstpipeline(filesrc->decodeBin2->gstbin(ffmpegcolorspace->capsfilter->appsink)). The gstBin shown above has a ghost pad (mirroring sink pad of the converter). When I try to link the src pad of decodeBin2 (in the new-decoded-pad callback) with the sink ghostpad of the gstbin - the call gst_element_get_static_pad on the bin leads to a segmentation fault. Any idea why? Here's the code I'm using - * Ghost pad creation - ConvSinkPad = gst_element_get_static_pad(converter, "sink"); gst_element_add_pad(GST_ELEMENT(app_data->convBin), gst_ghost_pad_new("sink", ConvSinkPad)); gst_object_unref(ConvSinkPad); Linking - on_decodeBin_newpad(..., GstPad* pad) { .... ... convBinSinkPad = gst_element_get_static_pad(app_data->convBin, "sink"); gst_pad_link(pad, convBinSinkPad); gst_object_unref(convBinSinkPad); }* PS - I tried debugging but I don't get any logs even with level 5. Thanks Rohit -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Mon Jan 17 06:36:29 2011 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Mon, 17 Jan 2011 11:06:29 +0530 Subject: [gst-devel] Duration of Pipeline In-Reply-To: References: <1295124249.4478.32.camel@blacksheep> Message-ID: Time should be in nano-seconds . For more info; http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-eventsseek.html On Mon, Jan 17, 2011 at 1:04 AM, Umakant Goyal wrote: > Hi Thiago, > > I tried to run pipeline for specified period of time as suggested by you. > But it did not work for me. I want to run pipeline for 15 seconds for that i > have done > > bool flag = gst_element_seek (pipeline, 1.0, GST_FORMAT_TIME, > (GstSeekFlags)(GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE), > GST_SEEK_TYPE_NONE, 0, GST_SEEK_TYPE_SET, 15*GST_SECOND); > > i have also tried following line of code > > bool falg = gst_element_seek (vBin, 1.0, GST_FORMAT_TIME, > (GstSeekFlags)(GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE), > GST_SEEK_TYPE_SET, 0 * 15*GST_SECOND, GST_SEEK_TYPE_SET, 15*GST_SECOND); > > i have run above line of code in once in READY state and once in PAUSED > state. I have also tried GST_MSECOND instead of GST_SECOND. But nothing > works for me. > > Please help by telling where i am doing wrong while calling above line of > code. > > Thanks > > On Sun, Jan 16, 2011 at 2:14 AM, Thiago Sousa Santos < > thiago.sousa.santos at collabora.co.uk> wrote: > >> On Sun, 2011-01-16 at 01:12 +0530, Umakant Goyal wrote: >> > Hi All, >> > >> > Can i make a pipeline to run for specific period of time? >> >> Yes. You can send a seek event to the pipeline with a start and a stop >> time. Check >> >> http://www.gstreamer.net/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-seek >> >> -- >> Thiago >> >> > >> > Thanks >> > >> > >> ------------------------------------------------------------------------------ >> > Protect Your Site and Customers from Malware Attacks >> > Learn about various malware tactics and how to avoid them. Understand >> > malware threats, the impact they can have on your business, and how you >> > can protect your company and customers by using code signing. >> > 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 >> >> >> >> >> ------------------------------------------------------------------------------ >> Protect Your Site and Customers from Malware Attacks >> Learn about various malware tactics and how to avoid them. Understand >> malware threats, the impact they can have on your business, and how you >> can protect your company and customers by using code signing. >> 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 >> > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 rohitratri at gmail.com Mon Jan 17 06:55:02 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Mon, 17 Jan 2011 11:25:02 +0530 Subject: [gst-devel] get_*_static_pad () on GstBin throws a seg fault In-Reply-To: References: Message-ID: Bummer! I was using a deprecated signal - new-decoded-pad. On a separate note, the documentation here needs an updated signal list - http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-decodebin2.html Thanks Rohit On Mon, Jan 17, 2011 at 10:12 AM, Rohit Atri wrote: > Hi, > > I've a pipeline like this - > gstpipeline(filesrc->decodeBin2->gstbin(ffmpegcolorspace->capsfilter->appsink)). > > The gstBin shown above has a ghost pad (mirroring sink pad of the > converter). When I try to link the src pad of decodeBin2 (in the > new-decoded-pad callback) with the sink ghostpad of the gstbin - the call > gst_element_get_static_pad on the bin leads to a segmentation fault. Any > idea why? > > Here's the code I'm using - > * > Ghost pad creation - > ConvSinkPad = gst_element_get_static_pad(converter, "sink"); > gst_element_add_pad(GST_ELEMENT(app_data->convBin), > gst_ghost_pad_new("sink", ConvSinkPad)); > gst_object_unref(ConvSinkPad); > > Linking - > on_decodeBin_newpad(..., GstPad* pad) > { > .... > ... > convBinSinkPad = gst_element_get_static_pad(app_data->convBin, > "sink"); > gst_pad_link(pad, convBinSinkPad); > gst_object_unref(convBinSinkPad); > }* > > PS - I tried debugging but I don't get any logs even with level 5. > > Thanks > Rohit > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jesuas at gmail.com Mon Jan 17 07:17:32 2011 From: jesuas at gmail.com (Jesu Anuroop Suresh) Date: Sun, 16 Jan 2011 22:17:32 -0800 (PST) Subject: [gst-devel] audioresample In-Reply-To: <4D33A8CB.3080302@tieto.com> References: <1294823945910-3213586.post@n4.nabble.com> <4D2E59E5.6020905@tieto.com> <4D33A8CB.3080302@tieto.com> Message-ID: Hi Cai, Thanks for your response. Below is the code which works and converts any input stream into 22KHz S16LE. 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_print("Error: %s\n",error->message); g_error_free(error); g_main_loop_quit(loop); break; } case GST_STATE_CHANGE_READY_TO_NULL: default: //g_print("Unkown message 0x%x\n",GST_MESSAGE_TYPE(msg)); 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 *playerloop; GstBus *playerbus; GstElement *pipeline,*source, *demuxer, *decoder, *conv, *sink, *resample,*resmux; GstCaps *caps; playerloop = g_main_loop_new(NULL,FALSE); gst_init(&argc,&argv); /* Create gstreamer elements */ pipeline = gst_pipeline_new ("audio-player"); source = gst_element_factory_make ("filesrc", "file-source"); sink = gst_element_factory_make ("alsasink", "audio-output"); resample = gst_element_factory_make ("audioresample", "audio-resample"); conv = gst_element_factory_make ("audioconvert", "converter1"); resmux = gst_element_factory_make ("capsfilter", "filter"); caps = gst_caps_new_simple ("audio/x-raw-int", "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, "rate", G_TYPE_INT, 22050, "channels",G_TYPE_INT, 2, NULL ); if (!pipeline || !source || !sink || !resample || !caps || !conv || !resmux ) { g_print ("NO MEM Exiting.\n"); return 1; } /* we set the input filename to the source element */ g_object_set (G_OBJECT (source), "location", argv[1], NULL); demuxer = gst_element_factory_make ("id3demux", "id3-demuxer"); decoder = gst_element_factory_make ("mad", "mp3-decoder"); if (!demuxer || !decoder || !conv) { g_print ("NO MEM Exiting.\n"); return 1; } g_object_set (G_OBJECT (resmux), "caps", caps, NULL); gst_caps_unref (caps); /* file-source -> demuxer -> decoder -> alsa-output */ gst_bin_add_many (GST_BIN (pipeline), source, demuxer, decoder, conv, resample, resmux, sink, NULL); gst_element_link (source, demuxer); gst_element_link_many (decoder, conv, resample,resmux,NULL); if ( !gst_element_link_filtered(resmux,sink,caps) ){ g_printerr("Failed to link elements resample and alsa-sink"); } g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), decoder); playerbus = gst_pipeline_get_bus(GST_PIPELINE(pipeline)); gst_bus_add_watch(playerbus,bus_call,playerloop); gst_object_unref(playerbus); gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING); g_main_loop_run(playerloop); gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_NULL); gst_object_unref(GST_OBJECT(pipeline)); g_print("Exit\n"); return 0; } With Warm Regards Jesu Anuroop Suresh "Any intelligent fool can make things bigger, more complex, and more violent. It takes a touch of genius -- and a lot of courage -- to move in the opposite direction." "Anyone who has never made a mistake has never tried anything new." On Mon, Jan 17, 2011 at 7:54 AM, Cai Yuanqing [via GStreamer-devel] < ml-node+3220635-1402660758-203210 at n4.nabble.com > wrote: > Hi, > > On 01/13/2011 01:39 PM, Anuroop Jesu wrote: > > > Hi , > > > > I Tried the suggestion provided for the by Cai. Thanks for the code Cai. > > > > It works something like this It only allows to playback the 22KHz > > S16LE audio. > > > > What I was trying is to convert the any input format into a 22KHz > > S16LE so I can mux it with other stream of the same property and mux > > multiple streams using alsasink plug:dmix. > I see what you mean :-) > I tried pipeline like this: > gst-launch-0.10 filesrc location=yellow.mp3 ! id3demux ! mad ! > audioconvert ! audioresample ! > 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true' > > ! alsasink > > It works well to first decode any type of mp3 files into PCM,and then > re-sample them into > 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true. > > So you can playback this stream ,or you can replace 'alsasink' to other > elements: > > $ file yellow.mp3 > yellow.mp3: Audio file with ID3 version 2.3.0, contains: MPEG ADTS, > layer III, v1, 128 kbps, 44.1 kHz, JntStereo > > $ gst-launch-0.10 filesrc location=yellow.mp3 ! id3demux ! mad ! > audioconvert ! audioresample ! > 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true' > > ! lame ! filesink location=haha.mp3 > > $ file haha.mp3 > haha.mp3: MPEG ADTS, layer III, v2, 128 kbps, 22.05 kHz, JntStereo > > > The pipeline above turn your stream to encode int a mp3 file with > property as > 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true'. > > > Just add elements behind audioresample and caps in the C code. > > Hope it helps :-) > > Thanks. > > > > > > > > With Warm Regards > > Jesu Anuroop Suresh > > > > "Any intelligent fool can make things bigger, more complex, and more > > violent. It takes a touch of genius -- and a lot of courage -- to move > > in the opposite direction." > > "Anyone who has never made a mistake has never tried anything new." > > > > > > > > > > > > > > On Thu, Jan 13, 2011 at 9:51 AM, Jesu Anuroop Suresh <[hidden email] > > >> > wrote: > > > > Hi Cai, > > > > > > Thanks for you response, I Will try out your suggestion of using > > the filtered link. > > > > Sorry there was some typoerror in my code what I shared. > > > > I did initialized the 'resmux' as capasity filter and used the > > conv not conv1. > > > > The cocde works for me for mp3 playback in its original settings. > > > > > > resample = gst_element_factory_make ("audioresample", > > "audio-resample"); > > conv = gst_element_factory_make ("audioconvert", > > "converter1"); > > resmux = gst_element_factory_make ("capsfilter", "filter"); > > > > caps = gst_caps_new_simple ("audio/x-raw-int", > > "width", G_TYPE_INT, 16, > > "depth", G_TYPE_INT, 16, > > "rate", G_TYPE_INT, 22050, > > "channels",G_TYPE_INT, 2, NULL > > ); > > > > if (!musicPlayer.playPipeline || !source || !sink || > > !resample || !resmux || !caps || !conv) > > { > > g_print ("NO MEM Exiting.\n"); > > return 1; > > } > > > > /* we set the input filename to the source element */ > > g_object_set (G_OBJECT (source), "location", filePath, NULL); > > > > > demuxer = gst_element_factory_make ("id3demux", > > "id3-demuxer"); > > decoder = gst_element_factory_make ("mad", "mp3-decoder"); > > > > if (!demuxer || !decoder || !conv) > > { > > g_print ("NO MEM Exiting.\n"); > > return 1; > > } > > > > With Warm Regards > > Jesu Anuroop Suresh > > > > "Any intelligent fool can make things bigger, more complex, and > > more violent. It takes a touch of genius -- and a lot of courage > > -- to move in the opposite direction." > > "Anyone who has never made a mistake has never tried anything new." > > > > On Thu, Jan 13, 2011 at 7:16 AM, Cai Yuanqing [via > > GStreamer-devel] <[hidden email] > > > wrote: > > > > Hi Suresh: > > Your application have a little problem. :-) > > > > > > On 01/12/2011 08:41 PM, Jesu Anuroop Suresh wrote: > > > > > Hi Sean, > > > > > > Yes, what I was trying is to resample the decoded mp3 data > > to the > > > fixed (22KHZ S16LE) formate, > > > > > > no matter what is the input rate using a C application. > > > > > > Thanks for your response. > > > > > > Here is the piece of the code for the same but it does not > > work with > > > audioresample with the caps filter 'resmux'. This code does > > work > > > without the caps filter 'resmux'. > > > > > > GstElement *source, *demuxer, *decoder, *conv, *sink, > > > *resample, *resmux; > > > GstCaps *caps; > > > > > > gst_init(NULL, NULL); > > > > > > /* Create gstreamer elements */ > > > musicPlayer.playPipeline = gst_pipeline_new > > ("audio-player"); > > > source = gst_element_factory_make ("filesrc", > > "file-source"); > > > sink = gst_element_factory_make ("alsasink", > > "audio-output"); > > > resample = gst_element_factory_make ("audioresample", > > > "audio-resample"); > > > conv = gst_element_factory_make ("audioconvert", > > > "converter1"); > > > > > > caps = gst_caps_new_simple ("audio/x-raw-int", > > > "width", G_TYPE_INT, 16, > > > "depth", G_TYPE_INT, 16, > > > "rate", G_TYPE_INT, > > 22050, > > > "channels",G_TYPE_INT, > > 2, NULL > > > ); > > > > > > if (!musicPlayer.playPipeline || !source || !sink || > > > !resample || !resmux || !caps || !conv) > > > { > > > g_print ("NO MEM Exiting.\n"); > > > return 1; > > > } > > resmux is not initialized yet,here maybe some random > > value,you'd better > > remove it from check list. > > > > > > > > /* we set the input filename to the source element */ > > > g_object_set (G_OBJECT (source), "location", > > filePath, NULL); > > > > > > demuxer = gst_element_factory_make ("id3demux", > > "id3-demuxer"); > > > decoder = gst_element_factory_make ("mad", > > "mp3-decoder"); > > > > > > if (!demuxer || !decoder || !conv1) > > conv1 ? dose it should be conv? > > > > > { > > > g_print ("NO MEM Exiting.\n"); > > > return 1; > > > } > > > > > > g_object_set (G_OBJECT (resmux), "caps", caps, NULL); > > > gst_caps_unref (caps); > > > > > as I said before,resmux haven't initialized ,that's not quite > > right. > > and I suggest you to remove these two lines. > > > > > /* file-source -> demuxer -> decoder -> > > alsa-output */ > > > gst_bin_add_many (GST_BIN (musicPlayer.playPipeline), > > > source, demuxer, decoder, conv, > > resample, > > > resmux,sink, NULL); > > > > > > gst_element_link (source, demuxer); > > > gst_element_link_many (decoder, conv, > > resample,resmux,sink, NULL); > > You can use gst_element_link_filtered to link resample and > > sink with > > caps instead of this way. > > something like: > > gst_element_link (source, demuxer); > > gst_element_link_many (decoder, conv, resample, NULL); > > if ( !gst_element_link_filtered(resample,sink,caps) ){ > > g_printerr("Failed to link elements resample and > > alsa-sink"); > > } > > > > > > > g_signal_connect (demuxer, "pad-added", G_CALLBACK > > > (on_pad_added), decoder); > > > > > > GstBus *bus = > > > gst_pipeline_get_bus(GST_PIPELINE(musicPlayer.playPipeline)); > > > gst_bus_add_watch(bus, bus_call, NULL); > > > gst_object_unref(bus); > > > > > > > > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), > > > GST_STATE_PLAYING); > > > > > > musicPlayer.playLoop = g_main_loop_new(NULL, FALSE); > > > > > > g_main_loop_run(musicPlayer.playLoop); > > > > > > > > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline), > > > GST_STATE_NULL); > > > gst_object_unref(GST_OBJECT(musicPlayer.playPipeline)); > > > > > > > > > > > > > > > > With Warm Regards > > > Jesu Anuroop Suresh > > > > > > "Any intelligent fool can make things bigger, more complex, > > and more > > > violent. It takes a touch of genius -- and a lot of courage > > -- to move > > > in the opposite direction." > > > "Anyone who has never made a mistake has never tried > > anything new." > > > > > > > > I attached my modified source code ,you can try it. > > Hope it helps. > > > > Thanks. > > > > > > -- > > B.R > > > > Cai Yuanqing > > > > > > > ------------------------------------------------------------------------------ > > > > > Protect Your Site and Customers from Malware Attacks > > Learn about various malware tactics and how to avoid them. > > Understand > > malware threats, the impact they can have on your business, > > and how you > > can protect your company and customers by using code signing. > > 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/audioresample-tp3213586p3215098.html > > < > http://gstreamer-devel.966125.n4.nabble.com/audioresample-tp3213586p3215098.html?by-user=t> > > > > > > > To unsubscribe from audioresample, click here > > < > http://gstreamer-devel.966125.n4.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=3213586&code=amVzdWFzQGdtYWlsLmNvbXwzMjEzNTg2fDcwOTc3MzYyOA==&by-user=t>. > > > > > > > > > > > > ------------------------------------------------------------------------ > > View this message in context: Re: audioresample > > < > http://gstreamer-devel.966125.n4.nabble.com/audioresample-tp3213586p3215225.html> > > > > > > > Sent from the GStreamer-devel mailing list archive > > > > at Nabble.com. > > > > > ------------------------------------------------------------------------------ > > > Protect Your Site and Customers from Malware Attacks > > Learn about various malware tactics and how to avoid them. Understand > > > malware threats, the impact they can have on your business, and > > how you > > can protect your company and customers by using code signing. > > http://p.sf.net/sfu/oracle-sfdevnl > > _______________________________________________ > > gstreamer-devel mailing list > > [hidden email] > > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > -- > B.R > > Cai Yuanqing > > > ------------------------------------------------------------------------------ > > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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/audioresample-tp3213586p3220635.html > > To unsubscribe from audioresample, click here. > > -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/audioresample-tp3213586p3220749.html Sent from the GStreamer-devel mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jesuas at gmail.com Mon Jan 17 07:35:11 2011 From: jesuas at gmail.com (Anuroop Jesu) Date: Mon, 17 Jan 2011 12:05:11 +0530 Subject: [gst-devel] Duration of Pipeline In-Reply-To: References: <1295124249.4478.32.camel@blacksheep> Message-ID: Hi Umakant, Below code should solve your problem static gboolean cb_end_stream (GstElement *pipeline) { g_main_loop_quit(loop); } return TRUE; } Have the above function in your code and add "g_timeout_add" with required value before you do run on main loop. gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING); g_timeout_add (1000, (GSourceFunc) cb_print_position, pipeline); g_main_loop_run(playerloop); With Warm Regards Jesu Anuroop Suresh "Any intelligent fool can make things bigger, more complex, and more violent. It takes a touch of genius -- and a lot of courage -- to move in the opposite direction." "Anyone who has never made a mistake has never tried anything new." On Mon, Jan 17, 2011 at 11:06 AM, sudarshan bisht wrote: > Time should be in nano-seconds . > > For more info; > > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-eventsseek.html > > > On Mon, Jan 17, 2011 at 1:04 AM, Umakant Goyal wrote: > >> Hi Thiago, >> >> I tried to run pipeline for specified period of time as suggested by you. >> But it did not work for me. I want to run pipeline for 15 seconds for that i >> have done >> >> bool flag = gst_element_seek (pipeline, 1.0, GST_FORMAT_TIME, >> (GstSeekFlags)(GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE), >> GST_SEEK_TYPE_NONE, 0, GST_SEEK_TYPE_SET, 15*GST_SECOND); >> >> i have also tried following line of code >> >> bool falg = gst_element_seek (vBin, 1.0, GST_FORMAT_TIME, >> (GstSeekFlags)(GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE), >> GST_SEEK_TYPE_SET, 0 * 15*GST_SECOND, GST_SEEK_TYPE_SET, 15*GST_SECOND); >> >> i have run above line of code in once in READY state and once in PAUSED >> state. I have also tried GST_MSECOND instead of GST_SECOND. But nothing >> works for me. >> >> Please help by telling where i am doing wrong while calling above line of >> code. >> >> Thanks >> >> On Sun, Jan 16, 2011 at 2:14 AM, Thiago Sousa Santos < >> thiago.sousa.santos at collabora.co.uk> wrote: >> >>> On Sun, 2011-01-16 at 01:12 +0530, Umakant Goyal wrote: >>> > Hi All, >>> > >>> > Can i make a pipeline to run for specific period of time? >>> >>> Yes. You can send a seek event to the pipeline with a start and a stop >>> time. Check >>> >>> http://www.gstreamer.net/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-seek >>> >>> -- >>> Thiago >>> >>> > >>> > Thanks >>> > >>> > >>> ------------------------------------------------------------------------------ >>> > Protect Your Site and Customers from Malware Attacks >>> > Learn about various malware tactics and how to avoid them. Understand >>> > malware threats, the impact they can have on your business, and how you >>> > can protect your company and customers by using code signing. >>> > 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 >>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Protect Your Site and Customers from Malware Attacks >>> Learn about various malware tactics and how to avoid them. Understand >>> malware threats, the impact they can have on your business, and how you >>> can protect your company and customers by using code signing. >>> 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 >>> >> >> >> >> ------------------------------------------------------------------------------ >> Protect Your Site and Customers from Malware Attacks >> Learn about various malware tactics and how to avoid them. Understand >> malware threats, the impact they can have on your business, and how you >> can protect your company and customers by using code signing. >> 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 > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 Mon Jan 17 07:45:40 2011 From: umakantgoyal1 at gmail.com (Umakant Goyal) Date: Mon, 17 Jan 2011 12:15:40 +0530 Subject: [gst-devel] Duration of Pipeline In-Reply-To: References: <1295124249.4478.32.camel@blacksheep> Message-ID: Hi, Thanks, As i mentioned in my last reply i have also tried to give time unit as nanosecond. But it did not work for me. It would be great if you can help me by telling what i am doing wrong in seek method to specify the duration of pipeline. Thanks On 1/17/11, Anuroop Jesu wrote: > Hi Umakant, > > Below code should solve your problem > > static gboolean > cb_end_stream (GstElement *pipeline) > { > g_main_loop_quit(loop); > } > return TRUE; > } > > > Have the above function in your code and add "g_timeout_add" with required > value before you do run on main loop. > > gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING); > g_timeout_add (1000, (GSourceFunc) cb_print_position, pipeline); > g_main_loop_run(playerloop); > > With Warm Regards > Jesu Anuroop Suresh > > "Any intelligent fool can make things bigger, more complex, and more > violent. It takes a touch of genius -- and a lot of courage -- to move in > the opposite direction." > "Anyone who has never made a mistake has never tried anything new." > > > > > > > On Mon, Jan 17, 2011 at 11:06 AM, sudarshan bisht > wrote: > >> Time should be in nano-seconds . >> >> For more info; >> >> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-eventsseek.html >> >> >> On Mon, Jan 17, 2011 at 1:04 AM, Umakant Goyal >> wrote: >> >>> Hi Thiago, >>> >>> I tried to run pipeline for specified period of time as suggested by you. >>> But it did not work for me. I want to run pipeline for 15 seconds for >>> that i >>> have done >>> >>> bool flag = gst_element_seek (pipeline, 1.0, GST_FORMAT_TIME, >>> (GstSeekFlags)(GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE), >>> GST_SEEK_TYPE_NONE, 0, GST_SEEK_TYPE_SET, 15*GST_SECOND); >>> >>> i have also tried following line of code >>> >>> bool falg = gst_element_seek (vBin, 1.0, GST_FORMAT_TIME, >>> (GstSeekFlags)(GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE), >>> GST_SEEK_TYPE_SET, 0 * 15*GST_SECOND, GST_SEEK_TYPE_SET, >>> 15*GST_SECOND); >>> >>> i have run above line of code in once in READY state and once in PAUSED >>> state. I have also tried GST_MSECOND instead of GST_SECOND. But nothing >>> works for me. >>> >>> Please help by telling where i am doing wrong while calling above line of >>> code. >>> >>> Thanks >>> >>> On Sun, Jan 16, 2011 at 2:14 AM, Thiago Sousa Santos < >>> thiago.sousa.santos at collabora.co.uk> wrote: >>> >>>> On Sun, 2011-01-16 at 01:12 +0530, Umakant Goyal wrote: >>>> > Hi All, >>>> > >>>> > Can i make a pipeline to run for specific period of time? >>>> >>>> Yes. You can send a seek event to the pipeline with a start and a stop >>>> time. Check >>>> >>>> http://www.gstreamer.net/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-seek >>>> >>>> -- >>>> Thiago >>>> >>>> > >>>> > Thanks >>>> > >>>> > >>>> ------------------------------------------------------------------------------ >>>> > Protect Your Site and Customers from Malware Attacks >>>> > Learn about various malware tactics and how to avoid them. Understand >>>> > malware threats, the impact they can have on your business, and how >>>> > you >>>> > can protect your company and customers by using code signing. >>>> > 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 >>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Protect Your Site and Customers from Malware Attacks >>>> Learn about various malware tactics and how to avoid them. Understand >>>> malware threats, the impact they can have on your business, and how you >>>> can protect your company and customers by using code signing. >>>> 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 >>>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Protect Your Site and Customers from Malware Attacks >>> Learn about various malware tactics and how to avoid them. Understand >>> malware threats, the impact they can have on your business, and how you >>> can protect your company and customers by using code signing. >>> 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 >> >> >> ------------------------------------------------------------------------------ >> Protect Your Site and Customers from Malware Attacks >> Learn about various malware tactics and how to avoid them. Understand >> malware threats, the impact they can have on your business, and how you >> can protect your company and customers by using code signing. >> 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 jesuas at gmail.com Mon Jan 17 08:04:01 2011 From: jesuas at gmail.com (Anuroop Jesu) Date: Mon, 17 Jan 2011 12:34:01 +0530 Subject: [gst-devel] Duration of Pipeline In-Reply-To: References: <1295124249.4478.32.camel@blacksheep> Message-ID: Hi Umakant, In function 'g_timeout_add' you pass the duration for which you want to run your pipeline, On timeout you exit from main loop, you don't need to seek. seek will not exit from main loop and will continue to run till it reaches EOS. I have tested the exit using the code as given below static gboolean cb_print_position (GMainLoop *loop) { g_main_loop_quit(loop); return TRUE; } gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING); g_timeout_add (1000, (GSourceFunc) cb_print_position, playerloop); g_main_loop_run(playerloop); Correct me If I'm wrong, you want to play a stream for a X duration of time and then exit correct. Then above code will work for you. With Warm Regards Jesu Anuroop Suresh "Any intelligent fool can make things bigger, more complex, and more violent. It takes a touch of genius -- and a lot of courage -- to move in the opposite direction." "Anyone who has never made a mistake has never tried anything new." On Mon, Jan 17, 2011 at 12:15 PM, Umakant Goyal wrote: > Hi, > > Thanks, As i mentioned in my last reply i have also tried to give time > unit as nanosecond. But it did not work for me. > > It would be great if you can help me by telling what i am doing wrong > in seek method to specify the duration of pipeline. > > Thanks > > On 1/17/11, Anuroop Jesu wrote: > > Hi Umakant, > > > > Below code should solve your problem > > > > static gboolean > > cb_end_stream (GstElement *pipeline) > > { > > g_main_loop_quit(loop); > > } > > return TRUE; > > } > > > > > > Have the above function in your code and add "g_timeout_add" with > required > > value before you do run on main loop. > > > > gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING); > > g_timeout_add (1000, (GSourceFunc) cb_print_position, pipeline); > > g_main_loop_run(playerloop); > > > > With Warm Regards > > Jesu Anuroop Suresh > > > > "Any intelligent fool can make things bigger, more complex, and more > > violent. It takes a touch of genius -- and a lot of courage -- to move in > > the opposite direction." > > "Anyone who has never made a mistake has never tried anything new." > > > > > > > > > > > > > > On Mon, Jan 17, 2011 at 11:06 AM, sudarshan bisht < > bisht.sudarshan at gmail.com > >> wrote: > > > >> Time should be in nano-seconds . > >> > >> For more info; > >> > >> > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-eventsseek.html > >> < > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-eventsseek.html > > > >> > >> On Mon, Jan 17, 2011 at 1:04 AM, Umakant Goyal > >> wrote: > >> > >>> Hi Thiago, > >>> > >>> I tried to run pipeline for specified period of time as suggested by > you. > >>> But it did not work for me. I want to run pipeline for 15 seconds for > >>> that i > >>> have done > >>> > >>> bool flag = gst_element_seek (pipeline, 1.0, GST_FORMAT_TIME, > >>> (GstSeekFlags)(GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE), > >>> GST_SEEK_TYPE_NONE, 0, GST_SEEK_TYPE_SET, 15*GST_SECOND); > >>> > >>> i have also tried following line of code > >>> > >>> bool falg = gst_element_seek (vBin, 1.0, GST_FORMAT_TIME, > >>> (GstSeekFlags)(GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE), > >>> GST_SEEK_TYPE_SET, 0 * 15*GST_SECOND, GST_SEEK_TYPE_SET, > >>> 15*GST_SECOND); > >>> > >>> i have run above line of code in once in READY state and once in PAUSED > >>> state. I have also tried GST_MSECOND instead of GST_SECOND. But nothing > >>> works for me. > >>> > >>> Please help by telling where i am doing wrong while calling above line > of > >>> code. > >>> > >>> Thanks > >>> > >>> On Sun, Jan 16, 2011 at 2:14 AM, Thiago Sousa Santos < > >>> thiago.sousa.santos at collabora.co.uk> wrote: > >>> > >>>> On Sun, 2011-01-16 at 01:12 +0530, Umakant Goyal wrote: > >>>> > Hi All, > >>>> > > >>>> > Can i make a pipeline to run for specific period of time? > >>>> > >>>> Yes. You can send a seek event to the pipeline with a start and a stop > >>>> time. Check > >>>> > >>>> > http://www.gstreamer.net/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-seek > >>>> > >>>> -- > >>>> Thiago > >>>> > >>>> > > >>>> > Thanks > >>>> > > >>>> > > >>>> > ------------------------------------------------------------------------------ > >>>> > Protect Your Site and Customers from Malware Attacks > >>>> > Learn about various malware tactics and how to avoid them. > Understand > >>>> > malware threats, the impact they can have on your business, and how > >>>> > you > >>>> > can protect your company and customers by using code signing. > >>>> > 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 > >>>> > >>>> > >>>> > >>>> > >>>> > ------------------------------------------------------------------------------ > >>>> Protect Your Site and Customers from Malware Attacks > >>>> Learn about various malware tactics and how to avoid them. Understand > >>>> malware threats, the impact they can have on your business, and how > you > >>>> can protect your company and customers by using code signing. > >>>> 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 > >>>> > >>> > >>> > >>> > >>> > ------------------------------------------------------------------------------ > >>> Protect Your Site and Customers from Malware Attacks > >>> Learn about various malware tactics and how to avoid them. Understand > >>> malware threats, the impact they can have on your business, and how you > >>> can protect your company and customers by using code signing. > >>> 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 > >> > >> > >> > ------------------------------------------------------------------------------ > >> Protect Your Site and Customers from Malware Attacks > >> Learn about various malware tactics and how to avoid them. Understand > >> malware threats, the impact they can have on your business, and how you > >> can protect your company and customers by using code signing. > >> 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 > >> > >> > > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 bilboed at gmail.com Mon Jan 17 09:40:18 2011 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 17 Jan 2011 09:40:18 +0100 Subject: [gst-devel] playing live mpegts stream over udp In-Reply-To: References: Message-ID: <1295253618.2449.7.camel@deumeu> On Fri, 2011-01-14 at 20:08 +0100, Julien Isorce wrote: > Hi, > > I am trying to play nicely a usual mpegts stream from network. (720p > h264/ac3 mpegts over udp multicast) > The pipeline is: > gst-launch uridecodebin uri=239.102.101.3:1013 ! then audio and video > renderers. > My network interface is connected in 1GO. > > With default properties, the video has a lot of macro block. If I > record the network packets in a pcap file then > I can perfeclty play the stream using pcapparse. > So I think the problem comes from some buffering pbs. > > When configuring with some heuristics use-buffering=1, buffer-duration > and buffer-size properties of uridecodebin, > the video contains less macro blocks but it still contains enough > block to make me fill it's not normal. > > With VLC the stream is played almost perfectly. > > Am I doing something wrong ? Should I need a something like a > jitterbuffer (no rtp here) ? I had a look at the problem over the weekend (reproduced it by using vlc server streaming mpeg-ts over udp) and basically the timing algorithms in mpegtsdemux are completely bogus (which is something we knew, but never confirmed). By introducing an evil hack in mpegtsdemux for that specific setup I could get it to play much more smoothly. You also want to set the udpsrc buffer-size property to a much higher value (500kB should do it) if you're using high-ish bandwith streams. The solution ? * Stream using proper rtsp (when vlc server is configured that way it works flawlessly with GStreamer). * Fix mpegtsdemux (I'd recommend buying hospital-sized packs of aspirin if you do that) * Help work on the from-scratch tsdemux project (some people looked into a similar issue some time back) Edward > > Any help would be appreciated. > Sincerely > Julien Isorce > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 kapil.agl at gmail.com Mon Jan 17 11:10:03 2011 From: kapil.agl at gmail.com (Kapil Agrawal) Date: Mon, 17 Jan 2011 15:40:03 +0530 Subject: [gst-devel] playing live mpegts stream over udp In-Reply-To: References: Message-ID: Hello Julien, Based on my personal experience I think you might even try ffdemux_mpegts. Yeah fixing mpegtsdemux would surely need aspirin. Best Kapil On Sat, Jan 15, 2011 at 12:38 AM, Julien Isorce wrote: > Hi, > > I am trying to play nicely a usual mpegts stream from network. (720p > h264/ac3 mpegts over udp multicast) > The pipeline is: > gst-launch uridecodebin uri=239.102.101.3:1013 ! then audio and video > renderers. > My network interface is connected in 1GO. > > With default properties, the video has a lot of macro block. If I record > the network packets in a pcap file then > I can perfeclty play the stream using pcapparse. > So I think the problem comes from some buffering pbs. > > When configuring with some heuristics use-buffering=1, buffer-duration and > buffer-size properties of uridecodebin, > the video contains less macro blocks but it still contains enough block to > make me fill it's not normal. > > With VLC the stream is played almost perfectly. > > Am I doing something wrong ? Should I need a something like a jitterbuffer > (no rtp here) ? > > Any help would be appreciated. > Sincerely > Julien Isorce > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 > > -- 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 julien.isorce at gmail.com Mon Jan 17 12:31:38 2011 From: julien.isorce at gmail.com (Julien Isorce) Date: Mon, 17 Jan 2011 12:31:38 +0100 Subject: [gst-devel] playing live mpegts stream over udp In-Reply-To: References: Message-ID: Hi, Thx for your informations. I set the udpsrc buffer-size property to 20 000 000 bytes and it seems to be fine for now. (If I put 10 000 000 I am getting some macro blocks, same with 500 000) (the stream is around 13 MB/s) I cannot control the stream server (and I cannot select an other encapsulation) since it's a professional source of one of our customer. Sincerely Julien 2011/1/17 Kapil Agrawal > Hello Julien, > > Based on my personal experience I think you might even try ffdemux_mpegts. > > Yeah fixing mpegtsdemux would surely need aspirin. > > Best > Kapil > > On Sat, Jan 15, 2011 at 12:38 AM, Julien Isorce wrote: > >> Hi, >> >> I am trying to play nicely a usual mpegts stream from network. (720p >> h264/ac3 mpegts over udp multicast) >> The pipeline is: >> gst-launch uridecodebin uri=239.102.101.3:1013 ! then audio and video >> renderers. >> My network interface is connected in 1GO. >> >> With default properties, the video has a lot of macro block. If I record >> the network packets in a pcap file then >> I can perfeclty play the stream using pcapparse. >> So I think the problem comes from some buffering pbs. >> >> When configuring with some heuristics use-buffering=1, buffer-duration and >> buffer-size properties of uridecodebin, >> the video contains less macro blocks but it still contains enough block to >> make me fill it's not normal. >> >> With VLC the stream is played almost perfectly. >> >> Am I doing something wrong ? Should I need a something like a jitterbuffer >> (no rtp here) ? >> >> Any help would be appreciated. >> Sincerely >> Julien Isorce >> >> >> >> ------------------------------------------------------------------------------ >> Protect Your Site and Customers from Malware Attacks >> Learn about various malware tactics and how to avoid them. Understand >> malware threats, the impact they can have on your business, and how you >> can protect your company and customers by using code signing. >> 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 >> >> > > > -- > www.mediamagictechnologies.com (Gstreamer, ffmpeg, Red5, Streaming) > twitter handle: @gst_kaps > http://www.linkedin.com/in/kapilagrawal > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 lubensky at cognitec.com Mon Jan 17 14:22:47 2011 From: lubensky at cognitec.com (Andreas Lubensky) Date: Mon, 17 Jan 2011 14:22:47 +0100 Subject: [gst-devel] grabbing frames from running stream Message-ID: <201101171422.48703.lubensky@cognitec.com> Hello, I want to use gstreamer to play all kinds of video sources, especially webcams, video files and DVDs. In the framework i am using, video sources get grabbed image by image in a loop. Currently, i am not sure how to combine that with gstreamer. My idea is to write a video sink that always keeps a copy of the last image. But this seems rather ugly...i thought there might be some way to get a snapshot of a running stream. That might be no problem with webcams, but i couldn't figure out yet how to do it with video files. As i am new to gstreamer, i probably just need a nudge in the right direction... From kapil.agl at gmail.com Mon Jan 17 14:55:08 2011 From: kapil.agl at gmail.com (Kapil Agrawal) Date: Mon, 17 Jan 2011 19:25:08 +0530 Subject: [gst-devel] grabbing frames from running stream In-Reply-To: <201101171422.48703.lubensky@cognitec.com> References: <201101171422.48703.lubensky@cognitec.com> Message-ID: one way is add an identity element just before sink & using the handoff signal , save the buffer ? http://www.gstreamer.net/data/doc/gstreamer/0.10.3/gstreamer-plugins/html/gstreamer-plugins-identity.html Best Luck Kapil On Mon, Jan 17, 2011 at 6:52 PM, Andreas Lubensky wrote: > Hello, > > I want to use gstreamer to play all kinds of video sources, especially > webcams, video files and DVDs. > In the framework i am using, video sources get grabbed image by image in a > loop. Currently, i am not sure how to combine that with gstreamer. My idea > is > to write a video sink that always keeps a copy of the last image. But this > seems rather ugly...i thought there might be some way to get a snapshot of > a > running stream. That might be no problem with webcams, but i couldn't > figure > out yet how to do it with video files. > As i am new to gstreamer, i probably just need a nudge in the right > direction... > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 > -- 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 malloblenne at gmail.com Mon Jan 17 15:02:18 2011 From: malloblenne at gmail.com (Mauro Brenna) Date: Mon, 17 Jan 2011 15:02:18 +0100 Subject: [gst-devel] [OT] RTSP: SET_PARAMETER. Which type of parameters are/can define other than text/parameters? Message-ID: Hello, I am implementing a rtspu server/client (rtsp based on UDP) and I will use gstreamer to receive the data in RTP. I would like to set/get some application specific parameters on my server. Can I use or extend the type of parameters acceptable? I mean, I would like to use something like float/parameter, byte/parameter ushort/parameter ... instead of text/parameter. I prefer send binary data if it is possible, even if it is directly mapped in a readable character. Is there any documents where I can find some standardized type of parameter other than text/parameter for RTSP? I am asking this question to your mailing list because gstreamer users helped me to solve another problem about rtsp. Thanks, Mauro From malloblenne at gmail.com Mon Jan 17 15:11:40 2011 From: malloblenne at gmail.com (Mauro Brenna) Date: Mon, 17 Jan 2011 15:11:40 +0100 Subject: [gst-devel] Is RTSPU implemented in gstreamer? Can I use RTSP over UDP? In-Reply-To: <1294934317.1875.48.camel@meany> References: <1294934317.1875.48.camel@meany> Message-ID: Thanks for the reply. I think about your reply and I will implement a little rtspu (over UDP) client for my application since the server does not implement the TCP transport protocol. Reading the RFC, I believe that rtsp means RTSP commands are sent over TCP, RTSPU over UDP. The way to specify if the RTP data packets would be sent (TCP/UDP) is in the Transport Header as lower-transport. See RFC 2326, in section "12.39 Transport". I think the two things are independently so you could have: RTSP (over TCP) + RTP data over TCP or UDP RTSPu (over UDP) + RTP data over TCP or UDP. Do you think I misunderstood something? Bye, Mauro From wl2776 at gmail.com Mon Jan 17 15:13:45 2011 From: wl2776 at gmail.com (wl2776) Date: Mon, 17 Jan 2011 06:13:45 -0800 (PST) Subject: [gst-devel] What is 'stream time of the segment'? Message-ID: <1295273625981-3221217.post@n4.nabble.com> Unfortunately, the article in the GTK-docs about GstSegment doesn't reveal too much details about GstSegment::time field. It says that it is the 'stream time of the segment'. part-segments.txt on the gstreamer/dosc/design/ doesn't mention it also. part-synchronization.txt says that the "stream time is also known as the position in the stream and is a value between 0 and the total duration of the media file". So, could anyone clarify if that definition of the stream time applies to the segment stream time? If I am right, then GstSegment::time should be a moment of the segment start, provided stream playback starts from 0. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-is-stream-time-of-the-segment-tp3221217p3221217.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bilboed at gmail.com Mon Jan 17 15:42:28 2011 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 17 Jan 2011 15:42:28 +0100 Subject: [gst-devel] Third pre-release of GStreamer Editing Services 0.10.0.4 In-Reply-To: <1294775102.5430.3.camel@deumeu> References: <1294671763.23475.24.camel@localhost> <1294775102.5430.3.camel@deumeu> Message-ID: <1295275348.11672.1.camel@localhost> Hi all, The third pre-release for the GStreamer Editing Services is available at: http://gstreamer.freedesktop.org/data/src/gst-editing-services/pre/ Documentation can be found online at: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-editing-services/html/ This third pre-release contains: * Usage fixes for ges-launch-0.10 Ubuntu users might also be able to test this pre-release by using the GStreamer PPA (as soon as they get uploaded there). Please test it thoroughly, and let us know of any important issues by filing blocker bugs in bugzilla: http://gstreamer.freedesktop.org/bugs/ (Component: gst-editing-services) Further pre-releases will follow as needed. If all goes well, 0.10.1 release should happen later this week when GStreamer core/-base 0.10.32 are out, as per http://gstreamer.freedesktop.org/wiki/ReleasePlanning2011 Edward From malloblenne at gmail.com Tue Jan 18 09:47:12 2011 From: malloblenne at gmail.com (Mauro Brenna) Date: Tue, 18 Jan 2011 09:47:12 +0100 Subject: [gst-devel] Is RTSPU implemented in gstreamer? Can I use RTSP over UDP? In-Reply-To: <1294934317.1875.48.camel@meany> References: <1294934317.1875.48.camel@meany> Message-ID: Thanks for the reply. I think about your reply and I will implement a little rtspu (over UDP) client for my application since the server does not implement the TCP transport protocol. Reading the RFC, I believe that rtsp means RTSP commands are sent over TCP, RTSPU over UDP. The way to specify if the RTP data packets would be sent (TCP/UDP) is in the Transport Header as lower-transport. See RFC 2326, in section "12.39 Transport". I think the two things are independently so you could have: RTSP (over TCP) + RTP data over TCP or UDP RTSPu (over UDP) + RTP data over TCP or UDP. Do you think I misunderstood something? Bye, Mauro From malloblenne at gmail.com Tue Jan 18 09:50:20 2011 From: malloblenne at gmail.com (Mauro Brenna) Date: Tue, 18 Jan 2011 09:50:20 +0100 Subject: [gst-devel] [OT] RTSP: SET_PARAMETER. Which type of parameters are/can define other than text/parameters? In-Reply-To: References: Message-ID: Hello, I am implementing a rtspu server/client (rtsp based on UDP) and I will use gstreamer to receive the data in RTP. I would like to set/get some application specific parameters on my server. Can I use or extend the type of parameters acceptable? I mean, I would like to use something like float/parameter, byte/parameter ushort/parameter ... instead of text/parameter. I prefer send binary data if it is possible, even if it is directly mapped in a readable character. Is there any documents where I can find some standardized type of parameter other than text/parameter for RTSP? I am asking this question to your mailing list because gstreamer users helped me to solve another problem about rtsp. Thanks, Mauro From jeetu.golani at gmail.com Tue Jan 18 11:02:01 2011 From: jeetu.golani at gmail.com (jeetu.golani at gmail.com) Date: Tue, 18 Jan 2011 15:02:01 +0500 Subject: [gst-devel] Timestamping a video file with gstreamer Message-ID: Hello, First, I'm new to gstreamer and video manipulation and file formats in general, therefore forgive me if what I ask is obvious :) I would like to timestamp a video file sort of like how the timeoverlay and clockoverlay elements do. However, I would like the time to begin from the time the file was recorded. I'm using MythTV to record video and upon recording I would like the file (or a clip) to be transcoded with the timestamp - am open to having the final recording in any of the popular container formats. My questions: - Is there a way for me to overlay a user defined time? Can I somehow manipulate the buffer or using some means introduce a time into the stream using the timeoverlay/clockoverlay elements? - Can I somehow introduce a timestamp in the file container itself and then this gets used as the start time? Google has so far not been able to provide me with my answers (or maybe I've been looking in the wrong place). There is mention of a patch for mencoder/mplayer that provided this functionality but that's obsolete now. Same for ffmpeg that provided something similar using a vhook plugin. I would sincerely appreciate your help on this. I'm definitely on my wits end :) Thanks Bye for now -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Tue Jan 18 14:18:47 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 18 Jan 2011 13:18:47 +0000 Subject: [gst-devel] even more pre-releases! GStreamer core, -base, -good, -ugly, -bad In-Reply-To: <1294790539.27422.19.camel@zingle> References: <1294488912.3488.9.camel@zingle> <1294790539.27422.19.camel@zingle> Message-ID: <1295356727.30497.15.camel@zingle> On Wed, 2011-01-12 at 00:02 +0000, Tim-Philipp M?ller wrote: Hi, A third and hopefully last round of pre-releases for GStreamer core, gst-plugins-base, gst-plugins-good, gst-plugins-ugly and gst-plugins-bad is now available at the following locations: da27c2e0ea12b3821cb2d1d96a2fe2f7 gstreamer-0.10.31.4.tar.gz bf6cef222fa8eab79e89435c68d8afa4 gstreamer-0.10.31.4.tar.bz2 http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.31.4.tar.gz http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.31.4.tar.bz2 91c9d64907161a97c10a046d2f440376 gst-plugins-base-0.10.31.4.tar.gz 095a05fa16bb375b8f8271a0a07f7dd4 gst-plugins-base-0.10.31.4.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.31.4.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.31.4.tar.bz2 59c11db25d4086224a119bac110d106c gst-plugins-good-0.10.26.4.tar.gz 86b6a1c6de6191d2e1e16784cd1b65f6 gst-plugins-good-0.10.26.4.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-good/pre/gst-plugins-good-0.10.26.4.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-good/pre/gst-plugins-good-0.10.26.4.tar.bz2 5be3213b67517af196afa5d03858d782 gst-plugins-ugly-0.10.16.4.tar.gz 61fcb502410fcd88d5f390b3179d0413 gst-plugins-ugly-0.10.16.4.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-ugly/pre/gst-plugins-ugly-0.10.16.4.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-ugly/pre/gst-plugins-ugly-0.10.16.4.tar.bz2 8b232990d56933661d2c5d1696b5d837 gst-plugins-bad-0.10.20.4.tar.gz 42e3de6dd7f79ee450e33f6d7335ded4 gst-plugins-bad-0.10.20.4.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.20.4.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.20.4.tar.bz2 ----- Ubuntu users will also be able to test these pre-releases by using the GStreamer PPA where they will be available shortly. ----- Changes from last pre-releases: Core: - misc. gobject-introspection build fixes - gobject-introspection requirement bumped to >= 0.9.12 - minor controller build fix gst-plugins-base: - minor typefinding caps fix - make gdp buffer metadata writable where needed - app: export GType for GstAppStreamType enum - pbutils: misc. discoverer fixes - pbutils. encoding profile saving fixes (NB: gep keyfile header changed) gst-plugins-good: - v4l2: fix compilation with old kernel headers - avimux, matroskamux: set src caps on startup, so they're not lost when the element is re-used - pulsesink: Uncork stream while flushing the ringbuffer - deinterlace: fix some off-by-one bugs gst-plugins-ugly: - mpeg2dec: fix cropping in Y42B format gst-plugins-bad: - y4mdec: default to I420 if no pixel format is specified - use g_error_free() for GErrors, not g_free() - dcaparse: fix sync word for 14-bit little endian coding - camerabin: set pad-negotiation-mode to active for output-selector - camerabin: enable more conversion flags by default - katetiger: ensure the kate pad does not shoot ahead of the video pad - qtmux: set caps on sink pad on start-up to fix re-use - audioparsers: fix refcounting issue ----- PACKAGERS: note that plugins were moved in this release cycle: the selector and valve plugins from -bad were merged into corelements in core, and the jack plugin was moved to gst-plugins-good. ----- Please test them thoroughly with all your favourite apps, and let us know of any regressions or other important issues by filing blocker bugs in bugzilla: http://gstreamer.freedesktop.org/bugs/ Further pre-releases will follow as needed, but hopefully this is the last pre-release, with final releases happening in the next few days. Happy testing! Cheers -Tim From wim.taymans at gmail.com Tue Jan 18 17:40:37 2011 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 18 Jan 2011 17:40:37 +0100 Subject: [gst-devel] Is RTSPU implemented in gstreamer? Can I use RTSP over UDP? In-Reply-To: References: <1294934317.1875.48.camel@meany> Message-ID: On Tue, Jan 18, 2011 at 9:47 AM, Mauro Brenna wrote: > Thanks for the reply. > I think about your reply and I will implement a little rtspu (over > UDP) client for my application since the server does not implement the > TCP transport protocol. > > Reading the RFC, I believe that rtsp means RTSP commands are sent over > TCP, RTSPU over UDP. > The way to specify if the RTP data packets would be sent (TCP/UDP) is > in the Transport Header as lower-transport. See RFC 2326, in section > "12.39 Transport". I think you misunderstood that part of the RFC, it talks about the transport of RTP packets, not RTSP. > > I think the two things are independently so you could have: > > RTSP (over TCP) + RTP data over TCP or UDP > RTSPu (over UDP) + RTP data over TCP or UDP. So how do you think it would work? you send the RTSP request over UDP? How would you get the reply? How does the server know where to send the reply to? How is this going to work over NAT? What will happen when the UDP packets get lost, duplicated or reordered? What would you do if the RTSP request is too big to fit in the MTU? What would you do if the reply is too big for the MTU? If you manage to get RTSP over UDP working, how and when are you going to open the TCP connection when you decide to do RTP over TCP? You need to answer those questions first before any of what you talk about makes sense. As I said: "I am not aware of any player doing the RTSP requests over UDP nor is it clear from the spec how this is supposed to work, or how a server should support it. I don't know why that part is in the spec or what it means." Wim > > Do you think I misunderstood something? > > Bye, > > Mauro > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 lane at brooks.nu Tue Jan 18 17:16:41 2011 From: lane at brooks.nu (Lane Brooks) Date: Tue, 18 Jan 2011 09:16:41 -0700 Subject: [gst-devel] kenburns plugin update - now a general video transform plugin Message-ID: <4D35BCE9.9000506@brooks.nu> I have an updated kenburns plugin with the following changes: - I dropped the internal controller. Now you use a GstContoller to control the 'xpos', 'ypos', and 'zpos'. - I added rotation transforms as well. You can now also control an 'xrot', 'yrot', and 'zrot' parameter. You can check out a video I put together that show cases these features at http://www.youtube.com/watch?v=JFm8Rra4xzQ. This video first shows the translation and rotation features with the 'xpos', 'ypos', 'zpos', 'xrot', 'yrot',and 'zrot' parameters hooked up to LFO's with an stmpe video test source. It then shows some sample kenburns effects, which use the 'xpos', 'ypos', and 'zpos' parameters hooked up to linear interpolating GstControllers. It then shows some sample video transitions that also make use of the position and rotation parameters hooked up to linear interpolating GstControllers. You can still clone the source code at: https://github.com/dirjud/kenburns Given that the feature set has grown beyond Ken Burns effects, I want to rename this element to something else. I was thinking 'videotransform' but am open to other suggestions. One reason against videotransform is that there is already a base class in gstreamer call GstVideoTransform. It is not exposed to the user, so it only a potential confusion for developers. But if there is a good name that is not a potential confusion, I would go with that instead. After I decide on a new name and hear any additional feedback, I will post a patch to gst-plugins-bad. Thanks, Lane From gibrovacco at gmail.com Tue Jan 18 21:43:29 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Tue, 18 Jan 2011 22:43:29 +0200 Subject: [gst-devel] Getting converted frame from playbin2 element In-Reply-To: <1294668953244-3207234.post@n4.nabble.com> References: <1294053159060-3171830.post@n4.nabble.com> <1294668953244-3207234.post@n4.nabble.com> Message-ID: Hi, looks like my time machine for jumping 11 days in the future eventually works.. (lame excuse for my late reply) Maybe you've already fixed your issue, but I still have a few ideas. On Mon, Jan 10, 2011 at 4:15 PM, vaisaari wrote: > >> imo this should not happen, as the signal handler (delegated to >> playsink) internally calls gst_play_sink_get_last_frame, which locks >> playsink. Can you post to pastebin the output you get after setting >> GST_DEBUG=playsink:5 ? > > You can find the output from ?http://pastebin.com/raw.php?i=HgD1S8Uz here ?. > The output looks the same whether or not the conversion is successful. yep, it looks good. > >> Which video decoder are you using? > > I have tried it with both uncompressed and MPEG4 encoded videos. > >> Besides, if you're trying to render a video clip in Qt, you may >> consider using QtGStreamer.. > > It looks very promising. At some point, I most certainly will. here: http://doc.qt.nokia.com/qt-maemo-4.7/qimage.html#QImage-4 is stated the validity of the data into memory must last at least as long as the QImage itself and in the list you've reported it appears you're not handling buffer references at all. At best, you're leaking memory like there's no tomorrow, at worst (depending on the pov) the buffer "buffer" gets unreffed somewhere by chances and then you might observed teared images (depending on the color space) or even segfaults. Another good reason for using QtGstreamer ;) Even if it worked, I would anyway not use such an approach to convert all the frames in a clip (check what does the gst_video_convert_frame function do to be convinced). It's imo better to use a color space conversion element (e.g. ffmpegcolorspace) at the end of the pipeline or after a tee and an appsink element. Regards, Marco > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Getting-converted-frame-from-playbin2-element-tp3171830p3207234.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. ? Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 jorney_dong at asus.com Wed Jan 19 04:15:30 2011 From: jorney_dong at asus.com (Jorney_dong) Date: Wed, 19 Jan 2011 11:15:30 +0800 Subject: [gst-devel] amrparse & amrwbdec Message-ID: <4D365752.30001@asus.com> Hi I am writing a amr-wb player, using amrparse & amrwbdec. There is a long time blocking (10s or even more than 1min) some time while seeking. I found that amrparse's gst_amrparse_check_valid_frame() data is nomal while blocking, but the amrwbdec's chain function's data is abnormal meanwhile. then could I say the problem is in amrparse? It's important, Thanks a lot! ===================================================================================================================================== This email and any attachments to it contain confidential information and are intended solely for the use of the individual to whom it is addressed.If you are not the intended recipient or receive it accidentally, please immediately notify the sender by e-mail and delete the message and any attachments from your computer system, and destroy all hard copies. If any, please be advised that any unauthorized disclosure, copying, distribution or any action taken or omitted in reliance on this, is illegal and prohibited. Furthermore, any views or opinions expressed are solely those of the author and do not represent those of ASUSTeK. Thank you for your cooperation. ===================================================================================================================================== From malloblenne at gmail.com Wed Jan 19 09:59:14 2011 From: malloblenne at gmail.com (Mauro Brenna) Date: Wed, 19 Jan 2011 09:59:14 +0100 Subject: [gst-devel] Is RTSPU implemented in gstreamer? Can I use RTSP over UDP? In-Reply-To: References: <1294934317.1875.48.camel@meany> Message-ID: Hi, As regards the possibility to transmit RTSP packets over UDP you can also read in the RFC (2326): " There is no notion of an RTSP connection; instead, a server maintains a session labeled by an identifier. An RTSP session is in no way tied to a transport-level connection such as a TCP connection. During an RTSP session, an RTSP client may open and close many reliable transport connections to the server to issue RTSP requests. Alternatively, it may use a connectionless transport protocol such as UDP." So, I am practically convinced you could send also the RTSP commands, not only the RTP data, via UDP. Anywat, I don't want to start a post-reply war. I would like only to understand what it is correct and what it is not. My application is very specific and I will implement a very very simple RTSP over UDP protocol. Now, to answer your question: - So how do you think it would work? you send the RTSP request over UDP? Yes, I would do like that. Note the difference between sessionless commands and commands that refers to a session. You need to mantain a state for each session, if you would like to stream in unicast to more than one client, otherwise you can use a single 'instance' of RTSP server state and refuse to SETUP if a second client asks you so. - How would you get the reply? Via UDP, you send the SESSION id in the reply. - How does the server know where to send the reply to? I do not understand the question. The server should know specific parameters via SETUP method. Or simply replies to the IP of the sender. Which is the difference between UDP and TCP in this case in your opinion? - How is this going to work over NAT? I don't use NAT. In any case when you use SETUP method you explicity declare the client and server port to enable the firewall to learn about them. That is what I understood by the RFC: "A firewall may need to understand the SETUP method to open a "hole" for the UDP media stream." "For the benefit of any intervening firewalls, a client must indicate the transport parameters even if it has no influence over these parameters, for example, where the server advertises a fixed multicast address. Since SETUP includes all transport initialization information, firewalls and other intermediate network devices (which need this information) are spared the more arduous task of parsing the DESCRIBE response, which has been reserved for media initialization." This could be a point in favour of TCP: " Unlike HTTP, RTSP allows the media server to send requests to the media client. However, this is only supported for persistent connections, as the media server otherwise has no reliable way of reaching the client. Also, this is the only way that requests from media server to client are likely to traverse firewalls." - What will happen when the UDP packets get lost, duplicated or reordered? It is not clearly explained. In any case there is a section dedicated to reliability: "9.2 Reliability and Acknowledgements Requests are acknowledged by the receiver unless they are sent to a multicast group. If there is no acknowledgement, the sender may resend the same message after a timeout of one round-trip time (RTT). The round-trip time is estimated as in TCP (RFC 1123) [18], with an initial round-trip value of 500 ms. An implementation MAY cache the last RTT measurement as the initial value for future connections. If a reliable transport protocol is used to carry RTSP, requests MUST NOT be retransmitted; the RTSP application MUST instead rely on the underlying transport to provide reliability." - What would you do if the RTSP request is too big to fit in the MTU? It does not happen in my case. If possible, you simply constraint the client/server to use at maximum an MTU and send more than one request. How is it implemented in TCP? Since the ordering is correct, you simply send the RTSP packet fragmented and the server/client would recompose it? How can the client/server understand the end of the RTSP packet? - What would you do if the reply is too big for the MTU? Same as above. - If you manage to get RTSP over UDP working, how and when are you going to open the TCP connection when you decide to do RTP over TCP? I would open a TCP socket for RTP when the client issues a SETUP, before sending the OK reply. I would close the TCP connection associated to the UDP RTSP session number X and ip Y, then the client with session X and ip Y issues the command TEARDOWN. I think I had formulated satisfactory answers but the one regarding the fragmentation. I'm looking forward to hearing from you your comments, Mauro From hardwick.stewart at googlemail.com Wed Jan 19 11:43:26 2011 From: hardwick.stewart at googlemail.com (Stewart Hardwick) Date: Wed, 19 Jan 2011 10:43:26 +0000 Subject: [gst-devel] Elements in rtspsrc pipeline Message-ID: Hi Folks I'm seeing a level of packet loss sufficient to cause macroblocking when I play a stream from an RTSP server. If I turn on debug for the jitterbuffer, I see warnings about packet loss coincident with the macroblocking. I have experimented with a minimal pipeline consisting of rtspsrc ! fakesink and I still see packet loss. I'm trying to figure out what elements are created by rtspsrc and how they are connected, in order to track down in which element the packets may be being lost. I took a look at the README in the gstrtspsrc folder and that talks about creating an rtpdec element, however I think that may be a bit out of date. Can anyone help me with a block diagram of which elements make up an rtspsrc and how they are connected, assuming the simplest case? Thanks Stewart -------------- next part -------------- An HTML attachment was scrubbed... URL: From wl2776 at gmail.com Wed Jan 19 11:56:33 2011 From: wl2776 at gmail.com (wl2776) Date: Wed, 19 Jan 2011 02:56:33 -0800 (PST) Subject: [gst-devel] Elements in rtspsrc pipeline In-Reply-To: References: Message-ID: <1295434593141-3225021.post@n4.nabble.com> Stewart Hardwick wrote: > > Can anyone help me with a block diagram of which elements make up an > rtspsrc > and how they are connected, assuming the > simplest case? > Something like this: http://dl.dropbox.com/u/11507187/player-rtsp-success.png Try writing a simple program, which would create a pipeline, set it to play, then call GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN(pipeline), GST_DEBUG_GRAPH_SHOW_ALL, filename); macro. Set environment variable GST_DEBUG_DUMP_DOT_DIR to some path before running this program. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Elements-in-rtspsrc-pipeline-tp3224990p3225021.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wim.taymans at gmail.com Wed Jan 19 12:34:28 2011 From: wim.taymans at gmail.com (Wim Taymans) Date: Wed, 19 Jan 2011 12:34:28 +0100 Subject: [gst-devel] Is RTSPU implemented in gstreamer? Can I use RTSP over UDP? In-Reply-To: References: <1294934317.1875.48.camel@meany> Message-ID: It sounds like you have it plan that could work. Wim On Wed, Jan 19, 2011 at 9:59 AM, Mauro Brenna wrote: > Hi, > > As regards the possibility to transmit RTSP packets over UDP you can > also read in the RFC (2326): > > " There is no notion of an RTSP connection; instead, a server maintains > ? a session labeled by an identifier. An RTSP session is in no way tied > ? to a transport-level connection such as a TCP connection. During an > ? RTSP session, an RTSP client may open and close many reliable > ? transport connections to the server to issue RTSP requests. > ? Alternatively, it may use a connectionless transport protocol such as > ? UDP." > > So, I am practically convinced you could send also the RTSP commands, > not only the RTP data, via UDP. > Anywat, I don't want to start a post-reply war. I would like only to > understand what it is correct and what it is not. My application is > very specific and I will implement a very very simple RTSP over UDP > protocol. > > Now, to answer your question: > > - So how do you think it would work? you send the RTSP request over UDP? > Yes, I would do like that. Note the difference between sessionless > commands and commands that refers to a session. You need to mantain a > state for each session, if you would like to stream in unicast to more > than one client, otherwise you can use a single 'instance' of RTSP > server state and refuse to SETUP if a second client asks you so. > > - How would you get the reply? > Via UDP, you send the SESSION id in the reply. > > - How does the server know where to send the reply to? > I do not understand the question. The server should know specific > parameters via SETUP method. Or simply replies to the IP of the > sender. Which is the difference between UDP and TCP in this case in > your opinion? > > - How is this going to work over NAT? > I don't use NAT. In any case when you use SETUP method you explicity > declare the client and server port to enable the firewall to learn > about them. That is what I understood by the RFC: > > "A firewall may need to ?understand the SETUP method to open a "hole" > for the UDP media stream." > > "For the benefit of any > ? intervening firewalls, a client must indicate the transport > ? parameters even if it has no influence over these parameters, for > ? example, where the server advertises a fixed multicast address. > > ? ? Since SETUP includes all transport initialization information, > ? ? firewalls and other intermediate network devices (which need this > ? ? information) are spared the more arduous task of parsing the > ? ? DESCRIBE response, which has been reserved for media > ? ? initialization." > > This could be a point in favour of TCP: > " Unlike HTTP, RTSP allows the media server to send requests to the > ? media client. However, this is only supported for persistent > ? connections, as the media server otherwise has no reliable way of > ? reaching the client. Also, this is the only way that requests from > ? media server to client are likely to traverse firewalls." > > > - What will happen when the UDP packets get lost, duplicated or reordered? > It is not clearly explained. In any case there is a section dedicated > to reliability: > > "9.2 Reliability and Acknowledgements > > ? Requests are acknowledged by the receiver unless they are sent to a > ? multicast group. If there is no acknowledgement, the sender may > ? resend the same message after a timeout of one round-trip time (RTT). > ? The round-trip time is estimated as in TCP (RFC 1123) [18], with an > ? initial round-trip value of 500 ms. An implementation MAY cache the > ? last RTT measurement as the initial value for future connections. > > ? If a reliable transport protocol is used to carry RTSP, requests MUST > ? NOT be retransmitted; the RTSP application MUST instead rely on the > ? underlying transport to provide reliability." > > > - What would you do if the RTSP request is too big to fit in the MTU? > It does not happen in my case. If possible, you simply constraint the > client/server to use at maximum an MTU and send more than one request. > How is it implemented in TCP? Since the ordering is correct, you > simply send the RTSP packet fragmented and the server/client would > recompose it? How can the client/server understand the end of the RTSP > packet? > > > - What would you do if the reply is too big for the MTU? > Same as above. > > - If you manage to get RTSP over UDP working, how and when are you > going to open the TCP connection when > you decide to do RTP over TCP? > > I would open a TCP socket for RTP when the client issues a SETUP, > before sending the OK reply. > I would close the TCP connection associated to the UDP RTSP session > number X and ip Y, then the client with session X and ip Y issues the > command TEARDOWN. > > > > I think I had formulated satisfactory answers but the one regarding > the fragmentation. > > > I'm looking forward to hearing from you your comments, > > Mauro > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 wim.taymans at gmail.com Wed Jan 19 12:37:15 2011 From: wim.taymans at gmail.com (Wim Taymans) Date: Wed, 19 Jan 2011 12:37:15 +0100 Subject: [gst-devel] Elements in rtspsrc pipeline In-Reply-To: <1295434593141-3225021.post@n4.nabble.com> References: <1295434593141-3225021.post@n4.nabble.com> Message-ID: Also a good trick is to increase the default kernel UDP buffer size. The next version of rtspsrc will by default set the size much bigger, which helps a lot. Wim On Wed, Jan 19, 2011 at 11:56 AM, wl2776 wrote: > > > Stewart Hardwick wrote: >> >> Can anyone help me with a block diagram of which elements make up an >> rtspsrc >> and how they are connected, assuming the >> simplest case? >> > > Something like this: > http://dl.dropbox.com/u/11507187/player-rtsp-success.png > > Try writing a simple program, which would create a pipeline, set it to play, > then call > GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN(pipeline), > GST_DEBUG_GRAPH_SHOW_ALL, filename); > macro. > > Set environment variable GST_DEBUG_DUMP_DOT_DIR to some path before running > this program. > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Elements-in-rtspsrc-pipeline-tp3224990p3225021.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 hardwick.stewart at googlemail.com Wed Jan 19 12:52:20 2011 From: hardwick.stewart at googlemail.com (Stewart Hardwick) Date: Wed, 19 Jan 2011 11:52:20 +0000 Subject: [gst-devel] Elements in rtspsrc pipeline Message-ID: > Something like this: > http://dl.dropbox.com/u/11507187/player-rtsp-success.png Thanks, that's just what I had in mind Cheers Stewart -------------- next part -------------- An HTML attachment was scrubbed... URL: From acandido at hi-iberia.es Wed Jan 19 16:17:54 2011 From: acandido at hi-iberia.es (Andres Gonzalez) Date: Wed, 19 Jan 2011 16:17:54 +0100 Subject: [gst-devel] Simple ogg pipeline doesn't work In-Reply-To: <4D09D54B.5010600@hi-iberia.es> References: <4CFE74D1.5090903@hi-iberia.es> <1292467536.19370.37.camel@abacate> <4D09D54B.5010600@hi-iberia.es> Message-ID: <4D3700A2.3030104@hi-iberia.es> On 16/12/10 10:00, Andres Gonzalez wrote: > 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 >> Great idea, the multiqueue worked! The resulting OGG file is correct (playable by VLC, no problem). The only remarkable fact (but not very important) is that this file's pages are smaller in average; which makes the file a little bit bigger than the original (672070 vs. 662135 bytes). Many thanks, Andr?s P.D.: program attached -- Andr?s Gonz?lez T: 91 458 51 19 (ext:165) M: 620 344 358 HI-IBERIA INGENIERIA Y PROYECTOS C/ Bolivia 5 - Madrid - 28016 -------------- next part -------------- A non-text attachment was scrubbed... Name: copyogg-multiq.py Type: text/x-python Size: 3128 bytes Desc: not available URL: From hardwick.stewart at googlemail.com Wed Jan 19 16:19:23 2011 From: hardwick.stewart at googlemail.com (Stewart Hardwick) Date: Wed, 19 Jan 2011 15:19:23 +0000 Subject: [gst-devel] Elements in rtspsrc pipeline In-Reply-To: References: Message-ID: > Also a good trick is to increase the default kernel UDP buffer size. > The next version of rtspsrc > will by default set the size much bigger, which helps a lot. I took a look at the latest plugins_good rtspsrc, and modified mine to add g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size", ??????? (100 * 1024), NULL); in gst_rtspsrc_stream_configure_udp. However, the debug from udpsrc still shows a buffer size of 50K (note confusing factor of 2 due to kernel overhead): udpsrc gstudpsrc.c:784:gst_udpsrc_start: allocating socket for 0.0.0.0:0 udpsrc gstudpsrc.c:795:gst_udpsrc_start: got socket 37 udpsrc gstudpsrc.c:797:gst_udpsrc_start: setting reuse 0 udpsrc gstudpsrc.c:804:gst_udpsrc_start: binding on port 0 udpsrc gstudpsrc.c:873:gst_udpsrc_start: have udp buffer of 106496 bytes udpsrc gstudpsrc.c:906:gst_udpsrc_start: bound, on port 32836 udpsrc gstudpsrc.c:909:gst_udpsrc_start: notifying port 32836 I was expecting to see "setting udp buffer of 102400 bytes" somewhere. Looking at the udpsrc code, in gst_udpsrc_start it looks like src->buffer_size must be zero. Possibly the call in rtspsrc is being made after gst_udpsrc_start has run and set the buffer size? Cheers Stewart From dsd at laptop.org Wed Jan 19 17:21:45 2011 From: dsd at laptop.org (Daniel Drake) Date: Wed, 19 Jan 2011 16:21:45 +0000 Subject: [gst-devel] unreliable xvimagesink rendering to GTK+ elements Message-ID: Hi, I am attempting to create a pipeline which includes an xvimagesink, where the output will be mapped onto a GTK+ element. Something like the video player example here: http://pygstdocs.berlios.de/pygst-tutorial/capabilities.html But, this example (plus my own app) does not run reliably. Sometimes it works, most of the time it doesn't. A whole assortment of errors are possible, including: python: xcb_io.c:140: dequeue_pending_request: Assertion `req == dpy->xcb->pending_requests' failed. test.py: Fatal IO error 11 (Resource temporarily unavailable) on X server :0.0. python: xcb_io.c:221: poll_for_event: Assertion `(((long) (event_sequence) - (long) (dpy->request)) <= 0)' failed. Gdk-ERROR **: The program 'test.py' received an X Window System error. This probably reflects a bug in the program. The error was 'BadWindow (invalid Window parameter)'. (Details: serial 195 error_code 3 request_code 2 minor_code 0) (Note to programmers: normally, X errors are reported asynchronously; that is, you will receive the error a while after causing it. To debug your program, run it with the --sync command line option to change this behavior. You can then get a meaningful backtrace from your debugger if you break on the gdk_x_error() function.) Here is a test program which you can use to see this: http://dev.laptop.org/~dsd/20110119/gst_xv_gtk.txt It shows xvimagesink on top of a GtkDrawingArea for 2 seconds, then exits. Run it in a loop to catch the errors: # while python gst_xv_gtk.txt; do :; done Interestingly, if I instead try to render on a GtkWindow, everything works reliably. (see the commented out section of on_sync_message(). Am I doing something wrong in my code? Is this a bug? Can other people reproduce it? Thanks, Daniel From t.i.m at zen.co.uk Wed Jan 19 19:39:09 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 19 Jan 2011 18:39:09 +0000 Subject: [gst-devel] unreliable xvimagesink rendering to GTK+ elements In-Reply-To: References: Message-ID: <1295462349.11953.0.camel@zingle> On Wed, 2011-01-19 at 16:21 +0000, Daniel Drake wrote: Hi Daniel, have you read through the docs at http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstxoverlay.html#gst-plugins-base-libs-gstxoverlay.description yet? (Sure, it's C, but the same applies in pygtk really) Cheers -Tim From dsd at laptop.org Wed Jan 19 20:31:48 2011 From: dsd at laptop.org (Daniel Drake) Date: Wed, 19 Jan 2011 19:31:48 +0000 Subject: [gst-devel] unreliable xvimagesink rendering to GTK+ elements In-Reply-To: <1295462349.11953.0.camel@zingle> References: <1295462349.11953.0.camel@zingle> Message-ID: On 19 January 2011 18:39, Tim-Philipp M?ller wrote: > On Wed, 2011-01-19 at 16:21 +0000, Daniel Drake wrote: > > Hi Daniel, > > have you read through the docs at > http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstxoverlay.html#gst-plugins-base-libs-gstxoverlay.description yet? > > (Sure, it's C, but the same applies in pygtk really) Thanks Tim & whoever wrote that page, thats exactly what I needed. Makes perfect sense. Daniel From 4ernov at gmail.com Wed Jan 19 22:17:53 2011 From: 4ernov at gmail.com (Alexey Chernov) Date: Thu, 20 Jan 2011 00:17:53 +0300 Subject: [gst-devel] How to make newly added header be parsed by gtk-doc Message-ID: <201101200017.53587.4ernov@gmail.com> Hello, I'm adding some functionality to v4l2 plugin in gst-plugins-good (radio element to be more exact, it was previously discussed here). The task is near the end, but I can't add my header (it is sys/v4l2/gstv4l2radio.h) to be parsed by gtk-doc when --enable-gtk-doc option is on. It seem to be simply ignored but I can't understand where it should be added to be parsed - in docs/ directory many files seem to be generated automatically. Could someone please give me any reference where I should add this header file's path or something to make it be parsed by gtk-doc? Thanks in advance. From stuart68 at gmail.com Thu Jan 20 02:28:13 2011 From: stuart68 at gmail.com (stuart68) Date: Wed, 19 Jan 2011 17:28:13 -0800 (PST) Subject: [gst-devel] flac files that will playback, but will not stream Message-ID: <1295486893640-3225461.post@n4.nabble.com> I have flac files that will playback using playbin2 uri=file:/// , but when I try to stream these files using playbin2 uri=http:// I get the following error: ERROR: from element /GstPlayBin2:playbin20/GstURIDecodeBin:uridecodebin0/GstSoupHTTPSrc:source: Internal data flow error. Additional debug info: gstbasesrc.c(2507): gst_base_src_loop (): /GstPlayBin2:playbin20/GstURIDecodeBin:uridecodebin0/GstSoupHTTPSrc:source: streaming task paused, reason error (-5) ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... This is using Ubuntu 10.04, with lighttpd as the web server. On the same machine I have a load of other flac files that all stream perfectly. Thanks for any help that can be provided -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/flac-files-that-will-playback-but-will-not-stream-tp3225461p3225461.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From nathanael at gnat.ca Thu Jan 20 08:25:39 2011 From: nathanael at gnat.ca (Nathanael D. Noblet) Date: Thu, 20 Jan 2011 00:25:39 -0700 Subject: [gst-devel] kenburns plugin update - now a general video transform plugin In-Reply-To: <4D35BCE9.9000506@brooks.nu> References: <4D35BCE9.9000506@brooks.nu> Message-ID: <4D37E373.4090108@gnat.ca> On 01/18/2011 09:16 AM, Lane Brooks wrote: > I have an updated kenburns plugin with the following changes: > > - I dropped the internal controller. Now you use a GstContoller to > control the 'xpos', 'ypos', and 'zpos'. > - I added rotation transforms as well. You can now also control an > 'xrot', 'yrot', and 'zrot' parameter. > > You can check out a video I put together that show cases these features > at http://www.youtube.com/watch?v=JFm8Rra4xzQ. This video first shows > the translation and rotation features with the 'xpos', 'ypos', 'zpos', > 'xrot', 'yrot',and 'zrot' parameters hooked up to LFO's with an stmpe > video test source. It then shows some sample kenburns effects, which use > the 'xpos', 'ypos', and 'zpos' parameters hooked up to linear > interpolating GstControllers. It then shows some sample video > transitions that also make use of the position and rotation parameters > hooked up to linear interpolating GstControllers. > > You can still clone the source code at: > https://github.com/dirjud/kenburns > > Given that the feature set has grown beyond Ken Burns effects, I want to > rename this element to something else. I was thinking 'videotransform' > but am open to other suggestions. One reason against videotransform is > that there is already a base class in gstreamer call GstVideoTransform. > It is not exposed to the user, so it only a potential confusion for > developers. But if there is a good name that is not a potential > confusion, I would go with that instead. > > After I decide on a new name and hear any additional feedback, I will > post a patch to gst-plugins-bad. I saw the youtube video. Impressive! This looks quite useful. From umakantgoyal1 at gmail.com Thu Jan 20 12:22:26 2011 From: umakantgoyal1 at gmail.com (Umakant Goyal) Date: Thu, 20 Jan 2011 16:52:26 +0530 Subject: [gst-devel] Not able to play some of 3gp files Message-ID: Hi All, I have made a small programe to stream 3gp files on network using gstreamer. But i am not able to stream all the 3gp files. After debugging, i have found that gstreamer is not sending GST_MESSAGE_ASYNC_DONE & GST_MESSAGE_NEW_CLOCK events to app for those 3gp files that it is not able to play. I tried lot to figure out the reason for the same. But i could not succeed. Please help me out by telling the reason for the same. PS: 3GP files that are not working with gstreamer, are working fine using VLC and other Media players. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From umakantgoyal1 at gmail.com Thu Jan 20 13:25:24 2011 From: umakantgoyal1 at gmail.com (qwerty-1) Date: Thu, 20 Jan 2011 04:25:24 -0800 (PST) Subject: [gst-devel] Not able to play some of 3gp files Message-ID: <1295526324490-3225499.post@n4.nabble.com> Hi All, I have made a small programe to stream 3gp files on network using gstreamer. But i am not able to stream all the 3gp files. After debugging, i have found that gstreamer is not sending GST_MESSAGE_ASYNC_DONE & GST_MESSAGE_NEW_CLOCK events to app for those 3gp files that it is not able to play. I tried lot to figure out the reason for the same. But i could not succeed. Please help me out by telling the reason for the same. PS: 3GP files that are not working with gstreamer, are working fine using VLC and other Media players. Thanks -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Not-able-to-play-some-of-3gp-files-tp3225499p3225499.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From kobalt_thebest at yahoo.co.uk Thu Jan 13 14:20:40 2011 From: kobalt_thebest at yahoo.co.uk (kobalt) Date: Thu, 13 Jan 2011 05:20:40 -0800 (PST) Subject: [gst-devel] Playbin stuck on buffering Message-ID: <1294924840294-3215817.post@n4.nabble.com> I'm trying to play an video file on an arm board using the following command: gst-launch playbin uri=http://192.168.0.1/audio/mediaexample.mpg -v The problem is that it stucks on buffering. here is the output of the command: Setting pipeline to PAUSED ... Pipeline is PREROLLING ... /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:src: caps = video/mpeg, systemstream=(boolean)true, mpegversion=(int)1 /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstMpegPSDemux:mpegpsdemux0.GstPad:sink: caps = video/mpeg, systemstream=(boolean)true, mpegversion=(int)1 /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstQueue:queue0.GstPad:sink: caps = video/mpeg, mpegversion=(int)1, systemstream=(boolean)false /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstQueue:queue0.GstPad:src: caps = video/mpeg, mpegversion=(int)1, systemstream=(boolean)false /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstQueue:queue1.GstPad:sink: caps = audio/mpeg, mpegversion=(int)1 /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstQueue:queue1.GstPad:src: caps = audio/mpeg, mpegversion=(int)1 /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstMPEGAudioParse:mpegaudioparse0.GstPad:sink: caps = audio/mpeg, mpegversion=(int)1 /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstMPEGAudioParse:mpegaudioparse0.GstPad:src: caps = audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)1, layer=(int)2, rate=(int)48000, channels=(e /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstOmxMp2Dec:omxmp2dec0.GstPad:sink: caps = audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)1, layer=(int)2, rate=(int)48000, channels=(int)2, pae /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstOmxMpeg2Dec:omxmpeg2dec0.GstPad:sink: caps = video/mpeg, mpegversion=(int)1, systemstream=(boolean)false /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstOmxMpeg2Dec:omxmpeg2dec0.GstPad:sink: caps = video/mpeg, mpegversion=(int)1, systemstream=(boolean)false /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstQueue:queue0: max-size-bytes = 18352 /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstQueue:queue0: max-size-bytes = 36704 /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstQueue:queue0: max-size-bytes = 73408 /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstQueue:queue0: max-size-bytes = 146816 /GstPlayBin:playbin0/GstDecodeBin:decodebin0.GstGhostPad:src1: caps = audio/x-raw-int, width=(int)16, depth=(int)16, rate=(int)48000, signed=(boolean)true, endianness=(int)1234, channels=(int)2 /GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstOmxMp2Dec:omxmp2dec0.GstPad:src: caps = audio/x-raw-int, width=(int)16, depth=(int)16, rate=(int)48000, signed=(boolean)true, endianness=(int)1234, 2 /GstPlayBin:playbin0/GstStreamSelector:selector_audio_src1.GstPlaybinSelectorPad:sink0: caps = audio/x-raw-int, width=(int)16, depth=(int)16, rate=(int)48000, signed=(boolean)true, endianness=(int2 /GstPlayBin:playbin0/GstDecodeBin:decodebin0.GstGhostPad:src1.GstProxyPad:proxypad2: caps = audio/x-raw-int, width=(int)16, depth=(int)16, rate=(int)48000, signed=(boolean)true, endianness=(int)122 /GstPlayBin:playbin0/GstStreamSelector:selector_audio_src1.GstPad:src: caps = audio/x-raw-int, width=(int)16, depth=(int)16, rate=(int)48000, signed=(boolean)true, endianness=(int)1234, channels=(2 /GstPlayBin:playbin0/GstStreamSelector:selector_audio_src1.GstPad:src: caps = audio/x-raw-int, width=(int)16, depth=(int)16, rate=(int)48000, signed=(boolean)true, endianness=(int)1234, channels=(2 /GstPlayBin:playbin0/GstQueue:preroll_audio_src1.GstPad:sink: caps = audio/x-raw-int, width=(int)16, depth=(int)16, rate=(int)48000, signed=(boolean)true, endianness=(int)1234, channels=(int)2 buffering... 0% If i try the same command but using playbin2 everything works perfectly. Is there a solution to make this work on playbin, or is this an issue known by gstreamer-dev team? Thank You, Cornel -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Playbin-stuck-on-buffering-tp3215817p3215817.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From holzer.st at akfree.it Thu Jan 13 15:21:10 2011 From: holzer.st at akfree.it (bonkers) Date: Thu, 13 Jan 2011 06:21:10 -0800 (PST) Subject: [gst-devel] gst_tag_list_get_string Message-ID: <1294928470699-3215915.post@n4.nabble.com> Hello! when i try to get some informations about the track using the GST_MESSAGE_TAG i get nothing. int MyStreamer_bus_call(GstBus *bus, GstMessage *msg, gpointer data) { GstTagList *taglist=gst_tag_list_new(); char *c,*artist; GMainLoop *loop = (GMainLoop*) data; switch(GST_MESSAGE_TYPE(msg)){ gst_message_parse_tag(msg, &taglist); if (gst_tag_list_get_string (taglist, GST_TAG_ARTIST, &artist)) { printf("got artist: %s\n",artist); } else{ printf("no artist!! \n"); } } return 0; } I always get "no artist"!! I?m sure that I set the tags of the mp3 file. can you help me?? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/gst-tag-list-get-string-tp3215915p3215915.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From crazylinux at 163.com Mon Jan 17 06:43:53 2011 From: crazylinux at 163.com (Felix) Date: Sun, 16 Jan 2011 21:43:53 -0800 (PST) Subject: [gst-devel] [SPAM] RTSP issue gst_rtspsrc_async_open and gst_rtspsrc_open Message-ID: <1295243033807-3220724.post@n4.nabble.com> There is a asynchronous function named gst_rtspsrc_async_open right after gst_rtspsrc_open in gstrtspsrc.c. Could anybody tell me why this open operation is a synchronize? Who gst_rtspsrc_async_open been disabled in the latest gstreamer release http://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-0.10.26.tar.gz ? Is that possible if I'd like to use the open in a way of async? Because sometimes the timeout is too long for application. It's 20s now! thx. #if 0 static gboolean gst_rtspsrc_async_open (GstRTSPSrc * src) { GError *error = NULL; gboolean res = TRUE; timestamp(); printf("cnh01286::: gst_rtspsrc_async_open\n"); src->thread = g_thread_create ((GThreadFunc) gst_rtspsrc_open, src, TRUE, &error); if (error != NULL) { GST_ELEMENT_ERROR (src, RESOURCE, INIT, (NULL), ("Could not start async thread (%s).", error->message)); } timestamp(); printf("cnh01286::: gst_rtspsrc_async_open return\n"); return res; } #endif -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/RTSP-issue-gst-rtspsrc-async-open-and-gst-rtspsrc-open-tp3220724p3220724.html Sent from the GStreamer-devel mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: From holzer.st at akfree.it Mon Jan 17 13:40:41 2011 From: holzer.st at akfree.it (bonkers) Date: Mon, 17 Jan 2011 04:40:41 -0800 (PST) Subject: [gst-devel] GThread-ERROR Message-ID: <1295268041055-3221147.post@n4.nabble.com> hello! when i start my gstreamer (implemented in c) i got this error: GThread-ERROR **: file gthread-posix.c: line 348 (g_thread_create_posix_impl): error 'Invalid argument' during 'pthread_attr_setschedparam (&attr, &sched)' aborting... Aborted can anyone help me? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/GThread-ERROR-tp3221147p3221147.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From kredfern at tycoint.com Thu Jan 20 10:54:57 2011 From: kredfern at tycoint.com (Redfern, Keith) Date: Thu, 20 Jan 2011 09:54:57 +0000 Subject: [gst-devel] how do I packet pace rtp network traffic? Message-ID: <8E459EED6B356D469071FDC814554FC43FF356C3@COR1MBX01.emea.tsp.ad> We are developing a gstreamer based pipeline to read a video source, rtp payload the video frame and then udpsink the rtp packets. We are finding that the pipeline bursts the rtp data unto the network, creating network spikes. In one example, the interpacket network delay is about 15micro seconds for approx 150 packets, meaning the whole video frame is delivered unto the network in 2.25ms. Is there some way in which to 'pace' the rtp packets so that it takes 20-30ms to send the same frame? Keith ________________________________ Tyco Safety Products/CEM Systems Ltd. Registered in Northern Ireland: NI 25728. Registered Office: 195 Airport Road West, Belfast, BT3 9ED. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of the company. This email and any attachments are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the sender immediately and delete any copies in your possession. -------------- next part -------------- An HTML attachment was scrubbed... URL: From smcnam at gmail.com Thu Jan 20 15:06:25 2011 From: smcnam at gmail.com (Sean McNamara) Date: Thu, 20 Jan 2011 09:06:25 -0500 Subject: [gst-devel] GThread-ERROR In-Reply-To: <1295268041055-3221147.post@n4.nabble.com> References: <1295268041055-3221147.post@n4.nabble.com> Message-ID: On Mon, Jan 17, 2011 at 7:40 AM, bonkers wrote: > > hello! > > when i start my gstreamer (implemented in c) i got this error: > > GThread-ERROR **: file gthread-posix.c: line 348 > (g_thread_create_posix_impl): error 'Invalid argument' during > 'pthread_attr_setschedparam (&attr, &sched)' > aborting... > Aborted > > can anyone help me? Need more info: 1. Which platform? Include version information, not just the name. 2. What version of gstreamer? If it's the stock version from the package repositories of a Linux distro and you don't know the answer, you can omit this, but you have to include the release version of the distro then. 3. Provide your code -- at a minimum, include anything you do with gstreamer, glib, gtk, or pthreads. That said, a quick google suggests that you might hit this issue if you run a GThread based app or API (Gstreamer is one such API) in a parent process that is real-time. See https://bugs.launchpad.net/ubuntu/+source/glib2.0/+bug/453898 and https://bugzilla.gnome.org/show_bug.cgi?id=599079 There doesn't seem to be a clear-cut resolution accepted by upstream yet (i.e. GLib maintainers), but what you can do is run your application without real-time scheduling... if that works, then just do it like that. If you think you need gstreamer to run real-time, then you most likely should reexamine your design and either improve performance, or reduce system load somehow, so that the pipeline doesn't stall with normal scheduling. > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/GThread-ERROR-tp3221147p3221147.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 jim.muchow at updatelogic.com Thu Jan 20 17:58:59 2011 From: jim.muchow at updatelogic.com (Jim Muchow) Date: Thu, 20 Jan 2011 11:58:59 -0500 Subject: [gst-devel] TCP Plugins w/ OpenSSL Message-ID: <12AA9965E245EB42A88235C0C7ECF4AB0AD2E3552B@P1EC3EVS01.HMC1.COMCAST.NET> We've implemented OpenSSL in the TCP portion of the Base Plugins. We would like to contribute this code to the Gstreamer project, but we don't know how. We also have some questions/concerns and some unfinished work. Rather than dive into the technical details, I'll just leave it here. Comments? Questions? Jim Muchow -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian.droege at collabora.co.uk Thu Jan 20 18:30:25 2011 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Thu, 20 Jan 2011 18:30:25 +0100 Subject: [gst-devel] TCP Plugins w/ OpenSSL In-Reply-To: <12AA9965E245EB42A88235C0C7ECF4AB0AD2E3552B@P1EC3EVS01.HMC1.COMCAST.NET> References: <12AA9965E245EB42A88235C0C7ECF4AB0AD2E3552B@P1EC3EVS01.HMC1.COMCAST.NET> Message-ID: <1295544625.3117.23.camel@odin.lan> On Thu, 2011-01-20 at 11:58 -0500, Jim Muchow wrote: > We?ve implemented OpenSSL in the TCP portion of the Base Plugins. We > would like to contribute this code to the Gstreamer project, but we > don?t know how. We also have some questions/concerns and some > unfinished work. > > Rather than dive into the technical details, I?ll just leave it here. > > Comments? Questions? You mean to tcpserver{sink,src} and tcpclient{sink,src}? I guess that's a good idea in general but the problem here is, that the OpenSSL license is not GPL compatible and as such this code can't live in gst-plugins-base. As alternative you could add a compile-time option to use either OpenSSL or GnuTLS. Apart from that it might make sense to create new elements for this in -base, that share a lot of code with the TCP elements... instead of having a enable-SSL/TLS property on the elements. -------------- 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 jim.muchow at updatelogic.com Thu Jan 20 19:26:19 2011 From: jim.muchow at updatelogic.com (Jim Muchow) Date: Thu, 20 Jan 2011 13:26:19 -0500 Subject: [gst-devel] TCP Plugins w/ OpenSSL In-Reply-To: <1295544625.3117.23.camel@odin.lan> References: <12AA9965E245EB42A88235C0C7ECF4AB0AD2E3552B@P1EC3EVS01.HMC1.COMCAST.NET> <1295544625.3117.23.camel@odin.lan> Message-ID: <12AA9965E245EB42A88235C0C7ECF4AB0AD2E35553@P1EC3EVS01.HMC1.COMCAST.NET> > From: Sebastian Dr?ge [mailto:sebastian.droege at collabora.co.uk] > Sent: Thursday, January 20, 2011 11:30 > > On Thu, 2011-01-20 at 11:58 -0500, Jim Muchow wrote: > > We?ve implemented OpenSSL in the TCP portion of the Base > Plugins. We > > would like to contribute this code to the Gstreamer project, but > we > > don?t know how. We also have some questions/concerns and some > > unfinished work. > > > > Rather than dive into the technical details, I?ll just leave it > here. > > > > Comments? Questions? > > You mean to tcpserver{sink,src} and tcpclient{sink,src}? I guess I do. We've implemented and tested the changes to tcpclientsink & tcpserversrc. Each of the functions in gsttcp.c now have an OpenSSL analog. > that's a good idea in general but the problem here is, that the > OpenSSL license is not GPL compatible and as such this code can't > live in gst-plugins-base. > > As alternative you could add a compile-time option to use either > OpenSSL or GnuTLS. The code has been implemented (in the 0.10.29 release) using a conditional compile. If not included, a resulting build binary is identical to one prior to this feature addition. If included in a build, however, the use of OpenSSL can be enabled or disabled via the set_property mechanism. > Apart from that it might make sense to create new elements for > this in -base, that share a lot of code with the TCP elements... > instead of having a enable-SSL/TLS property on the elements. There at least a couple of approaches to how integrate with TCP. One is to use the OpenSSL BIO abstraction. The other way is to let a sockets-based environment "do its thing" (socket(), listen(), accept(), connect()) and then bolt on an SSL session. We chose the latter. If former is preferred, then yes, a whole new set of elements would make sense. From bertd at tplogic.com Thu Jan 20 20:00:42 2011 From: bertd at tplogic.com (Bert Douglas) Date: Thu, 20 Jan 2011 11:00:42 -0800 Subject: [gst-devel] opengl on top of video Message-ID: Hello, I have a python-gstreamer app that shows video. It uses xvimagesink and links it to a gtk.DrawingArea. This works great. Now I want to overlay some graphics, dynamically generated from real time data. Opengl seems like a good choice to render the graphics. But I have no clue how to combine the video and graphics. I don't think they are made to work with each other. Thanks, Bert Douglas -------------- next part -------------- An HTML attachment was scrubbed... URL: From nathanael at gnat.ca Thu Jan 20 20:39:07 2011 From: nathanael at gnat.ca (Nathanael D. Noblet) Date: Thu, 20 Jan 2011 12:39:07 -0700 Subject: [gst-devel] opengl on top of video In-Reply-To: References: Message-ID: <4D388F5B.1060501@gnat.ca> On 01/20/2011 12:00 PM, Bert Douglas wrote: > Hello, > > I have a python-gstreamer app that shows video. > It uses xvimagesink and links it to a gtk.DrawingArea. > > This works great. > > Now I want to overlay some graphics, dynamically generated from real > time data. > Opengl seems like a good choice to render the graphics. > > But I have no clue how to combine the video and graphics. > I don't think they are made to work with each other. I'm working on the same issue right now.. no solution as of yet. I'm looking at GtkEventBox and the following example. http://library.gnome.org/devel/gdk/2.22/gdk-Windows.html#composited-window-example However nothing is working yet... Let me know if you find a solution From t.i.m at zen.co.uk Thu Jan 20 20:54:49 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 20 Jan 2011 19:54:49 +0000 Subject: [gst-devel] How to make newly added header be parsed by gtk-doc In-Reply-To: <201101200017.53587.4ernov@gmail.com> References: <201101200017.53587.4ernov@gmail.com> Message-ID: <1295553289.23989.1.camel@zingle> On Thu, 2011-01-20 at 00:17 +0300, Alexey Chernov wrote: > I'm adding some functionality to v4l2 plugin in gst-plugins-good (radio > element to be more exact, it was previously discussed here). The task is near > the end, but I can't add my header (it is sys/v4l2/gstv4l2radio.h) to be > parsed by gtk-doc when --enable-gtk-doc option is on. It seem to be simply > ignored but I can't understand where it should be added to be parsed - in > docs/ directory many files seem to be generated automatically. > > Could someone please give me any reference where I should add this header > file's path or something to make it be parsed by gtk-doc? Maybe doc/plugins/Makefile.am, but it shouldn't really be necessary (might make the private element struct show up though). Cheers -Tim From t.i.m at zen.co.uk Thu Jan 20 20:57:03 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 20 Jan 2011 19:57:03 +0000 Subject: [gst-devel] flac files that will playback, but will not stream In-Reply-To: <1295486893640-3225461.post@n4.nabble.com> References: <1295486893640-3225461.post@n4.nabble.com> Message-ID: <1295553423.23989.3.camel@zingle> On Wed, 2011-01-19 at 17:28 -0800, stuart68 wrote: Hi, > I have flac files that will playback using playbin2 uri=file:/// , but > when I try to stream these files using playbin2 uri=http:// I get the > following error: > > ERROR: from element > /GstPlayBin2:playbin20/GstURIDecodeBin:uridecodebin0/GstSoupHTTPSrc:source: > Internal data flow error. > Additional debug info: > gstbasesrc.c(2507): gst_base_src_loop (): > /GstPlayBin2:playbin20/GstURIDecodeBin:uridecodebin0/GstSoupHTTPSrc:source: > streaming task paused, reason error (-5) > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... Some element returned GST_FLOW_ERROR but didn't post a proper error message on the bus, not very nice. > This is using Ubuntu 10.04, with lighttpd as the web server. > > On the same machine I have a load of other flac files that all stream > perfectly. > > Thanks for any help that can be provided Could you file a bug please, and make that file available for download/testing? Does it error out immediately, or only after a while? Have you tried the GStreamer PPA for ubuntu (which has more recent versions of GStreamer)? Could you attach a GST_DEBUG=*:5 log to the bug report (gzipped)? Thanks! Cheers -Tim From t.i.m at zen.co.uk Thu Jan 20 21:01:31 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 20 Jan 2011 20:01:31 +0000 Subject: [gst-devel] Not able to play some of 3gp files In-Reply-To: <1295526324490-3225499.post@n4.nabble.com> References: <1295526324490-3225499.post@n4.nabble.com> Message-ID: <1295553691.23989.5.camel@zingle> On Thu, 2011-01-20 at 04:25 -0800, qwerty-1 wrote: > I have made a small programe to stream 3gp files on network using gstreamer. > But i am not able to stream all the 3gp files. After debugging, i have found > that gstreamer is not sending > GST_MESSAGE_ASYNC_DONE & GST_MESSAGE_NEW_CLOCK events to app for those 3gp > files that it is not able to play. I tried lot to figure out the reason for > the same. But i could not succeed. Please help me out by telling the reason > for the same. You don't say anything about your pipeline, nor did you attach your code. Does gst-launch-0.10 playbin2 uri=file:///path/to/foo.3gp work (replace with appropriate uri if not local files)? If not, what's the output? Do you get any GST_MESSAGE_ERROR messages on the bus? What's the URI that fails? (if it's public) Cheers -Tim From paper at gmx.net Thu Jan 20 22:12:10 2011 From: paper at gmx.net (Kony) Date: Fri, 21 Jan 2011 01:12:10 +0400 Subject: [gst-devel] mp4 file - rtp Message-ID: <1295557930.21163.5.camel@kk-laptop> i want to open a flv file, read h264 nals, package them in rtp packages and send over network... Can i do it without to touch nals or they should be transfered in a special way? I mean, i want to do that without any libs. flv file is encoded with gstreamer and x264 is not in a bytestream modus... Thanks in advance, kony From bmschwar at fas.harvard.edu Thu Jan 20 23:19:04 2011 From: bmschwar at fas.harvard.edu (Benjamin M. Schwartz) Date: Thu, 20 Jan 2011 17:19:04 -0500 Subject: [gst-devel] Least Buggy API for Totem Mozilla Plugin Message-ID: <4D38B4D8.8000701@fas.harvard.edu> The Totem developers have very helpfully provided reimplementations of the VLC, WMP, QuickTime, and DivX plugin javascript APIs. Which, if any, of these is preferred/less buggy/more usable? Thanks, Ben Schwartz -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: OpenPGP digital signature URL: From 4ernov at gmail.com Thu Jan 20 23:53:09 2011 From: 4ernov at gmail.com (Alexey Chernov) Date: Fri, 21 Jan 2011 01:53:09 +0300 Subject: [gst-devel] How to make newly added header be parsed by gtk-doc In-Reply-To: <1295553289.23989.1.camel@zingle> References: <201101200017.53587.4ernov@gmail.com> <1295553289.23989.1.camel@zingle> Message-ID: <201101210153.09337.4ernov@gmail.com> Thanks, Tim, I finally found the right files, don't really sure if it is necessary to include adding of documentation to build in the patch. For any case, the files are the following (in docs/plugins): -plugins-section.txt - one should add separate
to make xml be built; -plugins-docs.sgml - one should add xi:include here to keep a reference to xml; inspect/plugin-.xml - one should add separate section for each element with documentation to generate, the section contains brief description and some details. On Thursday 20 January 2011 22:54:49 you wrote: > On Thu, 2011-01-20 at 00:17 +0300, Alexey Chernov wrote: > > I'm adding some functionality to v4l2 plugin in gst-plugins-good (radio > > element to be more exact, it was previously discussed here). The task is > > near the end, but I can't add my header (it is sys/v4l2/gstv4l2radio.h) > > to be parsed by gtk-doc when --enable-gtk-doc option is on. It seem to > > be simply ignored but I can't understand where it should be added to be > > parsed - in docs/ directory many files seem to be generated > > automatically. > > > > Could someone please give me any reference where I should add this header > > file's path or something to make it be parsed by gtk-doc? > > Maybe doc/plugins/Makefile.am, but it shouldn't really be necessary > (might make the private element struct show up though). > > Cheers > -Tim From hadess at hadess.net Fri Jan 21 01:51:38 2011 From: hadess at hadess.net (Bastien Nocera) Date: Fri, 21 Jan 2011 00:51:38 +0000 Subject: [gst-devel] Least Buggy API for Totem Mozilla Plugin In-Reply-To: <4D38B4D8.8000701@fas.harvard.edu> References: <4D38B4D8.8000701@fas.harvard.edu> Message-ID: <1295571101.28153.32.camel@novo.hadess.net> On Thu, 2011-01-20 at 17:19 -0500, Benjamin M. Schwartz wrote: > The Totem developers have very helpfully provided reimplementations of the > VLC, WMP, QuickTime, and DivX plugin javascript APIs. Which, if any, of > these is preferred/less buggy/more usable? The short answer is that the VLC API is what you should use if you want to support Totem specifically (using an object with the "application/x-totem-plugin" mime-type). Otherwise I'd say that the QuickTime plugin is probably the most feature complete, if you're looking for a "compatible" API. If you find bugs in either APIs with the latest stable, or unstable, releases, please file a bug, it should get thrashed out pretty quickly. Cheers PS: gnome-multimedia is probably a better mailing-list than gstreamer-devel for Totem questions. From jorney_dong at asus.com Fri Jan 21 05:43:10 2011 From: jorney_dong at asus.com (Jorney_dong) Date: Fri, 21 Jan 2011 12:43:10 +0800 Subject: [gst-devel] amr-wb seek problem In-Reply-To: References: <1294652354379-3206831.post@n4.nabble.com> Message-ID: <4D390EDE.2030802@asus.com> Thanks for everyone's help! I have got the issue fixed, it's the amrwbdec: just add one line in the amrwbdec.c of gst-plugins-ugly-0.10.16. 288: if (!block || gst_adapter_available (amrwbdec->adapter)< block) + { + gst_adapter_clear (amrwbdec->adapter); break; + } Because when there are incorrect data, if you don't clear it, the next checking is still incorrect. On 2011?01?10? 21:00, sudarshan bisht wrote: > > It seems amrwbdec does not handle src pad event (seek ) and thats why > there is a blocking. There is a lack of implementation of src event > handling thats why kept in gst-plugins-ugly. > > > On Mon, Jan 10, 2011 at 11:39 AM, jorney > wrote: > > > Hi > I am writing amr-wb player with gstreamer on Marvell pxa310. > When I seek to a position by time format, there is a long time > blocking > (10 sec or even to 1 min), and there is no voice during the block, > then > an EOS message was emitted, > but the position I seeked is far away to the end of the file. > > Bytes format has the same problem! here is my seek code: > > breslt = gst_element_seek(pipeline, > 1.0, > GST_FORMAT_BYTES, > GST_SEEK_FLAG_FLUSH,//|GST_SEEK_FLAG_KEY_UNIT|GST_SEEK_FLAG_SEGMENT, > GST_SEEK_TYPE_SET, > bytePos,//GST_SECOND*nForw*count, > GST_SEEK_TYPE_NONE, > GST_CLOCK_TIME_NONE > ); > > I use gstreamer-0.10.29 and gst-plugins-ugly-0.10.15's amrwbdec, > gst-plugins-bad-0.10.13's amrparse. like this: > > gst-launch filesrc location=abc.amr ! amrparse ! amrwbdec ! alsasink > > > And the codec is opencore-amr-0.1.2 > Could any body help? > > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/amr-wb-seek-problem-tp3206831p3206831.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of > any company > that requires sensitive data to be transmitted over the Web. > Learn how to > best implement a security strategy that keeps consumers' > information secure > and instills the confidence they need to proceed with transactions. > 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 > > > ------------------------------------------------------------------------------ > Gaining the trust of online customers is vital for the success of any company > that requires sensitive data to be transmitted over the Web. Learn how to > best implement a security strategy that keeps consumers' information secure > and instills the confidence they need to proceed with transactions. > 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 ===================================================================================================================================== This email and any attachments to it contain confidential information and are intended solely for the use of the individual to whom it is addressed.If you are not the intended recipient or receive it accidentally, please immediately notify the sender by e-mail and delete the message and any attachments from your computer system, and destroy all hard copies. If any, please be advised that any unauthorized disclosure, copying, distribution or any action taken or omitted in reliance on this, is illegal and prohibited. Furthermore, any views or opinions expressed are solely those of the author and do not represent those of ASUSTeK. Thank you for your cooperation. ===================================================================================================================================== -------------- next part -------------- An HTML attachment was scrubbed... URL: From amitpandya at itimes.com Fri Jan 21 07:28:59 2011 From: amitpandya at itimes.com (amitpandya) Date: Thu, 20 Jan 2011 22:28:59 -0800 (PST) Subject: [gst-devel] AV synchronization problem with elementary AV (AAC/H264) streaming Message-ID: <1295591339101-3229054.post@n4.nabble.com> Hi all, I am doing the elementary streaming of AAC/H264 data using RTP over UDP with gstreamer. In my case, I have two separate pipelines for Audio & video as below. video Case:(pipeline1) appsrc->input_capsfilter->packatizer->output_capsfilter->gstrtpbin. where, appsrc is pushing H264 encoded data to pipeline Audio Case:(pipeline2) appsrc->input_capsfilter->packatizer->output_capsfilter->gstrtpbin. where, appsrc is pushing AAC encoded data to pipeline Now , the problem is at clientside(QT player) both Audio & Video playback is happening fine but audio is delayed by several seconds (8-10 sec precisely) than the video. After primary debugging found that audio pipeline is taking more time in processing and sending Audio Packets a bit late than the Video packets.Here both the Audio - video processing is happening in different pipelines and both pipelines have their own processing time, which is unfortunately different. In sort, Video pipeline start sending video packets much earlier & after some time Audio pipeline starts sending Audio packets to the N/W.that is why at client side video playbach happens first and after some delay audio starts playing. I have done following testing to resolve the problem. 1)Instead of two pipelines I used only single pipeline for both AV but it is not making any difference. 2)With single pipeline instead of using two different gstrtpbin (one for Audio & one for Video) used only one gstrtpbin but that is not making any difference. 3) Used queue elements in the pipeline to delay video pipeline still no difference. My question is how AV synchronization can be handled within the pipeline for these case? or I have to put some buffering mechanism outside gstreamer pipeline which can handle such AV syncronization?If so how to do it? Any help is greatly appreciated. Thanks, Amity Pandya -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/AV-synchronization-problem-with-elementary-AV-AAC-H264-streaming-tp3229054p3229054.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From anuprabhu32 at gmail.com Thu Jan 20 19:29:50 2011 From: anuprabhu32 at gmail.com (anuradha prabhu) Date: Thu, 20 Jan 2011 23:59:50 +0530 Subject: [gst-devel] Regarding GST_sink_Event Message-ID: HI, in the mad plugin it has been observed that gst_mad_sink_event is trigered before first chain function. The event is GST_EVENT_NEWSEGMENT with format as GST_FORMAT_TIME. How this sink_event has been sent with proper format as GST_FORMAT_TIME.. ? Which module sends this sink event. regards anuradha -------------- next part -------------- An HTML attachment was scrubbed... URL: From olivier.aubert at liris.cnrs.fr Fri Jan 21 09:43:37 2011 From: olivier.aubert at liris.cnrs.fr (Olivier Aubert) Date: Fri, 21 Jan 2011 09:43:37 +0100 Subject: [gst-devel] opengl on top of video In-Reply-To: References: Message-ID: <1295599417.3252.374.camel@abbeyroad.dnsalias.org> Hello Depending on the nature of your visualization, you may want to try the rsvgoverlay element, which should be included in the latest gst-plugins-bad pre-release. Insert the rsvgoverlay element in your pipeline, and feed it SVG data (through a property or through a data_sink pad) generated from your real-time data. Olivier On Thu, 2011-01-20 at 11:00 -0800, Bert Douglas wrote: > Hello, > > I have a python-gstreamer app that shows video. > It uses xvimagesink and links it to a gtk.DrawingArea. > > This works great. > > Now I want to overlay some graphics, dynamically generated from real > time data. > Opengl seems like a good choice to render the graphics. > > But I have no clue how to combine the video and graphics. > I don't think they are made to work with each other. > > Thanks, > Bert Douglas > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 acandido at hi-iberia.es Fri Jan 21 10:40:13 2011 From: acandido at hi-iberia.es (Andres Gonzalez) Date: Fri, 21 Jan 2011 10:40:13 +0100 Subject: [gst-devel] opengl on top of video In-Reply-To: References: Message-ID: <4D39547D.9050109@hi-iberia.es> Hi, On 20/01/11 20:00, Bert Douglas wrote: > Hello, > > I have a python-gstreamer app that shows video. > It uses xvimagesink and links it to a gtk.DrawingArea. > > This works great. > > Now I want to overlay some graphics, dynamically generated from real > time data. > Opengl seems like a good choice to render the graphics. > > But I have no clue how to combine the video and graphics. > I don't think they are made to work with each other. > > Thanks, > Bert Douglas Just an idea: have you tried using gst-plugins-gl for the graphics, and videomixer to overlay them on the video? Regards, Andr?s -- Andr?s Gonz?lez T: 91 458 51 19 (ext:165) HI-IBERIA INGENIERIA Y PROYECTOS C/ Bolivia 5 - Madrid - 28016 From bilboed at gmail.com Fri Jan 21 11:15:37 2011 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 21 Jan 2011 11:15:37 +0100 Subject: [gst-devel] Regarding GST_sink_Event In-Reply-To: References: Message-ID: <1295604937.11672.44.camel@localhost> On Thu, 2011-01-20 at 23:59 +0530, anuradha prabhu wrote: > HI, > > in the mad plugin it has been observed that gst_mad_sink_event is > trigered before first chain function. > The event is GST_EVENT_NEWSEGMENT with format as GST_FORMAT_TIME. Buffers must always be preceded by a NEWSEGMENT event. This is normal, you will see this same behaviour everywhere else. > > How this sink_event has been sent with proper format as > GST_FORMAT_TIME.. ? Which module sends this sink event. An upstream element, maybe a parser, maybe a demuxer, maybe rtspsrc, who knows ? Edward > > regards > anuradha > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From nvineeth at gmail.com Fri Jan 21 11:25:32 2011 From: nvineeth at gmail.com (vineeth) Date: Fri, 21 Jan 2011 15:55:32 +0530 Subject: [gst-devel] gst_tag_list_get_string In-Reply-To: <1294928470699-3215915.post@n4.nabble.com> References: <1294928470699-3215915.post@n4.nabble.com> Message-ID: Initially you can check if the tags are set properly using : id3v2 --list mp3-filename --vineeth On Thu, Jan 13, 2011 at 7:51 PM, bonkers wrote: > > Hello! > > when i try to get some informations about the track using the > GST_MESSAGE_TAG i get nothing. > > int MyStreamer_bus_call(GstBus *bus, GstMessage *msg, gpointer data) > { > GstTagList *taglist=gst_tag_list_new(); > char *c,*artist; > GMainLoop *loop = (GMainLoop*) data; > > switch(GST_MESSAGE_TYPE(msg)){ > gst_message_parse_tag(msg, &taglist); > if (gst_tag_list_get_string (taglist, GST_TAG_ARTIST, &artist)) > { > printf("got artist: %s\n",artist); > } > else{ > printf("no artist!! \n"); > } > } > return 0; > } > > I always get "no artist"!! I?m sure that I set the tags of the mp3 file. > > can you help me?? > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/gst-tag-list-get-string-tp3215915p3215915.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 nvineeth at gmail.com Fri Jan 21 11:29:53 2011 From: nvineeth at gmail.com (vineeth) Date: Fri, 21 Jan 2011 15:59:53 +0530 Subject: [gst-devel] gstreamer benchmark tests Message-ID: Hi, I was curious to know if we have a make target to run the benchmark tests ( under tests/benchmark ) similar to make check which executes all the unit tests. --vineeth -------------- next part -------------- An HTML attachment was scrubbed... URL: From thiago.sousa.santos at collabora.co.uk Fri Jan 21 12:12:18 2011 From: thiago.sousa.santos at collabora.co.uk (Thiago Sousa Santos) Date: Fri, 21 Jan 2011 08:12:18 -0300 Subject: [gst-devel] gst_tag_list_get_string In-Reply-To: References: <1294928470699-3215915.post@n4.nabble.com> Message-ID: <1295608338.2236.5.camel@blacksheep> On Fri, 2011-01-21 at 15:55 +0530, vineeth wrote: > Initially you can check if the tags are set properly using : > id3v2 --list mp3-filename > > --vineeth > > > On Thu, Jan 13, 2011 at 7:51 PM, bonkers wrote: > > Hello! > > when i try to get some informations about the track using the > GST_MESSAGE_TAG i get nothing. Your code has some flaws. > > int MyStreamer_bus_call(GstBus *bus, GstMessage *msg, gpointer > data) > { > GstTagList *taglist=gst_tag_list_new(); No need to create a taglist here. > char *c,*artist; > GMainLoop *loop = (GMainLoop*) data; > > switch(GST_MESSAGE_TYPE(msg)){ You need to add a case for the tag message type before attempting to parse it. > gst_message_parse_tag(msg, &taglist); > if (gst_tag_list_get_string (taglist, GST_TAG_ARTIST, > &artist)) > { > printf("got artist: %s\n",artist); > } > else{ > printf("no artist!! \n"); > } > } > return 0; > } > > I always get "no artist"!! I?m sure that I set the tags of the > mp3 file. > > can you help me?? -- Thiago > -- > View this message in context: > http://gstreamer-devel.966125.n4.nabble.com/gst-tag-list-get-string-tp3215915p3215915.html > Sent from the GStreamer-devel mailing list archive at > Nabble.com. > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. > Understand > malware threats, the impact they can have on your business, > and how you > can protect your company and customers by using code signing. > 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 > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From thiago.sousa.santos at collabora.co.uk Fri Jan 21 12:29:29 2011 From: thiago.sousa.santos at collabora.co.uk (Thiago Sousa Santos) Date: Fri, 21 Jan 2011 08:29:29 -0300 Subject: [gst-devel] [camerabin] buggy webcams with no autoexposure In-Reply-To: <1295170670.6629.29.camel@zwerg> References: <1295170670.6629.29.camel@zwerg> Message-ID: <1295609369.2236.18.camel@blacksheep> On Sun, 2011-01-16 at 10:37 +0100, Alexey Fisher wrote: > Hi, > i thing camerabin is the right place to discuss this problem, > > some build in webcams on some laptops provide only static framereate > without autoexposure. With 30fps i get goot frame rate but bad image, > with 10fps i get good picture, but camerabin will take 30fps by default. > > For example the cam on fujisu lifebook a530 is more tricky. It support > only 30fps on 640x480 frame size and about 9fps on 1280x1024. So to get > good picture for video chat i should do fallowing workaround: > > gst-launch-0.10 v4l2src ! ffmpegcolorspace ! > video/x-raw-yuv,width=1280,height=1024,framerate=9/1 ! videoscale ! > video/x-raw-yuv,width=320,height=240 ! fpsdisplaysink > > I have fallowing questions: > How can we find if the cam is buggy? Add v4l2_ctrl_quirk? Or if cam do > nor have exposure controls this mean it use static exposure time? > For video chat we need not more than 15fps, so if we have only choice > between 30fps and 10fps, best decision will be probably 10fps. My suggestion here is to have these possible caps ordered and make camerabin prefer the first structure on caps. In any case, this looks tricky to represent in caps in a generic way. You could solve that at application level. > > Less fps -> less videotmpnoise -> better image quality -> less load for > encoder -> less traffic. -- Thiago From katcipis at inf.ufsc.br Fri Jan 21 13:38:23 2011 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Fri, 21 Jan 2011 10:38:23 -0200 Subject: [gst-devel] Cross compiling on blackfin Message-ID: Hi all, i don't know if this is related to blackfin or if it is a build issue, so I'm posting on the gst-devel instead of gst-embedded. while i was cross compiling gstreamer-core, i disabled check (because i really don't want it on blackfin, it uses fork, and for me it is a problem, and while I'm cross compiling i cant run tests). i run configure like that: ./configure --prefix /home/katcipis/blackfin --libdir /home/katcipis/blackfin/lib --build=i686-pc-linux-gnu --host=bfin-linux-uclibc --enable-shared --disable-check --disable-tests --disable-failing-tests --disable-examples gstreamer core and gstreamer-base cross compile just ok.... but on gstreamer-good i configured like that: ./configure --prefix /home/katcipis/blackfin --libdir /home/katcipis/blackfin/lib --build=i686-pc-linux-gnu --host=bfin-linux-uclibc --enable-shared --without-x --without-libv4l2 --disable-libpng --disable-gdk_pixbuf --disable-oss4 --disable-soup --disable-gst_v4l2 --disable-annodex --disable-oss --disable-shout2 --disable-cairo --disable-examples --disable-schemas-install --disable-gtk-doc and everything compiles just great... but then...it tries to compile tests/check, i don't know why the configure process think i have libgstcheck (it is not on the lib dir since i disabled it on gstreamer core configuration) and tries to link with gst-check and to run the tests. the error is the following: make[3]: Entering directory `/home/katcipis/blackfin/src/gst-plugins-good-bfin/tests/check' CC autodetect.o CCLD elements/autodetect /opt/uClinux/bfin-linux-uclibc/lib/gcc/bfin-linux-uclibc/4.3.5/../../../../bfin-linux-uclibc/bin/ld: cannot find -lgstcheck-0.10 collect2: ld returned 1 exit status config.log shows that the configure process really thinks i have check: HAVE_GST_CHECK_FALSE='#' HAVE_GST_CHECK_TRUE='' i tried to configure with --disable-tests, --disable-check...none of this options exists on configure... and the internal tests that configure run to check com gst-check for some kind of reason pass...and it tries to use check...and i don't know how i can force gst-plugins-good build to don't use gst-check. Am i doing something wrong? is there a way to disable the run of the check tests on gst-plugins-good? best regards, Tiago Katcipis -------------- next part -------------- An HTML attachment was scrubbed... URL: From stringarray at gmail.com Sat Jan 22 01:46:42 2011 From: stringarray at gmail.com (laura la) Date: Fri, 21 Jan 2011 21:46:42 -0300 Subject: [gst-devel] gst_parse_bin_from_description and pad creation Message-ID: Hi All, I'm trying to use rsvgoverlay plugin from gst-plugins-bad, it overlays a .svg file above a video. It has two sinks, "data_sink" for the .svg data, and "sink" for video data. The gst-launch command works fine: gst-launch v4l2src device=/dev/video0 ! ffmpegcolorspace ! rsvgoverlay location=/usr/share/icons/gnome/scalable/places/gnome-foot.svg ! ffmpegcolorspace ! videoscale ! video/x-raw-yuv,width=640,height=480 ! ffmpegcolorspace ! ximagesink or gst-launch -v videotestsrc ! ffmpegcolorspace ! rsvgoverlay location=foo.svg ! ffmpegcolorspace ! autovideosink Here are a couple more examples in the plugin code itself: http://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/ext/rsvg/gstrsvgoverlay.c I need to create the GstElement using gst_parse_bin_from_description: GstElement * overlay = gst_parse_bin_from_description ("rsvgoverlay location=foo.svg", TRUE, NULL); and then link it to other elements of the stream. This isn't working. Here is a simplified code snippet with working and not working ways to create the GstElement. http://pastebin.com/uiqDZR1Y This works: overlay = gst_element_factory_make ("rsvgoverlay", "overlay"); g_object_set (overlay, "location", "foo.svg", NULL); But I using gst_parse_bin_from_description I get errors about negotiating format. I think during linking it's trying to use "data_sink" instead of "sink" in rsvgoverlay element. But I'm not 100% sure. I tried variations of the pipeline description, for example ffmpegcolorspace ! rsvgoverlay location=foo.svg ! ffmpegcolorspace but it doesn't work either. Here is a GST_DEBUG=4 log file in case you want to take a look: http://ubuntuone.com/p/Z4L I'm a participant in the GNOME Outreach program for Women 2010, I'm working on Cheese application, and the idea is to add funny .svg hats that follow you around with face recognition. See more here: http://programmermusings.blogspot.com/ http://projects.gnome.org/cheese/ Bye! Laura From jiqingkai at 126.com Sat Jan 22 04:07:48 2011 From: jiqingkai at 126.com (jiqingkai) Date: Sat, 22 Jan 2011 11:07:48 +0800 (CST) Subject: [gst-devel] How to compile Gstreamer in VS2010? In-Reply-To: References: Message-ID: <306aa6ef.c157.12dabb122d7.Coremail.jiqingkai@126.com> Hi All, I'm trying to build gstreamer with Visual Studio 2010. I installed these tools listed in readme.txt successfully. But fatal error occurs when I start Visual Studio 2010 to compile the Gstreamer. The following is I done: >. Windows Installer XML (WiX) 3.5 or later (e.g. http://wix.sourceforge.net/releases/3.5.2430.0/Wix35.msi) >. Perl 5.10+ (32-bit, x86-compatible) a. You can skip this step if you're only interested in building GStreamer itself and not the dependencies. b. Recommended ActiveState Perl (http://www.activestate.com/downloads/) on Windows 1. Download the LibXML package (these steps may be different in future releases) a. Load the perl package manager ("ppm" command) b. Select the Edit > Preferences menu c. Under "Add Repository" select "University of Winnipeg" from the "suggested" list and select "Add" d. Select "OK" e. Scroll down the list and right click on "XML-LibXML" and "XML-LibXML-Common" both and select install f. Select File > Run Marked Actions >. DirectX SDK (November 2008 or later) a. Download this, for example, from http://www.microsoft.com/downloads/info.aspx?na=90&p=&SrcDisplayLang=en&SrcCategoryId=&SrcFamilyId=5493f76a-6d37-478d-ba17-28b1cca4865a&u=http%3a%2f%2fdownload.microsoft.com%2fdownload%2f5%2f8%2f2%2f58223f79-689d-47ae-bdd0-056116ee8d16%2fDXSDK_Nov08.exe >. Python 2.6 (32-bit, x86-compatible) b. Download, for example, from http://www.python.org/download/releases/2.6.6/ c. Install to C:\Python26 >. PyGobject (2.14) >.Setup Visual Studio 2008+ Add Custom Rules Path 1. Select the Tools > Options... menu. 2. Expand "Projects and Solutions" and select "VC++ Project Settings" 3. In "Rule File Search Paths", add the top-level directory for this code (e.g. C:\OSSBuild\) a. If needed, separate other search paths with a semicolon (;). -------------- next part -------------- An HTML attachment was scrubbed... URL: From gibrovacco at gmail.com Sat Jan 22 10:04:07 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sat, 22 Jan 2011 11:04:07 +0200 Subject: [gst-devel] how do I packet pace rtp network traffic? In-Reply-To: <8E459EED6B356D469071FDC814554FC43FF356C3@COR1MBX01.emea.tsp.ad> References: <8E459EED6B356D469071FDC814554FC43FF356C3@COR1MBX01.emea.tsp.ad> Message-ID: Hi, On Thu, Jan 20, 2011 at 11:54 AM, Redfern, Keith wrote: > We are developing a gstreamer based pipeline to read a video source, rtp > payload the video frame and then udpsink the rtp packets. > > We are finding that the pipeline bursts the rtp data unto the network, > creating network spikes. In one example, the interpacket network delay is > about 15micro seconds for approx 150 packets, meaning the whole video frame > is delivered unto the network in 2.25ms. It's conceptually right that fragmented packets, having all the same timestamp, are sent as fast as possible over the network, the bottleneck being the network and physical layer setting. It's usually bad to try and artificially delay packets, as you will accrue network latencies over the time. If your problem is the number of packets (not the size of them) it's good to try with an higher MTU, you can usually set it from the payloader. If the MTU is already set appropriately (it really depends on the network you're using) or you're actually concerned about the # of bytes sent across the network, then you're encoding with too high parameters / quality. Try one or more of the following: - Reduce frame resolution. - Reduce frame rate. - Reduce encoder output bitrate. - Reduce encoder output quality. - Use a better encoder (e.g. h264 is usually better than mpeg-2). - Increase the number of P frames between I frames (or, that is the same, decrease the frequency of I frames). - Already said, increase the MTU (it will reduce the overhead caused by RTP, UDP and IP on the payload). > > Is there some way in which to ?pace? the rtp packets so that it takes > 20-30ms to send the same frame? Probably, but you're going through a lot of problems doing that ;) . Regards, Marco > > > > Keith > > > > > ________________________________ > Tyco Safety Products/CEM Systems?Ltd. > Registered in Northern Ireland: NI 25728.? Registered Office: 195 Airport > Road West, Belfast, BT3 9ED. > > Please note that any views or opinions presented in this email are solely > those of the author and do not necessarily represent those of the company. > This email and any attachments are confidential and intended solely for the > use of the individual or entity to whom they are addressed. If you have > received this email in error please notify the sender immediately and delete > any copies in your possession. > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 Sat Jan 22 10:08:54 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sat, 22 Jan 2011 11:08:54 +0200 Subject: [gst-devel] Playbin stuck on buffering In-Reply-To: <1294924840294-3215817.post@n4.nabble.com> References: <1294924840294-3215817.post@n4.nabble.com> Message-ID: Hi, On Thu, Jan 13, 2011 at 3:20 PM, kobalt wrote: ..snip.. > > buffering... 0% > > If i try the same command but using playbin2 everything works perfectly. Is > there a solution to make this work on playbin, or is this an issue known by > gstreamer-dev team? It is, point your browser to: http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-playbin.html and scroll until the first note in a green box. You'll read: "This element is deprecated and no longer supported. You should use the playbin2 element instead." Google is your friend ;). Regards, Marco > > Thank You, > Cornel > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Playbin-stuck-on-buffering-tp3215817p3215817.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 Sat Jan 22 10:20:56 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sat, 22 Jan 2011 11:20:56 +0200 Subject: [gst-devel] AV synchronization problem with elementary AV (AAC/H264) streaming In-Reply-To: <1295591339101-3229054.post@n4.nabble.com> References: <1295591339101-3229054.post@n4.nabble.com> Message-ID: Hi, On Fri, Jan 21, 2011 at 8:28 AM, amitpandya wrote: > > Hi all, > > I am doing the elementary streaming of AAC/H264 data using RTP over UDP with > gstreamer. > > In my case, I have two separate pipelines for Audio & video as below. > > video Case:(pipeline1) > appsrc->input_capsfilter->packatizer->output_capsfilter->gstrtpbin. > > ? ? ? ?where, appsrc is pushing H264 encoded data to pipeline > > Audio Case:(pipeline2) > appsrc->input_capsfilter->packatizer->output_capsfilter->gstrtpbin. > > ? ? ? ?where, appsrc is pushing AAC encoded data to pipeline It is usually bad to use separate pipelines for interrelated streams. You should make somehow sure that the clocks are perfectly synchronised and skew is kept under very low threshold. Btw I read below you've tried with a single pipeline as well, so you might have more than one problem here. > > Now , the problem is at clientside(QT player) both Audio & Video playback is > happening fine but audio is delayed by several seconds (8-10 sec precisely) > than the video. > > After primary debugging found that audio pipeline is taking more time in > processing and sending Audio Packets a bit late than the Video packets.Here > both the Audio - video processing is happening in different pipelines and > both pipelines have their own processing time, which is unfortunately > different. > > In sort, Video pipeline start sending video packets much earlier & after > some time Audio pipeline starts sending Audio packets to the N/W.that is why > at client side video playbach happens first and after some delay audio > starts playing. > > I have done following testing to resolve the problem. > 1)Instead of two pipelines I used only single pipeline for both AV but it is > not making any difference. This is a first step forward :) > 2)With single pipeline instead of using two different gstrtpbin (one for > Audio & one for Video) used only one gstrtpbin but that is not making any > difference. Another step in the right direction. > 3) Used queue elements in the pipeline to delay video pipeline still no > difference. A queue does not delay playback on the remote end. at least not as you're expecting it to happen. The player will always refer to timestamps, assigned from the source and not modified by the queue. > > My question is how AV synchronization can be handled within the pipeline for > these case? Are you setting proper timestamps and durations in the buffers you're generating with the two appsrcs? Are the timestamps synchronised to the pipeline clock (e.g. are there no drifts between the two streams)? Regards, Marco or > I have to put some buffering mechanism outside gstreamer pipeline which can > handle such AV syncronization?If so how to do it? > > Any help is greatly appreciated. > > Thanks, > Amity Pandya > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/AV-synchronization-problem-with-elementary-AV-AAC-H264-streaming-tp3229054p3229054.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > 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 Sat Jan 22 12:17:59 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sat, 22 Jan 2011 11:17:59 +0000 Subject: [gst-devel] RELEASE: GStreamer 0.10.32 "Take Me to the Bonus Level" Message-ID: <1295695079.7585.8.camel@zingle> This mail announces the release of GStreamer 0.10.32 "Take Me to the Bonus Level". 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 core elements: valve, input-selector, output-selector (moved from -bad) * Performance improvements for many heavily-used code paths (making more things lock-free etc.) * gobject-introspection annotations for most core API 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-0.10.32.tar.bz2 http://gstreamer.freedesktop.org/src/gstreamer/gstreamer-0.10.32.tar.gz MD5 sums (for tarballs downloaded from gstreamer.freedesktop.org): bcc72c77bf00e183102f6302e627aa20 gstreamer-0.10.32.tar.gz 442bc3d37b8511a73379143e7531d726 gstreamer-0.10.32.tar.bz2 Enjoy! -------------- next part -------------- Release notes for GStreamer?0.10.32 "Take Me to the Bonus Level" 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. 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 * GLib requirement is now >= 2.22 * New core element: valve (moved from -bad) * New core element: input-selector (N.B. without "select-all" property, use fsfunnel instead) (moved from -bad) * New core element: output-selector (with different negotiation behaviour by default, set pad-negotiation-mode=active for previous behaviour) (moved from -bad) * Performance improvements for many heavily-used code paths: GstPad, GstPoll, GstClock, GstTask, basesink, basesrc, queue2, multiqueue * gobject-introspection: add annotations for most core API * clock: make sync clock wait lockfree * fdsrc/fdsink: reenable on MSVC * registry: fix GStatBuf definition for win32 when building against older glib (fixes unnecessary rescanning of plugins at start-up) * element: add a more flexible way to get request pads from elements * multiqueue: return upon input when already eos * object: fix creation of default name (when creating more than 100000 elements) * pluginloader: fix hangs on OSX * poll: fixes for (p)select backend (used e.g. on OSX) * poll: refactor and make more lockfree; fixes for win32 and OSX (pselect backend) * registry: don't replace valid existing plugins by blacklisted ones * tags: don't produce duplicated entries when merging same value twice * basesink: preroll fixes for async=false case * basesink: rework position reporting code * basetransform: handle downstream giving a buffer with new caps but invalid size Bugs fixed in this release * 635785 : basesrc: fix deadlock * 638599 : GST_PTR_FORMAT causes crashes if GLib-internal printf is used but system provides register_printf_specifier * 503592 : gstpad.c does many ref/unref of peer pad in dataflow * 564056 : Protect against umask(0177) * 607513 : input-selector segfaults in g_object_notify() * 632168 : [gsttask] MSVC thread names for task debugging * 632447 : reduce static memory allocated by the registry * 632557 : [macros] Define restrict keyword if not available * 632778 : Optimisations to GstBaseSink * 632779 : gstdataqueue: Only emit g_cond_signal when needed * 632780 : queue: Remove useless checks from e406f7 * 633918 : [fakesink] [PATCH] print sink-message events like a message and its structure * 634965 : sinks render buffers in state PAUSED when async is FALSE * 635001 : basesink: fix position reporting in PAUSED * 636268 : configure test fails on FreeBSD * 636455 : basesrc: Avoid taking object locks for just checking tag presence * 637057 : [plugin-scanner] gstpoll fails with select backend * 637300 : [API] request pad based on caps * 637549 : build fails: ./.libs/libgstbase-0.10.so: undefined reference to `gst_clock_single_shot_id_reinit' * 637776 : merging the same tag values again produces lists containing duplicates * 638381 : {input,output}-selector: double-check API before release * 638399 : a few typos in GStreamer * 638900 : [GstPoll] Doesn't compile with MinGW * 638941 : registry scan/loading race and inconsistency API changed in this release - API additions: * gst_clock_single_shot_id_reinit() * gst_element_request_pad() * GstElementClass::request_new_pad_full() * gst_poll_get_read_gpollfd() * gst_value_list_merge() * GST_CLOCK_DONE 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. Contributors to this release * Andoni Morales Alastruey * Benjamin Gaignard * Benjamin Otte * David Hoyt * David Schleef * Edward Hervey * Havard Graff * Jan Schmidt * Kipp Cannon * Koop Mast * Lasse Laukkanen * Mark Nauwelaerts * Michael Smith * Olivier Crete * Olivier Cr?te * Raimo J?rvi * Sebastian Dr?ge * Stefan Kost * Thiago Santos * Tim-Philipp M?ller * Tommi My?h?nen * Wim Taymans * Zhang Wanming * Christian Schaller * Sjoerd Simons ? From t.i.m at zen.co.uk Sat Jan 22 12:21:32 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sat, 22 Jan 2011 11:21:32 +0000 Subject: [gst-devel] RELEASE: GStreamer Base Plug-ins 0.10.32 "Your Life You Like It Well" Message-ID: <1295695292.7585.12.camel@zingle> This mail announces the release of GStreamer Base Plug-ins 0.10.32 "Your Life You Like It Well". 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: * New encodebin element * New encoding profile and encoding targets API in pbutils * 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-0.10.32.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-0.10.32.tar.gz MD5 sums (for tarballs downloaded from gstreamer.freedesktop.org): ddfd2d71cfa594529dfa7a8e0fb37b79 gst-plugins-base-0.10.32.tar.gz 2920af2b3162f3d9fbaa7fabc8ed4d38 gst-plugins-base-0.10.32.tar.bz2 Enjoy! -------------- next part -------------- Release notes for GStreamer Base Plug-ins?0.10.32 "Your Life You Like It Well" 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. 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 network: tcp typefind functions audio processing: audioconvert, adder, audiorate, audioresample, volume visualisation: libvisual video processing: ffmpegcolorspace aggregate elements: uridecodebin, playbin2, decodebin2, decodebin, playbin, encodebin libraries: app, audio, cdda, fft, interfaces, netbuffer, pbutils, riff, rtp, rtsp, sdp, tag, video 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 * GLib requirement is now >= 2.22, gobject-introspection >= 0.9.12 * New encodebin element * New encoding profile and encoding targets API in pbutils * audioresample: corrected buffer duration calculation to account for nonzero initial timestamp * audioresample: provide as much valid output ts and offset as valid input * audioresample: push half a history length, instead of a full history length, at end-of-stream so that output segment and input segment have same duration * decodebin2: deprecate new-decoded-pad and removed-decoded-pad signals (use "pad-added" and "pad-removed" instead) * multifdsink: add first and last buffer's timestamp to the stats; only keep last valid timestamp * oggdemux: extract more tags (vorbis comment headers, Kate) * oggdemux: ignore header pages when looking for keyframe; set headers on caps * oggdemux: fix interpretation of Theora granule position and parsing of Theora size * oggparse: Set DELTA_UNIT on buffers * playbin2: delay stream-changed messages, fixing KATE subtitle recognition * textoverlay: make text, xpos, ypos, color, and silent properties controllable * typefinding: (E)AC-3 and ISO typefinder improvements; add yuv4mpeg typefinder * typefinding: add "stream-format" to h264 caps, and framed=false to DTS caps * typefinding: assume EBML files without doctype are matroska * videorate: fix behaviour for frame rate cap changes * vorbisdec: avoid using invalid timestamps; keep timestamps when no decoded output * ximagesink, xvimagesink: add read-only window-width and window-height properties * baseaudiopay: fix timestamps on buffer lists * baseaudiosink: protect against ringbuffer disappearing while in a query * basedepay: add support for buffer lists in the depayloader * basertppay: use RTP base time when invalid timestamps * rtpbuffer: relax arrangement for RTP bufferlists * rtpdepayloader: add support for getting events * rtppayload: copy applied rate to segment * sdp: add method to check for multicast addresses * sdp: only parse TTL for IP4 addresses * video: add 8-bit paletted RGB, YUV9, YVU9 and IYU1 video formats * video: return correct component width/height for A420 Bugs fixed in this release * 619778 : oggdemux: fails on zero-length pages with Patent_Absurdity_HD_3540kbit.ogv * 586570 : Add GAP Flag support to audioresample * 623413 : pbutils: Add/Fix some media descriptions * 627476 : New profile library and encoding plugin * 629349 : [oggdemux] extract stream tags for tagreadbin and transcoding * 632667 : [ximagesink] added read-only properties window-width and window-height * 634397 : [multifdsink] [PATCH] Add the timestamp of the first and last buffer to the stats * 634522 : gst-visualize-m.m imports but doesn't use File::Basename * 635231 : baseaudiosink: protect against ringbuffer disappearing while in a query * 636198 : decodebin2: " removed-decoded-pad " signal never fired * 636769 : [appsink] Flushing property is never reset * 636827 : Usage of gst_caps_interset where gst_caps_can_intersect was intended? * 637324 : oggdemux: unable to demux Ogg files with Skeleton in push mode * 637377 : timeoverlay: add missing break * 637519 : ogg: implement packet duration query for kate streams * 637586 : playbin2 fails to recognize subtitle caps from katedec * 637735 : [encoding-profile] automatic load/save support and registry * 637758 : [exiftag] Generates buffers with uninitialized data during taglist- > exif buffer serialization * 637822 : oggdemux: allocate buffers using gst_buffer_new_and_alloc * 637927 : oggdemux: set headers on caps * 638200 : [oggdemux] fails to playback video file * 638276 : oggstream: when the last keyframe position is not known, do not use -1 * 638859 : textoverlay: make misc. properties controllable * 638901 : [encodebin] proper element documentation * 638903 : [encodebin] missing-plugin support * 638961 : Small configure bashism 0.10.31.2 * 639039 : gobject-introspection: GstPbutils gir scanner fails to link with gold linker * 639121 : oggdemux: outdated comment for gst_ogg_demux_submit_buffer() * 639215 : examples: Allow building with newer GTK+ * 639730 : discoverer: Validate timeouts before processing them * 639755 : discoverer: Clean up callbacks in dispose() * 639778 : discoverer: Drop new stream tags once preroll is done * 639790 : [gdp] Fix metadata g_warning * 639747 : Please export GST_TYPE_APP_STREAM_TYPE * 553244 : theoraparse doesn't work at all (throws criticals and asserts) API changed in this release - API additions: * gst_app_stream_type_get_type() * gst_discoverer_info_get_seekable() * gst_encoding_audio_profile_get_type() * gst_encoding_audio_profile_new() * gst_encoding_container_profile_add_profile() * gst_encoding_container_profile_contains_profile() * gst_encoding_container_profile_get_profiles() * gst_encoding_container_profile_get_type() * gst_encoding_container_profile_new() * gst_encoding_list_all_targets() * gst_encoding_list_available_categories() * gst_encoding_profile_find() * gst_encoding_profile_get_description() * gst_encoding_profile_get_format() * gst_encoding_profile_get_input_caps() * gst_encoding_profile_get_name() * gst_encoding_profile_get_presence() * gst_encoding_profile_get_preset() * gst_encoding_profile_get_restriction() * gst_encoding_profile_get_type() * gst_encoding_profile_get_type_nick() * gst_encoding_profile_is_equal() * gst_encoding_profile_set_description() * gst_encoding_profile_set_format() * gst_encoding_profile_set_name() * gst_encoding_profile_set_presence() * gst_encoding_profile_set_preset() * gst_encoding_profile_set_restriction() * gst_encoding_target_add_profile() * gst_encoding_target_get_category() * gst_encoding_target_get_description() * gst_encoding_target_get_name() * gst_encoding_target_get_profile() * gst_encoding_target_get_profiles() * gst_encoding_target_get_type() * gst_encoding_target_load() * gst_encoding_target_load_from_file() * gst_encoding_target_new() * gst_encoding_target_save() * gst_encoding_target_save_to_file() * gst_encoding_video_profile_get_pass() * gst_encoding_video_profile_get_type() * gst_encoding_video_profile_get_variableframerate() * gst_encoding_video_profile_new() * gst_encoding_video_profile_set_pass() * gst_encoding_video_profile_set_variableframerate() * gst_base_rtp_depayload_push_list() * gst_rtsp_url_decode_path_components() * gst_sdp_address_is_multicast() * gst_video_parse_caps_palette() 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 * Andoni Morales Alastruey * Andrea Sebastianutti * Andy Wingo * Arun Raghavan * Bastien Nocera * Benjamin Gaignard * Byeong-ryeol Kim * David Schleef * Edward Hervey * Evan Broder * Gavin Stark * Havard Graff * Koop Mast * Lane Brooks * Leo Singer * Mark Nauwelaerts * Michael Smith * Ren? Stadler * Rob Clark * Robert Swain * Sebastian Dr?ge * Sreerenj Balachandran * Stefan Kost * Thiago Santos * Tim-Philipp M?ller * Vincent Penquerc'h * Wim Taymans * Yang Xichuan * Zeeshan Ali (Khattak) * christian schaller ? From t.i.m at zen.co.uk Sat Jan 22 12:31:56 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sat, 22 Jan 2011 11:31:56 +0000 Subject: [gst-devel] RELEASE: GStreamer Good Plug-ins 0.10.27 "Some Kind of Temporal Blend" Message-ID: <1295695916.7585.23.camel@zingle> This mail announces the release of GStreamer Good Plug-ins 0.10.27 "Some Kind of Temporal Blend". 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: * JACK audio plugin moved to -good from -bad * qtdemux: add support for fragmented mp4 * deinterlacer improvements * RTP: new AC-3 payloader, fix many payloaders for seeking * v4l2src: flag interlaced video properly * 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-0.10.27.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-0.10.27.tar.gz MD5 sums (for tarballs downloaded from gstreamer.freedesktop.org): b6bc734152535388765f17d61278a6e2 gst-plugins-good-0.10.27.tar.gz c25d6c51916424009dfde4ee42ae1feb gst-plugins-good-0.10.27.tar.bz2 Enjoy! -------------- next part -------------- Release notes for GStreamer Good Plug-ins?0.10.27 "Some Kind of Temporal Blend" 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. "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 * avidemux: add workaround for buggy list size; extract datetime tags * cacasink: fix masks and strides * deinterlace: change the default to linear * deinterlace: avoid infinite loop draining * deinterlace: rewrite/fix how neighboring scan lines are calculated * flvdemux: use aac codec-data to adjust samplerate if needed * flvmux: Fix for nellymoser codecid setting * icydemux: Add 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag * id3demux: fix parsing of ID3v2.4 genre frames with multiple genres * imagefreeze: pass along eos if received before buffer arrives * jpegdec: add "max-errors" property to ignore decoding errors * jpegdec: avoid infinite loop when resyncing; discard incomplete image * matroskademux: add stream-format and alignment properties for h264 * matroskademux: assume matroska if no doctype is specified * matroskademux: increase allowed max. block size for push mode from 10M to 15M * matroskademux: normalize empty Cues to no Cues * matroskamux: add support for DTS and E-AC3 audio * matroskamux: try to write timestamps in all the outgoing buffers * multifilesink: send stream headers in key-frame mode * multiudpsink: add buffer-size property * navseek: add basic support to change playback rate * pulsemixer: Implement MIXER_FLAG_AUTO_NOTIFICATIONS * pulsesink: flush remaining buffered samples on EOS * pulsesink: make corking during pause synchronous; don't uncork in _start * pulsesink: Uncork stream while flushing the ringbuffer * pulsesrc: add "client" property * qtdemux: add support for fragmented mp4 * qtdemux: add support for (E)AC-3, WMA and VC-1 audio * qtdemux: allow pulling atoms with unknown size * qtdemux: fix flow return aggregation and handling of near end-of-file corner cases * qtdemux: parse and use creation time tag from mvhd * rtpbin: copy buffering stats * rtpbin: correctly calculate RTCP packet size * rtp: fix rank of payloaders and depayloaders * rtp: flush state on flush-stop for seek handling for many (de)payloaders * rtp ac3pay: add AC3 payloader * rtp h264depay: determine output h264 layout using caps negotiation * rtp h264pay: implement full bytestream scan mode * rtp j2kdepay: add support for buffer lists; make depayloader more resilient * rtp j2kpay: use buffer lists for better performance * rtp j2kpay: handle EOC correctly; stop scanning when we reached the end * rtp j2kpay: use SOP markers to split bitstream * rtp jitterbuffer: provide a clock; get better buffering level * rtp jpegdepay: fix framerate parsing for locales that use a comma as floating point * rtp mp4adepay: improve timestamps on outgoing packets * rtpsession: also emit RTCP activity on SR * rtpsession: remember last sent RB values * rtspsrc: add and use auto buffering mode * rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response * rtspsrc: include range request for all streams with non-aggregate control * rtspsrc: increase udp buffer size * rtspsrc: reset session manager base time when flushing * rtspsrc: select multicast transports in a smarter way * souphttpsrc: don't send seeks behind the end of file to the server * v4l2sink: add navigation support; properties to control crop * vrawdepay: fix length check * wavparse: detect DTS advertised as PCM correctly in some more cases * ximagesrc: change from XGetImage to XGetSubImage dependant on a property Bugs fixed in this release * 596321 : qtdemux: add support for fragmented MP4 and " mfra " boxes * 618389 : [pulsemixer] Should implement MIXER_FLAG_AUTO_NOTIFICATIONS interface * 618652 : [effectv] Use of uninitialised value in unit test * 620283 : Support for Adobe's F4F missing * 621929 : [PLUGIN-MOVE] move jack plugin from -bad to -good * 623178 : [matroskademux] error message for unrecognised FOURCC codes should be improved * 625825 : cannot link rtpmp4adepay ! aacparse * 629418 : progressreport: add support for determining stream position from buffer timestamps instead of using queries * 631516 : [navseek] Add support to change playback rate * 632654 : [matroskamux] try to write timestamps in most of the outgoing buffers * 632897 : flvmux does not set the correct nellymoser codec id * 633280 : [icydemux][PATCH] icydemux: Send 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag * 634314 : pngdec hangs on faulty pngs * 634391 : [v4l2src] add interlaced field to caps * 634393 : v4l2src: Set top field first for interlaced captures * 634910 : [rtph264pay] Implement bytestream scan mode * 634928 : [qtdemux] report creation/modification time via metadata tag * 635734 : jpegdec: infinite loop when playing back motion jpeg stream * 636049 : ximagesrc: fix remote X and off by ones * 636172 : imagefreeze: eos is not passed before a buffer arrives * 636234 : [wavparse] dts 6ch played as stereo 16 bit pcm if DTS frame starts at non-zero offset * 636621 : flvdemux: doesn't set the right sample rate for aac audio * 636784 : [qtdemux] GST_QUERY_CONVERT implementation for qtdemux * 637060 : matroskademux: errors out on 13MB blocks when streaming * 637686 : [jpegenc] Improve sinkpad getcaps results * 638019 : [matroskademux] some matroska files are not specifying DocType * 638072 : build failure: rtpsource.c: error: 'have_rb' may be used uninitialized in this function * 638535 : id3demux: multiple genres as per ID3v2.4 not supported correctly * 638569 : cacasink crashes when given 15-bit video. * 639240 : pulsesink: PLAYING- > PAUSED- > PLAYING transition causes dropout * 639321 : deinterlace: field{1,3} scanline pointers seem to be off by one field line * 639339 : v4l2: fails to build with older kernels due to missing V4L_FIELD_INTERLACED_{TB,BT} * 639516 : muxers: fix setting src pad caps * 639740 : [pulsesink] doesn't uncork in some cases during reverse playback * 640028 : [qtdemux] crash on malformed mov stream * 640063 : rtph264depay: leaks codec data buffer in byte-stream=false mode * 640064 : rtspsrc memory leak * 640080 : rtspsrc: fails to error out properly on network failure * 623063 : [jpegdec] add " max-errors " property 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 * Andoni Morales Alastruey * Andy Wingo * Arun Raghavan * Benjamin Gaignard * Benjamin Otte * Christian Schaller * David Hoyt * David Schleef * Edward Hervey * Erich Schubert * Guillaume Emont * Iain Holmes * Jan Schmidt * Janne Grunau * Johan Dahlin * Kishore Arepalli * Leif Johnson * Marc-Andr? Lureau * Mark Nauwelaerts * Paul Davis * Rob Clark * Ronald S. Bultje * Sebastian Dr?ge * Stefan Kost * Steve Baker * St?phane Loeuillet * Tambet Ingo * Thiago Santos * Thomas Vander Stichele * Tim-Philipp M?ller * Tom Janiszewski * Tristan Matthews * Vincent Penquerc'h * Wim Taymans ? From t.i.m at zen.co.uk Sat Jan 22 12:38:14 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sat, 22 Jan 2011 11:38:14 +0000 Subject: [gst-devel] RELEASE: GStreamer Ugly Plug-ins 0.10.17 "Raised by Wolves" Message-ID: <1295696294.7585.30.camel@zingle> This mail announces the release of GStreamer Ugly Plug-ins 0.10.17 "Raised by Wolves". GStreamer Ugly Plug-ins is a set of plug-ins that have good quality and correct functionality, but distributing them might pose problems. The license on either the plug-ins or the supporting libraries might not be how we'd like. The code might be widely known to present patent problems. Highlights of this release: * mad mp3 decoder: support for reverse playback * dvdreadsrc: fix handling of multi-angle streams * x264enc: work around rate control issue in old versions of x264 For more information, see http://gstreamer.freedesktop.org/modules/gst-plugins-ugly.html To file bugs, request features or submit patches, please go to https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-plugins-ugly Direct links: http://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-0.10.17.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-0.10.17.tar.bz2 MD5 sums (for tarballs downloaded from gstreamer.freedesktop.org): 504ed6508c3a8f035a2002ed69d68506 gst-plugins-ugly-0.10.17.tar.bz2 7df1fb46ce2403426d98cc9db773a31a gst-plugins-ugly-0.10.17.tar.gz Enjoy! -------------- next part -------------- Release notes for GStreamer Ugly Plug-ins?0.10.17 "Raised by Wolves" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer Ugly Plug-ins. The 0.10.x series is a stable series targeted at end users. "When you have to shoot, shoot. Don't talk." There are times when the world needs a color between black and white. Quality code to match the good's, but two-timing, backstabbing and ready to sell your freedom down the river. These plug-ins might have a patent noose around their neck, or a lock-up license, or any other problem that makes you think twice about shipping them. We don't call them ugly because we like them less. Does a mother love her son less because he's not as pretty as the other ones ? No - she commends him on his great personality. These plug-ins are the life of the party. And we'll still step in and set them straight if you report any unacceptable behaviour - because there are two kinds of people in the world, my friend: those with a rope around their neck and the people who do the cutting. This module contains a set of plug-ins that have good quality and correct functionality, but distributing them might pose problems. The license on either the plug-ins or the supporting libraries might not be how we'd like. The code might be widely known to present patent problems. Distributors should check if they want/can ship these plug-ins. Other modules containing plug-ins are: 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-bad contains a set of less supported plug-ins that haven't passed the rigorous quality testing we expect Features of this release * asfdemux: Handle new type of DRM'd asf files * dvdreadsrc: fix handling of multi-angle streams * dvdreadsrc: improve error messages on read errors * mad: if gst_pad_alloc_buffer() returns a buffer with the wrong size allocate a new one * mad: support reverse playback * mpeg2dec: convert the position to stream time before answering to a position query * mpeg2dec: fix position query by trusting upstream * mpeg2dec: refactor cropping code to use libgstvideo functions * mpeg2dec: use gst_pad_alloc_buffer() when cropping buffers * mpegparse: re-fix flow return handling * rmdemux: set GST_BUFFER_FLAG_DELTA_UNIT properly * x264enc: add a note to the docs about encoder latency and queues * x264enc: also accept YV12 input * x264enc: Handle codec/encoder tags * x264enc: Work around a rate control issue in libx264 * x264: use pkg-config to locate libx264 * x264: vbv-buf-capacity should have a minimum of 0 * xingmux: Don't ignore WRONG_STATE and NOT_LINKED when pushing data downstream Bugs fixed in this release * 539254 : [dvdreadsrc] DVDs with multiple angles switch angles during read * 571146 : mpeg2dec: possibly uses wrong strides for 4:2:2 and 4:4:4 YUV with unusual display width or height * 599515 : Fix dvb for DVB API 3.3 * 621465 : [x264enc] Video codec information copied incorrectly to output * 632549 : [mpeg2dec] answers to position queries are wrong for DVDs * 632861 : [x264enc] really bad quality with tune=zerolatency * 634840 : x264: use pkg-config to locate dependencies for plugin * 635291 : x264enc: " vbv-buf-capacity " property should have minimum value of 0 * 635461 : Crash in mad decoder when changing number of audio output channels in Totem while media is running * 636107 : x264enc: docs should indicate specifics of using with queues * 637093 : rmdemux: set GST_BUFFER_FLAG_DELTA_UNIT properly * 637932 : realmedia: do not use the pad buffer allocation functions in demuxers * 639226 : [asfdemux] Doesn't handle new PlayReady DRM files Download You can find source releases of gst-plugins-ugly in the download directory: http://gstreamer.freedesktop.org/src/gst-plugins-ugly/ 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 * Benjamin Gaignard * David Schleef * Edward Hervey * Guillaume Emont * Jan Schmidt * Leo Singer * Mark Nauwelaerts * Rob Clark * Robert Swain * Sebastian Dr?ge * Stefan Kost * Thiago Santos * Thomas Green * Tim-Philipp M?ller * Tristan Matthews * Vincent Penquerc'h * Wim Taymans ? From t.i.m at zen.co.uk Sat Jan 22 12:52:20 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sat, 22 Jan 2011 11:52:20 +0000 Subject: [gst-devel] RELEASE: GStreamer Bad Plug-ins 0.10.21 "Pink Noise" Message-ID: <1295697140.7585.44.camel@zingle> This mail announces the release of GStreamer Bad Plug-ins 0.10.21 "Pink Noise". GStreamer Bad Plugins is a set of plugins that aren't up to par compared to the rest. They might be close to being good quality, but they're missing something - be it a good code review, some documentation, a set of tests, a real live maintainer, or some actual wide use. Highlights of this release: * removed alsaspdifsink element (use alsasink device=spdif) * removed metadata plugin (image tag handling is done differently now) * moved jack plugin to gst-plugins-good (0.10.27) * moved selector to core (0.10.32) * moved valve to core (0.10.32) * new applemedia plugin for Apple multimedia APIs (avfvideosrc, celvideosrc, qtkitvideosrc, miovideosrc, vth264decbin, vth264encbin, vtdec, vtenc) * new y4mdec element * new dcaparse element (dts/dca audio parser) * new mpegaudioparse element (mp3parse replacement) * new camerabin2 element (work-in-progress, experimental, not built by default) * new dvbsuboverlay element to overlay DVB subtitle * new opencv plugin (facedetect, faceblur, edgedetect, cvsobel, cvsmooth, cvlaplace, cverode, cvequalizehist, cvdilate, textwrite, templatematch, pyramidsegment elements) * new interlace element * new rotate element in geometrictransform plugin * new jp2kdecimator element (JPEG2000 decimator, drops details without reencoding) * new autovideoconvert element, an autoconvert based video format converter * new checksumsink and chopmydata debug elements * new rsvgoverlay element for scalable and relative svg overlay with cairo * new ismlmux element, for fragmented ISML mp4 in qtmux plugin * many audio parser improvements (reverse playback, seek tables, index support) * chromahold effect in coloreffects plugin * many other improvements and bug fixes For more information, see http://gstreamer.freedesktop.org/modules/gst-plugins-bad.html To file bugs, request features or submit patches, please go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-plugins-bad Direct links: http://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad-0.10.21.tar.gz http://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad-0.10.21.tar.bz2 MD5 sums (for tarballs downloaded from gstreamer.freedesktop.org): d19af641ab0b9a5c6cbb66ba19c35b16 gst-plugins-bad-0.10.21.tar.gz f501336ab1d18d2565f47c36ce653a82 gst-plugins-bad-0.10.21.tar.bz2 Enjoy! -------------- next part -------------- Release notes for GStreamer Bad Plug-ins?0.10.21 "Pink Noise" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer Bad 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. "That an accusation?" No perfectly groomed moustache or any amount of fine clothing is going to cover up the truth - these plug-ins are Bad with a capital B. They look fine on the outside, and might even appear to get the job done, but at the end of the day they're a black sheep. Without a golden-haired angel to watch over them, they'll probably land in an unmarked grave at the final showdown. Don't bug us about their quality - exercise your Free Software rights, patch up the offender and send us the patch on the fastest steed you can steal from the Confederates. Because you see, in this world, there's two kinds of people, my friend: those with loaded guns and those who dig. You dig. This module contains a set of plug-ins that aren't up to par compared to the rest. They might be close to being good quality, but they're missing something - be it a good code review, some documentation, a set of tests, a real live maintainer, or some actual wide use. If the blanks are filled in they might be upgraded to become part of either gst-plugins-good or gst-plugins-ugly, depending on the other factors. If the plug-ins break, you can't complain - instead, you can fix the problem and send us a patch, or bribe someone into fixing them for you. New contributors can start here for things to work on. Other modules containing plug-ins are: 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 Features of this release * alsaspdif: removed alsaspdifsink element (replaced by alsasink device=spdif) * metadata: remove metadata plugin * jack: jack plugin has moved to gst-plugins-good (0.10.27) * selector: input-selector and output-selector have moved to GStreamer core (0.10.32) * valve: has moved to gstreamer core (0.10.32) * applemedia: new plugin for Apple multimedia APIs (avfvideosrc, celvideosrc, qtkitvideosrc, miovideosrc, vth264decbin, vth264encbin, vtdec, vtenc) * applemedia: new iOS video source based on AVFoundation * y4mdec: new y4mdec element * dcaparse: new dts/dca audio parser * camerabin2: new camerabin element (work-in-progress, experimental) * opencv: new plugin with elements facedetect, faceblur, edgedetect, cvsobel, cvsmooth, cvlaplace, cverode, cvequalizehist, cvdilate, textwrite, templatematch, pyramidsegment * interlace: new element * geometrictransform: new rotate element * jp2kdecimator: add a JPEG2000 decimator element (drop details without reencoding) * audioparsers: add dcaparse, a dts/dca parser, and mpegaudioparse (mp3parse replacement) * autoconvert: add autovideoconvert, an autoconvert based video format converter * checksumsink, chopmydata: new debug elements * dvbsuboverlay: new element to overlay DVB subtitle * rsvgoverlay: new element for scalable and relative svg overlay with cairo * qtmux: add new ismlmux element, for fragmented isml major brand * ac3parse: properly parse E-AC3 frame header and use proper EAC-3 caps * ac3parse: relax BSID checking, performance improvements * applemedia mtapi: update to reflect new API on iOS 4.x * applemedia vtenc: bump H.264 level from 1.3 to 3.0 * applemedia vtenc: remove keyframe enforcement workaround * applemedia celapi: update to reflect new API on iOS 4.x * applemedia cvapi: add wrapper for IO surface access * audioparse: add support for setting the channel-positions * audioparsers: increase ranks to enable auto-plugging (incl. new mp3 parser) * baseparse: enhancements for timestamp marked framed formats * baseparse: increase keyframe awareness * baseparse: perform bitrate handling and posting after newsegment sending * baseparse: post duration message if average bitrates is updated * baseparse: prevent indefinite resyncing * baseparse: add index support, seek table and accurate seek support * baseparse: support reverse playback * baseparse: use determined seekability in answering SEEKING query * basevideo: Add handling of GstForceKeyUnit events * basevideodecoder: add capture pattern handling; don't blow away field information * bayer2rgb: add format=bggr/etc. to caps, add framerate to the sink caps * camerabin: add "preview-source-filter","ready-for-capture", "idle" properties * camerabin: change "zoom" property from int to float * camerabin: enable all conversion flags by default to make it work out-of-the-box everywhere * coloreffects: Add chromahold effect * cog: improvements in colorspace and scaler; add fast paths for colorspace conversion * colorspace: revive element and add support for many more pixellayouts/colorspaces * colorspace: add support for SDTV/HDTV YUV conversions * dtmfsrc: Make the dtmfsrc accept events sent with gst_element_send_event * tools: misc. improvements to element-maker tool * faac: handle trailing encoder output * faad: support reverse playback; cater for decoder delay and renegotiation * faad: tweak output buffer timestamping * flacparse: don't drop the last frame if it is followed by garbage * flacparse: don't parse the complete FLAC frames but only look for valid frame headers (for performance) * flacparse: fix picture parsing, fix parsing with unknown framesizes * flacparse: parse seektable * frei0r: add support for the FREI0R_PATH environment variable * frei0r: fix crashes if plugins don't provide correct property information * frei0r: fix scanning of plugin subdirectories and support different vendors * frei0r: update frei0r interface specification to 1.2 * gaudieffects: avoid divide by 0 in burn element, make filter parameters dynamic and controllable * id3mux: map GST_TAG_ALBUM_ARTIST, give PRIMARY rank * invtelecine: Fix name of 30p/60i pattern * jasperdec: don't fail hard on decoding error * jifmux: detect EOI correctly; do not limit the size of the image on 16bits * jp2kenc: Emit SOP markers in every codestream packet * jpegparse: avoid infinite loop when resyncing; discard incomplete image * kate: add segment tracking, and various other improvements * kate: ensure the kate pad does not shoot ahead of the video pad * mpegtsdemux: extract language for DVB subtitle streams * mpegtsdemux: enable gather_pes only for DVB subtitle private streams * mpegtsdemux: fix re-syncing on invalid data after seek * mpegtsmux: Initialize PES packet before getting the header size * mpegtsmux: Set adaptation flag when appropriate * mpegtsmux: Set random_access_indicator for keyframes * mpegtsparse: send TDT tables messages in a serialized event downstream * ofa: Call g_object_notify() after the fingerprint was created * pcapparse: Add support for Linux Cooked Capture (SLL) frames * photography: add missing property and cabability flag for noise reduction * photography: Add "zoom" and "image-preview-supported-caps" interface properties * photography: add gst_photography_{set,get}_noise_reduction() and CAPS_NOISE_REDUCTION flag * qtmux: add support for fragmented mp4 * qtmux: add "streamable" property to avoid building fragmented mfra index * qtmux: timestamp tracking fixes and many other fixes * resindvd: attempt to use glib language setting for DVD menus/audio * resindvd: improve error messages on read errors; button state tracking fixes * rfbsrc: fail more gracefully if source gets disconnected or geometry changes * sdlvideosink: re-enable YVYU and UYVY support * sdpdemux: error out if no streams found in sdp message * sdpdemux: redirect SDP with an rtsp control URL and add property to disable redirect * ssim: add I420 support * tiger: outline text by default, to make it easier to read * winks: add property probing support; fix framerate fraction range mapping * winks: ignore unsupported formats; work around shutdown deadlock * winks: performance improvements * zbar: make scanner cache optional, disable it by default * zbar: use correct strides, support more formats Bugs fixed in this release * 628609 : The qtwrapperaudiodec_samr decoder doesn't handle buffers containing many AMR frames properly * 639296 : [y4mdec] Doesn't handle files which don't specify a colorspace * 613379 : camerabin: Do not use audio clock after stopping video capture * 636279 : REGRESSION: Video often freezes during playback of mpeg2 files * 630255 : [examples] camerabin example still uses old GstXOverlay interface * 631232 : [colorspace] AYUV/ARGB handling broken on big endian systems * 627229 : fpsdisplaysink should not measure fps relative to pipeline clock * 435120 : cairosvgoverlay * 486659 : xmp/exif metadata handling * 578629 : libgstphotography missing exports for MSVC * 587223 : mpegtsdemux seg.fault due to invalid PMT_pid * 598078 : osxvideosrc doesn't build in snow leopard x86_64 * 600929 : [kate] tiger element doesn't handle segments and text/video synchronization properly * 603063 : camerabin example fails to start recording * 611428 : [gdiscreencapsrc] leaks memory (ICONINFO) * 613633 : [resindvd] scrambled DVDs yield useless generic error message if dvdcss is not available * 616814 : Photography interface extension: colour tone mode and noise reduction settings * 616923 : camerabin: remove photography interface implementation * 618045 : [cogcolorspace] No Y41B support * 618542 : DVB subtitles support * 625558 : [basevideoencoder] [vp8] encoder timestamps are wrong when there are gaps * 626425 : cog_virt_frame_new_convert_u8 has bogus source data * 627134 : photography interface: add API for capture correction * 627211 : jpegformat: Push tags after setting srcpad caps * 627253 : [mpegtsparse] Post tags of channel and currently running event * 627992 : dtmfsrc doesn't support gst_element_send_event * 628326 : vdpau: don't change structure in setcaps function * 628527 : videoanalyse: classificication is wrong * 628548 : [mpegtsmux] Initialize PES packet before getting the header size * 628570 : cogcolorspace: element classificication is wrong * 629554 : dvbsrc: Fix example usage, bandwidth=8 not 8MHz * 629897 : [cog] Ignores --disable-orc and always requires orc * 629910 : jpegparse: properly clean up comment string * 629917 : [output-selector] Recheck pending_pad after pushing a buffer * 630046 : sdpdemux: Add optional support for rtspsrc as session element * 630253 : [sdl] Still uses old GstXOverlay interface * 630254 : [vdpau] Still uses old GstXOverlay interface * 630783 : [frei0r] Crashes if broken plugins don't give correct property information * 630808 : valve: move to core * 631200 : flacparse: major performance improvements * 631389 : [flacparse] backport/check corner case fixes done in flacdec * 631449 : [audioparse] doest not support several channel numbers * 631501 : [faad] failed to dynamically switch from 2 audio channels to 6 * 631814 : [flacparse] unit test failures * 631993 : [flacparse] imperfect timestamping * 632070 : qtmux: infinite loop * 632668 : Gaudi Effects [review] * 632885 : Gaudi Effects dynamically controllable parameters [review] * 632911 : qtmux: add fragmented mp4 support (isml brand) * 633466 : [katedec] Pushes events before the final caps are known * 633917 : [mpegtsparse] [PATCH] Send TDT messages in an serialized event downstream * 635200 : [dvbbasebin] [PATCH] Add TDT to the initial pids filter for dvbsrc * 635202 : mpeg4videoparse: Minor fixes * 635204 : mpeg4videoparse: Set srcpad caps before forwarding newsegment * 635205 : h264parse: Set srcpad caps before forwarding newsegment * 635229 : celtenc: uninitialized tags variable can cause segfault * 635304 : [opencv] fix caps issues and extend supported caps for some elements [PATCH] * 635397 : rfbsrc: avoid infinite loop if source gets disconnected and don't crash if frame geometry changes * 635529 : interlace: Add pattern offset and fix timestamps * 635720 : vp8enc incorrectly sets timestamps based on theoretical framerate * 635786 : [audioparse] Fix division-by-zero exception * 635878 : [qtmux] gst_qtmux_check_difference subtract 2 unsigned numbers without taking care of the result sign * 636106 : autocolorspace: new plugin for auto space convertor selection * 636109 : [SSIM] klass should be Filter/Effect/Video * 636185 : qtmux: msvc incompatibility * 637062 : [ac3parse] parse problems on some MTS streams * 637224 : [bayer2rgb] missing framerate in sink caps * 637308 : gst-plugins-bad did not find xvidcore on my box * 637359 : Internal GStreamer error, during pcap to mp4 conversion * 637403 : qtmux do not store 1st TS when detect 1st buffer * 637486 : qtmux: error if no buffers have arrived to one pad at EOS * 637532 : applemedia: redundant declaration of 'parent_class' * 637590 : [PATCH] fix gst-plugins-bad compile against latest gtk+ * 637824 : mpeg4videoparse: gst_buffer_is_metadata_writable warning (regression) * 637929 : mve: do not use the pad buffer allocation functions in demuxers * 637931 : mpegdemux: do not use the pad buffer allocation functions in demuxers * 638004 : tiger: fallback on headers in caps to initialize if headers are absent * 638288 : qtmux: fails to handle out-of-order buffers without duration * 638412 : kate: reenable the sending of a message for tags * 638527 : tiger: outline text by default, to make it easier to read * 638604 : basecamerasrc isn't build by default but camerabin2 is and requires it * 639063 : mpegtsparse: fix (re)sync with invalid data at beginning * 639338 : [qtmux] Protect against copying a null caps * 639413 : Camerabin should use output-selector:pad-negotiation-mode=active * 639456 : [camerabin] Should have all conversion flags enabled by default * 639555 : [audioparsers] Be careful to not lose the event ref * 639950 : flacparse: avoid unref'ing NULL buffer * 630830 : zbar: Fixes, single frame scan and width/stride fix * 635281 : [mpegtsparse] TDT packets are only parsed for odd hours Download You can find source releases of gst-plugins-bad in the download directory: http://gstreamer.freedesktop.org/src/gst-plugins-bad/ 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 * Alejandro Gonzalez * Aleksey Lim * Alessandro Decina * Andoni Morales Alastruey * Andres Colubri * Andr? Dieb Martins * Arun Raghavan * Benjamin Gaignard * Carl-Anton Ingmarsson * Christian Berentsen * Damien Lespiau * David Hoyt * David Schleef * Edward Hervey * Felipe Contreras * Francis Rammeloo * Hoseok Chang * Hu Gang * Jan Schmidt * Janne Grunau * Jonathan Rosser * Josh Doe * Kaj-Michael Lang * Karol Sobczak * Knut Inge Hvidsten * Lasse Laukkanen * Leo Singer * Luciana Fujii Pontello * Luis de Bethencourt * Marc-Andr? Lureau * Mark Nauwelaerts * Mart Raudsepp * Matthew Ife * Mike Sheldon * Noam * Ole Andr? Vadla Ravn?s * Olivier Aubert * Olivier Cr?te * Rob Clark * Robert Swain * Saleem Abdulrasool * Sebastian Dr?ge * Sebastian P?lsterl * Sreerenj Balachandran * Stefan Kost * Teemu Katajisto * Thiago Santos * Thibault Saunier * Thijs Vermeir * Tim-Philipp M?ller * Tristan Matthews * Vincent Penquerc'h * V?ctor Manuel J?quez Leal * Wim Taymans * Youness Alaoui * benjamin gaignard * kapil ? From ensonic at hora-obscura.de Sat Jan 22 19:27:14 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 22 Jan 2011 20:27:14 +0200 Subject: [gst-devel] gst_parse_bin_from_description and pad creation In-Reply-To: References: Message-ID: <4D3B2182.7080500@hora-obscura.de> Am 22.01.2011 02:46, schrieb laura la: > Hi All, > > I'm trying to use rsvgoverlay plugin from gst-plugins-bad, it overlays > a .svg file above a video. It has two sinks, "data_sink" for the .svg > data, and "sink" for video data. > > The gst-launch command works fine: > > gst-launch v4l2src device=/dev/video0 ! ffmpegcolorspace ! rsvgoverlay > location=/usr/share/icons/gnome/scalable/places/gnome-foot.svg ! > ffmpegcolorspace ! videoscale ! video/x-raw-yuv,width=640,height=480 ! > ffmpegcolorspace ! ximagesink > > or > > gst-launch -v videotestsrc ! ffmpegcolorspace ! rsvgoverlay > location=foo.svg ! ffmpegcolorspace ! autovideosink > > Here are a couple more examples in the plugin code itself: > > http://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/ext/rsvg/gstrsvgoverlay.c > > I need to create the GstElement using gst_parse_bin_from_description: > > GstElement * overlay = gst_parse_bin_from_description ("rsvgoverlay > location=foo.svg", TRUE, NULL); > > and then link it to other elements of the stream. This isn't working. > > Here is a simplified code snippet with working and not working ways to > create the GstElement. http://pastebin.com/uiqDZR1Y > > This works: > overlay = gst_element_factory_make ("rsvgoverlay", "overlay"); > g_object_set (overlay, "location", "foo.svg", NULL); > This code is much more recommended for applications as its more directly expression what you like to do. gst_parse_bin_from_description() might struggle with the two sinj-pads. Stefan > But I using gst_parse_bin_from_description I get errors about > negotiating format. I think during linking it's trying to use > "data_sink" instead of "sink" in rsvgoverlay element. But I'm not 100% > sure. > > I tried variations of the pipeline description, for example > ffmpegcolorspace ! rsvgoverlay location=foo.svg ! ffmpegcolorspace but > it doesn't work either. > > Here is a GST_DEBUG=4 log file in case you want to take a look: > http://ubuntuone.com/p/Z4L > > I'm a participant in the GNOME Outreach program for Women 2010, I'm > working on Cheese application, and the idea is to add funny .svg hats > that follow you around with face recognition. See more here: > http://programmermusings.blogspot.com/ > http://projects.gnome.org/cheese/ > > Bye! > Laura > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From gibrovacco at gmail.com Sat Jan 22 21:09:20 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sat, 22 Jan 2011 22:09:20 +0200 Subject: [gst-devel] kenburns plugin update - now a general video transform plugin In-Reply-To: <4D35BCE9.9000506@brooks.nu> References: <4D35BCE9.9000506@brooks.nu> Message-ID: Hi, On Tue, Jan 18, 2011 at 6:16 PM, Lane Brooks wrote: ..snip.. > Given that the feature set has grown beyond Ken Burns effects, I want to > rename this element to something else. I was thinking 'videotransform' > but am open to other suggestions. What about videoeffects? It probably explains better what's the target use of the component. An alternative could be to split the component in several functional elements sharing big portions of code (more or less like qtdemux is doing) and name them something like "rotation", "transition", etc. Regards From gibrovacco at gmail.com Sat Jan 22 21:30:44 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sat, 22 Jan 2011 22:30:44 +0200 Subject: [gst-devel] Timestamping a video file with gstreamer In-Reply-To: References: Message-ID: Hi, late reply, I hope it will help anyway.. On Tue, Jan 18, 2011 at 12:02 PM, jeetu.golani at gmail.com wrote: > Hello, > > First, I'm new to gstreamer and video manipulation and file formats in > general, therefore forgive me if what I ask is obvious :) > > I would like to timestamp a video file sort of like how the timeoverlay and > clockoverlay elements do. However, I would like the time to begin from the > time the file was recorded. I'm using MythTV to record video and upon > recording I would like the file (or a clip) to be transcoded with the > timestamp - am open to having the final recording in any of the popular > container formats. My questions: > > - Is there a way for me to overlay a user defined time? Can I somehow > manipulate the buffer or using some means introduce a time into the stream > using the timeoverlay/clockoverlay elements? >From what I see in timeoverlay and clockoverlay sources, you'd need to hack the elements in order to add an initial offset. If you make a property out of it, there could be some interest for upstream. As an alternative, you can use a textoverlay and a timer in your application which makes it change the content (for instance) each second to the time you prefer. > - Can I somehow introduce a timestamp in the file container itself and then > this gets used as the start time? It looks really hackish and results may be not what you really want when you try to play back the file again. Regards > > Google has so far not been able to provide me with my answers (or maybe I've > been looking in the wrong place). There is mention of a patch for > mencoder/mplayer that provided this functionality but that's obsolete now. > Same for ffmpeg that provided something similar using a vhook plugin. > > I would sincerely appreciate your help on this. I'm definitely on my wits > end :) > > Thanks > > Bye for now > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 arnout at mind.be Sat Jan 22 21:10:44 2011 From: arnout at mind.be (Arnout Vandecappelle) Date: Sat, 22 Jan 2011 21:10:44 +0100 Subject: [gst-devel] HTTP video source serves individual JPEG images In-Reply-To: <4D21A711.1060504@ingelabs.com> References: <4D21A711.1060504@ingelabs.com> Message-ID: <201101222110.45472.arnout@mind.be> On Monday 03 January 2011 11:38:09, Guillermo Rodriguez Garcia wrote: > I am trying to use gstreamer to stream video (sort of) from a HTTP source > that serves individual JPEG images. This is not a MJPEG stream -- instead, > a new HTTP request is required for each frame that is to be retrieved. You'd need to modify souphttpsrc to support this. We wouldn't want to have the souphttpsrc element reconnect by default after the video ends, of course, so it should be indicated with a property. Implementing this is unfortunately non-trivial, because of the multithread behaviour of souphttpsrc. In the gst_soup_http_src_got_body_cb() and gst_soup_http_src_finished_cb(), you need to re-send the message rather than sending EOS. You can try to set src->read_position to 0 and src->retry to TRUE, but I'm afraid it won't be that simple... Regards, Arnout -- Arnout Vandecappelle arnout at mind be Senior Embedded Software Architect +32-16-286540 Essensium/Mind http://www.mind.be G.Geenslaan 9, 3001 Leuven, Belgium BE 872 984 063 RPR Leuven LinkedIn profile: http://www.linkedin.com/in/arnoutvandecappelle GPG fingerprint: 31BB CF53 8660 6F88 345D 54CC A836 5879 20D7 CF43 -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Sat Jan 22 21:55:18 2011 From: wim.taymans at gmail.com (Wim Taymans) Date: Sat, 22 Jan 2011 21:55:18 +0100 Subject: [gst-devel] HTTP video source serves individual JPEG images In-Reply-To: <4D21A711.1060504@ingelabs.com> References: <4D21A711.1060504@ingelabs.com> Message-ID: For those sources I would suggest to periodically set the pipeline to NULL when you get an EOS event. When you want to get a new image, set it back to PLAYING, and then you repeat this. Wim On Mon, Jan 3, 2011 at 11:38 AM, Guillermo Rodriguez Garcia wrote: > Hello all, > > I am trying to use gstreamer to stream video (sort of) from a HTTP source > that serves individual JPEG images. This is not a MJPEG stream -- instead, > a new HTTP request is required for each frame that is to be retrieved. > > This pipeline works for MJPEG: > > gst-launch souphttpsrc location=http://user:pass at hostname/path ! jpegdec ! xvimagesink > > If I try to use the same pipeline for an HTTP source that serves individual > JPEG images, only one image is shown (since the HTTP response actually > contains just one single image). > > Is there any way to have souphttpsrc make additional requests periodically > from the specified location? > > Is there any other way to achieve this? > > Thanks, > > GRodriguez > > > ------------------------------------------------------------------------------ > 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 arnout at mind.be Sat Jan 22 22:00:45 2011 From: arnout at mind.be (Arnout Vandecappelle) Date: Sat, 22 Jan 2011 22:00:45 +0100 Subject: [gst-devel] HTTP video source serves individual JPEG images In-Reply-To: References: <4D21A711.1060504@ingelabs.com> Message-ID: <201101222200.46427.arnout@mind.be> Good idea, but wouldn't that wreak havoc on the output file in case it's a filesink instead of an xvimagesink? Regards, Arnout On Saturday 22 January 2011 21:55:18, Wim Taymans wrote: > For those sources I would suggest to periodically set the pipeline to > NULL when you > get an EOS event. When you want to get a new image, set it back to PLAYING, > and then you repeat this. > > Wim > > On Mon, Jan 3, 2011 at 11:38 AM, Guillermo Rodriguez Garcia > > wrote: > > Hello all, > > > > I am trying to use gstreamer to stream video (sort of) from a HTTP source > > that serves individual JPEG images. This is not a MJPEG stream -- > > instead, a new HTTP request is required for each frame that is to be > > retrieved. > > > > This pipeline works for MJPEG: > > > > gst-launch souphttpsrc location=http://user:pass at hostname/path ! jpegdec > > ! xvimagesink > > > > If I try to use the same pipeline for an HTTP source that serves > > individual JPEG images, only one image is shown (since the HTTP response > > actually contains just one single image). > > > > Is there any way to have souphttpsrc make additional requests > > periodically from the specified location? > > > > Is there any other way to achieve this? > > > > Thanks, > > > > GRodriguez > > > > > > ------------------------------------------------------------------------- > > ----- 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 > > --------------------------------------------------------------------------- > --- Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better > price-free! Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Arnout Vandecappelle arnout at mind be Senior Embedded Software Architect +32-16-286540 Essensium/Mind http://www.mind.be G.Geenslaan 9, 3001 Leuven, Belgium BE 872 984 063 RPR Leuven LinkedIn profile: http://www.linkedin.com/in/arnoutvandecappelle GPG fingerprint: 31BB CF53 8660 6F88 345D 54CC A836 5879 20D7 CF43 -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Sat Jan 22 22:04:40 2011 From: wim.taymans at gmail.com (Wim Taymans) Date: Sat, 22 Jan 2011 22:04:40 +0100 Subject: [gst-devel] HTTP video source serves individual JPEG images In-Reply-To: <201101222200.46427.arnout@mind.be> References: <4D21A711.1060504@ingelabs.com> <201101222200.46427.arnout@mind.be> Message-ID: I think the file would just be overwritten with new contents because the filesink would reopen (not in append mode) the file. Wim On Sat, Jan 22, 2011 at 10:00 PM, Arnout Vandecappelle wrote: > Good idea, but wouldn't that wreak havoc on the output file in case it's a > filesink instead of an xvimagesink? > > Regards, > > Arnout > > On Saturday 22 January 2011 21:55:18, Wim Taymans wrote: > >> For those sources I would suggest to periodically set the pipeline to > >> NULL when you > >> get an EOS event. When you want to get a new image, set it back to >> PLAYING, > >> and then you repeat this. > >> > >> Wim > >> > >> On Mon, Jan 3, 2011 at 11:38 AM, Guillermo Rodriguez Garcia > >> > >> wrote: > >> > Hello all, > >> > > >> > I am trying to use gstreamer to stream video (sort of) from a HTTP >> > source > >> > that serves individual JPEG images. This is not a MJPEG stream -- > >> > instead, a new HTTP request is required for each frame that is to be > >> > retrieved. > >> > > >> > This pipeline works for MJPEG: > >> > > >> > gst-launch souphttpsrc location=http://user:pass at hostname/path ! jpegdec > >> > ! xvimagesink > >> > > >> > If I try to use the same pipeline for an HTTP source that serves > >> > individual JPEG images, only one image is shown (since the HTTP response > >> > actually contains just one single image). > >> > > >> > Is there any way to have souphttpsrc make additional requests > >> > periodically from the specified location? > >> > > >> > Is there any other way to achieve this? > >> > > >> > Thanks, > >> > > >> > GRodriguez > >> > > >> > > >> > >> > ------------------------------------------------------------------------- > >> > ----- 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 > >> > >> >> --------------------------------------------------------------------------- > >> --- Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > >> Finally, a world-class log management solution at an even better > >> price-free! Download using promo code Free_Logger_4_Dev2Dev. Offer expires > >> February 28th, so secure your free ArcSight Logger TODAY! > >> http://p.sf.net/sfu/arcsight-sfd2d > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- > > Arnout Vandecappelle arnout at mind be > > Senior Embedded Software Architect +32-16-286540 > > Essensium/Mind http://www.mind.be > > G.Geenslaan 9, 3001 Leuven, Belgium BE 872 984 063 RPR Leuven > > LinkedIn profile: http://www.linkedin.com/in/arnoutvandecappelle > > GPG fingerprint: 31BB CF53 8660 6F88 345D 54CC A836 5879 20D7 CF43 > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From wim.taymans at gmail.com Sat Jan 22 22:09:46 2011 From: wim.taymans at gmail.com (Wim Taymans) Date: Sat, 22 Jan 2011 22:09:46 +0100 Subject: [gst-devel] HTTP video source serves individual JPEG images In-Reply-To: References: <4D21A711.1060504@ingelabs.com> <201101222200.46427.arnout@mind.be> Message-ID: You could also wrap the source and decoder in a bin, patch up a buffer timestamp, catch the EOS event, set things to NULL internally and schedule a timeout for the next frame. You could then more easily mux things. Wim On Sat, Jan 22, 2011 at 10:04 PM, Wim Taymans wrote: > I think the file would just be overwritten with new contents because > the filesink > would reopen (not in append mode) the file. > > Wim > > On Sat, Jan 22, 2011 at 10:00 PM, Arnout Vandecappelle wrote: >> Good idea, but wouldn't that wreak havoc on the output file in case it's a >> filesink instead of an xvimagesink? >> >> Regards, >> >> Arnout >> >> On Saturday 22 January 2011 21:55:18, Wim Taymans wrote: >> >>> For those sources I would suggest to periodically set the pipeline to >> >>> NULL when you >> >>> get an EOS event. When you want to get a new image, set it back to >>> PLAYING, >> >>> and then you repeat this. >> >>> >> >>> Wim >> >>> >> >>> On Mon, Jan 3, 2011 at 11:38 AM, Guillermo Rodriguez Garcia >> >>> >> >>> wrote: >> >>> > Hello all, >> >>> > >> >>> > I am trying to use gstreamer to stream video (sort of) from a HTTP >>> > source >> >>> > that serves individual JPEG images. This is not a MJPEG stream -- >> >>> > instead, a new HTTP request is required for each frame that is to be >> >>> > retrieved. >> >>> > >> >>> > This pipeline works for MJPEG: >> >>> > >> >>> > gst-launch souphttpsrc location=http://user:pass at hostname/path ! jpegdec >> >>> > ! xvimagesink >> >>> > >> >>> > If I try to use the same pipeline for an HTTP source that serves >> >>> > individual JPEG images, only one image is shown (since the HTTP response >> >>> > actually contains just one single image). >> >>> > >> >>> > Is there any way to have souphttpsrc make additional requests >> >>> > periodically from the specified location? >> >>> > >> >>> > Is there any other way to achieve this? >> >>> > >> >>> > Thanks, >> >>> > >> >>> > GRodriguez >> >>> > >> >>> > >> >>> > >>> > ------------------------------------------------------------------------- >> >>> > ----- 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 >> >>> >> >>> >>> --------------------------------------------------------------------------- >> >>> --- Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> >>> Finally, a world-class log management solution at an even better >> >>> price-free! Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> >>> February 28th, so secure your free ArcSight Logger TODAY! >> >>> http://p.sf.net/sfu/arcsight-sfd2d >> >>> _______________________________________________ >> >>> gstreamer-devel mailing list >> >>> gstreamer-devel at lists.sourceforge.net >> >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> -- >> >> Arnout Vandecappelle arnout at mind be >> >> Senior Embedded Software Architect +32-16-286540 >> >> Essensium/Mind http://www.mind.be >> >> G.Geenslaan 9, 3001 Leuven, Belgium BE 872 984 063 RPR Leuven >> >> LinkedIn profile: http://www.linkedin.com/in/arnoutvandecappelle >> >> GPG fingerprint: 31BB CF53 8660 6F88 345D 54CC A836 5879 20D7 CF43 >> >> ------------------------------------------------------------------------------ >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> Finally, a world-class log management solution at an even better price-free! >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> February 28th, so secure your free ArcSight Logger TODAY! >> http://p.sf.net/sfu/arcsight-sfd2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > From jeetu.golani at gmail.com Sun Jan 23 03:59:36 2011 From: jeetu.golani at gmail.com (jeetu.golani at gmail.com) Date: Sun, 23 Jan 2011 08:29:36 +0530 Subject: [gst-devel] Timestamping a video file with gstreamer In-Reply-To: References: Message-ID: Hi Marco, Thanks so much for the reply. For now I'm using a non-gstreamer workaround by using the subtitle approach to solve my problem since I needed something quick. I still think that your suggestion of reworking timeoverlay and adding a property will probably be better and may even help get a frame level based accuracy (?) If I do get a chance I'll defi look at the code and see what it takes to put in the functionality :) Thanks again :) Bye for now On Sun, Jan 23, 2011 at 2:00 AM, Marco Ballesio wrote: > Hi, > > late reply, I hope it will help anyway.. > > On Tue, Jan 18, 2011 at 12:02 PM, jeetu.golani at gmail.com > wrote: >> Hello, >> >> First, I'm new to gstreamer and video manipulation and file formats in >> general, therefore forgive me if what I ask is obvious :) >> >> I would like to timestamp a video file sort of like how the timeoverlay and >> clockoverlay elements do. However, I would like the time to begin from the >> time the file was recorded. I'm using MythTV to record video and upon >> recording I would like the file (or a clip) to be transcoded with the >> timestamp - am open to having the final recording in any of the popular >> container formats. My questions: >> >> - Is there a way for me to overlay a user defined time? Can I somehow >> manipulate the buffer or using some means introduce a time into the stream >> using the timeoverlay/clockoverlay elements? > > >From what I see in timeoverlay and clockoverlay sources, you'd need to > hack the elements in order to add an initial offset. If you make a > property out of it, there could be some interest for upstream. > > As an alternative, you can use a textoverlay and a timer in your > application which makes it change the content (for instance) each > second to the time you prefer. > >> - Can I somehow introduce a timestamp in the file container itself and then >> this gets used as the start time? > > It looks really hackish and results may be not what you really want > when you try to play back the file again. > > Regards > >> >> Google has so far not been able to provide me with my answers (or maybe I've >> been looking in the wrong place). There is mention of a patch for >> mencoder/mplayer that provided this functionality but that's obsolete now. >> Same for ffmpeg that provided something similar using a vhook plugin. >> >> I would sincerely appreciate your help on this. I'm definitely on my wits >> end :) >> >> Thanks >> >> Bye for now >> >> >> ------------------------------------------------------------------------------ >> Protect Your Site and Customers from Malware Attacks >> Learn about various malware tactics and how to avoid them. Understand >> malware threats, the impact they can have on your business, and how you >> can protect your company and customers by using code signing. >> 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 >> >> > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From bilboed at gmail.com Sun Jan 23 08:17:05 2011 From: bilboed at gmail.com (Edward Hervey) Date: Sun, 23 Jan 2011 08:17:05 +0100 Subject: [gst-devel] how do I packet pace rtp network traffic? In-Reply-To: <8E459EED6B356D469071FDC814554FC43FF356C3@COR1MBX01.emea.tsp.ad> References: <8E459EED6B356D469071FDC814554FC43FF356C3@COR1MBX01.emea.tsp.ad> Message-ID: <1295767025.2471.1.camel@deumeu> On Thu, 2011-01-20 at 09:54 +0000, Redfern, Keith wrote: > We are developing a gstreamer based pipeline to read a video source, > rtp payload the video frame and then udpsink the rtp packets. > > We are finding that the pipeline bursts the rtp data unto the network, > creating network spikes. In one example, the interpacket network delay > is about 15micro seconds for approx 150 packets, meaning the whole > video frame is delivered unto the network in 2.25ms. > > Is there some way in which to ?pace? the rtp packets so that it takes > 20-30ms to send the same frame? You can pace (or throttle) the time at which the rtp packets are sent by adjusting the outgoing timestamps of all buffers (i.e. the ones coming out of the payloaders and going into the udpsinks). The packets without timestamps will be sent as soon as they are received. Wim is also working on a way to throttle that based on buffer(list?) duration also fwiw. Edward > > > > Keith > > > > > > > ______________________________________________________________________ > Tyco Safety Products/CEM Systems Ltd. > Registered in Northern Ireland: NI 25728. Registered Office: 195 > Airport Road West, Belfast, BT3 9ED. > > Please note that any views or opinions presented in this email are > solely those of the author and do not necessarily represent those of > the company. This email and any attachments are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. If you have received this email in error please notify > the sender immediately and delete any copies in your possession. > > > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 Sun Jan 23 09:21:07 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sun, 23 Jan 2011 10:21:07 +0200 Subject: [gst-devel] What is 'stream time of the segment'? In-Reply-To: <1295273625981-3221217.post@n4.nabble.com> References: <1295273625981-3221217.post@n4.nabble.com> Message-ID: Hi, On Mon, Jan 17, 2011 at 4:13 PM, wl2776 wrote: > > Unfortunately, the article in the GTK-docs about GstSegment doesn't reveal > too much details about GstSegment::time field. > It says that it is the 'stream time of the segment'. > part-segments.txt on the gstreamer/dosc/design/ doesn't mention it also. > part-synchronization.txt says that the "stream time is also known as the > position in the stream and is a value between 0 and the total duration of > the media file". > > So, could anyone clarify if that definition of the stream time applies to > the segment stream time? I tend to agree with this definition. To rephrase what's in the documentation, the "time" field in the GstSegment structure can be seen as the position in the stream at which the segment takes place. A better description on hot the start and time fields are to be interpreted can be found at: http://www.gstreamer.net/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-new-segment-full Regards > > If I am right, then GstSegment::time should be a moment of the segment > start, provided stream playback starts from 0. > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/What-is-stream-time-of-the-segment-tp3221217p3221217.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Protect Your Site and Customers from Malware Attacks > Learn about various malware tactics and how to avoid them. Understand > malware threats, the impact they can have on your business, and how you > can protect your company and customers by using code signing. > 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 bilboed at gmail.com Mon Jan 24 11:48:01 2011 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 24 Jan 2011 11:48:01 +0100 Subject: [gst-devel] RELEASE: GStreamer Editing Services 0.10.1 "This is Scandinavian one-o-one" Message-ID: <1295866081.3821.3.camel@localhost> This mail announces the release of GStreamer Editing Services 0.10.1 "This is Scandinavian one-o-one". The GStreamer Editing Services is a high-level library to assist in creating non-linear audio/video editors. Highlights of this release: * Highly portable and lightweight, written in C/GObject * High-level API for simple applications * Full-depth API for advanced applications * gobject-introspection support for easy language bindings creation * Audio/Video/Image file support * Audio/Video Transitions * Title/Credits * Title overlay * Saving/Loading of timelines to custom file format * Playback/Rendering of timelines with convenience GstPipeline To file bugs, request features or submit patches, please go to https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-editing-services Direct links: http://gstreamer.freedesktop.org/src/gst-editing-services/gstreamer-editing-services-0.10.1.tar.bz2 http://gstreamer.freedesktop.org/src/gst-editing-services/gstreamer-editing-services-0.10.1.tar.gz Enjoy! -------------- next part -------------- Release notes for GStreamer Editing Services?0.10.1 "This is Scandinavian one-o-one" The GStreamer team is proud to announce the first release in the 0.10.x stable series of the GStreamer Editing Services. Features of this release * Highly portable and lightweight, written in C/GObject * High-level API for simple applications * Full-depth API for advanced applications * gobject-introspection support for easy language bindings creation * Audio/Video/Image file support * Audio/Video Transitions * Title/Credits * Title overlay * Saving/Loading of timelines to custom file format * Playback/Rendering of timelines with convenience GstPipeline * Bugs fixed in this release * 639218 : Exports private symbols * 639219 : ges-internal.h shouldn't be installed * 639222 : Rename ges-launch to ges-launch-0.10 Download You can find source releases of gst-editing-services in the download directory: http://gstreamer.freedesktop.org/src/gst-editing-services/ 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 * Brandon Lewis * Edward Hervey * Jarkko Pallviainen * Mark Nauwelaerts * Sebastian Dr?ge * Stefan Kost * Thiago Santos * Thibault Saunier * Tim-Philipp M?ller ? From bilboed at gmail.com Mon Jan 24 12:01:12 2011 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 24 Jan 2011 12:01:12 +0100 Subject: [gst-devel] RELEASE: GNonLin non-linear editing plugins 0.10.17 "Flight from Hawaii coming out of the sun" Message-ID: <1295866872.3821.5.camel@localhost> This mail announces the release of the GnonLin non-linear editing plugins 0.10.17 "Flight from Hawaii coming out of the sun" This module contains a set of plug-ins for GStreamer to ease the creation of multimedia editors, or any other application where a timeline-oriented use of GStreamer makes sense. For more information, see http://gstreamer.freedesktop.org/modules/gnonlin.html To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gnonlin -------------- next part -------------- Release notes for GNonLin Non-Linear Editing Plug-ins?0.10.17 "Flight from Hawaii coming out of the sun" The GStreamer team is proud to announce a new release in the 0.10.x stable series of GNonLin. This module contains a set of plug-ins for GStreamer to ease the creation of multimedia editors, or any other application where a timeline-oriented use of GStreamer makes sense. These elements include: gnlsource An element for using source elements/bins in a GnlComposition gnlfilesource A higher-level element for using a uri in a GnlComposition gnlcomposition A container element that handles GNonLin objects gnloperation An element for using filters in a GnlComposition Features of this release * documentation and debugging fixes * Use glib 2.22 API if available Bugs fixed in this release * 628943 : make check fails if videomixer element can't be found * 633721 : tests: gnl/simple: test_one_under_another failure / timeout Download You can find source releases of gnonlin in the download directory: http://gstreamer.freedesktop.org/src/gnonlin/ 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 GNonLin is primarily used by PiTiVi (http://www.pitivi.org/) and Jokosher (http://www.jokosher.org/). Contributors to this release * David Schleef * Edward Hervey * Jan Schmidt * Sebastian Dr?ge * Stefan Kost * Tim-Philipp M?ller ? From bilboed at gmail.com Mon Jan 24 12:04:47 2011 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 24 Jan 2011 12:04:47 +0100 Subject: [gst-devel] RELEASE: GStreamer Python Bindings 0.10.21 "She used to be an ironhorse, twenty years ago" Message-ID: <1295867087.3821.7.camel@localhost> This mail announces the release of GStreamer Python Bindings 0.10.19 "She used to be an ironhorse, twenty years ago". GStreamer Python Bindings is a set of Python bindings for GStreamer, using the same system as PyGTK. It also comes with a number of examples. For more information, see http://gstreamer.freedesktop.org/modules/gst-python.html To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-python -------------- next part -------------- Release notes for GStreamer Python bindings?0.10.21 "She used to be an ironhorse, twenty years ago" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer Python Bindings. 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. Features of this release * gstreamer core/base 0.10.32 API additions * Fix GstDiscover methods Bugs fixed in this release * 639293 : gst-python 0.10.20 does not build with gstreamer < 0.10.31 * 639632 : GST_TAG_APPLICATION_NAME not bound Download You can find source releases of gst-python in the download directory: http://gstreamer.freedesktop.org/src/gst-python/ 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 * Arun Raghavan * Edward Hervey * Stefan Kost * Tim-Philipp M?ller * christian schaller ? From giorgio9 at libero.it Mon Jan 24 12:08:45 2011 From: giorgio9 at libero.it (giorgino) Date: Mon, 24 Jan 2011 03:08:45 -0800 (PST) Subject: [gst-devel] source port of udp sink Message-ID: <1295867325134-3233756.post@n4.nabble.com> Hi all, I have a question for you. There is a way in order to know or to set the source port of an udpsink or a rtpbin (that i think works over udpsink)? Thanks in advance -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/source-port-of-udp-sink-tp3233756p3233756.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From wim.taymans at gmail.com Mon Jan 24 12:21:04 2011 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 24 Jan 2011 12:21:04 +0100 Subject: [gst-devel] source port of udp sink In-Reply-To: <1295867325134-3233756.post@n4.nabble.com> References: <1295867325134-3233756.post@n4.nabble.com> Message-ID: Did you try the 'port' property? Use gst-inspect udpsrc and gst-inspect udpsink to find the properties or read the element documentation on the website. Wim On Mon, Jan 24, 2011 at 12:08 PM, giorgino wrote: > > Hi all, > ? ? ? ? I have a question for you. There is a way in order to know or to > set the source port of an udpsink or a rtpbin (that i think works over > udpsink)? > > Thanks in advance > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/source-port-of-udp-sink-tp3233756p3233756.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From wl2776 at gmail.com Mon Jan 24 12:34:34 2011 From: wl2776 at gmail.com (wl2776) Date: Mon, 24 Jan 2011 03:34:34 -0800 (PST) Subject: [gst-devel] HTTP video source serves individual JPEG images In-Reply-To: <4D21A711.1060504@ingelabs.com> References: <4D21A711.1060504@ingelabs.com> Message-ID: <1295868874908-3233804.post@n4.nabble.com> There's the curlsrc plugin, which was implemented by its original author for the same purpose. You could download and try it. The discussion and links to the source code are here: https://bugzilla.gnome.org/show_bug.cgi?id=558450 -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/HTTP-video-source-serves-individual-JPEG-images-tp3171825p3233804.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From giorgio9 at libero.it Mon Jan 24 12:40:08 2011 From: giorgio9 at libero.it (giorgino) Date: Mon, 24 Jan 2011 03:40:08 -0800 (PST) Subject: [gst-devel] source port of udp sink In-Reply-To: References: <1295867325134-3233756.post@n4.nabble.com> Message-ID: <1295869208757-3233814.post@n4.nabble.com> Hi and thanks for the answer. I tried but there isn't anything. I don't need anything on udpsrc infact my problem is to know and then to set, the source port of the udpsink before the pipe starts. I think that usually gstreamer assigns the source port in a random way, but for my purposes I need to have the management of this thing in order to set manually the source port. G. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/source-port-of-udp-sink-tp3233756p3233814.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From michael at panicking.kicks-ass.org Mon Jan 24 12:52:26 2011 From: michael at panicking.kicks-ass.org (Michael Trimarchi) Date: Mon, 24 Jan 2011 12:52:26 +0100 Subject: [gst-devel] source port of udp sink In-Reply-To: <1295869208757-3233814.post@n4.nabble.com> References: <1295867325134-3233756.post@n4.nabble.com> <1295869208757-3233814.post@n4.nabble.com> Message-ID: <4D3D67FA.2070202@panicking.kicks-ass.org> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 01/24/2011 12:40 PM, giorgino wrote: > > Hi and thanks for the answer. I tried but there isn't anything. I don't need > anything on udpsrc infact my problem is to know and then to set, the source > port of the udpsink before the pipe starts. I think that usually gstreamer > assigns the source port in a random way, but for my purposes I need to have > the management of this thing in order to set manually the source port. > > G. g_object_set(G_OBJECT(udpsrc), "sockfd", fd_sock, "port", port, NULL); Michael -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJNPWf6AAoJEJJLw+x5tkpwv84H/1sKXMiJOl6e7rol4GsYu9OF z2SccHxqHkWDfqyYMuN85PIWsBZsf6PFU4OgRl9Cw7rUH8h0gg9XKHvn8+6LGs9k t4uVp8NckwonPYaHqgmsU0vsSsd28BJOUKT4vkXBX8t6VZK2KYU9VoooKLds+B3m 5rmrrdtjfODMtaPA/mN83vvFuYBe1w23sgI2qoA4DqfAbc7znUg3PigORW9eOhRd 4J/6yf6Hx2sAUipzUEsg4Aj9ZbqU/COyRkPATmDtQYVeCy4L6FJ+7ux8aAN8STCw wQUNDMzdlYFvHs8RvnI6TzoJ8/6fJHUJvZ0Nb9kjAIuSFK2yVbgBBa85kSFRHns= =NgqR -----END PGP SIGNATURE----- From kiagiadakis.george at gmail.com Mon Jan 24 13:59:25 2011 From: kiagiadakis.george at gmail.com (George Kiagiadakis) Date: Mon, 24 Jan 2011 14:59:25 +0200 Subject: [gst-devel] RELEASE: QtGStreamer 0.10.1 Message-ID: This mail announces the release of QtGStreamer 0.10.1. QtGStreamer is a set of libraries providing C++ bindings for GStreamer with a Qt-style API, plus some helper classes and elements for integrating GStreamer better in Qt applications. This first release marks the beginning of the stable 0.10 series of QtGStreamer. For more information, go to http://gstreamer.freedesktop.org/modules/qt-gstreamer.html Documentation can be found at http://gstreamer.freedesktop.org/data/doc/gstreamer/head/qt-gstreamer/html/index.html To file bugs, request features or submit patches, please go to https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=qt-gstreamer Direct links to tarballs: http://gstreamer.freedesktop.org/src/qt-gstreamer/qt-gstreamer-0.10.1.tar.bz2 http://gstreamer.freedesktop.org/src/qt-gstreamer/qt-gstreamer-0.10.1.tar.gz Enjoy! From ikt011 at gmail.com Mon Jan 24 14:31:10 2011 From: ikt011 at gmail.com (Kocsis Tibor) Date: Mon, 24 Jan 2011 14:31:10 +0100 Subject: [gst-devel] gst-launch -e on windows Message-ID: Hi, how can I use the -e option for gst-launch on windows? On linux it works fine, forces the eos but on windows when I press ctrl+c it nothing happens. Maybe there is a trick to use it somehow? Thanks, Tibor From thiago.sousa.santos at collabora.co.uk Mon Jan 24 15:09:54 2011 From: thiago.sousa.santos at collabora.co.uk (Thiago Sousa Santos) Date: Mon, 24 Jan 2011 11:09:54 -0300 Subject: [gst-devel] gst-launch -e on windows In-Reply-To: References: Message-ID: <1295878194.17238.0.camel@blacksheep> On Mon, 2011-01-24 at 14:31 +0100, Kocsis Tibor wrote: > Hi, > > how can I use the -e option for gst-launch on windows? On linux it > works fine, forces the eos but on windows when I press ctrl+c it > nothing happens. Maybe there is a trick to use it somehow? It should work, does your version on windows have the -e option? Does the same pipeline work on linux? What's the pipeline? -- Thiago > > Thanks, > Tibor > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ikt011 at gmail.com Mon Jan 24 16:06:35 2011 From: ikt011 at gmail.com (Kocsis Tibor) Date: Mon, 24 Jan 2011 16:06:35 +0100 Subject: [gst-devel] gst-launch -e on windows In-Reply-To: <1295878194.17238.0.camel@blacksheep> References: <1295878194.17238.0.camel@blacksheep> Message-ID: The cmd is something like this: gst-launch -e rtspsrc location="..." ! some elements ! filesink location="..." Yes, it works on linux. gst-launch --help shows the -e option. I'm using the last build from here: http://code.google.com/p/ossbuild/downloads/list# Regards, Tibor On Mon, Jan 24, 2011 at 3:09 PM, Thiago Sousa Santos wrote: > On Mon, 2011-01-24 at 14:31 +0100, Kocsis Tibor wrote: >> Hi, >> >> how can I use the -e option for gst-launch on windows? On linux it >> works fine, forces the eos but on windows when I press ctrl+c it >> nothing happens. Maybe there is a trick to use it somehow? > > It should work, does your version on windows have the -e option? Does > the same pipeline work on linux? > > What's the pipeline? > > > -- > Thiago > >> >> Thanks, >> Tibor >> >> ------------------------------------------------------------------------------ >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> Finally, a world-class log management solution at an even better price-free! >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> February 28th, so secure your free ArcSight Logger TODAY! >> http://p.sf.net/sfu/arcsight-sfd2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > 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 Jan 24 16:18:37 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 24 Jan 2011 15:18:37 +0000 Subject: [gst-devel] gst-launch -e on windows In-Reply-To: <1295878194.17238.0.camel@blacksheep> References: <1295878194.17238.0.camel@blacksheep> Message-ID: <1295882317.17817.2.camel@zingle> On Mon, 2011-01-24 at 11:09 -0300, Thiago Sousa Santos wrote: > > how can I use the -e option for gst-launch on windows? On linux it > > works fine, forces the eos but on windows when I press ctrl+c it > > nothing happens. Maybe there is a trick to use it somehow? > > It should work, does your version on windows have the -e option? Does > the same pipeline work on linux? Note this code in gst-launch.c: /* FIXME: hack alert */ #ifdef HAVE_WIN32 #define DISABLE_FAULT_HANDLER #endif Someone needs to implement/test this for win32 I guess.. Cheers -Tim From ikt011 at gmail.com Mon Jan 24 16:28:48 2011 From: ikt011 at gmail.com (Kocsis Tibor) Date: Mon, 24 Jan 2011 16:28:48 +0100 Subject: [gst-devel] gst-launch -e on windows In-Reply-To: <1295882317.17817.2.camel@zingle> References: <1295878194.17238.0.camel@blacksheep> <1295882317.17817.2.camel@zingle> Message-ID: Bad news :( Thanks, Tibor On Mon, Jan 24, 2011 at 4:18 PM, Tim-Philipp M?ller wrote: > On Mon, 2011-01-24 at 11:09 -0300, Thiago Sousa Santos wrote: > >> > how can I use the -e option for gst-launch on windows? On linux it >> > works fine, forces the eos but on windows when I press ctrl+c it >> > nothing happens. Maybe there is a trick to use it somehow? >> >> It should work, does your version on windows have the -e option? Does >> the same pipeline work on linux? > > Note this code in gst-launch.c: > > /* FIXME: hack alert */ > #ifdef HAVE_WIN32 > #define DISABLE_FAULT_HANDLER > #endif > > Someone needs to implement/test this for win32 I guess.. > > ?Cheers > ?-Tim > > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From paper at gmx.net Mon Jan 24 17:04:02 2011 From: paper at gmx.net (Kony) Date: Mon, 24 Jan 2011 20:04:02 +0400 Subject: [gst-devel] ffdec_h264 and vdpau Message-ID: <1295885042.17192.4.camel@kk-laptop> Does ffdec_h264 plugin support vdpau through ffmpeg? From kapil.agl at gmail.com Mon Jan 24 17:23:25 2011 From: kapil.agl at gmail.com (Kapil Agrawal) Date: Mon, 24 Jan 2011 21:53:25 +0530 Subject: [gst-devel] ffdec_h264 and vdpau In-Reply-To: <1295885042.17192.4.camel@kk-laptop> References: <1295885042.17192.4.camel@kk-laptop> Message-ID: No, it doesnt. If you are looking towards using vdpau accelration via ffmpeg, i dont think you can currently do with any gst-ffmpeg plugin, but yes you may wan to try vdpau plugins inside gst-plugins-bad. ffdec_h264 is a software decoder that uses ffmpeg h264 decoding. On Mon, Jan 24, 2011 at 9:34 PM, Kony wrote: > Does ffdec_h264 plugin support vdpau through ffmpeg? > > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better > price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > 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 Mon Jan 24 17:36:44 2011 From: lazyt at gmx.net (Thomas Loewe) Date: Mon, 24 Jan 2011 08:36:44 -0800 (PST) Subject: [gst-devel] How to use GstVideoMixerPad? Message-ID: <1295887004426-3234360.post@n4.nabble.com> I would like to use the videomixer element to add picture in picture support into my app. According to the docs it should be configured via GstVideoMixerPad. Is there an example in c/c++ available? I found only something in http://www.oz9aec.net/index.php/gstreamer/354-gstreamer-compositing-using-the-gstvideomixerpad-properties python but i don't understand it... Thanks, Thomas -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-use-GstVideoMixerPad-tp3234360p3234360.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From konstantin.kauz at googlemail.com Mon Jan 24 16:58:36 2011 From: konstantin.kauz at googlemail.com (Konstantin Kauz) Date: Mon, 24 Jan 2011 19:58:36 +0400 Subject: [gst-devel] ffdec_h264 and vdpau Message-ID: <1295884716.17192.2.camel@kk-laptop> Does ffdec_h264 plugin support vdpau decoding? From t.i.m at zen.co.uk Mon Jan 24 18:19:42 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 24 Jan 2011 17:19:42 +0000 Subject: [gst-devel] How to use GstVideoMixerPad? In-Reply-To: <1295887004426-3234360.post@n4.nabble.com> References: <1295887004426-3234360.post@n4.nabble.com> Message-ID: <1295889582.17817.5.camel@zingle> On Mon, 2011-01-24 at 08:36 -0800, Thomas Loewe wrote: > I would like to use the videomixer element to add picture in picture support > into my app. > > According to the docs it should be configured via GstVideoMixerPad. Is there > an example in c/c++ available? > > I found only something in > http://www.oz9aec.net/index.php/gstreamer/354-gstreamer-compositing-using-the-gstvideomixerpad-properties > python but i don't understand it... What exactly do you not understand / is unclear? Videomixer pads are sub-classes of GstPad, so they provide additional functionality/API. This extra functionality is exposed via GObject properties. In C/C++ you'll have to use g_object_set() to set those. The GstVideoMixerPad object structure is not available outside the plugin, so you'll have to treat it as a normal GstPad with additional properties. Cheers -Tim From paper at gmx.net Mon Jan 24 20:54:00 2011 From: paper at gmx.net (Kony) Date: Mon, 24 Jan 2011 23:54:00 +0400 Subject: [gst-devel] mp4mux Message-ID: <1295898840.22458.11.camel@kk-laptop> I'm parsing a mpeg4 file(generated with gstreamer mp4mux), through references it should look, like this parser do: http://code.google.com/p/jcodec/source/browse/trunk/src/main/java/org/jcodec/containers/mp4/boxes/AudioSampleEntryBox.java?r=50 but my file is looking like: b - box length f - full box fields s - Sample Description Box fields |b|b|b|b|s|t|s|d|f|f|f|f|s|s|s|s| should be six bytes reserved 0 from Sample Entry(ISO doc), but is: |0|0|0|136|a|v|c|1| The ISO and Apple docs describe the parser mentioned above way. Can send the file per email, can somebody please explain this issue? From paper at gmx.net Mon Jan 24 21:02:42 2011 From: paper at gmx.net (Kony) Date: Tue, 25 Jan 2011 00:02:42 +0400 Subject: [gst-devel] mp4mux Message-ID: <1295899362.22709.1.camel@kk-laptop> Sorry, forget it. i not saw the abstract "SampleEntry" extends a box :( From lazyt at gmx.net Tue Jan 25 08:28:48 2011 From: lazyt at gmx.net (Thomas Loewe) Date: Mon, 24 Jan 2011 23:28:48 -0800 (PST) Subject: [gst-devel] How to use GstVideoMixerPad? In-Reply-To: <1295889582.17817.5.camel@zingle> References: <1295887004426-3234360.post@n4.nabble.com> <1295889582.17817.5.camel@zingle> Message-ID: <1295940528431-3235584.post@n4.nabble.com> > In C/C++ you'll have to use g_object_set() to set those. Ah ok, didn't realized that g_object_set() can be used on a pad - used it only for element properties so far. Works great now, Thanks! -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/How-to-use-GstVideoMixerPad-tp3234360p3235584.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From julien.isorce at gmail.com Tue Jan 25 14:29:53 2011 From: julien.isorce at gmail.com (Julien Isorce) Date: Tue, 25 Jan 2011 14:29:53 +0100 Subject: [gst-devel] G722 in gstreamer In-Reply-To: <1285846255.2520.15.camel@metal> References: <1285846255.2520.15.camel@metal> Message-ID: 2010/9/30 Wim Taymans > On Wed, 2010-09-29 at 16:16 +0200, Antoni Silvestre Padr?s wrote: > > Hello, > > > > > > I've been looking for it and I guess there isn't one but just in case > > I've missed it somehow I'll ask it here, is there any G.722 codec and > > payloader for gstreamer? > > There is a G722 encoder/decoder in ffmpeg git so it could be added to > gst-ffmpeg too. I'll commit an G722 payloader/depayloader soon. > Hi, Would it be possible to add G722 and G729 to gst-ffmpeg ? The codec_id are: CODEC_ID_G729 CODEC_ID_ADPCM_G722 Sincerely Julien > > Wim > > > > > > > Thanks, > > Toni Silvestre > > > ------------------------------------------------------------------------------ > > Start uncovering the many advantages of virtual appliances > > and start using them to simplify application deployment and > > accelerate your shift to cloud computing. > > http://p.sf.net/sfu/novell-sfdev2dev > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Start uncovering the many advantages of virtual appliances > and start using them to simplify application deployment and > accelerate your shift to cloud computing. > http://p.sf.net/sfu/novell-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 jobbagy.robert at gmail.com Tue Jan 25 14:58:05 2011 From: jobbagy.robert at gmail.com (Robert Jobbagy) Date: Tue, 25 Jan 2011 14:58:05 +0100 Subject: [gst-devel] gst-opencv multi thread problem Message-ID: Hi Guys, I use one pipeline with 4 rtsp src and 4 instance of own filter. I think the different threads caused the segfault when different object call opencv functions. And I fixed this with mutex,but I realized that different objects can access to other objects frame buffer in opencv. When different objects call opencv functions then next objects what call opencv function find some frames in buffer from previous cam of object. But I dont understand how. What do you think about this. Thanks your time. example: gst-launch rtspsrc location="rtsp://admin:12345 at 10.30.0.126:554/h.264/ch1/sub" latency=1000 ! decodebin ! ffmpegcolorspace ! motioncells ! ffmpegcolorspace ! xvimagesink rtspsrc location="rtsp://admin:12345 at 10.30.0.122:554/h.264/ch1/sub" latency=1000 ! decodebin ! ffmpegcolorspace ! motioncells ! ffmpegcolorspace ! xvimagesink rtspsrc location="rtsp://admin:12345 at 10.30.0.125:554/h.264/ch1/sub" latency=1000 ! decodebin ! ffmpegcolorspace ! motioncells ! ffmpegcolorspace ! xvimagesink rtspsrc location="rtsp://admin:12345 at 10.30.0.115:554/h.264/ch1/sub" latency=1000 ! decodebin ! ffmpegcolorspace ! motioncells ! ffmpegcolorspace ! xvimagesink my filter name is motioncells -- Best Regards, Robert From bilboed at gmail.com Tue Jan 25 15:28:23 2011 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 25 Jan 2011 15:28:23 +0100 Subject: [gst-devel] G722 in gstreamer In-Reply-To: References: <1285846255.2520.15.camel@metal> Message-ID: <1295965703.3821.10.camel@localhost> On Tue, 2011-01-25 at 14:29 +0100, Julien Isorce wrote: > > > 2010/9/30 Wim Taymans > On Wed, 2010-09-29 at 16:16 +0200, Antoni Silvestre Padr?s > wrote: > > Hello, > > > > > > I've been looking for it and I guess there isn't one but > just in case > > I've missed it somehow I'll ask it here, is there any G.722 > codec and > > payloader for gstreamer? > > > There is a G722 encoder/decoder in ffmpeg git so it could be > added to > gst-ffmpeg too. I'll commit an G722 payloader/depayloader > soon. > > > > > Hi, > > > Would it be possible to add G722 and G729 to gst-ffmpeg ? > > > The codec_id are: > > > CODEC_ID_G729 > CODEC_ID_ADPCM_G722 I updated gst-ffmpeg yesterday to track last svn revision of ffmpeg. I still haven't figured out what proper gstreamer mediatype to use for those two codecs. Once I've done that (or someone proposes such types) I'll push it. Edward > > > Sincerely > Julien > > > > > > > > > Wim > > > > > > > Thanks, > > Toni Silvestre > > > ------------------------------------------------------------------------------ > > Start uncovering the many advantages of virtual appliances > > and start using them to simplify application deployment and > > accelerate your shift to cloud computing. > > http://p.sf.net/sfu/novell-sfdev2dev > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Start uncovering the many advantages of virtual appliances > and start using them to simplify application deployment and > accelerate your shift to cloud computing. > http://p.sf.net/sfu/novell-sfdev2dev > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From olivier.crete at collabora.co.uk Tue Jan 25 16:06:11 2011 From: olivier.crete at collabora.co.uk (Olivier =?ISO-8859-1?Q?Cr=EAte?=) Date: Tue, 25 Jan 2011 10:06:11 -0500 Subject: [gst-devel] G722 in gstreamer In-Reply-To: <1295965703.3821.10.camel@localhost> References: <1285846255.2520.15.camel@metal> <1295965703.3821.10.camel@localhost> Message-ID: <1295967971.15660.1.camel@TesterBox.tester.ca> On Tue, 2011-01-25 at 15:28 +0100, Edward Hervey wrote: > On Tue, 2011-01-25 at 14:29 +0100, Julien Isorce wrote: > > Wim Taymans > > On Wed, 2010-09-29 at 16:16 +0200, Antoni Silvestre Padr?s > > wrote: > > > I've been looking for it and I guess there isn't one but > > just in case > > > I've missed it somehow I'll ask it here, is there any G.722 > > codec and > > > payloader for gstreamer? > > > > There is a G722 encoder/decoder in ffmpeg git so it could be > > added to > > gst-ffmpeg too. I'll commit an G722 payloader/depayloader > > soon. > > > > Would it be possible to add G722 and G729 to gst-ffmpeg ? > > > > The codec_id are: > > > > CODEC_ID_G729 > > CODEC_ID_ADPCM_G722 > > I updated gst-ffmpeg yesterday to track last svn revision of ffmpeg. I > still haven't figured out what proper gstreamer mediatype to use for > those two codecs. Once I've done that (or someone proposes such types) > I'll push it. From good/gst/rtp/gstrtpg72?.c: audio/G722 and audio/G729 -- Olivier Cr?te olivier.crete at collabora.co.uk Collabora Ltd -------------- 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 grodriguez at ingelabs.com Tue Jan 25 16:09:42 2011 From: grodriguez at ingelabs.com (grodriguez at ingelabs.com) Date: Tue, 25 Jan 2011 16:09:42 +0100 Subject: [gst-devel] gstreamer-devel Digest, Vol 56, Issue 75 In-Reply-To: References: Message-ID: <4D3EE7B6.8060808@ingelabs.com> Hello, Thank you for the pointer although from the discussion linked it seems that it never moved beyond alpha stage. I'll have a look anyway. Thanks you. El 24/01/2011 12:40, gstreamer-devel-request at lists.sourceforge.net escribi?: > There's the curlsrc plugin, which was implemented by its original author for > the same purpose. > You could download and try it. > > The discussion and links to the source code are here: > https://bugzilla.gnome.org/show_bug.cgi?id=558450 From grodriguez at ingelabs.com Tue Jan 25 16:02:55 2011 From: grodriguez at ingelabs.com (grodriguez at ingelabs.com) Date: Tue, 25 Jan 2011 16:02:55 +0100 Subject: [gst-devel] HTTP video source serves individual JPEG images In-Reply-To: References: Message-ID: <4D3EE61F.1030202@ingelabs.com> Dear Wim, Arnout, Thank you very much for your answers, I'll have a look and if I get somewhere I'll post the results back. Thanks, G Rodriguez El 23/01/2011 9:21, gstreamer-devel-request at lists.sourceforge.net escribi?: > I think the file would just be overwritten with new contents because > the filesink > would reopen (not in append mode) the file. > > Wim > > On Sat, Jan 22, 2011 at 10:00 PM, Arnout Vandecappelle wrote: >> Good idea, but wouldn't that wreak havoc on the output file in case it's a >> filesink instead of an xvimagesink? >> >> Regards, >> >> Arnout >> >> On Saturday 22 January 2011 21:55:18, Wim Taymans wrote: >> >>> For those sources I would suggest to periodically set the pipeline to >> >>> NULL when you >> >>> get an EOS event. When you want to get a new image, set it back to >>> PLAYING, >> >>> and then you repeat this. >> >>> >> >>> Wim From bilboed at gmail.com Tue Jan 25 16:36:26 2011 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 25 Jan 2011 16:36:26 +0100 Subject: [gst-devel] G722 in gstreamer In-Reply-To: <1295967971.15660.1.camel@TesterBox.tester.ca> References: <1285846255.2520.15.camel@metal> <1295965703.3821.10.camel@localhost> <1295967971.15660.1.camel@TesterBox.tester.ca> Message-ID: <1295969786.3821.13.camel@localhost> On Tue, 2011-01-25 at 10:06 -0500, Olivier Cr?te wrote: > On Tue, 2011-01-25 at 15:28 +0100, Edward Hervey wrote: > > On Tue, 2011-01-25 at 14:29 +0100, Julien Isorce wrote: > > > Wim Taymans > > > On Wed, 2010-09-29 at 16:16 +0200, Antoni Silvestre Padr?s > > > wrote: > > > > I've been looking for it and I guess there isn't one but > > > just in case > > > > I've missed it somehow I'll ask it here, is there any G.722 > > > codec and > > > > payloader for gstreamer? > > > > > > There is a G722 encoder/decoder in ffmpeg git so it could be > > > added to > > > gst-ffmpeg too. I'll commit an G722 payloader/depayloader > > > soon. > > > > > > Would it be possible to add G722 and G729 to gst-ffmpeg ? > > > > > > The codec_id are: > > > > > > CODEC_ID_G729 > > > CODEC_ID_ADPCM_G722 > > > > I updated gst-ffmpeg yesterday to track last svn revision of ffmpeg. I > > still haven't figured out what proper gstreamer mediatype to use for > > those two codecs. Once I've done that (or someone proposes such types) > > I'll push it. > > From good/gst/rtp/gstrtpg72?.c: > audio/G722 and audio/G729 I pushed the required code go to add audio/G722 features in gst-ffmpeg. As for G729 it's not available in the list of publicly available codecs exported from libavcodec, therefore we can't wrap it. Edward > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ 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 Jan 25 16:46:14 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 25 Jan 2011 15:46:14 +0000 Subject: [gst-devel] gst-opencv multi thread problem In-Reply-To: References: Message-ID: <1295970374.2171.16.camel@zingle> On Tue, 2011-01-25 at 14:58 +0100, Robert Jobbagy wrote: > What do you think about this. > I think you should file a bug :) http://gstreamer.freedesktop.org/bugs/ Cheers -tim From wl2776 at gmail.com Tue Jan 25 17:45:31 2011 From: wl2776 at gmail.com (wl2776) Date: Tue, 25 Jan 2011 08:45:31 -0800 (PST) Subject: [gst-devel] Does filesrc provide any caching internally? Message-ID: <1295973931986-3236487.post@n4.nabble.com> When seeking, MPEG-PS demuxer (mpegpsdemux) repeatedly requests data from its upstream peer element using gst_pad_pull_range(). However, seeking is performed rather quickly in my playbin2 pipeline. Is it because filesrc caches data? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Does-filesrc-provide-any-caching-internally-tp3236487p3236487.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From lazyt at gmx.net Tue Jan 25 18:06:17 2011 From: lazyt at gmx.net (Thomas Loewe) Date: Tue, 25 Jan 2011 09:06:17 -0800 (PST) Subject: [gst-devel] problems with 0.10.32 core release (cpuload, deinterlacer) Message-ID: <1295975177921-3236547.post@n4.nabble.com> Hi, have a strange problem with the 0.10.32 release: it causes high cpu-load here (i7 clocks to 2400MHz/15% load instead 933MHz/5% like before). Replacing the libgstreamer-0.10.dll with the one from 0.10.31 and the problem is gone. Next thing is the deinterlacer: it can not longer switched on/off via the "mode" property (nothing happens). This problem is also fixed with replacing the dll. Can anyone confirm this or have an idea what the problem could be? Thanks, Thomas -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/problems-with-0-10-32-core-release-cpuload-deinterlacer-tp3236547p3236547.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From paper at gmx.net Tue Jan 25 18:18:57 2011 From: paper at gmx.net (Kony) Date: Tue, 25 Jan 2011 21:18:57 +0400 Subject: [gst-devel] mp4 or matroska Message-ID: <1295975937.18279.7.camel@kk-laptop> i want to take video clips and make a big moov atom, then playing as a one file the clips. For reach that i need to fake the movie length in the moov atom and manipulate the frame tables etc. It is possible with mp4 container format or maybe with matroska, i write the library from scratch for this purposes(streamable property have to be to false for matroska, player doesn't not supports this option)? Thanks you in advice for hints. From bilboed at gmail.com Tue Jan 25 18:46:05 2011 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 25 Jan 2011 18:46:05 +0100 Subject: [gst-devel] problems with 0.10.32 core release (cpuload, deinterlacer) In-Reply-To: <1295975177921-3236547.post@n4.nabble.com> References: <1295975177921-3236547.post@n4.nabble.com> Message-ID: <1295977565.3821.14.camel@localhost> On Tue, 2011-01-25 at 09:06 -0800, Thomas Loewe wrote: > Hi, > > have a strange problem with the 0.10.32 release: it causes high cpu-load > here (i7 clocks to 2400MHz/15% load instead 933MHz/5% like before). > > Replacing the libgstreamer-0.10.dll with the one from 0.10.31 and the > problem is gone. > > Next thing is the deinterlacer: it can not longer switched on/off via the > "mode" property (nothing happens). This problem is also fixed with replacing > the dll. > > Can anyone confirm this or have an idea what the problem could be? Looks like you might not have compiled it with a recent enough ORC. Edward > > Thanks, > Thomas From lazyt at gmx.net Tue Jan 25 19:27:22 2011 From: lazyt at gmx.net (Thomas Loewe) Date: Tue, 25 Jan 2011 10:27:22 -0800 (PST) Subject: [gst-devel] problems with 0.10.32 core release (cpuload, deinterlacer) In-Reply-To: <1295977565.3821.14.camel@localhost> References: <1295975177921-3236547.post@n4.nabble.com> <1295977565.3821.14.camel@localhost> Message-ID: <1295980042396-3236739.post@n4.nabble.com> > Looks like you might not have compiled it with a recent enough ORC. Using the latest 0.4.11 compiled under ubuntu 10.10 with mingw32. Thomas -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/problems-with-0-10-32-core-release-cpuload-deinterlacer-tp3236547p3236739.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From gibrovacco at gmail.com Tue Jan 25 20:17:56 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Tue, 25 Jan 2011 21:17:56 +0200 Subject: [gst-devel] G722 in gstreamer In-Reply-To: <1295969786.3821.13.camel@localhost> References: <1285846255.2520.15.camel@metal> <1295965703.3821.10.camel@localhost> <1295967971.15660.1.camel@TesterBox.tester.ca> <1295969786.3821.13.camel@localhost> Message-ID: Hi, On Tue, Jan 25, 2011 at 5:36 PM, Edward Hervey wrote: > On Tue, 2011-01-25 at 10:06 -0500, Olivier Cr?te wrote: >> On Tue, 2011-01-25 at 15:28 +0100, Edward Hervey wrote: >> > On Tue, 2011-01-25 at 14:29 +0100, Julien Isorce wrote: >> > > Wim Taymans >> > > ? ? ? ? On Wed, 2010-09-29 at 16:16 +0200, Antoni Silvestre Padr?s >> > > ? ? ? ? wrote: >> > > ? ? ? ? > I've been looking for it and I guess there isn't one but >> > > ? ? ? ? just in case >> > > ? ? ? ? > I've missed it somehow I'll ask it here, is there any G.722 >> > > ? ? ? ? codec and >> > > ? ? ? ? > payloader for gstreamer? >> > > >> > > ? ? ? ? There is a G722 encoder/decoder in ffmpeg git so it could be >> > > ? ? ? ? added to >> > > ? ? ? ? gst-ffmpeg too. I'll commit an G722 payloader/depayloader >> > > ? ? ? ? soon. >> > > >> > > Would it be possible to add G722 and G729 to gst-ffmpeg ? >> > > >> > > The codec_id ?are: >> > > >> > > CODEC_ID_G729 >> > > CODEC_ID_ADPCM_G722 >> > >> > ?I updated gst-ffmpeg yesterday to track last svn revision of ffmpeg. I >> > still haven't figured out what proper gstreamer mediatype to use for >> > those two codecs. Once I've done that (or someone proposes such types) >> > I'll push it. >> >> From good/gst/rtp/gstrtpg72?.c: >> audio/G722 and audio/G729 > > ?I pushed the required code go to add audio/G722 features in > gst-ffmpeg. > > ?As for G729 it's not available in the list of publicly available > codecs exported from libavcodec, therefore we can't wrap it. > for g729, we already have the "native" implementation available: http://gitorious.org/gladstone The only thing impeding me to contribute it in gst-plugins-ugly is the redistribution of the reference code. I've already asked to ITU-T whether it's possible (or not) to do so, but no replies so far... :( Regards > ? ?Edward > >> >> >> ------------------------------------------------------------------------------ >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> Finally, a world-class log management solution at an even better price-free! >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> February 28th, so secure your free ArcSight Logger TODAY! >> http://p.sf.net/sfu/arcsight-sfd2d >> _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From nathanael at gnat.ca Wed Jan 26 07:22:28 2011 From: nathanael at gnat.ca (Nathanael D. Noblet) Date: Tue, 25 Jan 2011 23:22:28 -0700 Subject: [gst-devel] problem with pa_stream_make_writeable_size() failed connection terminated Message-ID: <4D3FBDA4.204@gnat.ca> Hello Everyone, So I've built an application based on gstreamer that's been working well so far. I've recently run into an issue that I don't understand. I'm hoping someone here will know what I should look at. So I read from an IP camera via rtspsrc. It sends video in MP4 format and Audio in AAC. We're running a beta with our client and recently after installing it on location I've run into issues with one location. After 5-9 seconds I get pa_stream_make_writeable_size() failed connection terminated this causes the stream to pause if I'm recording, but not if I'm just viewing. After taking a look I found the following differences. The cameras that worked were set to have audio at 96Kbps vs 16Kbps. I only noticed when I compared the caps that each pad-added pad from the decoder had. The one that has the problem was: application/x-rtp, media=(string)audio, payload=(int)97, clock-rate=(int)16000, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)2, streamtype=(string)5, profile-level-id=(string)15, mode=(string)AAC-hbr, config=(string)1410, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, ctsdeltalength=(string)0, dtsdeltalength=(string)0, a-charset=(string)Shift_JIS, a-etag=(string)1234567890, clock-base=(uint)0, seqnum-base=(uint)0, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1; and the one that worked: application/x-rtp, media=(string)audio, payload=(int)97, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)2, streamtype=(string)5, profile-level-id=(string)15, mode=(string)AAC-hbr, config=(string)1210, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, ctsdeltalength=(string)0, dtsdeltalength=(string)0, a-charset=(string)Shift_JIS, a-etag=(string)1234567890, clock-base=(uint)0, seqnum-base=(uint)0, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1; A diff of the two was what tipped me off to the difference between the two cameras: @@ -1,13 +1,13 @@ Structure application/x-rtp, media=(string)audio, payload=(int)97, -clock-rate=(int)16000, +clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)2, streamtype=(string)5, profile-level-id=(string)15, mode=(string)AAC-hbr, -config=(string)1410, +config=(string)1210, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, Once I changed the settings on the camera it started working again, lowering the bitrate causes the error again. So I'm at a total loss as to what the error I get about pa_stream_make_writeable_size() has to do with the audio bitrate. I'm *really* hoping someone could shed some light on this so I can understand what the issue is and how to work around it. -- Nathanael d. Noblet From t.i.m at zen.co.uk Wed Jan 26 12:29:44 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 26 Jan 2011 11:29:44 +0000 Subject: [gst-devel] problems with 0.10.32 core release (cpuload, deinterlacer) In-Reply-To: <1295975177921-3236547.post@n4.nabble.com> References: <1295975177921-3236547.post@n4.nabble.com> Message-ID: <1296041384.18773.1.camel@zingle> On Tue, 2011-01-25 at 09:06 -0800, Thomas Loewe wrote: > have a strange problem with the 0.10.32 release: it causes high cpu-load > here (i7 clocks to 2400MHz/15% load instead 933MHz/5% like before). > > Replacing the libgstreamer-0.10.dll with the one from 0.10.31 and the > problem is gone. > > Next thing is the deinterlacer: it can not longer switched on/off via the > "mode" property (nothing happens). This problem is also fixed with replacing > the dll. > > Can anyone confirm this or have an idea what the problem could be? Could you file (separate) bugs for these things please, so we can investigate them? Some more details on how to reproduce both cases (esp. the first) would be nice. Thanks! Cheers -Tim From loren.rogers at gmail.com Wed Jan 26 19:25:47 2011 From: loren.rogers at gmail.com (Loren Rogers) Date: Wed, 26 Jan 2011 13:25:47 -0500 Subject: [gst-devel] Should RTSPSRC correct for time drift when converting the RTP timestamp to gstreamer clock? Message-ID: Hey gang, I have a gstreamer program that records audio from an RTP audio source. I then timestamp the audio frame on the recorder based on the gstreamer clock (RTSPSRC should have already done the RTP TS ->gstreamer clock TS conversion). I am finding that the time drifts (i.e. the most recent gstreamer-clock-based timestamp on this audio frame might be increasingly early or late relative to timestamp the current recorder thinks it should be) when timestamping the frames end to end. I am wondering if this drift may be linked to the clock drift between the audio device and my recording device. Should the RTSPSRC element already be correcting for such time drift when converting the RTP timestamp to the gstreamer clock? From bossart.nospam at gmail.com Wed Jan 26 18:21:16 2011 From: bossart.nospam at gmail.com (pl bossart) Date: Wed, 26 Jan 2011 09:21:16 -0800 Subject: [gst-devel] problem with pa_stream_make_writeable_size() failed connection terminated In-Reply-To: <4D3FBDA4.204@gnat.ca> References: <4D3FBDA4.204@gnat.ca> Message-ID: > ? So I read from an IP camera via rtspsrc. It sends video in MP4 format > and Audio in AAC. We're running a beta with our client and recently > after installing it on location I've run into issues with one location. > After 5-9 seconds I get > > pa_stream_make_writeable_size() failed connection terminated This is an issue in the connection with PulseAudio. Since it's PCM at that point, I don't think it's an issue with the bitrate of the AAC payload. Can you try and capture the pulseaudio log (pulseaudio -vvvv 2> log.txt). Don't send the log, which can be pretty long, just a link on something like pastebin. -Pierre From nathanael at gnat.ca Thu Jan 27 07:55:01 2011 From: nathanael at gnat.ca (Nathanael D. Noblet) Date: Wed, 26 Jan 2011 23:55:01 -0700 Subject: [gst-devel] problem with pa_stream_make_writeable_size() failed connection terminated In-Reply-To: References: <4D3FBDA4.204@gnat.ca> Message-ID: <4D4116C5.2000107@gnat.ca> On 01/26/2011 10:21 AM, pl bossart wrote: >> So I read from an IP camera via rtspsrc. It sends video in MP4 format >> and Audio in AAC. We're running a beta with our client and recently >> after installing it on location I've run into issues with one location. >> After 5-9 seconds I get >> >> pa_stream_make_writeable_size() failed connection terminated > > This is an issue in the connection with PulseAudio. Since it's PCM at > that point, I don't think it's an issue with the bitrate of the AAC > payload. > Can you try and capture the pulseaudio log (pulseaudio -vvvv 2> > log.txt). Don't send the log, which can be pretty long, just a link on > something like pastebin. Yeah I can't seem to figure out what it could be either. I do have SELinux rules in place, however the messages don't stop when SELinux is disabled. Also in testing, my x86_64 box doesn't exhibit the same behaviour *nearly* as often/reliably as the little netbooks do. In anycase, here's the output from a pulseaudio session where I did get this message.... http://fpaste.org/j0Jo/ I don't see anything obvious but with that many lines I really didn't have a clue what to look for at all. Ask for whatever output you need and I'll do my best to make it happen. Since you feel it is permissions based I'll also explain that my app is setgid in the areas that are starting the gstreamer threads. So it is quite possible that for one reason or another gstreamer/pulse interaction is having issues, however they do go away with higher bitrates which is really odd to me... Perhaps the bitrates/clock-rate cause a change request or something that doesn't happen when they are higher but I'm really at a loss. I may try to debug using gdb to see if I can get a bt... Thanks for the help thus far, I'd really like to understand what is going on... -- Nathanael d. Noblet From bilboed at gmail.com Thu Jan 27 09:09:40 2011 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 27 Jan 2011 09:09:40 +0100 Subject: [gst-devel] Should RTSPSRC correct for time drift when converting the RTP timestamp to gstreamer clock? In-Reply-To: References: Message-ID: <1296115780.2443.9.camel@deumeu> On Wed, 2011-01-26 at 13:25 -0500, Loren Rogers wrote: > Hey gang, > I have a gstreamer program that records audio from an RTP audio > source. I then timestamp the audio frame on the recorder based on the > gstreamer clock (RTSPSRC should have already done the RTP TS > ->gstreamer clock TS conversion). I am finding that the time drifts > (i.e. the most recent gstreamer-clock-based timestamp on this audio > frame might be increasingly early or late relative to timestamp the > current recorder thinks it should be) when timestamping the frames end > to end. I am wondering if this drift may be linked to the clock drift > between the audio device and my recording device. Should the RTSPSRC > element already be correcting for such time drift when converting the > RTP timestamp to the gstreamer clock? The drift should normally be taken into account (i.e. detecting that the remote clock is faster or slower than the local gstreamer clock being used) and compensated in rtspsrc (actually in the jitterbuffer contained within that element). What you might be experiencing though is jitter which is somewhat compensated but not completely. A small test to do to see if it's indeed jitter you're getting to replace your encoding part with " ! queue ! autoaudiosink". If it plays back fine (no noise/pops) then it means you're experiencing jitter (which the audiosinks automatically correct up to 40ms difference). If it's jitter indeed, you need to insert a audiorate with a specified tolerance value before your encoder. Try with values ranging from 10ms to 40ms until it records chunks of 10seconds without glitches. That's because encoders expect a perfect incoming audio stream (without any jitter/gaps/overlaps). Edward > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From david.henningsson at canonical.com Thu Jan 27 11:36:22 2011 From: david.henningsson at canonical.com (David Henningsson) Date: Thu, 27 Jan 2011 11:36:22 +0100 Subject: [gst-devel] problem with pa_stream_make_writeable_size() failed connection terminated In-Reply-To: <4D4116C5.2000107@gnat.ca> References: <4D3FBDA4.204@gnat.ca> <4D4116C5.2000107@gnat.ca> Message-ID: <4D414AA6.9000609@canonical.com> On 2011-01-27 07:55, Nathanael D. Noblet wrote: > On 01/26/2011 10:21 AM, pl bossart wrote: >>> So I read from an IP camera via rtspsrc. It sends video in MP4 format >>> and Audio in AAC. We're running a beta with our client and recently >>> after installing it on location I've run into issues with one location. >>> After 5-9 seconds I get >>> >>> pa_stream_make_writeable_size() failed connection terminated >> >> This is an issue in the connection with PulseAudio. Since it's PCM at >> that point, I don't think it's an issue with the bitrate of the AAC >> payload. >> Can you try and capture the pulseaudio log (pulseaudio -vvvv 2> >> log.txt). Don't send the log, which can be pretty long, just a link on >> something like pastebin. > > Yeah I can't seem to figure out what it could be either. I do have > SELinux rules in place, however the messages don't stop when SELinux is > disabled. Also in testing, my x86_64 box doesn't exhibit the same > behaviour *nearly* as often/reliably as the little netbooks do. In > anycase, here's the output from a pulseaudio session where I did get > this message.... > > http://fpaste.org/j0Jo/ > > > I don't see anything obvious but with that many lines I really didn't > have a clue what to look for at all. Ask for whatever output you need > and I'll do my best to make it happen. > > Since you feel it is permissions based I'll also explain that my app is > setgid in the areas that are starting the gstreamer threads. So it is > quite possible that for one reason or another gstreamer/pulse > interaction is having issues, however they do go away with higher > bitrates which is really odd to me... Perhaps the bitrates/clock-rate > cause a change request or something that doesn't happen when they are > higher but I'm really at a loss. I may try to debug using gdb to see if > I can get a bt... > > Thanks for the help thus far, I'd really like to understand what is > going on... PulseAudio eats too much CPU in RT-prio time and ends up getting killed by the kernel. One way to counteract this is for gstreamer to send few big packets instead of many small ones, so soes the patch I posted here recently help to resolve this symptom? See http://gstreamer-devel.966125.n4.nabble.com/attachment/3207515/0/default-buffer-size.patch -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic From idan_kahlon at hotmail.com Thu Jan 27 11:55:28 2011 From: idan_kahlon at hotmail.com (Idan) Date: Thu, 27 Jan 2011 02:55:28 -0800 (PST) Subject: [gst-devel] streaming f4v stream to rtmp server(fms) Message-ID: <1296125728192-3241961.post@n4.nabble.com> Hi all I am trying to stream an mp4 stream to an fms server using librtmp. i am sending only the data that follows the mdat box, and it dosen't work(i am sending data to the server but unable to watch the stream) i am using the following pipeline: gst-launch-0.10 -v --gst-debug-level=2 videotestsrc ! x264enc ! mp4mux ! customelement ip=host stream="my_stream2" where customelement is an element i am using to transfer data to the rtmp server any help will be appreciated? Thanks Idan -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/streaming-f4v-stream-to-rtmp-server-fms-tp3241961p3241961.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From giorgio9 at libero.it Thu Jan 27 12:34:30 2011 From: giorgio9 at libero.it (giorgino) Date: Thu, 27 Jan 2011 03:34:30 -0800 (PST) Subject: [gst-devel] C code for rtp h264 decoding. I can't find how to solve the error. Read is insteresting Message-ID: <1296128070433-3242017.post@n4.nabble.com> Hi All. I have two pc where one is the sender and the other the receiver. In the sender I use this pipe gst-launch -v gstrtpbin name=rtpbin v4l2src ! video/x-raw-yuv,width=320,height=240! queue ! videorate ! ffmpegcolorspace ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.100.196 ts-offset=0 name=vrtpsink rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.100.196 sync=false async=false name=vrtcpsink udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 and it works well. If in the receiver I use the following pipe the receiver works well st-launch -v gstrtpbin name=rtpbin latency=200 udpsrc caps=application/x-rtp,media=video,clock-rate=90000,encoding-name=H264 port=5000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! ffdec_h264 ! ffmpegcolorspace ! autovideosink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=127.0.0.1 sync=false async=false Now I need to map this pipe into a C file and I use the following source #include #include #include /* the caps of the sender RTP stream. This is usually negotiated out of band with * SDP or RTSP. */ #define VIDEO_CAPS "application/x-rtp,media=(string)video,clock-rate=(int)9000,encoding-name=(string)H264" //#define VIDEO_CAPS "application/x-rtp,media=video,clock-rate=9000,encoding-name=H264" #define VIDEO_DEPAY "rtph264depay" #define VIDEO_DEC "ffdec_h264" #define VIDEO_SINK "autovideosink" /* the destination machine to send RTCP to. This is the address of the sender and * is used to send back the RTCP reports of this receiver. If the data is sent * from another machine, change this address. */ #define DEST_HOST "127.0.0.1" /* print the stats of a source */ static void print_source_stats (GObject * source) { GstStructure *stats; gchar *str; g_return_if_fail (source != NULL); /* get the source stats */ g_object_get (source, "stats", &stats, NULL); /* simply dump the stats structure */ str = gst_structure_to_string (stats); g_print ("source stats: %s\n", str); gst_structure_free (stats); g_free (str); } /* will be called when gstrtpbin signals on-ssrc-active. It means that an RTCP * packet was received from another source. */ static void on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint ssrc, GstElement * depay) { GObject *session, *isrc, *osrc; g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc); /* get the right session */ g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session); /* get the internal source (the SSRC allocated to us, the receiver */ g_object_get (session, "internal-source", &isrc, NULL); print_source_stats (isrc); /* get the remote source that sent us RTCP */ g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc); print_source_stats (osrc); } /* will be called when rtpbin has validated a payload that we can depayload */ static void pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay) { GstPad *sinkpad; GstPadLinkReturn lres; g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad)); sinkpad = gst_element_get_static_pad (depay, "sink"); g_assert (sinkpad); lres = gst_pad_link (new_pad, sinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (sinkpad); } int main (int argc, char *argv[]) { GstElement *rtpbin, *rtpsrc, *rtcpsrc, *rtcpsink; GstElement *videodepay, *videodec, //*videores, *videoconv, *videosink; GstElement *pipeline; GMainLoop *loop; GstCaps *caps; gboolean res; GstPadLinkReturn lres; GstPad *srcpad, *sinkpad; /* always init first */ gst_init (&argc, &argv); /* the pipeline to hold everything */ pipeline = gst_pipeline_new (NULL); g_assert (pipeline); /* the udp src and source we will use for RTP and RTCP */ rtpsrc = gst_element_factory_make ("udpsrc", "rtpsrc"); g_assert (rtpsrc); g_object_set (rtpsrc, "port", 5000, NULL); /* we need to set caps on the udpsrc for the RTP data */ caps = gst_caps_from_string (VIDEO_CAPS); g_object_set (rtpsrc, "caps", caps, NULL); gst_caps_unref (caps); rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc"); g_assert (rtcpsrc); g_object_set (rtcpsrc, "port", 5001, NULL); rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink"); g_assert (rtcpsink); g_object_set (rtcpsink, "port", 5005, "host", DEST_HOST, NULL); /* no need for synchronisation or preroll on the RTCP sink */ g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL); gst_bin_add_many (GST_BIN (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL); /* the depayloading and decoding */ videodepay = gst_element_factory_make (VIDEO_DEPAY, "videodepay"); g_assert (videodepay); videodec = gst_element_factory_make (VIDEO_DEC, "videodec"); g_assert (videodec); /* the audio playback and format conversion */ videoconv = gst_element_factory_make ("ffmpegcolorspace", "videoconv"); g_assert (videoconv); /* audiores = gst_element_factory_make ("audioresample", "audiores"); g_assert (audiores); */ videosink = gst_element_factory_make (VIDEO_SINK, "videosink"); g_assert (videosink); /* add depayloading and playback to the pipeline and link */ gst_bin_add_many (GST_BIN (pipeline), videodepay, videodec, videoconv, /*videores,*/ videosink, NULL); res = gst_element_link_many (videodepay, videodec, videoconv, /*videores,*/videosink, NULL); g_assert (res == TRUE); /* the rtpbin element */ rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin"); g_assert (rtpbin); g_object_set (G_OBJECT (rtpbin),"latency",200,NULL); gst_bin_add (GST_BIN (pipeline), rtpbin); /* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */ srcpad = gst_element_get_static_pad (rtpsrc, "src"); sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0"); lres = gst_pad_link (srcpad, sinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (srcpad); /* get an RTCP sinkpad in session 0 */ srcpad = gst_element_get_static_pad (rtcpsrc, "src"); sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0"); lres = gst_pad_link (srcpad, sinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (srcpad); gst_object_unref (sinkpad); /* get an RTCP srcpad for sending RTCP back to the sender */ srcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0"); sinkpad = gst_element_get_static_pad (rtcpsink, "sink"); lres = gst_pad_link (srcpad, sinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (sinkpad); /* the RTP pad that we have to connect to the depayloader will be created * dynamically so we connect to the pad-added signal, pass the depayloader as * user_data so that we can link to it. */ g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), videodepay); /* give some stats when we receive RTCP */ //g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_cb),videodepay); /* set the pipeline to playing */ g_print ("starting receiver pipeline\n"); gst_element_set_state (pipeline, GST_STATE_PLAYING); /* we need to run a GLib main loop to get the messages */ loop = g_main_loop_new (NULL, FALSE); g_main_loop_run (loop); g_print ("stopping receiver pipeline\n"); gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); return 0; } When I launch it I receive the following error ERROR:rtpclient.c::pad_added_cb: assertion failed: (lres == GST_PAD_LINK_OK) How I can solve the problem? Do you have any ideas? G. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/C-code-for-rtp-h264-decoding-I-can-t-find-how-to-solve-the-error-Read-is-insteresting-tp3242017p3242017.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ensonic at hora-obscura.de Thu Jan 27 12:59:40 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Thu, 27 Jan 2011 13:59:40 +0200 Subject: [gst-devel] Does filesrc provide any caching internally? In-Reply-To: <1295973931986-3236487.post@n4.nabble.com> References: <1295973931986-3236487.post@n4.nabble.com> Message-ID: <4D415E2C.90203@hora-obscura.de> On 25.01.2011 18:45, wl2776 wrote: > When seeking, MPEG-PS demuxer (mpegpsdemux) repeatedly requests data from its > upstream peer element using gst_pad_pull_range(). > However, seeking is performed rather quickly in my playbin2 pipeline. > Is it because filesrc caches data? filesrc isn't but usually the filesystem does. Stefan From ensonic at hora-obscura.de Thu Jan 27 13:05:13 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Thu, 27 Jan 2011 14:05:13 +0200 Subject: [gst-devel] gst-opencv multi thread problem In-Reply-To: <1295970374.2171.16.camel@zingle> References: <1295970374.2171.16.camel@zingle> Message-ID: <4D415F79.1010105@hora-obscura.de> On 25.01.2011 17:46, Tim-Philipp M?ller wrote: > On Tue, 2011-01-25 at 14:58 +0100, Robert Jobbagy wrote: > >> What do you think about this. >> > I think you should file a bug :) > > http://gstreamer.freedesktop.org/bugs/ > > Cheers > -tim > and attach your patch :) Stefan From jobbagy.robert at gmail.com Thu Jan 27 13:10:21 2011 From: jobbagy.robert at gmail.com (Robert Jobbagy) Date: Thu, 27 Jan 2011 13:10:21 +0100 Subject: [gst-devel] gst-opencv multi thread problem In-Reply-To: <4D415F79.1010105@hora-obscura.de> References: <1295970374.2171.16.camel@zingle> <4D415F79.1010105@hora-obscura.de> Message-ID: I realised the code what I use , it has some bug. 2011/1/27 Stefan Kost : > On 25.01.2011 17:46, Tim-Philipp M?ller wrote: >> On Tue, 2011-01-25 at 14:58 +0100, Robert Jobbagy wrote: >> >>> What do you think about this. >>> >> I think you should file a bug :) >> >> http://gstreamer.freedesktop.org/bugs/ >> >> Cheers >> ?-tim >> > and attach your patch :) > > Stefan > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Best Regards, Robert From t.i.m at zen.co.uk Thu Jan 27 13:40:08 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 27 Jan 2011 12:40:08 +0000 Subject: [gst-devel] Let's move mailing lists to freedesktop.org! Message-ID: <1296132008.32303.11.camel@zingle> Hi, I think we should move our mailing lists to freedesktop.org, for two reasons: 1. it's the only bit of GStreamer infrastructure still on SourceForge (we host code and website on fdo, bugzilla on gnome.org and mailing lists on SF), and our "project identity" is freedesktop.org-based. It's just less confusing. 2. we've had reliability problems with the SF mailing lists, some of which we don't expect to be able to get sorted out in the near future, if ever (ie. the SourceForge mail servers regularly reject commit mails coming from the fdo mail server because they receive infected mail from that host they believe to contain virus attachements). From bilboed at gmail.com Thu Jan 27 13:52:50 2011 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 27 Jan 2011 13:52:50 +0100 Subject: [gst-devel] Let's move mailing lists to freedesktop.org! In-Reply-To: <1296132008.32303.11.camel@zingle> References: <1296132008.32303.11.camel@zingle> Message-ID: <1296132770.31112.3.camel@localhost> On Thu, 2011-01-27 at 12:40 +0000, Tim-Philipp M?ller wrote: > Hi, > > I think we should move our mailing lists to freedesktop.org, for two > reasons: > > 1. it's the only bit of GStreamer infrastructure still on SourceForge > (we host code and website on fdo, bugzilla on gnome.org and mailing > lists on SF), and our "project identity" is freedesktop.org-based. It's > just less confusing. > > 2. we've had reliability problems with the SF mailing lists, some of > which we don't expect to be able to get sorted out in the near future, > if ever (ie. the SourceForge mail servers regularly reject commit mails > coming from the fdo mail server because they receive infected mail from > that host they believe to contain virus attachements). > \o/ YES YES YES YES YES YES YES \o/ Edward > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > 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 Jan 27 13:56:03 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 27 Jan 2011 12:56:03 +0000 Subject: [gst-devel] Let's move mailing lists to freedesktop.org! In-Reply-To: <1296132008.32303.11.camel@zingle> References: <1296132008.32303.11.camel@zingle> Message-ID: <1296132963.32303.27.camel@zingle> (Looks like that got sent off unfinished, so here goes the rest:) Hi, I think we should move our mailing lists to freedesktop.org, for two reasons: 1. it's the only bit of GStreamer infrastructure still on SourceForge (we host code and website on fdo, bugzilla on gnome.org and mailing lists on SF), and our "project identity" is freedesktop.org-based. It's just less confusing. 2. we've had reliability problems with the SF mailing lists, some of which we don't expect to be able to get sorted out in the near future, if ever (ie. the SourceForge mail servers regularly reject commit mails coming from the fdo mail server because they receive infected mail from that host they believe to contain virus attachements). So, in detail: gstreamer-announce => gstreamer-announce at lists.freedesktop.org gstreamer-bugs => gstreamer-bugs at lists.freedesktop.org gstreamer-cvs => gstreamer-commits at lists.freedesktop.org gstreamer-openmax => gstreamer-openmax at lists.freedesktop.org gstreamer-daily => unused, don't move These two I'm not sure about - obviously we want to move them over, but maybe not in the current form: gstreamer-devel => gstreamer-devel at lists.freedesktop.org (?) gstreamer-embedded => gstreamer-embedded at lists.freedesktop.org (?) First the -devel list is a bit of a misnomer, because we don't really use it at all for development purposes. Maybe it should just be renamed to gstreamer-discuss? The list is fairly high traffic and high noise, so it would be nice if there was a way to split it up a bit, but I can't think of something suitable. Maybe there should be a -newbies list? Or an app-development vs. plugin-development list? But then that's unlikely to change the quality or overall volume of the posts, so maybe there's not so much point. As for the gstreamer-embedded list, do we want to keep it? It seems to have some traffic, but many questions are just cross-posted to -devel anyway and embedded topics haven't moved from -devel to -embedded, nor do I have the feeling that some kind of distinct embedded community has developed there. I haven't compared the subscriber lists though to see if people subscribed to -embedded are mostly subscribed to -devel as well. Cheers -Tim From lane at brooks.nu Thu Jan 27 14:53:42 2011 From: lane at brooks.nu (Lane Brooks) Date: Thu, 27 Jan 2011 06:53:42 -0700 Subject: [gst-devel] Let's move mailing lists to freedesktop.org! In-Reply-To: <1296132963.32303.27.camel@zingle> References: <1296132008.32303.11.camel@zingle> <1296132963.32303.27.camel@zingle> Message-ID: <4D4178E6.5040805@brooks.nu> On 01/27/2011 05:56 AM, Tim-Philipp M?ller wrote: > (Looks like that got sent off unfinished, so here goes the rest:) > > Hi, > > I think we should move our mailing lists to freedesktop.org, > > These two I'm not sure about - obviously we want to move them over, but > maybe not in the current form: > > gstreamer-devel => gstreamer-devel at lists.freedesktop.org (?) > gstreamer-embedded => gstreamer-embedded at lists.freedesktop.org (?) > > > First the -devel list is a bit of a misnomer, because we don't really > use it at all for development purposes. Maybe it should just be renamed > to gstreamer-discuss? > A lot of projects have an -users list for user related questions/discussion and a -devel list for development discussion. Can you move the archives as well? Lane From ensonic at hora-obscura.de Thu Jan 27 14:55:16 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Thu, 27 Jan 2011 15:55:16 +0200 Subject: [gst-devel] Let's move mailing lists to freedesktop.org! In-Reply-To: <1296132008.32303.11.camel@zingle> References: <1296132008.32303.11.camel@zingle> Message-ID: <4D417944.6090102@hora-obscura.de> Hi, On 27.01.2011 14:40, Tim-Philipp M?ller wrote: > Hi, > > I think we should move our mailing lists to freedesktop.org, for two > reasons: > > 1. it's the only bit of GStreamer infrastructure still on SourceForge > (we host code and website on fdo, bugzilla on gnome.org and mailing > lists on SF), and our "project identity" is freedesktop.org-based. It's > just less confusing. > > 2. we've had reliability problems with the SF mailing lists, some of > which we don't expect to be able to get sorted out in the near future, > if ever (ie. the SourceForge mail servers regularly reject commit mails > coming from the fdo mail server because they receive infected mail from > that host they believe to contain virus attachements). > just for the info, the mails sf.net got from fdo did indeed contained viruses. I have talked to fdo admins and they confirmed the issue and looking into it. Maybe you want to confirm its fixed before moving the lists. Stefan From xavier.bestel at free.fr Thu Jan 27 15:11:25 2011 From: xavier.bestel at free.fr (Xavier Bestel) Date: Thu, 27 Jan 2011 15:11:25 +0100 Subject: [gst-devel] Let's move mailing lists to freedesktop.org! In-Reply-To: <4D4178E6.5040805@brooks.nu> References: <1296132008.32303.11.camel@zingle> <1296132963.32303.27.camel@zingle> <4D4178E6.5040805@brooks.nu> Message-ID: <1296137485.27398.115.camel@skunk> On Thu, 2011-01-27 at 06:53 -0700, Lane Brooks wrote: > On 01/27/2011 05:56 AM, Tim-Philipp M?ller wrote: > > (Looks like that got sent off unfinished, so here goes the rest:) > > > > Hi, > > > > I think we should move our mailing lists to freedesktop.org, > > > > These two I'm not sure about - obviously we want to move them over, but > > maybe not in the current form: > > > > gstreamer-devel => gstreamer-devel at lists.freedesktop.org (?) > > gstreamer-embedded => gstreamer-embedded at lists.freedesktop.org (?) > > > > > > First the -devel list is a bit of a misnomer, because we don't really > > use it at all for development purposes. Maybe it should just be renamed > > to gstreamer-discuss? > > > > A lot of projects have an -users list for user related > questions/discussion and a -devel list for development discussion. ... and I think for projects like Gst where users are developers it's a flawed way. Witness gtk-devel which is regularly the landing spot for plain users which just want to know how to use Gtk+. I think a better name for gst developers is called. Something like gstreamer-hacking. Xav From gibrovacco at gmail.com Thu Jan 27 15:13:07 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Thu, 27 Jan 2011 16:13:07 +0200 Subject: [gst-devel] Should RTSPSRC correct for time drift when converting the RTP timestamp to gstreamer clock? In-Reply-To: <1296115780.2443.9.camel@deumeu> References: <1296115780.2443.9.camel@deumeu> Message-ID: Hi, On Thu, Jan 27, 2011 at 10:09 AM, Edward Hervey wrote: > On Wed, 2011-01-26 at 13:25 -0500, Loren Rogers wrote: >> Hey gang, >> I have a gstreamer program that records audio from an RTP audio >> source. ?I then timestamp the audio frame on the recorder based on the >> gstreamer clock (RTSPSRC should have already done the RTP TS >> ->gstreamer clock TS conversion). ?I am finding that the time drifts >> (i.e. the most recent gstreamer-clock-based timestamp on this audio >> frame might be increasingly early or late relative to timestamp the >> current recorder thinks it should be) when timestamping the frames end >> to end. ?I am wondering if this drift may be linked to the clock drift >> between the audio device and my recording device. ?Should the RTSPSRC >> element already be correcting for such time drift when converting the >> RTP timestamp to the gstreamer clock? > > ?The drift should normally be taken into account (i.e. detecting that > the remote clock is faster or slower than the local gstreamer clock > being used) and compensated in rtspsrc (actually in the jitterbuffer > contained within that element). On top of this, a better synchronisation between sender and receiver clock can be granted using the RTCP NTP feature. You'll need to enable it from the server and set the "ntp-sync" on the gstrtpbin used as session in rtspsrc. Regards > ?What you might be experiencing though is jitter which is somewhat > compensated but not completely. > > ?A small test to do to see if it's indeed jitter you're getting to > replace your encoding part with " ! queue ! autoaudiosink". If it plays > back fine (no noise/pops) then it means you're experiencing jitter > (which the audiosinks automatically correct up to 40ms difference). > > ?If it's jitter indeed, you need to insert a audiorate with a specified > tolerance value before your encoder. Try with values ranging from 10ms > to 40ms until it records chunks of 10seconds without glitches. That's > because encoders expect a perfect incoming audio stream (without any > jitter/gaps/overlaps). > > ? ?Edward > >> >> ------------------------------------------------------------------------------ >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> Finally, a world-class log management solution at an even better price-free! >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> February 28th, so secure your free ArcSight Logger TODAY! >> http://p.sf.net/sfu/arcsight-sfd2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > 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 Jan 27 15:19:30 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 27 Jan 2011 14:19:30 +0000 Subject: [gst-devel] Let's move mailing lists to freedesktop.org! In-Reply-To: <4D4178E6.5040805@brooks.nu> References: <1296132008.32303.11.camel@zingle> <1296132963.32303.27.camel@zingle> <4D4178E6.5040805@brooks.nu> Message-ID: <1296137970.32303.33.camel@zingle> On Thu, 2011-01-27 at 06:53 -0700, Lane Brooks wrote: > A lot of projects have an -users list for user related > questions/discussion and a -devel list for development discussion. True, but most of those projects probably actually use that -devel list then. We don't really use the mailing list for development discussions at all, and I don't see that changing (no follow-ups arguing that we should please, it's just how it is). Also, since we're a library, almost all of our users are actually developers, so it's not all that clear cut, and many people will just post wherever they think they'll get some attention. But mostly I don't see the point because I expect a -devel list to have no meaningful traffic (other than misdirected -users posts). At the moment I'm leaning towards just renaming -devel to -discuss. > Can you move the archives as well? Of course. Cheers -Tim From lubensky at cognitec.com Thu Jan 27 15:28:08 2011 From: lubensky at cognitec.com (Andreas Lubensky) Date: Thu, 27 Jan 2011 15:28:08 +0100 Subject: [gst-devel] Reconnecting to a rtsp stream after EOS Message-ID: <201101271528.10141.lubensky@cognitec.com> Hello, I am using an uridecobin to decode all kinds of videos, including rtsp streams. Usually I want videos to loop seamlessly, so I seek the video with GST_SEEK_FLAG_SEGMANT to get a GST_MESSAGE_SEGMENT_DONE and seek back to the beginning. I use vlc to broadcast the rtsp stream and there looping basically means that the stream is closed and restarted again. That gives me a GST_MESSAGE_EOS. Unfortunately just seeking to the beginning or setting the pipeline state to GST_STATE_NULL and GST_STATE_PLAYING to reconnect the stream has no effect. I guess I can tear down and build up the whole pipeline again, but is there a cleaner way? best regards, Andreas Lubensky From bilboed at gmail.com Thu Jan 27 15:32:48 2011 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 27 Jan 2011 15:32:48 +0100 Subject: [gst-devel] Let's move mailing lists to freedesktop.org! In-Reply-To: <1296132963.32303.27.camel@zingle> References: <1296132008.32303.11.camel@zingle> <1296132963.32303.27.camel@zingle> Message-ID: <1296138768.31112.8.camel@localhost> On Thu, 2011-01-27 at 12:56 +0000, Tim-Philipp M?ller wrote: > (Looks like that got sent off unfinished, so here goes the rest:) > > Hi, > > I think we should move our mailing lists to freedesktop.org, for two > reasons: > > 1. it's the only bit of GStreamer infrastructure still on SourceForge > (we host code and website on fdo, bugzilla on gnome.org and mailing > lists on SF), and our "project identity" is freedesktop.org-based. It's > just less confusing. > > 2. we've had reliability problems with the SF mailing lists, some of > which we don't expect to be able to get sorted out in the near future, > if ever (ie. the SourceForge mail servers regularly reject commit mails > coming from the fdo mail server because they receive infected mail from > that host they believe to contain virus attachements). > > So, in detail: > > gstreamer-announce => gstreamer-announce at lists.freedesktop.org > gstreamer-bugs => gstreamer-bugs at lists.freedesktop.org > gstreamer-cvs => gstreamer-commits at lists.freedesktop.org > gstreamer-openmax => gstreamer-openmax at lists.freedesktop.org > > gstreamer-daily => unused, don't move > > > These two I'm not sure about - obviously we want to move them over, but > maybe not in the current form: > > gstreamer-devel => gstreamer-devel at lists.freedesktop.org (?) > gstreamer-embedded => gstreamer-embedded at lists.freedesktop.org (?) > > > First the -devel list is a bit of a misnomer, because we don't really > use it at all for development purposes. Maybe it should just be renamed > to gstreamer-discuss? > > The list is fairly high traffic and high noise, so it would be nice if > there was a way to split it up a bit, but I can't think of something > suitable. Maybe there should be a -newbies list? Or an app-development > vs. plugin-development list? But then that's unlikely to change the > quality or overall volume of the posts, so maybe there's not so much > point. > > As for the gstreamer-embedded list, do we want to keep it? It seems to > have some traffic, but many questions are just cross-posted to -devel > anyway and embedded topics haven't moved from -devel to -embedded, nor > do I have the feeling that some kind of distinct embedded community has > developed there. I haven't compared the subscriber lists though to see > if people subscribed to -embedded are mostly subscribed to -devel as > well. In the past three years, there have been around 1300 mail on gst-embedded. In the equivalent period, there were over 13000 on gst-devel. I'd be fine with just having one mailing list for all gstreamer discussions. As for the name, gstreamer-devel is fine by me too. Edward > > Cheers > -Tim > > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From nathanael at gnat.ca Thu Jan 27 17:50:03 2011 From: nathanael at gnat.ca (Nathanael D. Noblet) Date: Thu, 27 Jan 2011 09:50:03 -0700 Subject: [gst-devel] Reconnecting to a rtsp stream after EOS In-Reply-To: <201101271528.10141.lubensky@cognitec.com> References: <201101271528.10141.lubensky@cognitec.com> Message-ID: <4D41A23B.50000@gnat.ca> On 01/27/2011 07:28 AM, Andreas Lubensky wrote: > Hello, > > I am using an uridecobin to decode all kinds of videos, including rtsp > streams. > Usually I want videos to loop seamlessly, so I seek the video with > GST_SEEK_FLAG_SEGMANT to get a GST_MESSAGE_SEGMENT_DONE > and seek back to the beginning. > I use vlc to broadcast the rtsp stream and there looping basically means that > the stream is closed and restarted again. That gives me a > GST_MESSAGE_EOS. Unfortunately just seeking to the beginning or setting > the pipeline state to GST_STATE_NULL and GST_STATE_PLAYING to > reconnect the stream has no effect. Don't set state to NULL, AFAIK that removes all settings (like the uri etc). Have you tried setting it to READY and then PLAYING? From nathanael at gnat.ca Thu Jan 27 17:52:13 2011 From: nathanael at gnat.ca (Nathanael D. Noblet) Date: Thu, 27 Jan 2011 09:52:13 -0700 Subject: [gst-devel] C code for rtp h264 decoding. I can't find how to solve the error. Read is insteresting In-Reply-To: <1296128070433-3242017.post@n4.nabble.com> References: <1296128070433-3242017.post@n4.nabble.com> Message-ID: <4D41A2BD.8030801@gnat.ca> On 01/27/2011 04:34 AM, giorgino wrote: > When I launch it I receive the following error > ERROR:rtpclient.c::pad_added_cb: assertion failed: (lres == GST_PAD_LINK_OK) > > How I can solve the problem? Do you have any ideas? How bout you compare the GST_PAD_X return to something so you know why it failed? http://library.gnome.org/devel/gstreamer/unstable/GstPad.html#GstPadLinkReturn From nathanael at gnat.ca Thu Jan 27 18:00:03 2011 From: nathanael at gnat.ca (Nathanael D. Noblet) Date: Thu, 27 Jan 2011 10:00:03 -0700 Subject: [gst-devel] problem with pa_stream_make_writeable_size() failed connection terminated In-Reply-To: <4D414AA6.9000609@canonical.com> References: <4D3FBDA4.204@gnat.ca> <4D4116C5.2000107@gnat.ca> <4D414AA6.9000609@canonical.com> Message-ID: <4D41A493.7090903@gnat.ca> On 01/27/2011 03:36 AM, David Henningsson wrote: >> Thanks for the help thus far, I'd really like to understand what is >> going on... > > PulseAudio eats too much CPU in RT-prio time and ends up getting killed > by the kernel. One way to counteract this is for gstreamer to send few > big packets instead of many small ones, so soes the patch I posted here > recently help to resolve this symptom? > > See > http://gstreamer-devel.966125.n4.nabble.com/attachment/3207515/0/default-buffer-size.patch I don't think pulse is dying as I've killed pulse manually and the error message I get is different, however I will patch my copy with the above and let you know if it helps at all. -- Nathanael d. Noblet From gibrovacco at gmail.com Thu Jan 27 18:28:50 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Thu, 27 Jan 2011 19:28:50 +0200 Subject: [gst-devel] C code for rtp h264 decoding. I can't find how to solve the error. Read is insteresting In-Reply-To: <1296128070433-3242017.post@n4.nabble.com> References: <1296128070433-3242017.post@n4.nabble.com> Message-ID: Hi, On Thu, Jan 27, 2011 at 1:34 PM, giorgino wrote: ..snip.. > #define VIDEO_CAPS > "application/x-rtp,media=(string)video,clock-rate=(int)9000,encoding-name=(string)H264" typo, s/9000/90000 or, that is the same: "application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" Regards > //#define VIDEO_CAPS > "application/x-rtp,media=video,clock-rate=9000,encoding-name=H264" > > #define VIDEO_DEPAY "rtph264depay" > #define VIDEO_DEC ? "ffdec_h264" > #define VIDEO_SINK ?"autovideosink" > > /* the destination machine to send RTCP to. This is the address of the > sender and > ?* is used to send back the RTCP reports of this receiver. If the data is > sent > ?* from another machine, change this address. */ > #define DEST_HOST "127.0.0.1" > > /* print the stats of a source */ > static void print_source_stats (GObject * source) { > ?GstStructure *stats; > ?gchar *str; > > ?g_return_if_fail (source != NULL); > > ?/* get the source stats */ > ?g_object_get (source, "stats", &stats, NULL); > > ?/* simply dump the stats structure */ > ?str = gst_structure_to_string (stats); > ?g_print ("source stats: %s\n", str); > > ?gst_structure_free (stats); > ?g_free (str); > } > > /* will be called when gstrtpbin signals on-ssrc-active. It means that an > RTCP > ?* packet was received from another source. */ > static void on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint > ssrc, GstElement * depay) { > > ? ?GObject *session, *isrc, *osrc; > ? ?g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc); > > ?/* get the right session */ > ?g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session); > > ?/* get the internal source (the SSRC allocated to us, the receiver */ > ?g_object_get (session, "internal-source", &isrc, NULL); > ?print_source_stats (isrc); > > ?/* get the remote source that sent us RTCP */ > ?g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc); > ?print_source_stats (osrc); > } > > /* will be called when rtpbin has validated a payload that we can depayload > */ > static void > pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay) > { > ?GstPad *sinkpad; > ?GstPadLinkReturn lres; > > ?g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad)); > > ?sinkpad = gst_element_get_static_pad (depay, "sink"); > ?g_assert (sinkpad); > > ?lres = gst_pad_link (new_pad, sinkpad); > ?g_assert (lres == GST_PAD_LINK_OK); > ?gst_object_unref (sinkpad); > > } > > > int main (int argc, char *argv[]) > { > ?GstElement *rtpbin, *rtpsrc, *rtcpsrc, *rtcpsink; > ?GstElement *videodepay, > ? ? ? ? ? ? *videodec, > ? ? ? ? ? ? //*videores, > ? ? ? ? ? ? *videoconv, > ? ? ? ? ? ? *videosink; > > ?GstElement *pipeline; > ?GMainLoop *loop; > ?GstCaps *caps; > ?gboolean res; > ?GstPadLinkReturn lres; > ?GstPad *srcpad, *sinkpad; > > ?/* always init first */ > ?gst_init (&argc, &argv); > > ?/* the pipeline to hold everything */ > ?pipeline = gst_pipeline_new (NULL); > ?g_assert (pipeline); > > ?/* the udp src and source we will use for RTP and RTCP */ > ?rtpsrc = gst_element_factory_make ("udpsrc", "rtpsrc"); > ?g_assert (rtpsrc); > ?g_object_set (rtpsrc, "port", 5000, NULL); > ?/* we need to set caps on the udpsrc for the RTP data */ > ?caps = gst_caps_from_string (VIDEO_CAPS); > ?g_object_set (rtpsrc, "caps", caps, NULL); > ?gst_caps_unref (caps); > > ?rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc"); > ?g_assert (rtcpsrc); > ?g_object_set (rtcpsrc, "port", 5001, NULL); > > ?rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink"); > ?g_assert (rtcpsink); > ?g_object_set (rtcpsink, "port", 5005, "host", DEST_HOST, NULL); > ?/* no need for synchronisation or preroll on the RTCP sink */ > ?g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL); > > ?gst_bin_add_many (GST_BIN (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL); > > ?/* the depayloading and decoding */ > ?videodepay = gst_element_factory_make (VIDEO_DEPAY, "videodepay"); > ?g_assert (videodepay); > ?videodec = gst_element_factory_make (VIDEO_DEC, "videodec"); > ?g_assert (videodec); > ?/* the audio playback and format conversion */ > ?videoconv = gst_element_factory_make ("ffmpegcolorspace", "videoconv"); > ?g_assert (videoconv); > /* > ?audiores = gst_element_factory_make ("audioresample", "audiores"); > ?g_assert (audiores); > */ > ?videosink = gst_element_factory_make (VIDEO_SINK, "videosink"); > ?g_assert (videosink); > > ?/* add depayloading and playback to the pipeline and link */ > ?gst_bin_add_many (GST_BIN (pipeline), videodepay, videodec, videoconv, > /*videores,*/ videosink, NULL); > > ?res = gst_element_link_many (videodepay, videodec, videoconv, > /*videores,*/videosink, NULL); > ?g_assert (res == TRUE); > > ?/* the rtpbin element */ > ?rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin"); > ?g_assert (rtpbin); > > ?g_object_set (G_OBJECT (rtpbin),"latency",200,NULL); > > ?gst_bin_add (GST_BIN (pipeline), rtpbin); > > ?/* now link all to the rtpbin, start by getting an RTP sinkpad for session > 0 */ > ?srcpad = gst_element_get_static_pad (rtpsrc, "src"); > ?sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0"); > ?lres = gst_pad_link (srcpad, sinkpad); > ?g_assert (lres == GST_PAD_LINK_OK); > ?gst_object_unref (srcpad); > > ?/* get an RTCP sinkpad in session 0 */ > ?srcpad = gst_element_get_static_pad (rtcpsrc, "src"); > ?sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0"); > ?lres = gst_pad_link (srcpad, sinkpad); > ?g_assert (lres == GST_PAD_LINK_OK); > ?gst_object_unref (srcpad); > ?gst_object_unref (sinkpad); > > ?/* get an RTCP srcpad for sending RTCP back to the sender */ > ?srcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0"); > ?sinkpad = gst_element_get_static_pad (rtcpsink, "sink"); > ?lres = gst_pad_link (srcpad, sinkpad); > ?g_assert (lres == GST_PAD_LINK_OK); > ?gst_object_unref (sinkpad); > > ?/* the RTP pad that we have to connect to the depayloader will be created > ? * dynamically so we connect to the pad-added signal, pass the depayloader > as > ? * user_data so that we can link to it. */ > ?g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), > videodepay); > > ?/* give some stats when we receive RTCP */ > ?//g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK > (on_ssrc_active_cb),videodepay); > > ?/* set the pipeline to playing */ > ?g_print ("starting receiver pipeline\n"); > ?gst_element_set_state (pipeline, GST_STATE_PLAYING); > > ?/* we need to run a GLib main loop to get the messages */ > ?loop = g_main_loop_new (NULL, FALSE); > ?g_main_loop_run (loop); > > ?g_print ("stopping receiver pipeline\n"); > ?gst_element_set_state (pipeline, GST_STATE_NULL); > > ?gst_object_unref (pipeline); > > ?return 0; > } > > > When I launch it I receive the following error > ERROR:rtpclient.c::pad_added_cb: assertion failed: (lres == GST_PAD_LINK_OK) > > How I can solve the problem? Do you have any ideas? > > G. > > > -- > View this message in context: http://gstreamer-devel.966125.n4.nabble.com/C-code-for-rtp-h264-decoding-I-can-t-find-how-to-solve-the-error-Read-is-insteresting-tp3242017p3242017.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From nathanael at gnat.ca Thu Jan 27 19:55:22 2011 From: nathanael at gnat.ca (Nathanael D. Noblet) Date: Thu, 27 Jan 2011 11:55:22 -0700 Subject: [gst-devel] problem with pa_stream_make_writeable_size() failed connection terminated In-Reply-To: <4D414AA6.9000609@canonical.com> References: <4D3FBDA4.204@gnat.ca> <4D4116C5.2000107@gnat.ca> <4D414AA6.9000609@canonical.com> Message-ID: <4D41BF9A.1030206@gnat.ca> On 01/27/2011 03:36 AM, David Henningsson wrote: > PulseAudio eats too much CPU in RT-prio time and ends up getting killed > by the kernel. One way to counteract this is for gstreamer to send few > big packets instead of many small ones, so soes the patch I posted here > recently help to resolve this symptom? > > See > http://gstreamer-devel.966125.n4.nabble.com/attachment/3207515/0/default-buffer-size.patch This does indeed fix it. Patching the latest release with that patch, and upgrading to it I don't see the error anymore, downgrading and its back. So is there a bug someplace I can voice my experience thus far? From olivier.crete at collabora.co.uk Thu Jan 27 19:58:52 2011 From: olivier.crete at collabora.co.uk (Olivier =?ISO-8859-1?Q?Cr=EAte?=) Date: Thu, 27 Jan 2011 13:58:52 -0500 Subject: [gst-devel] Let's move mailing lists to freedesktop.org! In-Reply-To: <1296132963.32303.27.camel@zingle> References: <1296132008.32303.11.camel@zingle> <1296132963.32303.27.camel@zingle> Message-ID: <1296154732.2892.24.camel@TesterTop4> On Thu, 2011-01-27 at 12:56 +0000, Tim-Philipp M?ller wrote: > gstreamer-bugs => gstreamer-bugs at lists.freedesktop.org What about using a @gnome.bugs fake-email that people can just watch in bugzilla like all the other projects on bugzilla.gnome.org... When bgo migrated from mailing lists to using bugzilla watch, they had some scripts to add all the people who were previously subscribed to the mailing list, I guess someone probably still has it somewhere so we can minimal disruption. -- Olivier Cr?te olivier.crete at collabora.co.uk -------------- 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 t.i.m at zen.co.uk Thu Jan 27 20:12:25 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 27 Jan 2011 19:12:25 +0000 Subject: [gst-devel] bugzilla watch alias (was: Re: Let's move mailing lists to freedesktop.org!) In-Reply-To: <1296154732.2892.24.camel@TesterTop4> References: <1296132008.32303.11.camel@zingle> <1296132963.32303.27.camel@zingle> <1296154732.2892.24.camel@TesterTop4> Message-ID: <1296155545.32303.38.camel@zingle> On Thu, 2011-01-27 at 13:58 -0500, Olivier Cr?te wrote: > > gstreamer-bugs => gstreamer-bugs at lists.freedesktop.org > > What about using a @gnome.bugs fake-email that people can just watch in > bugzilla like all the other projects on bugzilla.gnome.org... That could be done as well (I just watch the gstreamer-bugs at lists.sourceforge.net user in bugzilla, fwiw). A mailing list allows users more options, e.g. receive mail in digests. Cheers -Tim From giorgio9 at libero.it Fri Jan 28 09:07:20 2011 From: giorgio9 at libero.it (giorgino) Date: Fri, 28 Jan 2011 00:07:20 -0800 (PST) Subject: [gst-devel] C code for rtp h264 decoding. I can't find how to solve the error. Read is insteresting In-Reply-To: References: <1296128070433-3242017.post@n4.nabble.com> Message-ID: <1296202040297-3243913.post@n4.nabble.com> Yes you identified the error. The value is 90000 for that reason previuosly it doesn't work. Thanks and have a nice day G. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/C-code-for-rtp-h264-decoding-I-can-t-find-how-to-solve-the-error-Read-is-insteresting-tp3242017p3243913.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From giorgio9 at libero.it Fri Jan 28 09:09:43 2011 From: giorgio9 at libero.it (giorgino) Date: Fri, 28 Jan 2011 00:09:43 -0800 (PST) Subject: [gst-devel] source port of udp sink In-Reply-To: <4D3D67FA.2070202@panicking.kicks-ass.org> References: <1295867325134-3233756.post@n4.nabble.com> <1295869208757-3233814.post@n4.nabble.com> <4D3D67FA.2070202@panicking.kicks-ass.org> Message-ID: <1296202183919-3243915.post@n4.nabble.com> Hi Micheal I have not yet tried your solution but I guess it works, so many thanks. G. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/source-port-of-udp-sink-tp3233756p3243915.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From jobbagy.robert at gmail.com Fri Jan 28 13:02:33 2011 From: jobbagy.robert at gmail.com (Robert Jobbagy) Date: Fri, 28 Jan 2011 13:02:33 +0100 Subject: [gst-devel] gst-opencv app compile error with OpenCV 2.2 Message-ID: Hi Guys, I used my gst-opencv filter with OpenCV 2.0 and worked fine. I tried update to OpenCV 2.1 because I found a bug. But OpenCV 2.1 has the bug too and tried update to OpenCV 2.2. And I gave this compile error: In file included from /usr//include/opencv/cvaux.h:61, from gstfacedetect.h:52, from gstfacedetect.c:66: /usr/local/include/opencv2/legacy/blobtrack.hpp:82: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvVSModule? /usr/local/include/opencv2/legacy/blobtrack.hpp:133: error: expected ?;?, ?,? or ?)? before ?=? token /usr/local/include/opencv2/legacy/blobtrack.hpp:137: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvFGDetector? /usr/local/include/opencv2/legacy/blobtrack.hpp:148: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:149: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:160: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?cvBlob? In file included from /usr//include/opencv/cvaux.h:61, from gstfacedetect.h:52, from gstfacedetect.c:66: /usr/local/include/opencv2/legacy/blobtrack.hpp:180: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobSeq? /usr/local/include/opencv2/legacy/blobtrack.hpp:270: error: expected specifier-qualifier-list before ?CvBlobSeq? /usr/local/include/opencv2/legacy/blobtrack.hpp:273: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackSeq? /usr/local/include/opencv2/legacy/blobtrack.hpp:294: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobDetector? /usr/local/include/opencv2/legacy/blobtrack.hpp:308: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:311: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:312: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:314: error: expected identifier or ?(? before ?:? token /usr/local/include/opencv2/legacy/blobtrack.hpp:319: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?cvDetectedBlob? /usr/local/include/opencv2/legacy/blobtrack.hpp:327: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvObjectDetector? In file included from /usr//include/opencv/cvaux.h:61, from gstfacedetect.h:52, from gstfacedetect.c:66: /usr/local/include/opencv2/legacy/blobtrack.hpp:391: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvImageDrawer? /usr/local/include/opencv2/legacy/blobtrack.hpp:410: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackGen? /usr/local/include/opencv2/legacy/blobtrack.hpp:420: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:427: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:428: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:433: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTracker? /usr/local/include/opencv2/legacy/blobtrack.hpp:503: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:507: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackerOne? /usr/local/include/opencv2/legacy/blobtrack.hpp:524: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:529: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:545: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:546: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:547: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:557: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:560: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:563: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:566: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:569: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:570: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:573: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:580: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token In file included from /usr//include/opencv/cvaux.h:61, from gstfacedetect.h:52, from gstfacedetect.c:66: /usr/local/include/opencv2/legacy/blobtrack.hpp:597: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackPostProc? /usr/local/include/opencv2/legacy/blobtrack.hpp:620: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:628: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackPostProcOne? /usr/local/include/opencv2/legacy/blobtrack.hpp:637: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:641: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:642: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:643: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:648: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackPredictor? /usr/local/include/opencv2/legacy/blobtrack.hpp:656: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:661: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackAnalysis? /usr/local/include/opencv2/legacy/blobtrack.hpp:676: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:684: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackFVGen? /usr/local/include/opencv2/legacy/blobtrack.hpp:701: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackAnalysisOne? /usr/local/include/opencv2/legacy/blobtrack.hpp:712: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:717: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:720: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:723: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:726: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token In file included from /usr//include/opencv/cvaux.h:61, from gstfacedetect.h:52, from gstfacedetect.c:66: /usr/local/include/opencv2/legacy/blobtrack.hpp:743: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:749: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:752: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackAnalysisHeight? /usr/local/include/opencv2/legacy/blobtrack.hpp:762: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackerAuto? /usr/local/include/opencv2/legacy/blobtrack.hpp:777: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:792: error: expected specifier-qualifier-list before ?CvFGDetector? /usr/local/include/opencv2/legacy/blobtrack.hpp:813: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:816: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token In file included from /usr//include/opencv/cvaux.h:61, from gstfacedetect.h:52, from gstfacedetect.c:66: /usr/local/include/opencv2/legacy/blobtrack.hpp:840: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:841: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:842: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:846: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvProb? /usr/local/include/opencv2/legacy/blobtrack.hpp:861: error: expected ?)? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:865: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:866: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:867: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:868: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token /usr/local/include/opencv2/legacy/blobtrack.hpp:874: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token In file included from /usr//include/opencv/cvaux.h:61, from gstfacedetect.h:52, from gstfacedetect.c:66: /usr/local/include/opencv2/legacy/blobtrack.hpp:910: error: expected ?;?, ?,? or ?)? before ?=? token In file included from /usr//include/opencv/cvaux.h:61, from gstfacedetect.h:52, from gstfacedetect.c:66: /usr/local/include/opencv2/legacy/blobtrack.hpp:940: error: expected ?;?, ?,? or ?)? before ?=? token make[3]: *** [libgstfacedetect_la-gstfacedetect.lo] Error 1 the bug why I updated: I would like to use alpha blending. I followed Shervin Emam instructions but it isnt help :( The alpha blending method: I split the input frame into 3 image from the 3 channel. and I use cvAddWeighted function on 3 image and add 3 other image with alpha and beta equal 0.5. and after I merge this images after this method the video is flicking and I see some sub image in the frame etc ... So I think this is a bug. And doesnt solved the update to newer OpenCV version. Any idea ? -- Best Regards, Robert From bsreerenj at gmail.com Fri Jan 28 14:04:28 2011 From: bsreerenj at gmail.com (sreerenj b) Date: Fri, 28 Jan 2011 15:04:28 +0200 Subject: [gst-devel] gst-opencv app compile error with OpenCV 2.2 In-Reply-To: References: Message-ID: Hi Robert, Currently gst-opencv supports only* "opencv >= 2.0.0 opencv <= 2.1.0".* On Fri, Jan 28, 2011 at 2:02 PM, Robert Jobbagy wrote: > Hi Guys, > > I used my gst-opencv filter with OpenCV 2.0 and worked fine. > I tried update to OpenCV 2.1 because I found a bug. > But OpenCV 2.1 has the bug too and tried update to OpenCV 2.2. > > And I gave this compile error: > > In file included from /usr//include/opencv/cvaux.h:61, > from gstfacedetect.h:52, > from gstfacedetect.c:66: > /usr/local/include/opencv2/legacy/blobtrack.hpp:82: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvVSModule? > /usr/local/include/opencv2/legacy/blobtrack.hpp:133: error: expected > ?;?, ?,? or ?)? before ?=? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:137: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvFGDetector? > /usr/local/include/opencv2/legacy/blobtrack.hpp:148: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:149: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:160: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?cvBlob? > In file included from /usr//include/opencv/cvaux.h:61, > from gstfacedetect.h:52, > from gstfacedetect.c:66: > /usr/local/include/opencv2/legacy/blobtrack.hpp:180: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobSeq? > /usr/local/include/opencv2/legacy/blobtrack.hpp:270: error: expected > specifier-qualifier-list before ?CvBlobSeq? > /usr/local/include/opencv2/legacy/blobtrack.hpp:273: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackSeq? > /usr/local/include/opencv2/legacy/blobtrack.hpp:294: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobDetector? > /usr/local/include/opencv2/legacy/blobtrack.hpp:308: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:311: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:312: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:314: error: expected > identifier or ?(? before ?:? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:319: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?cvDetectedBlob? > /usr/local/include/opencv2/legacy/blobtrack.hpp:327: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvObjectDetector? > In file included from /usr//include/opencv/cvaux.h:61, > from gstfacedetect.h:52, > from gstfacedetect.c:66: > /usr/local/include/opencv2/legacy/blobtrack.hpp:391: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvImageDrawer? > /usr/local/include/opencv2/legacy/blobtrack.hpp:410: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackGen? > /usr/local/include/opencv2/legacy/blobtrack.hpp:420: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:427: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:428: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:433: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTracker? > /usr/local/include/opencv2/legacy/blobtrack.hpp:503: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:507: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackerOne? > /usr/local/include/opencv2/legacy/blobtrack.hpp:524: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:529: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:545: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:546: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:547: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:557: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:560: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:563: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:566: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:569: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:570: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:573: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:580: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > In file included from /usr//include/opencv/cvaux.h:61, > from gstfacedetect.h:52, > from gstfacedetect.c:66: > /usr/local/include/opencv2/legacy/blobtrack.hpp:597: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackPostProc? > /usr/local/include/opencv2/legacy/blobtrack.hpp:620: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:628: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before > ?CvBlobTrackPostProcOne? > /usr/local/include/opencv2/legacy/blobtrack.hpp:637: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:641: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:642: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:643: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:648: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackPredictor? > /usr/local/include/opencv2/legacy/blobtrack.hpp:656: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:661: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackAnalysis? > /usr/local/include/opencv2/legacy/blobtrack.hpp:676: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:684: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackFVGen? > /usr/local/include/opencv2/legacy/blobtrack.hpp:701: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before > ?CvBlobTrackAnalysisOne? > /usr/local/include/opencv2/legacy/blobtrack.hpp:712: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:717: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:720: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:723: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:726: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > In file included from /usr//include/opencv/cvaux.h:61, > from gstfacedetect.h:52, > from gstfacedetect.c:66: > /usr/local/include/opencv2/legacy/blobtrack.hpp:743: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:749: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:752: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before > ?CvBlobTrackAnalysisHeight? > /usr/local/include/opencv2/legacy/blobtrack.hpp:762: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackerAuto? > /usr/local/include/opencv2/legacy/blobtrack.hpp:777: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:792: error: expected > specifier-qualifier-list before ?CvFGDetector? > /usr/local/include/opencv2/legacy/blobtrack.hpp:813: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:816: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > In file included from /usr//include/opencv/cvaux.h:61, > from gstfacedetect.h:52, > from gstfacedetect.c:66: > /usr/local/include/opencv2/legacy/blobtrack.hpp:840: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:841: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:842: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:846: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvProb? > /usr/local/include/opencv2/legacy/blobtrack.hpp:861: error: expected > ?)? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:865: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:866: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:867: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:868: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > /usr/local/include/opencv2/legacy/blobtrack.hpp:874: error: expected > ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > In file included from /usr//include/opencv/cvaux.h:61, > from gstfacedetect.h:52, > from gstfacedetect.c:66: > /usr/local/include/opencv2/legacy/blobtrack.hpp:910: error: expected > ?;?, ?,? or ?)? before ?=? token > In file included from /usr//include/opencv/cvaux.h:61, > from gstfacedetect.h:52, > from gstfacedetect.c:66: > /usr/local/include/opencv2/legacy/blobtrack.hpp:940: error: expected > ?;?, ?,? or ?)? before ?=? token > make[3]: *** [libgstfacedetect_la-gstfacedetect.lo] Error 1 > > > the bug why I updated: > > I would like to use alpha blending. > I followed Shervin Emam instructions but it isnt help :( > The alpha blending method: > > I split the input frame into 3 image from the 3 channel. > and I use cvAddWeighted function on 3 image and add 3 other image with > alpha and beta equal 0.5. > and after I merge this images > > after this method the video is flicking and I see some sub image in > the frame etc ... > > So I think this is a bug. And doesnt solved the update to newer OpenCV > version. > > Any idea ? > > -- > Best Regards, > > Robert > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better > price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Sreerenj Balachandran (Nokia-D R&D/Helsinki) Senior Engineer, Multimedia http://github.com/sreerenjb -------------- next part -------------- An HTML attachment was scrubbed... URL: From jobbagy.robert at gmail.com Fri Jan 28 14:10:10 2011 From: jobbagy.robert at gmail.com (Robert Jobbagy) Date: Fri, 28 Jan 2011 14:10:10 +0100 Subject: [gst-devel] gst-opencv app compile error with OpenCV 2.2 In-Reply-To: References: Message-ID: Thanks your quickly response. When will support opencv 2.2 ? 2011/1/28 sreerenj b : > Hi Robert, > Currently gst-opencv supports only "opencv >= 2.0.0 opencv <= 2.1.0". > > On Fri, Jan 28, 2011 at 2:02 PM, Robert Jobbagy > wrote: >> >> Hi Guys, >> >> I used my gst-opencv filter with OpenCV 2.0 and worked fine. >> I tried update to OpenCV 2.1 because I found a bug. >> But OpenCV 2.1 has the bug too and tried update to OpenCV 2.2. >> >> And I gave this compile error: >> >> In file included from /usr//include/opencv/cvaux.h:61, >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> /usr/local/include/opencv2/legacy/blobtrack.hpp:82: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvVSModule? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:133: error: expected >> ?;?, ?,? or ?)? before ?=? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:137: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvFGDetector? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:148: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:149: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:160: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?cvBlob? >> In file included from /usr//include/opencv/cvaux.h:61, >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> /usr/local/include/opencv2/legacy/blobtrack.hpp:180: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobSeq? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:270: error: expected >> specifier-qualifier-list before ?CvBlobSeq? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:273: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackSeq? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:294: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobDetector? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:308: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:311: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:312: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:314: error: expected >> identifier or ?(? before ?:? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:319: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?cvDetectedBlob? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:327: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvObjectDetector? >> In file included from /usr//include/opencv/cvaux.h:61, >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> /usr/local/include/opencv2/legacy/blobtrack.hpp:391: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvImageDrawer? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:410: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackGen? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:420: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:427: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:428: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:433: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTracker? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:503: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:507: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackerOne? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:524: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:529: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:545: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:546: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:547: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:557: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:560: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:563: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:566: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:569: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:570: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:573: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:580: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> In file included from /usr//include/opencv/cvaux.h:61, >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> /usr/local/include/opencv2/legacy/blobtrack.hpp:597: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackPostProc? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:620: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:628: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> ?CvBlobTrackPostProcOne? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:637: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:641: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:642: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:643: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:648: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackPredictor? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:656: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:661: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackAnalysis? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:676: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:684: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackFVGen? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:701: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> ?CvBlobTrackAnalysisOne? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:712: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:717: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:720: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:723: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:726: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> In file included from /usr//include/opencv/cvaux.h:61, >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> /usr/local/include/opencv2/legacy/blobtrack.hpp:743: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:749: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:752: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> ?CvBlobTrackAnalysisHeight? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:762: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackerAuto? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:777: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:792: error: expected >> specifier-qualifier-list before ?CvFGDetector? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:813: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:816: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> In file included from /usr//include/opencv/cvaux.h:61, >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> /usr/local/include/opencv2/legacy/blobtrack.hpp:840: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:841: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:842: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:846: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvProb? >> /usr/local/include/opencv2/legacy/blobtrack.hpp:861: error: expected >> ?)? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:865: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:866: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:867: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:868: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> /usr/local/include/opencv2/legacy/blobtrack.hpp:874: error: expected >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> In file included from /usr//include/opencv/cvaux.h:61, >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> /usr/local/include/opencv2/legacy/blobtrack.hpp:910: error: expected >> ?;?, ?,? or ?)? before ?=? token >> In file included from /usr//include/opencv/cvaux.h:61, >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> /usr/local/include/opencv2/legacy/blobtrack.hpp:940: error: expected >> ?;?, ?,? or ?)? before ?=? token >> make[3]: *** [libgstfacedetect_la-gstfacedetect.lo] Error 1 >> >> >> the bug why I updated: >> >> ?I would like to use alpha blending. >> I followed Shervin Emam instructions but it isnt help :( >> The alpha blending method: >> >> I split the input frame into 3 image from the 3 channel. >> and I use cvAddWeighted function on 3 image and add 3 other image with >> alpha and beta equal 0.5. >> and after I merge this images >> >> after this method the video is flicking and I see some sub image in >> the frame etc ... >> >> So I think this is a bug. And doesnt solved the update to newer OpenCV >> version. >> >> Any idea ? >> >> -- >> Best Regards, >> >> Robert >> >> >> ------------------------------------------------------------------------------ >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> Finally, a world-class log management solution at an even better >> price-free! >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> February 28th, so secure your free ArcSight Logger TODAY! >> http://p.sf.net/sfu/arcsight-sfd2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > -- > Sreerenj Balachandran (Nokia-D R&D/Helsinki) > Senior Engineer, Multimedia > http://github.com/sreerenjb > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Best Regards, Robert From bsreerenj at gmail.com Fri Jan 28 14:20:54 2011 From: bsreerenj at gmail.com (sreerenj b) Date: Fri, 28 Jan 2011 15:20:54 +0200 Subject: [gst-devel] gst-opencv app compile error with OpenCV 2.2 In-Reply-To: References: Message-ID: Hi Robert, Some works are going on with gst-opencv . We will update it as soon as possible. On Fri, Jan 28, 2011 at 3:10 PM, Robert Jobbagy wrote: > Thanks your quickly response. > When will support opencv 2.2 ? > > 2011/1/28 sreerenj b : > > Hi Robert, > > Currently gst-opencv supports only "opencv >= 2.0.0 opencv <= 2.1.0". > > > > On Fri, Jan 28, 2011 at 2:02 PM, Robert Jobbagy < > jobbagy.robert at gmail.com> > > wrote: > >> > >> Hi Guys, > >> > >> I used my gst-opencv filter with OpenCV 2.0 and worked fine. > >> I tried update to OpenCV 2.1 because I found a bug. > >> But OpenCV 2.1 has the bug too and tried update to OpenCV 2.2. > >> > >> And I gave this compile error: > >> > >> In file included from /usr//include/opencv/cvaux.h:61, > >> from gstfacedetect.h:52, > >> from gstfacedetect.c:66: > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:82: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvVSModule? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:133: error: expected > >> ?;?, ?,? or ?)? before ?=? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:137: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvFGDetector? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:148: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:149: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:160: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?cvBlob? > >> In file included from /usr//include/opencv/cvaux.h:61, > >> from gstfacedetect.h:52, > >> from gstfacedetect.c:66: > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:180: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobSeq? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:270: error: expected > >> specifier-qualifier-list before ?CvBlobSeq? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:273: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackSeq? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:294: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobDetector? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:308: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:311: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:312: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:314: error: expected > >> identifier or ?(? before ?:? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:319: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?cvDetectedBlob? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:327: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvObjectDetector? > >> In file included from /usr//include/opencv/cvaux.h:61, > >> from gstfacedetect.h:52, > >> from gstfacedetect.c:66: > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:391: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvImageDrawer? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:410: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackGen? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:420: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:427: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:428: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:433: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTracker? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:503: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:507: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackerOne? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:524: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:529: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:545: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:546: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:547: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:557: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:560: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:563: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:566: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:569: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:570: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:573: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:580: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> In file included from /usr//include/opencv/cvaux.h:61, > >> from gstfacedetect.h:52, > >> from gstfacedetect.c:66: > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:597: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackPostProc? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:620: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:628: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before > >> ?CvBlobTrackPostProcOne? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:637: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:641: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:642: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:643: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:648: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackPredictor? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:656: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:661: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackAnalysis? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:676: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:684: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackFVGen? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:701: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before > >> ?CvBlobTrackAnalysisOne? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:712: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:717: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:720: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:723: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:726: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> In file included from /usr//include/opencv/cvaux.h:61, > >> from gstfacedetect.h:52, > >> from gstfacedetect.c:66: > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:743: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:749: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:752: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before > >> ?CvBlobTrackAnalysisHeight? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:762: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackerAuto? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:777: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:792: error: expected > >> specifier-qualifier-list before ?CvFGDetector? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:813: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:816: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> In file included from /usr//include/opencv/cvaux.h:61, > >> from gstfacedetect.h:52, > >> from gstfacedetect.c:66: > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:840: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:841: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:842: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:846: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvProb? > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:861: error: expected > >> ?)? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:865: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:866: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:867: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:868: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:874: error: expected > >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token > >> In file included from /usr//include/opencv/cvaux.h:61, > >> from gstfacedetect.h:52, > >> from gstfacedetect.c:66: > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:910: error: expected > >> ?;?, ?,? or ?)? before ?=? token > >> In file included from /usr//include/opencv/cvaux.h:61, > >> from gstfacedetect.h:52, > >> from gstfacedetect.c:66: > >> /usr/local/include/opencv2/legacy/blobtrack.hpp:940: error: expected > >> ?;?, ?,? or ?)? before ?=? token > >> make[3]: *** [libgstfacedetect_la-gstfacedetect.lo] Error 1 > >> > >> > >> the bug why I updated: > >> > >> I would like to use alpha blending. > >> I followed Shervin Emam instructions but it isnt help :( > >> The alpha blending method: > >> > >> I split the input frame into 3 image from the 3 channel. > >> and I use cvAddWeighted function on 3 image and add 3 other image with > >> alpha and beta equal 0.5. > >> and after I merge this images > >> > >> after this method the video is flicking and I see some sub image in > >> the frame etc ... > >> > >> So I think this is a bug. And doesnt solved the update to newer OpenCV > >> version. > >> > >> Any idea ? > >> > >> -- > >> Best Regards, > >> > >> Robert > >> > >> > >> > ------------------------------------------------------------------------------ > >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > >> Finally, a world-class log management solution at an even better > >> price-free! > >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires > >> February 28th, so secure your free ArcSight Logger TODAY! > >> http://p.sf.net/sfu/arcsight-sfd2d > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > -- > > Sreerenj Balachandran (Nokia-D R&D/Helsinki) > > Senior Engineer, Multimedia > > http://github.com/sreerenjb > > > > > ------------------------------------------------------------------------------ > > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > > Finally, a world-class log management solution at an even better > price-free! > > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > > February 28th, so secure your free ArcSight Logger TODAY! > > http://p.sf.net/sfu/arcsight-sfd2d > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > -- > Best Regards, > > Robert > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better > price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Sreerenj Balachandran (Nokia-D R&D/Helsinki) Senior Engineer, Multimedia http://github.com/sreerenjb -------------- next part -------------- An HTML attachment was scrubbed... URL: From jobbagy.robert at gmail.com Fri Jan 28 14:35:50 2011 From: jobbagy.robert at gmail.com (Robert Jobbagy) Date: Fri, 28 Jan 2011 14:35:50 +0100 Subject: [gst-devel] gst-opencv app compile error with OpenCV 2.2 In-Reply-To: References: Message-ID: Okay, thanks your time. All best. 2011/1/28 sreerenj b : > Hi Robert, > Some works are going on with gst-opencv . We will update it as soon as > possible. > > On Fri, Jan 28, 2011 at 3:10 PM, Robert Jobbagy > wrote: >> >> Thanks your quickly response. >> When will support opencv 2.2 ? >> >> 2011/1/28 sreerenj b : >> > Hi Robert, >> > Currently gst-opencv supports only "opencv >= 2.0.0 opencv <= 2.1.0". >> > >> > On Fri, Jan 28, 2011 at 2:02 PM, Robert Jobbagy >> > >> > wrote: >> >> >> >> Hi Guys, >> >> >> >> I used my gst-opencv filter with OpenCV 2.0 and worked fine. >> >> I tried update to OpenCV 2.1 because I found a bug. >> >> But OpenCV 2.1 has the bug too and tried update to OpenCV 2.2. >> >> >> >> And I gave this compile error: >> >> >> >> In file included from /usr//include/opencv/cvaux.h:61, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:82: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvVSModule? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:133: error: expected >> >> ?;?, ?,? or ?)? before ?=? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:137: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvFGDetector? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:148: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:149: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:160: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?cvBlob? >> >> In file included from /usr//include/opencv/cvaux.h:61, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:180: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobSeq? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:270: error: expected >> >> specifier-qualifier-list before ?CvBlobSeq? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:273: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackSeq? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:294: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobDetector? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:308: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:311: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:312: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:314: error: expected >> >> identifier or ?(? before ?:? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:319: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?cvDetectedBlob? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:327: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvObjectDetector? >> >> In file included from /usr//include/opencv/cvaux.h:61, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:391: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvImageDrawer? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:410: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackGen? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:420: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:427: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:428: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:433: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTracker? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:503: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:507: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackerOne? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:524: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:529: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:545: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:546: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:547: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:557: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:560: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:563: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:566: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:569: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:570: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:573: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:580: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> In file included from /usr//include/opencv/cvaux.h:61, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:597: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackPostProc? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:620: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:628: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> >> ?CvBlobTrackPostProcOne? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:637: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:641: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:642: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:643: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:648: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackPredictor? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:656: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:661: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackAnalysis? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:676: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:684: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackFVGen? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:701: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> >> ?CvBlobTrackAnalysisOne? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:712: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:717: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:720: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:723: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:726: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> In file included from /usr//include/opencv/cvaux.h:61, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:743: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:749: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:752: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> >> ?CvBlobTrackAnalysisHeight? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:762: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvBlobTrackerAuto? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:777: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:792: error: expected >> >> specifier-qualifier-list before ?CvFGDetector? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:813: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:816: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> In file included from /usr//include/opencv/cvaux.h:61, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:840: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:841: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:842: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:846: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?CvProb? >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:861: error: expected >> >> ?)? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:865: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:866: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:867: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:868: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:874: error: expected >> >> ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?*? token >> >> In file included from /usr//include/opencv/cvaux.h:61, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:910: error: expected >> >> ?;?, ?,? or ?)? before ?=? token >> >> In file included from /usr//include/opencv/cvaux.h:61, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.h:52, >> >> ? ? ? ? ? ? ? ? from gstfacedetect.c:66: >> >> /usr/local/include/opencv2/legacy/blobtrack.hpp:940: error: expected >> >> ?;?, ?,? or ?)? before ?=? token >> >> make[3]: *** [libgstfacedetect_la-gstfacedetect.lo] Error 1 >> >> >> >> >> >> the bug why I updated: >> >> >> >> ?I would like to use alpha blending. >> >> I followed Shervin Emam instructions but it isnt help :( >> >> The alpha blending method: >> >> >> >> I split the input frame into 3 image from the 3 channel. >> >> and I use cvAddWeighted function on 3 image and add 3 other image with >> >> alpha and beta equal 0.5. >> >> and after I merge this images >> >> >> >> after this method the video is flicking and I see some sub image in >> >> the frame etc ... >> >> >> >> So I think this is a bug. And doesnt solved the update to newer OpenCV >> >> version. >> >> >> >> Any idea ? >> >> >> >> -- >> >> Best Regards, >> >> >> >> Robert >> >> >> >> >> >> >> >> ------------------------------------------------------------------------------ >> >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> >> Finally, a world-class log management solution at an even better >> >> price-free! >> >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> >> February 28th, so secure your free ArcSight Logger TODAY! >> >> http://p.sf.net/sfu/arcsight-sfd2d >> >> _______________________________________________ >> >> gstreamer-devel mailing list >> >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> > >> > >> > -- >> > Sreerenj Balachandran (Nokia-D R&D/Helsinki) >> > Senior Engineer, Multimedia >> > http://github.com/sreerenjb >> > >> > >> > ------------------------------------------------------------------------------ >> > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> > Finally, a world-class log management solution at an even better >> > price-free! >> > Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> > February 28th, so secure your free ArcSight Logger TODAY! >> > http://p.sf.net/sfu/arcsight-sfd2d >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> > >> >> >> >> -- >> Best Regards, >> >> Robert >> >> >> ------------------------------------------------------------------------------ >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> Finally, a world-class log management solution at an even better >> price-free! >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> February 28th, so secure your free ArcSight Logger TODAY! >> http://p.sf.net/sfu/arcsight-sfd2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > -- > Sreerenj Balachandran (Nokia-D R&D/Helsinki) > Senior Engineer, Multimedia > http://github.com/sreerenjb > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Best Regards, Robert From giorgio9 at libero.it Fri Jan 28 15:16:57 2011 From: giorgio9 at libero.it (giorgino) Date: Fri, 28 Jan 2011 06:16:57 -0800 (PST) Subject: [gst-devel] v4l2loopback and gstreamer Message-ID: <1296224217375-3244534.post@n4.nabble.com> Hi All I need your support in order to solve an issue. I had a usb camera that has to be used at the same time by two different application. This it is obviously a problem because when I launch the second application, system says me that the resource is busy. I thought that the solution can be to install v4l2loopback and execute it. In this way I have two devices that are /dev/video0 and /dev/video1. So I launched gst-launch v4l2src device=/dev/video0 ! autovideosink and after gst-launch v4l2src device=/dev/video1 ! autovideosink The first pipe works well but the second one goes on error with the following message Impostazione della pipeline a PAUSED ... libv4l2: error getting pixformat: Argomento non valido ERRORE: la pipeline non vuole mettersi in pausa. ERRORE: dall'elemento /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Could not negotiate format Informazioni di debug aggiuntive: gstbasesrc.c(2755): gst_base_src_start (): /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Check your filtered caps, if any Impostazione della pipeline a NULL ... Esecuzione di free sulla pipeline... How I can solve this issue? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/v4l2loopback-and-gstreamer-tp3244534p3244534.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From jesuas at gmail.com Fri Jan 28 15:49:18 2011 From: jesuas at gmail.com (Jesu Anuroop Suresh) Date: Fri, 28 Jan 2011 06:49:18 -0800 (PST) Subject: [gst-devel] id3demux set propery prefer-v1 fails Message-ID: <1296226158698-3244638.post@n4.nabble.com> Hi All, Does any one tried setting the property of the id3demux ? when I try to set it using the GstElement *demux = gst_element_factory_make ("id3demux", "id3-demuxer0"); if (!demux) { g_print ("NO MEM Exiting.\n"); return 1; } g_object_set (G_OBJECT (demuxer), "prefer-v1", "true", NULL); It gives the following error and does not work (:5543): GLib-GObject-WARNING **: value "TRUE" of type `gboolean' is invalid or out of range for property `prefer-v1' of type `gboolean' Regards Jesu Anuroop Suresh -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/id3demux-set-propery-prefer-v1-fails-tp3244638p3244638.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Fri Jan 28 15:55:27 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 28 Jan 2011 14:55:27 +0000 Subject: [gst-devel] id3demux set propery prefer-v1 fails In-Reply-To: <1296226158698-3244638.post@n4.nabble.com> References: <1296226158698-3244638.post@n4.nabble.com> Message-ID: <1296226527.7693.1.camel@zingle> On Fri, 2011-01-28 at 06:49 -0800, Jesu Anuroop Suresh wrote: > g_object_set (G_OBJECT (demuxer), "prefer-v1", "true", NULL); > > It gives the following error and does not work > > (:5543): GLib-GObject-WARNING **: value "TRUE" of type `gboolean' > is invalid or out of range for property `prefer-v1' of type `gboolean' Try: g_object_set (demuxer, "prefer-v1", TRUE, NULL); It's a vararg function, the type you pass needs to match the type shown in gst-inspect. (The gst-launch utility will convert the string automatically, but g_object_set() can't do that). Cheers -Tim From chombourger at gmail.com Fri Jan 28 16:02:48 2011 From: chombourger at gmail.com (Cedric Hombourger) Date: Fri, 28 Jan 2011 16:02:48 +0100 Subject: [gst-devel] Use of mp3parse in a commercial product Message-ID: <7A23F358-725E-4B8B-9BCD-33E4E92DC48E@gmail.com> Hello, We have created a GStreamer MP3 decoder plugin from a licensed proprietary library. It currently does not support seeking. We have found that the mp3parse plugin from gst-plugins-ugly could easily fix our problem. So is there any particular reason why this plugin is in -ugly? Do you foresee any issues with its use in a commercial product given that we will fulfill our obligations for MP3 patents? I will then have the same question for aacparse - same answer? Many thanks -Cedric From uraeus at linuxrising.org Fri Jan 28 16:21:44 2011 From: uraeus at linuxrising.org (Christian Fredrik Kalager Schaller) Date: Fri, 28 Jan 2011 15:21:44 +0000 Subject: [gst-devel] Use of mp3parse in a commercial product In-Reply-To: <7A23F358-725E-4B8B-9BCD-33E4E92DC48E@gmail.com> References: <7A23F358-725E-4B8B-9BCD-33E4E92DC48E@gmail.com> Message-ID: <1296228104.5079.7.camel@localhost.localdomain> Hi Cedric, I am not a lawyer, but the reason we put the plugin into ugly was that it could only be used with plugins we felt could have problems. That said we have actually discussed moving it to good at certain times as people who want to use it with a mp3 hardware decoder for instance, would find it useful. So as far as I can tell it would not be an issue for you to use this plugin in your product, same goes for the aacparser one. Christian On Fri, 2011-01-28 at 16:02 +0100, Cedric Hombourger wrote: > Hello, > > We have created a GStreamer MP3 decoder plugin from a licensed proprietary library. It currently does not support seeking. We have found that the mp3parse plugin from gst-plugins-ugly could easily fix our problem. So is there any particular reason why this plugin is in -ugly? Do you foresee any issues with its use in a commercial product given that we will fulfill our obligations for MP3 patents? I will then have the same question for aacparse - same answer? > > Many thanks > -Cedric > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From chombourger at gmail.com Fri Jan 28 23:21:39 2011 From: chombourger at gmail.com (Cedric Hombourger) Date: Fri, 28 Jan 2011 23:21:39 +0100 Subject: [gst-devel] Use of mp3parse in a commercial product In-Reply-To: <1296228104.5079.7.camel@localhost.localdomain> References: <7A23F358-725E-4B8B-9BCD-33E4E92DC48E@gmail.com> <1296228104.5079.7.camel@localhost.localdomain> Message-ID: <252A9E21-F3C6-4CDC-90CB-0CA0651F9D91@gmail.com> Thanks Christian; I'm getting my legal department to look into this. Have a great week-end, Bye -Cedric On Jan 28, 2011, at 4:21 PM, Christian Fredrik Kalager Schaller wrote: > Hi Cedric, > I am not a lawyer, but the reason we put the plugin into ugly was that > it could only be used with plugins we felt could have problems. That > said we have actually discussed moving it to good at certain times as > people who want to use it with a mp3 hardware decoder for instance, > would find it useful. So as far as I can tell it would not be an issue > for you to use this plugin in your product, same goes for the aacparser > one. > > Christian > > On Fri, 2011-01-28 at 16:02 +0100, Cedric Hombourger wrote: >> Hello, >> >> We have created a GStreamer MP3 decoder plugin from a licensed proprietary library. It currently does not support seeking. We have found that the mp3parse plugin from gst-plugins-ugly could easily fix our problem. So is there any particular reason why this plugin is in -ugly? Do you foresee any issues with its use in a commercial product given that we will fulfill our obligations for MP3 patents? I will then have the same question for aacparse - same answer? >> >> Many thanks >> -Cedric >> ------------------------------------------------------------------------------ >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> Finally, a world-class log management solution at an even better price-free! >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> February 28th, so secure your free ArcSight Logger TODAY! >> http://p.sf.net/sfu/arcsight-sfd2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bossart.nospam at gmail.com Fri Jan 28 23:58:49 2011 From: bossart.nospam at gmail.com (pl bossart) Date: Fri, 28 Jan 2011 16:58:49 -0600 Subject: [gst-devel] Use of mp3parse in a commercial product In-Reply-To: <1296228104.5079.7.camel@localhost.localdomain> References: <7A23F358-725E-4B8B-9BCD-33E4E92DC48E@gmail.com> <1296228104.5079.7.camel@localhost.localdomain> Message-ID: > I am not a lawyer, but the reason we put the plugin into ugly was that > it could only be used with plugins we felt could have problems. That > said we have actually discussed moving it to good at certain times as > people who want to use it with a mp3 hardware decoder for instance, > would find it useful. So as far as I can tell it would not be an issue > for you to use this plugin in your product, same goes for the aacparser > one. If you are using a 'legal' decoding library you somehow already paid for the right to use MP3-related patents. Seeking does not use any other patents. Same if a hardware decoder is used, a license fee was already paid to Thomson and others. If you didn't pay any kind of license you would be in gray area even before looking into seeking... -Pierre From nathanael at gnat.ca Sat Jan 29 00:09:19 2011 From: nathanael at gnat.ca (Nathanael D. Noblet) Date: Fri, 28 Jan 2011 16:09:19 -0700 Subject: [gst-devel] Use of mp3parse in a commercial product In-Reply-To: References: <7A23F358-725E-4B8B-9BCD-33E4E92DC48E@gmail.com> <1296228104.5079.7.camel@localhost.localdomain> Message-ID: <4D434C9F.3090001@gnat.ca> On 01/28/2011 03:58 PM, pl bossart wrote: >> I am not a lawyer, but the reason we put the plugin into ugly was that >> it could only be used with plugins we felt could have problems. That >> said we have actually discussed moving it to good at certain times as >> people who want to use it with a mp3 hardware decoder for instance, >> would find it useful. So as far as I can tell it would not be an issue >> for you to use this plugin in your product, same goes for the aacparser >> one. > > If you are using a 'legal' decoding library you somehow already paid > for the right to use MP3-related patents. Seeking does not use any > other patents. Same if a hardware decoder is used, a license fee was > already paid to Thomson and others. > If you didn't pay any kind of license you would be in gray area even > before looking into seeking... Hmmm got me wondering here... if I buy an IP camera that sends an MP4 stream, can I legally decode that stream with the software in gstreamer if my product is sold as part of the camera? From shamun.toha at gmail.com Sat Jan 29 01:33:25 2011 From: shamun.toha at gmail.com (Shamun toha md) Date: Sat, 29 Jan 2011 01:33:25 +0100 Subject: [gst-devel] Can you stream ? Message-ID: Hello, Can you stream audio/video to multi user to view from web or using other tcp/udp software or anything else ? For example capturing and showing via web link from laptop ? Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From alet at librelogiciel.com Sat Jan 29 04:39:38 2011 From: alet at librelogiciel.com (alet at librelogiciel.com) Date: Sat, 29 Jan 2011 04:39:38 +0100 Subject: [gst-devel] Can you stream ? In-Reply-To: References: Message-ID: <20110129033938.GB24658@vazy.pykota.com> On Sat, Jan 29, 2011 at 01:33:25AM +0100, Shamun toha md wrote: > Hello, > > Can you stream audio/video to multi user to view from web or using other > tcp/udp software or anything else ? For example capturing and showing via > web link from laptop ? Yes. See for example software like http://boxtream.unice.fr which makes easy use of gstreamer to stream audio and video. bye Jerome Alet From marc.leeman at gmail.com Sat Jan 29 09:01:53 2011 From: marc.leeman at gmail.com (Marc Leeman) Date: Sat, 29 Jan 2011 09:01:53 +0100 Subject: [gst-devel] GStreamer RTP e(X)tension bit and video specific fixed headers In-Reply-To: <20110128185226.GG30378@crichton.homelinux.org> References: <862617A579113B4C8AE1BD85EF38DD000104523C@KUUMEX03.barco.com> <20110128173951.GF30378@crichton.homelinux.org> <20110128185226.GG30378@crichton.homelinux.org> Message-ID: <20110129080153.GH30378@crichton.homelinux.org> I got a remark that our GStreamer based system does use 4 extra bytes after the RTP header while the X bit in the RTP header is not set. I did some digging and next to some remarks that the X bit is profile specific; I do not see in the relevant RFCs that the relevant bit needs to be set when adding the profile specific header. Mostly, the RFC (e.g. 2250 for MPEG2) says some fields need to be so and so; but not that the X bit needs to be set. I think that the remark originates (indirectly) from the other side using FFMPEG based RTP handling (skip 12 bytes, do something with the rest). Below is my assessment as I thought it to be after reading a number RFCs; I would apreciate any review of my comments into this and if I am wrong, we'll have to raise a gst-plugins-good bug on RTP. On Fri, Jan 28, 2011 at 07:52:26PM +0100, Marc Leeman wrote: > On Fri, Jan 28, 2011 at 06:39:51PM +0100, Marc Leeman wrote: > > > controlRtp does not process it correctly. hope you or Marc Leeman can > > > correct this on the MGS side. Please refer to RFC2250 for details. > > > > So for MPEG2; there is not even an option NOT to include the header; > > which explains why the X bit is profile dependent. > > I have been reading a bit more and as far as I read 2250; there are two > MPEG2 headers. > > 1/ must always be there, the MPEG fixed video header > 2/ MPEG header extension that is optional > > IMO the RFC 3550 extension is as such relevant to point 2; not to point 1. > MPEG2 is used; the 4 byte fixed header MUST always follow the 12 > byte fixed RTP header. If the RTP extension bit is set; then there is > another variable length extension header (that is MPEG2 specific). > > As such; we are sending 12 bytes RTP AND the mandatory 4 bytes MPEG2 > extension. Since we do not set the X bit; there are no MPEG2 header > extensions (on top of the 4 mandatory MPEG2 bytes). > > If the X were to be the fixed header for MPEG2; you would see that > the definition of the header as defined in 5.3.1 RTP Header Extension > (3550) would conflict with the MPEG2 fixed header as defined in 3.4 MPEG > Video-specific header (2250), > > Only when the MPEG2 fixed header is considered not covered by the X bit, > that there is no conflict. > > This reasoning seems to be confirmed by other RFCs too for other video > payloads (mentioned before). > > I'll check this next week on IRC. -- greetz, marc We must believe that it is the darkest before the dawn of a beautiful new world. We will see it when we believe it. -- Saul Alinsky 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 lazyt at gmx.net Sat Jan 29 13:07:59 2011 From: lazyt at gmx.net (Thomas Loewe) Date: Sat, 29 Jan 2011 04:07:59 -0800 (PST) Subject: [gst-devel] wrong horizontal position of dvbsuboverlay if used behind videomixer? Message-ID: <1296302879878-3246076.post@n4.nabble.com> Hi, since I use the videomixer to implement picture in picture the horizontal positioning of subtitles via dvbsuboverlay is incorrect when putting it behind the videomixer. For example: the video resolution is 720x576 in 16:9 and videomixer makes it 1024x576 (768x576 in 4:3). Subtitles are shown on the left and not centered because it uses 720 instead 1024. The position is of course correct when putting it before the videomixer, but then the subtitles are overlayd by the pip. Is this a bug? Thanks, Thomas -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/wrong-horizontal-position-of-dvbsuboverlay-if-used-behind-videomixer-tp3246076p3246076.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From lazyt at gmx.net Sat Jan 29 13:12:59 2011 From: lazyt at gmx.net (Thomas Loewe) Date: Sat, 29 Jan 2011 04:12:59 -0800 (PST) Subject: [gst-devel] problems with 0.10.32 core release (cpuload, deinterlacer) In-Reply-To: <1296041384.18773.1.camel@zingle> References: <1295975177921-3236547.post@n4.nabble.com> <1296041384.18773.1.camel@zingle> Message-ID: <1296303179575-3246081.post@n4.nabble.com> > Could you file (separate) bugs for these things please, so we can investigate them? Forget to post it was filed via https://bugzilla.gnome.org/show_bug.cgi?id=640675 640675 -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/problems-with-0-10-32-core-release-cpuload-deinterlacer-tp3236547p3246081.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ensonic at hora-obscura.de Sat Jan 29 14:27:21 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 29 Jan 2011 15:27:21 +0200 Subject: [gst-devel] v4l2loopback and gstreamer In-Reply-To: <1296224217375-3244534.post@n4.nabble.com> References: <1296224217375-3244534.post@n4.nabble.com> Message-ID: <4D4415B9.6090807@hora-obscura.de> Am 28.01.2011 16:16, schrieb giorgino: > > Hi All I need your support in order to solve an issue. I had a usb camera > that has to be used at the same time by two different application. This it > is obviously a problem because when I launch the second application, system > says me that the resource is busy. I thought that the solution can be to > install v4l2loopback and execute it. In this way I have two devices that are > /dev/video0 and /dev/video1. So I launched > > gst-launch v4l2src device=/dev/video0 ! autovideosink > > and after > > gst-launch v4l2src device=/dev/video1 ! autovideosink > > The first pipe works well but the second one goes on error with the > following message > > Impostazione della pipeline a PAUSED ... > libv4l2: error getting pixformat: Argomento non valido > ERRORE: la pipeline non vuole mettersi in pausa. > ERRORE: dall'elemento /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Could not > negotiate format > Informazioni di debug aggiuntive: > gstbasesrc.c(2755): gst_base_src_start (): > /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: > Check your filtered caps, if any > Impostazione della pipeline a NULL ... > Esecuzione di free sulla pipeline... > > > How I can solve this issue? > Please file a bug and attatch: - information of how to setup the v4l2loopback device - a (compressed) log for the 2nd GST_DEBUG="':5" gst-launch 2>debug.log v4l2src device=/dev/video1 ! autovideosink Stefan From 4ernov at gmail.com Sat Jan 29 20:18:59 2011 From: 4ernov at gmail.com (Alexey Chernov) Date: Sat, 29 Jan 2011 22:18:59 +0300 Subject: [gst-devel] new v4l2radio radio tuner element Message-ID: <201101292218.59183.4ernov@gmail.com> Hello, I finally completed merging radiotuner element which was discussed here previously with v4l2 GStreamer plugin. The element is used to control V4L2 radio devices to receive analog radio signal. As several people pointed out, there were many shared code between the two so adding another element to v4l2 plugin is really much better IMO than to add radiotuner separately. The bug report and the patch is here: https://bugzilla.gnome.org/show_bug.cgi?id=640118 I hope that this patch could be helpful for GStreamer plugins range and it would be possible to merge my patch to the main GStreamer development tree. From olivier.crete at collabora.co.uk Sat Jan 29 22:02:39 2011 From: olivier.crete at collabora.co.uk (Olivier =?ISO-8859-1?Q?Cr=EAte?=) Date: Sat, 29 Jan 2011 16:02:39 -0500 Subject: [gst-devel] GStreamer RTP e(X)tension bit and video specific fixed headers In-Reply-To: <20110129080153.GH30378@crichton.homelinux.org> References: <862617A579113B4C8AE1BD85EF38DD000104523C@KUUMEX03.barco.com> <20110128173951.GF30378@crichton.homelinux.org> <20110128185226.GG30378@crichton.homelinux.org> <20110129080153.GH30378@crichton.homelinux.org> Message-ID: <1296334959.15660.7.camel@TesterBox.tester.ca> On Sat, 2011-01-29 at 09:01 +0100, Marc Leeman wrote: > I got a remark that our GStreamer based system does use 4 extra bytes > after the RTP header while the X bit in the RTP header is not set. I did > some digging and next to some remarks that the X bit is profile > specific; I do not see in the relevant RFCs that the relevant bit needs > to be set when adding the profile specific header. There is RFC 5285 which you probably missed (it took me a while to discover its existence too). > Mostly, the RFC (e.g. 2250 for MPEG2) says some fields need to be so and > so; but not that the X bit needs to be set. The X bit is in many ways orthogonal to the format. So Payload-type specific RFCs like 2250 don't tell you anything about it. Unless you use an application specific extension of RFC 5285 based on, the X bit should be 0. -- Olivier Cr?te olivier.crete at collabora.co.uk Collabora Ltd -------------- 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 sholzer at my-gekko.com Fri Jan 28 10:30:47 2011 From: sholzer at my-gekko.com (gekko) Date: Fri, 28 Jan 2011 01:30:47 -0800 (PST) Subject: [gst-devel] Properties of a GST_OBJECT Message-ID: <1296207047364-3244042.post@n4.nabble.com> How can i get all properties of a GST_OBJECT? -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Properties-of-a-GST-OBJECT-tp3244042p3244042.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From mmikkone at paju.oulu.fi Sat Jan 29 21:19:44 2011 From: mmikkone at paju.oulu.fi (Marko Mikkonen) Date: Sat, 29 Jan 2011 22:19:44 +0200 Subject: [gst-devel] Can't build gst-ffmpeg on MinGW Message-ID: <4D447660.7040904@paju.oulu.fi> I have a little problem when building gst-ffmpeg-0.10.11 on MinGW. I get warning messages saying that shared libraries of avformat, avcodec and avutil are not built. Later I get a lot of linker errors about missing references to functions, all of which seem to reside in gst-libs/ext/ffmpeg subdirectories. How can I build shared libraries for avformat, avcodec and avutil? I'm not very familiar with autotools so this is a big problem for me. These are my commands: ./configure --prefix=/c/MinGW --enable-shared --disable-static make install Here are the warnings and error messages: Making install in ext make[1]: Entering directory `/home/Marko/gst-ffmpeg-0.10.11/ext' Making install in ffmpeg make[2]: Entering directory `/home/Marko/gst-ffmpeg-0.10.11/ext/ffmpeg' CCLD libgstffmpeg.la *** Warning: Trying to link with static lib archive ../../gst-libs/ext/ffmpeg/libavformat/libavformat.a. *** I have the capability to make that library automatically link in when *** you link to this library. But I can only do this if you have a *** shared version of the library, which you do not appear to have *** because the file extensions .a of this argument makes me believe *** that it is just a static archive that I should not use here. *** Warning: Trying to link with static lib archive ../../gst-libs/ext/ffmpeg/libavcodec/libavcodec.a. *** I have the capability to make that library automatically link in when *** you link to this library. But I can only do this if you have a *** shared version of the library, which you do not appear to have *** because the file extensions .a of this argument makes me believe *** that it is just a static archive that I should not use here. *** Warning: Trying to link with static lib archive ../../gst-libs/ext/ffmpeg/libavutil/libavutil.a. *** I have the capability to make that library automatically link in when *** you link to this library. But I can only do this if you have a *** shared version of the library, which you do not appear to have *** because the file extensions .a of this argument makes me believe *** that it is just a static archive that I should not use here. Info: resolving ___gst_debug_min by linking to __imp____gst_debug_min (auto-import) Info: resolving _GST_CAT_DEFAULT by linking to __imp__GST_CAT_DEFAULT (auto-import) Creating library file: .libs/libgstffmpeg.dll.a /mingw/bin/../lib/gcc/mingw32/4.5.0/../../../../mingw32/bin/ld.exe: warning: auto-importing has been activated without --enable-auto-import specified on the command line. This should work unless it involves constant data structures referencing symbols from auto-imported DLLs. .libs/libgstffmpeg_la-gstffmpeg.o: In function `plugin_init': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:134: undefined reference to `av_log_set_callback' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:139: undefined reference to `av_register_all' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:154: undefined reference to `register_protocol' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:155: undefined reference to `register_protocol' .libs/libgstffmpeg_la-gstffmpeg.o: In function `gst_ffmpeg_avcodec_open': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:52: undefined reference to `avcodec_open' .libs/libgstffmpeg_la-gstffmpeg.o: In function `gst_ffmpeg_avcodec_close': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:64: undefined reference to `avcodec_close' .libs/libgstffmpeg_la-gstffmpeg.o: In function `gst_ffmpeg_av_find_stream_info': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:76: undefined reference to `av_find_stream_info' .libs/libgstffmpeg_la-gstffmpegcodecmap.o: In function `gst_ffmpeg_get_palette': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:52: undefined reference to `av_free' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:53: undefined reference to `av_malloc' .libs/libgstffmpeg_la-gstffmpegcodecmap.o: In function `gst_ffmpeg_codecid_to_caps': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:1547: undefined reference to `avcodec_find_decoder' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:1547: undefined reference to `avcodec_find_encoder' .libs/libgstffmpeg_la-gstffmpegcodecmap.o: In function `gst_ffmpeg_caps_with_codecid': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2250: undefined reference to `av_free' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2276: undefined reference to `av_mallocz' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2325: undefined reference to `av_mallocz' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2291: undefined reference to `av_mallocz' .libs/libgstffmpeg_la-gstffmpegmake[2]: Leaving directory `/home/Marko/gst-ffmpeg-0.10.11/ext/ffmpeg' make[1]: Leaving directory `/home/Marko/gst-ffmpeg-0.10.11/ext' codecmap.o: In function `gst_ffmpeg_caps_to_codecid': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3288: undefined reference to `avcodec_find_decoder_by_name' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3288: undefined reference to `avcodec_find_encoder_by_name' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3300: undefined reference to `avcodec_find_decoder_by_name' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3300: undefined reference to `avcodec_find_encoder_by_name' .libs/libgstffmpeg_la-gstffmpegutils.o: In function `gst_ffmpeg_get_codecid_longname': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:31: undefined reference to `avcodec_find_decoder' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:31: undefined reference to `avcodec_find_encoder' .libs/libgstffmpeg_la-gstffmpegutils.o: In function `new_aligned_buffer': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:410: undefined reference to `av_malloc' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:412: undefined reference to `av_free' .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_init': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:248: undefined reference to `avcodec_alloc_context' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:249: undefined reference to `avcodec_alloc_frame' .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_encode_audio': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:860: undefined reference to `avcodec_encode_audio' .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_chain_audio': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:913: undefined reference to `av_get_bits_per_sample_format' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1024: undefined reference to `av_get_bits_per_sample' .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_flush_buffers': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1071: undefined reference to `avcodec_encode_video' .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpeg_time_gst_to_ff': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: undefined reference to `av_rescale_q' .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_chain_video': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:781: undefined reference to `avcodec_encode_video' .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_getcaps': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:448: undefined reference to `avcodec_alloc_context' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:487: undefined reference to `av_free' .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_setcaps': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:529: undefined reference to `avcodec_get_context_defaults' .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_finalize': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:298: undefined reference to `av_free' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:299: undefined reference to `av_free' .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_register': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1307: undefined reference to `av_codec_next' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1378: undefined reference to `av_codec_next' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1378: undefined reference to `av_codec_next' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_init': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:425: undefined reference to `avcodec_alloc_context' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:426: undefined reference to `avcodec_alloc_frame' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_flush_pcache': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2351: undefined reference to `av_parser_close' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2352: undefined reference to `av_parser_init' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_audio_frame': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2109: undefined reference to `avcodec_decode_audio2' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_video_frame': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1754: undefined reference to `avcodec_decode_video' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `get_output_buffer': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1568: undefined reference to `av_picture_copy' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_chain': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2510: undefined reference to `avcodec_flush_buffers' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2589: undefined reference to `av_parser_parse' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_close': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:587: undefined reference to `av_free' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:592: undefined reference to `av_free' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:601: undefined reference to `av_parser_close' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_setcaps': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:733: undefined reference to `avcodec_get_context_defaults' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_open': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:641: undefined reference to `av_parser_init' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:656: undefined reference to `av_parser_init' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_get_buffer': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:989: undefined reference to `avcodec_default_get_buffer' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1016: undefined reference to `avcodec_align_dimensions' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1032: undefined reference to `avcodec_default_get_buffer' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_finalize': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:456: undefined reference to `av_free' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:461: undefined reference to `av_free' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_sink_event': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2378: undefined reference to `avcodec_flush_buffers' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_register': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2850: undefined reference to `av_codec_next' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2973: undefined reference to `av_codec_next' .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_release_buffer': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1078: undefined reference to `avcodec_default_release_buffer' .libs/libgstffmpeg_la-gstffmpegcfg.o: In function `gst_ffmpeg_cfg_install_property': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcfg.c:769: undefined reference to `avcodec_alloc_context' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcfg.c:771: undefined reference to `avcodec_get_context_defaults' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcfg.c:882: undefined reference to `av_free' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_close': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:344: undefined reference to `av_close_input_file' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpeg_time_ff_to_gst': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: undefined reference to `av_rescale_q' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_do_seek': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:442: undefined reference to `av_find_default_stream_index' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpeg_time_gst_to_ff': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: undefined reference to `av_rescale_q' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_do_seek': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:468: undefined reference to `av_index_search_timestamp' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpeg_time_ff_to_gst': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: undefined reference to `av_rescale_q' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_do_seek': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:488: undefined reference to `av_seek_frame' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpeg_time_ff_to_gst': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: undefined reference to `av_rescale_q' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: undefined reference to `av_rescale_q' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_loop': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1345: undefined reference to `av_read_frame' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpeg_time_ff_to_gst': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: undefined reference to `av_rescale_q' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: undefined reference to `av_rescale_q' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_loop': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1473: undefined reference to `av_free_packet' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_open': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1162: undefined reference to `av_open_input_file' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_loop': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1436: undefined reference to `av_picture_copy' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1551: undefined reference to `av_free_packet' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1562: undefined reference to `av_free_packet' .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_register': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1865: undefined reference to `av_iformat_next' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:2025: undefined reference to `av_iformat_next' .libs/libgstffmpeg_la-gstffmpegmux.o: In function `gst_ffmpegmux_collected': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:638: undefined reference to `url_fopen' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:645: undefined reference to `av_set_parameters' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:652: undefined reference to `av_write_header' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:662: undefined reference to `put_flush_packet' .libs/libgstffmpeg_la-gstffmpegmux.o: In function `gst_ffmpeg_time_gst_to_ff': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: undefined reference to `av_rescale_q' .libs/libgstffmpeg_la-gstffmpegmux.o: In function `gst_ffmpegmux_collected': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:754: undefined reference to `av_write_frame' .libs/libgstffmpeg_la-gstffmpegmux.o: In function `gst_ffmpeg_time_gst_to_ff': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: undefined reference to `av_rescale_q' .libs/libgstffmpeg_la-gstffmpegmux.o: In function `gst_ffmpegmux_collected': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:735: undefined reference to `av_picture_copy' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:760: undefined reference to `av_write_trailer' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:762: undefined reference to `put_flush_packet' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:763: undefined reference to `url_fclose' .libs/libgstffmpeg_la-gstffmpegmux.o: In function `gst_ffmpegmux_change_state': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:801: undefined reference to `url_fclose' .libs/libgstffmpeg_la-gstffmpegmux.o: In function `gst_ffmpegmux_request_new_pad': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:453: undefined reference to `av_new_stream' .libs/libgstffmpeg_la-gstffmpegmux.o: In function `gst_ffmpegmux_register': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:877: undefined reference to `av_oformat_next' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:951: undefined reference to `av_oformat_next' .libs/libgstffmpeg_la-gstffmpegdeinterlace.o: In function `gst_ffmpegdeinterlace_chain': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:190: undefined reference to `avpicture_deinterlace' .libs/libgstffmpeg_la-gstffmpegdeinterlace.o: In function `gst_ffmpegdeinterlace_sink_setcaps': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:124: undefined reference to `avcodec_alloc_context' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:136: undefined reference to `av_free' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:139: undefined reference to `avpicture_get_size' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:145: undefined reference to `gst_caps_set_value' C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:130: undefined reference to `av_free' .libs/libgstffmpeg_la-gstffmpegaudioresample.o: In function `gst_ffmpegaudioresample_transform': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegaudioresample.c:278: undefined reference to `audio_resample' .libs/libgstffmpeg_la-gstffmpegaudioresample.o: In function `gst_ffmpegaudioresample_set_caps': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegaudioresample.c:251: undefined reference to `audio_resample_init' .libs/libgstffmpeg_la-gstffmpegaudioresample.o: In function `gst_ffmpegaudioresample_finalize': C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegaudioresample.c:154: undefined reference to `audio_resample_close' collect2: ld returned 1 exit status make[2]: *** [libgstffmpeg.la] Error 1 make[1]: *** [install-recursive] Error 1 make: *** [install-recursive] Error 1 -------------- next part -------------- An HTML attachment was scrubbed... URL: From clafil81 at gmail.com Sun Jan 30 09:57:00 2011 From: clafil81 at gmail.com (Lucifer) Date: Sun, 30 Jan 2011 00:57:00 -0800 (PST) Subject: [gst-devel] AV synchronization problem with elementary AV (AAC/H264) streaming In-Reply-To: <1295591339101-3229054.post@n4.nabble.com> References: <1295591339101-3229054.post@n4.nabble.com> Message-ID: <1296377820417-3246964.post@n4.nabble.com> Hello all, I have the same issue. I'm trying to capture video (v4l2src), encode H264 and send over the net. In addition, I have audio already encoded (using filesrc) and send it over the net. I'm using gstrtpbin pipeline, it works but the audio is delayed of about 2-3 seconds. How can I delay the video? how "ts-offset" property of udpsink work? My idea was to add a timestamp offset to the RTP video packets. Any idea? thanks! Lucy -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/AV-synchronization-problem-with-elementary-AV-AAC-H264-streaming-tp3229054p3246964.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From mnauw at users.sourceforge.net Sun Jan 30 11:41:35 2011 From: mnauw at users.sourceforge.net (Mark Nauwelaerts) Date: Sun, 30 Jan 2011 11:41:35 +0100 Subject: [gst-devel] Can't build gst-ffmpeg on MinGW In-Reply-To: <4D447660.7040904@paju.oulu.fi> References: <4D447660.7040904@paju.oulu.fi> Message-ID: <4D45405F.2050404@users.sourceforge.net> Marko Mikkonen wrote: > I have a little problem when building gst-ffmpeg-0.10.11 on MinGW. I > get warning messages saying that shared libraries of avformat, avcodec > and avutil are not built. Later I get a lot of linker errors about > missing references to functions, all of which seem to reside in > gst-libs/ext/ffmpeg subdirectories. How can I build shared libraries for > avformat, avcodec and avutil? I'm not very familiar with autotools so > this is a big problem for me. > > These are my commands: > ./configure --prefix=/c/MinGW --enable-shared --disable-static > make install You could try export lt_cv_deplibs_check_method=pass_all ./configure Important part is that the former variable should have the configure script convince libtool to also consider a real static lib for linking, rather than only a static import lib (with dll counterpart). Mark. > > Here are the warnings and error messages: > > Making install in ext > make[1]: Entering directory `/home/Marko/gst-ffmpeg-0.10.11/ext' > Making install in ffmpeg > make[2]: Entering directory `/home/Marko/gst-ffmpeg-0.10.11/ext/ffmpeg' > CCLD libgstffmpeg.la > > *** Warning: Trying to link with static lib archive > ../../gst-libs/ext/ffmpeg/libavformat/libavformat.a. > *** I have the capability to make that library automatically link in when > *** you link to this library. But I can only do this if you have a > *** shared version of the library, which you do not appear to have > *** because the file extensions .a of this argument makes me believe > *** that it is just a static archive that I should not use here. > > *** Warning: Trying to link with static lib archive > ../../gst-libs/ext/ffmpeg/libavcodec/libavcodec.a. > *** I have the capability to make that library automatically link in when > *** you link to this library. But I can only do this if you have a > *** shared version of the library, which you do not appear to have > *** because the file extensions .a of this argument makes me believe > *** that it is just a static archive that I should not use here. > > *** Warning: Trying to link with static lib archive > ../../gst-libs/ext/ffmpeg/libavutil/libavutil.a. > *** I have the capability to make that library automatically link in when > *** you link to this library. But I can only do this if you have a > *** shared version of the library, which you do not appear to have > *** because the file extensions .a of this argument makes me believe > *** that it is just a static archive that I should not use here. > Info: resolving ___gst_debug_min by linking to __imp____gst_debug_min > (auto-import) > Info: resolving _GST_CAT_DEFAULT by linking to __imp__GST_CAT_DEFAULT > (auto-import) > Creating library file: .libs/libgstffmpeg.dll.a > /mingw/bin/../lib/gcc/mingw32/4.5.0/../../../../mingw32/bin/ld.exe: > warning: auto-importing has been activated without --enable-auto-import > specified on the command line. > This should work unless it involves constant data structures referencing > symbols from auto-imported DLLs. > .libs/libgstffmpeg_la-gstffmpeg.o: In function `plugin_init': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:134: > undefined reference to `av_log_set_callback' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:139: > undefined reference to `av_register_all' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:154: > undefined reference to `register_protocol' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:155: > undefined reference to `register_protocol' > .libs/libgstffmpeg_la-gstffmpeg.o: In function `gst_ffmpeg_avcodec_open': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:52: > undefined reference to `avcodec_open' > .libs/libgstffmpeg_la-gstffmpeg.o: In function `gst_ffmpeg_avcodec_close': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:64: > undefined reference to `avcodec_close' > .libs/libgstffmpeg_la-gstffmpeg.o: In function > `gst_ffmpeg_av_find_stream_info': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:76: > undefined reference to `av_find_stream_info' > .libs/libgstffmpeg_la-gstffmpegcodecmap.o: In function > `gst_ffmpeg_get_palette': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:52: > undefined reference to `av_free' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:53: > undefined reference to `av_malloc' > .libs/libgstffmpeg_la-gstffmpegcodecmap.o: In function > `gst_ffmpeg_codecid_to_caps': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:1547: > undefined reference to `avcodec_find_decoder' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:1547: > undefined reference to `avcodec_find_encoder' > .libs/libgstffmpeg_la-gstffmpegcodecmap.o: In function > `gst_ffmpeg_caps_with_codecid': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2250: > undefined reference to `av_free' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2276: > undefined reference to `av_mallocz' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2325: > undefined reference to `av_mallocz' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2291: > undefined reference to `av_mallocz' > .libs/libgstffmpeg_la-gstffmpegmake[2]: Leaving directory > `/home/Marko/gst-ffmpeg-0.10.11/ext/ffmpeg' > make[1]: Leaving directory `/home/Marko/gst-ffmpeg-0.10.11/ext' > codecmap.o: In function `gst_ffmpeg_caps_to_codecid': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3288: > undefined reference to `avcodec_find_decoder_by_name' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3288: > undefined reference to `avcodec_find_encoder_by_name' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3300: > undefined reference to `avcodec_find_decoder_by_name' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3300: > undefined reference to `avcodec_find_encoder_by_name' > .libs/libgstffmpeg_la-gstffmpegutils.o: In function > `gst_ffmpeg_get_codecid_longname': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:31: > undefined reference to `avcodec_find_decoder' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:31: > undefined reference to `avcodec_find_encoder' > .libs/libgstffmpeg_la-gstffmpegutils.o: In function `new_aligned_buffer': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:410: > undefined reference to `av_malloc' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:412: > undefined reference to `av_free' > .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_init': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:248: > undefined reference to `avcodec_alloc_context' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:249: > undefined reference to `avcodec_alloc_frame' > .libs/libgstffmpeg_la-gstffmpegenc.o: In function > `gst_ffmpegenc_encode_audio': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:860: > undefined reference to `avcodec_encode_audio' > .libs/libgstffmpeg_la-gstffmpegenc.o: In function > `gst_ffmpegenc_chain_audio': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:913: > undefined reference to `av_get_bits_per_sample_format' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1024: > undefined reference to `av_get_bits_per_sample' > .libs/libgstffmpeg_la-gstffmpegenc.o: In function > `gst_ffmpegenc_flush_buffers': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1071: > undefined reference to `avcodec_encode_video' > .libs/libgstffmpeg_la-gstffmpegenc.o: In function > `gst_ffmpeg_time_gst_to_ff': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: > undefined reference to `av_rescale_q' > .libs/libgstffmpeg_la-gstffmpegenc.o: In function > `gst_ffmpegenc_chain_video': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:781: > undefined reference to `avcodec_encode_video' > .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_getcaps': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:448: > undefined reference to `avcodec_alloc_context' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:487: > undefined reference to `av_free' > .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_setcaps': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:529: > undefined reference to `avcodec_get_context_defaults' > .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_finalize': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:298: > undefined reference to `av_free' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:299: > undefined reference to `av_free' > .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_register': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1307: > undefined reference to `av_codec_next' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1378: > undefined reference to `av_codec_next' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1378: > undefined reference to `av_codec_next' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_init': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:425: > undefined reference to `avcodec_alloc_context' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:426: > undefined reference to `avcodec_alloc_frame' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function > `gst_ffmpegdec_flush_pcache': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2351: > undefined reference to `av_parser_close' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2352: > undefined reference to `av_parser_init' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function > `gst_ffmpegdec_audio_frame': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2109: > undefined reference to `avcodec_decode_audio2' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function > `gst_ffmpegdec_video_frame': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1754: > undefined reference to `avcodec_decode_video' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function `get_output_buffer': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1568: > undefined reference to `av_picture_copy' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_chain': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2510: > undefined reference to `avcodec_flush_buffers' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2589: > undefined reference to `av_parser_parse' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_close': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:587: > undefined reference to `av_free' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:592: > undefined reference to `av_free' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:601: > undefined reference to `av_parser_close' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_setcaps': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:733: > undefined reference to `avcodec_get_context_defaults' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_open': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:641: > undefined reference to `av_parser_init' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:656: > undefined reference to `av_parser_init' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function > `gst_ffmpegdec_get_buffer': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:989: > undefined reference to `avcodec_default_get_buffer' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1016: > undefined reference to `avcodec_align_dimensions' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1032: > undefined reference to `avcodec_default_get_buffer' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_finalize': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:456: > undefined reference to `av_free' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:461: > undefined reference to `av_free' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function > `gst_ffmpegdec_sink_event': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2378: > undefined reference to `avcodec_flush_buffers' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_register': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2850: > undefined reference to `av_codec_next' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2973: > undefined reference to `av_codec_next' > .libs/libgstffmpeg_la-gstffmpegdec.o: In function > `gst_ffmpegdec_release_buffer': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1078: > undefined reference to `avcodec_default_release_buffer' > .libs/libgstffmpeg_la-gstffmpegcfg.o: In function > `gst_ffmpeg_cfg_install_property': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcfg.c:769: > undefined reference to `avcodec_alloc_context' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcfg.c:771: > undefined reference to `avcodec_get_context_defaults' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcfg.c:882: > undefined reference to `av_free' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function > `gst_ffmpegdemux_close': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:344: > undefined reference to `av_close_input_file' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function > `gst_ffmpeg_time_ff_to_gst': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: > undefined reference to `av_rescale_q' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function > `gst_ffmpegdemux_do_seek': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:442: > undefined reference to `av_find_default_stream_index' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function > `gst_ffmpeg_time_gst_to_ff': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: > undefined reference to `av_rescale_q' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function > `gst_ffmpegdemux_do_seek': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:468: > undefined reference to `av_index_search_timestamp' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function > `gst_ffmpeg_time_ff_to_gst': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: > undefined reference to `av_rescale_q' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function > `gst_ffmpegdemux_do_seek': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:488: > undefined reference to `av_seek_frame' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function > `gst_ffmpeg_time_ff_to_gst': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: > undefined reference to `av_rescale_q' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: > undefined reference to `av_rescale_q' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_loop': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1345: > undefined reference to `av_read_frame' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function > `gst_ffmpeg_time_ff_to_gst': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: > undefined reference to `av_rescale_q' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: > undefined reference to `av_rescale_q' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_loop': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1473: > undefined reference to `av_free_packet' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_open': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1162: > undefined reference to `av_open_input_file' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_loop': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1436: > undefined reference to `av_picture_copy' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1551: > undefined reference to `av_free_packet' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1562: > undefined reference to `av_free_packet' > .libs/libgstffmpeg_la-gstffmpegdemux.o: In function > `gst_ffmpegdemux_register': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1865: > undefined reference to `av_iformat_next' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:2025: > undefined reference to `av_iformat_next' > .libs/libgstffmpeg_la-gstffmpegmux.o: In function > `gst_ffmpegmux_collected': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:638: > undefined reference to `url_fopen' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:645: > undefined reference to `av_set_parameters' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:652: > undefined reference to `av_write_header' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:662: > undefined reference to `put_flush_packet' > .libs/libgstffmpeg_la-gstffmpegmux.o: In function > `gst_ffmpeg_time_gst_to_ff': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: > undefined reference to `av_rescale_q' > .libs/libgstffmpeg_la-gstffmpegmux.o: In function > `gst_ffmpegmux_collected': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:754: > undefined reference to `av_write_frame' > .libs/libgstffmpeg_la-gstffmpegmux.o: In function > `gst_ffmpeg_time_gst_to_ff': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: > undefined reference to `av_rescale_q' > .libs/libgstffmpeg_la-gstffmpegmux.o: In function > `gst_ffmpegmux_collected': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:735: > undefined reference to `av_picture_copy' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:760: > undefined reference to `av_write_trailer' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:762: > undefined reference to `put_flush_packet' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:763: > undefined reference to `url_fclose' > .libs/libgstffmpeg_la-gstffmpegmux.o: In function > `gst_ffmpegmux_change_state': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:801: > undefined reference to `url_fclose' > .libs/libgstffmpeg_la-gstffmpegmux.o: In function > `gst_ffmpegmux_request_new_pad': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:453: > undefined reference to `av_new_stream' > .libs/libgstffmpeg_la-gstffmpegmux.o: In function `gst_ffmpegmux_register': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:877: > undefined reference to `av_oformat_next' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:951: > undefined reference to `av_oformat_next' > .libs/libgstffmpeg_la-gstffmpegdeinterlace.o: In function > `gst_ffmpegdeinterlace_chain': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:190: > undefined reference to `avpicture_deinterlace' > .libs/libgstffmpeg_la-gstffmpegdeinterlace.o: In function > `gst_ffmpegdeinterlace_sink_setcaps': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:124: > undefined reference to `avcodec_alloc_context' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:136: > undefined reference to `av_free' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:139: > undefined reference to `avpicture_get_size' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:145: > undefined reference to `gst_caps_set_value' > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:130: > undefined reference to `av_free' > .libs/libgstffmpeg_la-gstffmpegaudioresample.o: In function > `gst_ffmpegaudioresample_transform': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegaudioresample.c:278: > undefined reference to `audio_resample' > .libs/libgstffmpeg_la-gstffmpegaudioresample.o: In function > `gst_ffmpegaudioresample_set_caps': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegaudioresample.c:251: > undefined reference to `audio_resample_init' > .libs/libgstffmpeg_la-gstffmpegaudioresample.o: In function > `gst_ffmpegaudioresample_finalize': > C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegaudioresample.c:154: > undefined reference to `audio_resample_close' > collect2: ld returned 1 exit status > make[2]: *** [libgstffmpeg.la] Error 1 > make[1]: *** [install-recursive] Error 1 > make: *** [install-recursive] Error 1 > > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > > > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From gibrovacco at gmail.com Sun Jan 30 13:40:53 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Sun, 30 Jan 2011 14:40:53 +0200 Subject: [gst-devel] SILK wrappers ready for testing Message-ID: Hi all, a gstreamer plugin delivering the SILK speech codec elements is available at: http://gitorious.org/baco The elements wrap around reference code extracted from: http://tools.ietf.org/id/draft-vos-silk-01.txt, document which needs to be downloaded separately. Still RTP payloader and depayloader are missing, so real streaming is not yet actually possible. It would be good if who's interested could send me some comments / enhancements / suggestions to bring it at a good level and, possibly, to have one glorious day it accepted in gst-plugins-ugly. Regards From nico at inattendu.org Sun Jan 30 14:53:10 2011 From: nico at inattendu.org (Nicolas Bertrand) Date: Sun, 30 Jan 2011 17:53:10 +0400 Subject: [gst-devel] add_buffer_probe in python do nothing with gstreamer 0.10.32 Message-ID: <4D456D46.7050307@inattendu.org> Hi, I use gstreamer via pygst. with the version 0.10.32 from launchpad dev. ppa I did not success to execute add_buffer_probe = The callback is never executed. This works fine with an older version of python-gst / libgstreamer. (cf. version details below) Code Example : def _capture_start_cb(self, wbin, b) : pad = self.get_by_name("JpegEncoder").src_pads().next() self._grabprobe = pad.add_buffer_probe(self._cb_process_frame, True) def _cb_process_frame(self, pad, gst_buffer, ret): """ Callbak to inidicate that buffer is available from probe """ pad.remove_buffer_probe(self._grabprobe) return True Did somebody know if it is a misuse or a real problem ( in that case I fill a bug) ? cheers, nico Problem constated Versions : Ubuntu/lucid (64 bits) python-gst0.10 0.10.21-1~lucid1 libgstreamer0.10-0 0.10.32-1~lucid1 No problem constated version : Ubuntu/maverick (32 bits) python-gst0.10 0.10.19.4-1~lucid1 libgstreamer0.10-0 0.10.32-1~lucid1 From higher.orbital at gmail.com Sun Jan 30 18:25:19 2011 From: higher.orbital at gmail.com (Jonathan Lyons) Date: Sun, 30 Jan 2011 12:25:19 -0500 Subject: [gst-devel] Gstreamer and Cocoa run loops In-Reply-To: <4D456D46.7050307@inattendu.org> References: <4D456D46.7050307@inattendu.org> Message-ID: I'm having a lot of difficulty integrating the bus of a Gstreamer pipeline with a Cocoa run loop. Right now I'm using a CFMessagePort to turn a sync signal on one thread into an async event on the main thread. However, this results in weird delays and bugs. Has anyone on this list tried this? Would it be better to run all media operations on another thread which itself runs a glib main loop? I look forward to any response. Thank you. On Jan 30, 2011, at 8:53 AM, Nicolas Bertrand wrote: > Hi, > I use gstreamer via pygst. > with the version 0.10.32 from launchpad dev. ppa I did not success to > execute add_buffer_probe = The callback is never executed. > > This works fine with an older version of python-gst / libgstreamer. (cf. > version details below) > > Code Example : > > def _capture_start_cb(self, wbin, b) : > > pad = self.get_by_name("JpegEncoder").src_pads().next() > > self._grabprobe = pad.add_buffer_probe(self._cb_process_frame, True) > > > > def _cb_process_frame(self, pad, gst_buffer, ret): > > """ Callbak to inidicate that buffer is available from probe """ > > > > pad.remove_buffer_probe(self._grabprobe) > > > return True > > > Did somebody know if it is a misuse or a real problem ( in that case I > fill a bug) ? > > cheers, > nico > > Problem constated Versions : > Ubuntu/lucid (64 bits) > python-gst0.10 0.10.21-1~lucid1 > libgstreamer0.10-0 0.10.32-1~lucid1 > > No problem constated version : > Ubuntu/maverick (32 bits) > python-gst0.10 0.10.19.4-1~lucid1 > libgstreamer0.10-0 0.10.32-1~lucid1 > > > > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > 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 Jan 30 22:02:52 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sun, 30 Jan 2011 23:02:52 +0200 Subject: [gst-devel] Properties of a GST_OBJECT In-Reply-To: <1296207047364-3244042.post@n4.nabble.com> References: <1296207047364-3244042.post@n4.nabble.com> Message-ID: <4D45D1FC.6020703@hora-obscura.de> Am 28.01.2011 11:30, schrieb gekko: > > How can i get all properties of a GST_OBJECT? maybe you can be more specific. A GstObject is subclassed from GObject and there are thinks like g_object_get() or g_object_class_list_properties() Stefan From david.henningsson at canonical.com Mon Jan 31 06:07:22 2011 From: david.henningsson at canonical.com (David Henningsson) Date: Mon, 31 Jan 2011 06:07:22 +0100 Subject: [gst-devel] Fighting rewinds / pulseaudio crash - update Message-ID: <4D46438A.40403@canonical.com> So to recap a little on the fighting rewinds issue - the one that makes pulseaudio crash due to spending too much time in RT prio, and with the pulseaudio log filled with rewinds. A month ago I provided five patches in total, two in gstreamer[1] and three in pulseaudio. GStreamer #1: we have the "don't start the stream until we have written some samples" - this fix is in gstreamer HEAD, committed by Wim Taymans. GStreamer #2: send larger packages by removing a strange limit to gstreamer's "segment size". This fix has been posted to the gstreamer-devel mailinglist, but not committed AFAIK. I'm attaching a version of the patch which has proper git headers and against git head. Both GStreamer fixes have recently been uploaded to Ubuntu Natty, the development version of Ubuntu. PulseAudio #1 and #2: I believe these should be applied to PulseAudio upstream. PulseAudio #3: A person with alias h31 helped out here, and noticed this patch was the one causing a regression in Skype. While I haven't exactly figured out why, I do notice that this patch could cause problems for very short samples. So let's drop this one for the time being. For reference I resend the three patches (one in GStreamer, two in pulseaudio) that I would like to see applied. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic [1] Well, Wim Taymans has some credit for the helping out with the first one. -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-Fighting-rewinds-Seek-and-write-data-in-the-same-mes.patch Type: text/x-patch Size: 3596 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: 0002-Fighting-rewinds-Reduce-calls-to-handle_seek.patch Type: text/x-patch Size: 5225 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-Pulsesink-Allow-chunks-up-to-bufsize-instead-of-segs.patch Type: text/x-patch Size: 2344 bytes Desc: not available URL: From rohitratri at gmail.com Mon Jan 31 07:48:36 2011 From: rohitratri at gmail.com (Rohit Atri) Date: Mon, 31 Jan 2011 12:18:36 +0530 Subject: [gst-devel] Gstreamer and Cocoa run loops In-Reply-To: References: <4D456D46.7050307@inattendu.org> Message-ID: Actually I'm trying to do a similar thing with Qt. Right now I have a 0sec timer in Qtimer - every time it goes off, I manually pull messages from the bus. Problem is both exec() of a Qthread and main loops run can't co-exist. So what do people normally do it such scenarios? Thanks in advance! On Sun, Jan 30, 2011 at 10:55 PM, Jonathan Lyons wrote: > I'm having a lot of difficulty integrating the bus of a Gstreamer pipeline > with a Cocoa run loop. Right now I'm using a CFMessagePort to turn a sync > signal on one thread into an async event on the main thread. However, this > results in weird delays and bugs. > > Has anyone on this list tried this? Would it be better to run all media > operations on another thread which itself runs a glib main loop? > > I look forward to any response. Thank you. > > On Jan 30, 2011, at 8:53 AM, Nicolas Bertrand wrote: > > > Hi, > > I use gstreamer via pygst. > > with the version 0.10.32 from launchpad dev. ppa I did not success to > > execute add_buffer_probe = The callback is never executed. > > > > This works fine with an older version of python-gst / libgstreamer. (cf. > > version details below) > > > > Code Example : > > > > def _capture_start_cb(self, wbin, b) : > > > > pad = self.get_by_name("JpegEncoder").src_pads().next() > > > > self._grabprobe = pad.add_buffer_probe(self._cb_process_frame, > True) > > > > > > > > def _cb_process_frame(self, pad, gst_buffer, ret): > > > > """ Callbak to inidicate that buffer is available from probe """ > > > > > > > > pad.remove_buffer_probe(self._grabprobe) > > > > > > return True > > > > > > Did somebody know if it is a misuse or a real problem ( in that case I > > fill a bug) ? > > > > cheers, > > nico > > > > Problem constated Versions : > > Ubuntu/lucid (64 bits) > > python-gst0.10 0.10.21-1~lucid1 > > libgstreamer0.10-0 0.10.32-1~lucid1 > > > > No problem constated version : > > Ubuntu/maverick (32 bits) > > python-gst0.10 0.10.19.4-1~lucid1 > > libgstreamer0.10-0 0.10.32-1~lucid1 > > > > > > > > > > > > > ------------------------------------------------------------------------------ > > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > > Finally, a world-class log management solution at an even better > price-free! > > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > > February 28th, so secure your free ArcSight Logger TODAY! > > http://p.sf.net/sfu/arcsight-sfd2d > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better > price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > 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 Mon Jan 31 08:07:41 2011 From: msmith at xiph.org (Michael Smith) Date: Sun, 30 Jan 2011 23:07:41 -0800 Subject: [gst-devel] Gstreamer and Cocoa run loops In-Reply-To: References: <4D456D46.7050307@inattendu.org> Message-ID: I'm not familiar with the specifics of Cocoa, or what CFMessagePort does/how it works, so I can't really comment on that. However, in general, the approach you're using sounds very sensible. My advice is usually: - if appropriate to your platform, simply run a GMainLoop and integrate through that. Easy, and well documented. - Otherwise, connect to the sync-message signal on the bus. Then handle that directly for any messages you do need to handle synchronously (which is unusual), and dispatch to your native platform event loop to handle messages asynchronously. This sounds like what you're doing. Generally, the only tricky bit with that is getting the reference counting right - which isn't hard, but can sometimes be not quite obvious. If you're having "weird delays and bugs" you need to figure out what's causing that - your general approach is the right one. Mike On Sun, Jan 30, 2011 at 9:25 AM, Jonathan Lyons wrote: > I'm having a lot of difficulty integrating the bus of a Gstreamer pipeline with a Cocoa run loop. Right now I'm using a CFMessagePort to turn a sync signal on one thread into an async event on the main thread. However, this results in weird delays and bugs. > > Has anyone on this list tried this? Would it be better to run all media operations on another thread which itself runs a glib main loop? > > I look forward to any response. Thank you. > > On Jan 30, 2011, at 8:53 AM, Nicolas Bertrand wrote: > >> Hi, >> I use gstreamer via pygst. >> with the version 0.10.32 from launchpad dev. ppa I did not success to >> execute add_buffer_probe = The callback is never executed. >> >> This works fine with an older version of python-gst / libgstreamer. (cf. >> version details below) >> >> Code Example : >> >> ? def _capture_start_cb(self, wbin, b) : >> >> ? ? ? ? pad = self.get_by_name("JpegEncoder").src_pads().next() >> >> ? ? ? ? self._grabprobe = pad.add_buffer_probe(self._cb_process_frame, True) >> >> >> >> ? ? def ?_cb_process_frame(self, pad, gst_buffer, ret): >> >> ? ? ? ? """ Callbak to inidicate that buffer is available from probe """ >> >> >> >> ? ? ? ? pad.remove_buffer_probe(self._grabprobe) >> >> >> ? ? ? ? return True >> >> >> Did somebody know if it is a misuse or a real problem ( in that case I >> fill a bug) ? >> >> cheers, >> nico >> >> Problem constated Versions : >> Ubuntu/lucid (64 bits) >> python-gst0.10 0.10.21-1~lucid1 >> libgstreamer0.10-0 0.10.32-1~lucid1 >> >> No problem constated version : >> Ubuntu/maverick (32 bits) >> python-gst0.10 0.10.19.4-1~lucid1 >> libgstreamer0.10-0 0.10.32-1~lucid1 >> >> >> >> >> >> ------------------------------------------------------------------------------ >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> Finally, a world-class log management solution at an even better price-free! >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> February 28th, so secure your free ArcSight Logger TODAY! >> http://p.sf.net/sfu/arcsight-sfd2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From david.henningsson at canonical.com Mon Jan 31 11:46:15 2011 From: david.henningsson at canonical.com (David Henningsson) Date: Mon, 31 Jan 2011 11:46:15 +0100 Subject: [gst-devel] [pulseaudio-discuss] Fighting rewinds / pulseaudio crash - update In-Reply-To: References: <4D46438A.40403@canonical.com> Message-ID: <4D4692F7.1080307@canonical.com> On 2011-01-31 10:41, Colin Guthrie wrote: > Thanks for the update David :) You're welcome :-) >> PulseAudio #1 and #2: I believe these should be applied to PulseAudio >> upstream. > > OK, I think you've given it sufficient testing and exposer to confirm > this. Can I just ask if these are "safe" if the corresponding gst fixes > are not committed? (i.e. will GST generally bork or should it cope as > well as currently without it's patches?) Both are optimisations - gst patch only will be better than no patch at all, and PA patch only will be better than no patch at all. Most people will see it solved by either patch, but I guess that there are machines so slow that you'll need both. > Also, master only or master+s-q > (I'm personally favouring the latter). Me too. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic From zsfrenkel at yahoo.com Mon Jan 31 11:27:08 2011 From: zsfrenkel at yahoo.com (S F) Date: Mon, 31 Jan 2011 02:27:08 -0800 (PST) Subject: [gst-devel] Gstreamer Winbuilds installation problem Message-ID: <686429.38343.qm@web121403.mail.ne1.yahoo.com> Hi, Looking for a friendly multimedia software package under the windows 7 OS, I am trying to install Gstreamer Winbuilds. Although I had few issues (e.g. iconv.h missing) I managed to install this package (or at least it seems so...). Then, I compiled the sample "hellow world" program in your ducumentation and?linked it successfully with the required static libraries. I am using the Code::Blocks IDE. The GST environment path was set to point to the GST Library. Upon running the program, it crashed, displaying the attached error messages. It seems that the missing DLL is not registered. An attempt to register it manually terminated with "error #193". I also re-installed both the SDK and OSSbuild, with no change. I'll appreciate any help/direction in resolving the same. Thanks, SF -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: GST error message #1.jpg Type: image/jpeg Size: 30174 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: GST error message #2.jpg Type: image/jpeg Size: 99449 bytes Desc: not available URL: From mmikkone at paju.oulu.fi Sun Jan 30 19:53:30 2011 From: mmikkone at paju.oulu.fi (Marko Mikkonen) Date: Sun, 30 Jan 2011 20:53:30 +0200 Subject: [gst-devel] Can't build gst-ffmpeg on MinGW In-Reply-To: <4D45405F.2050404@users.sourceforge.net> References: <4D447660.7040904@paju.oulu.fi> <4D45405F.2050404@users.sourceforge.net> Message-ID: <4D45B3AA.4010003@paju.oulu.fi> Thank you! It worked! That was a really accurate tip. Actually I'm quite impressed :) I never would have found out that :) *Novice bows* -Marko 30.1.2011 12:41, Mark Nauwelaerts kirjoitti: > Marko Mikkonen wrote: >> I have a little problem when building gst-ffmpeg-0.10.11 on MinGW. I >> get warning messages saying that shared libraries of avformat, avcodec >> and avutil are not built. Later I get a lot of linker errors about >> missing references to functions, all of which seem to reside in >> gst-libs/ext/ffmpeg subdirectories. How can I build shared libraries for >> avformat, avcodec and avutil? I'm not very familiar with autotools so >> this is a big problem for me. >> >> These are my commands: >> ./configure --prefix=/c/MinGW --enable-shared --disable-static >> make install > You could try > export lt_cv_deplibs_check_method=pass_all > ./configure > > Important part is that the former variable should have the configure script > convince libtool to also consider a real static lib for linking, rather than > only a static import lib (with dll counterpart). > > Mark. > >> Here are the warnings and error messages: >> >> Making install in ext >> make[1]: Entering directory `/home/Marko/gst-ffmpeg-0.10.11/ext' >> Making install in ffmpeg >> make[2]: Entering directory `/home/Marko/gst-ffmpeg-0.10.11/ext/ffmpeg' >> CCLD libgstffmpeg.la >> >> *** Warning: Trying to link with static lib archive >> ../../gst-libs/ext/ffmpeg/libavformat/libavformat.a. >> *** I have the capability to make that library automatically link in when >> *** you link to this library. But I can only do this if you have a >> *** shared version of the library, which you do not appear to have >> *** because the file extensions .a of this argument makes me believe >> *** that it is just a static archive that I should not use here. >> >> *** Warning: Trying to link with static lib archive >> ../../gst-libs/ext/ffmpeg/libavcodec/libavcodec.a. >> *** I have the capability to make that library automatically link in when >> *** you link to this library. But I can only do this if you have a >> *** shared version of the library, which you do not appear to have >> *** because the file extensions .a of this argument makes me believe >> *** that it is just a static archive that I should not use here. >> >> *** Warning: Trying to link with static lib archive >> ../../gst-libs/ext/ffmpeg/libavutil/libavutil.a. >> *** I have the capability to make that library automatically link in when >> *** you link to this library. But I can only do this if you have a >> *** shared version of the library, which you do not appear to have >> *** because the file extensions .a of this argument makes me believe >> *** that it is just a static archive that I should not use here. >> Info: resolving ___gst_debug_min by linking to __imp____gst_debug_min >> (auto-import) >> Info: resolving _GST_CAT_DEFAULT by linking to __imp__GST_CAT_DEFAULT >> (auto-import) >> Creating library file: .libs/libgstffmpeg.dll.a >> /mingw/bin/../lib/gcc/mingw32/4.5.0/../../../../mingw32/bin/ld.exe: >> warning: auto-importing has been activated without --enable-auto-import >> specified on the command line. >> This should work unless it involves constant data structures referencing >> symbols from auto-imported DLLs. >> .libs/libgstffmpeg_la-gstffmpeg.o: In function `plugin_init': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:134: >> undefined reference to `av_log_set_callback' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:139: >> undefined reference to `av_register_all' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:154: >> undefined reference to `register_protocol' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:155: >> undefined reference to `register_protocol' >> .libs/libgstffmpeg_la-gstffmpeg.o: In function `gst_ffmpeg_avcodec_open': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:52: >> undefined reference to `avcodec_open' >> .libs/libgstffmpeg_la-gstffmpeg.o: In function `gst_ffmpeg_avcodec_close': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:64: >> undefined reference to `avcodec_close' >> .libs/libgstffmpeg_la-gstffmpeg.o: In function >> `gst_ffmpeg_av_find_stream_info': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpeg.c:76: >> undefined reference to `av_find_stream_info' >> .libs/libgstffmpeg_la-gstffmpegcodecmap.o: In function >> `gst_ffmpeg_get_palette': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:52: >> undefined reference to `av_free' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:53: >> undefined reference to `av_malloc' >> .libs/libgstffmpeg_la-gstffmpegcodecmap.o: In function >> `gst_ffmpeg_codecid_to_caps': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:1547: >> undefined reference to `avcodec_find_decoder' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:1547: >> undefined reference to `avcodec_find_encoder' >> .libs/libgstffmpeg_la-gstffmpegcodecmap.o: In function >> `gst_ffmpeg_caps_with_codecid': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2250: >> undefined reference to `av_free' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2276: >> undefined reference to `av_mallocz' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2325: >> undefined reference to `av_mallocz' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:2291: >> undefined reference to `av_mallocz' >> .libs/libgstffmpeg_la-gstffmpegmake[2]: Leaving directory >> `/home/Marko/gst-ffmpeg-0.10.11/ext/ffmpeg' >> make[1]: Leaving directory `/home/Marko/gst-ffmpeg-0.10.11/ext' >> codecmap.o: In function `gst_ffmpeg_caps_to_codecid': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3288: >> undefined reference to `avcodec_find_decoder_by_name' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3288: >> undefined reference to `avcodec_find_encoder_by_name' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3300: >> undefined reference to `avcodec_find_decoder_by_name' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcodecmap.c:3300: >> undefined reference to `avcodec_find_encoder_by_name' >> .libs/libgstffmpeg_la-gstffmpegutils.o: In function >> `gst_ffmpeg_get_codecid_longname': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:31: >> undefined reference to `avcodec_find_decoder' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:31: >> undefined reference to `avcodec_find_encoder' >> .libs/libgstffmpeg_la-gstffmpegutils.o: In function `new_aligned_buffer': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:410: >> undefined reference to `av_malloc' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.c:412: >> undefined reference to `av_free' >> .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_init': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:248: >> undefined reference to `avcodec_alloc_context' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:249: >> undefined reference to `avcodec_alloc_frame' >> .libs/libgstffmpeg_la-gstffmpegenc.o: In function >> `gst_ffmpegenc_encode_audio': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:860: >> undefined reference to `avcodec_encode_audio' >> .libs/libgstffmpeg_la-gstffmpegenc.o: In function >> `gst_ffmpegenc_chain_audio': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:913: >> undefined reference to `av_get_bits_per_sample_format' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1024: >> undefined reference to `av_get_bits_per_sample' >> .libs/libgstffmpeg_la-gstffmpegenc.o: In function >> `gst_ffmpegenc_flush_buffers': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1071: >> undefined reference to `avcodec_encode_video' >> .libs/libgstffmpeg_la-gstffmpegenc.o: In function >> `gst_ffmpeg_time_gst_to_ff': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: >> undefined reference to `av_rescale_q' >> .libs/libgstffmpeg_la-gstffmpegenc.o: In function >> `gst_ffmpegenc_chain_video': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:781: >> undefined reference to `avcodec_encode_video' >> .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_getcaps': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:448: >> undefined reference to `avcodec_alloc_context' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:487: >> undefined reference to `av_free' >> .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_setcaps': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:529: >> undefined reference to `avcodec_get_context_defaults' >> .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_finalize': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:298: >> undefined reference to `av_free' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:299: >> undefined reference to `av_free' >> .libs/libgstffmpeg_la-gstffmpegenc.o: In function `gst_ffmpegenc_register': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1307: >> undefined reference to `av_codec_next' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1378: >> undefined reference to `av_codec_next' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegenc.c:1378: >> undefined reference to `av_codec_next' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_init': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:425: >> undefined reference to `avcodec_alloc_context' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:426: >> undefined reference to `avcodec_alloc_frame' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function >> `gst_ffmpegdec_flush_pcache': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2351: >> undefined reference to `av_parser_close' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2352: >> undefined reference to `av_parser_init' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function >> `gst_ffmpegdec_audio_frame': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2109: >> undefined reference to `avcodec_decode_audio2' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function >> `gst_ffmpegdec_video_frame': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1754: >> undefined reference to `avcodec_decode_video' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function `get_output_buffer': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1568: >> undefined reference to `av_picture_copy' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_chain': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2510: >> undefined reference to `avcodec_flush_buffers' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2589: >> undefined reference to `av_parser_parse' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_close': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:587: >> undefined reference to `av_free' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:592: >> undefined reference to `av_free' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:601: >> undefined reference to `av_parser_close' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_setcaps': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:733: >> undefined reference to `avcodec_get_context_defaults' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_open': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:641: >> undefined reference to `av_parser_init' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:656: >> undefined reference to `av_parser_init' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function >> `gst_ffmpegdec_get_buffer': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:989: >> undefined reference to `avcodec_default_get_buffer' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1016: >> undefined reference to `avcodec_align_dimensions' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1032: >> undefined reference to `avcodec_default_get_buffer' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_finalize': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:456: >> undefined reference to `av_free' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:461: >> undefined reference to `av_free' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function >> `gst_ffmpegdec_sink_event': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2378: >> undefined reference to `avcodec_flush_buffers' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function `gst_ffmpegdec_register': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2850: >> undefined reference to `av_codec_next' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:2973: >> undefined reference to `av_codec_next' >> .libs/libgstffmpeg_la-gstffmpegdec.o: In function >> `gst_ffmpegdec_release_buffer': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdec.c:1078: >> undefined reference to `avcodec_default_release_buffer' >> .libs/libgstffmpeg_la-gstffmpegcfg.o: In function >> `gst_ffmpeg_cfg_install_property': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcfg.c:769: >> undefined reference to `avcodec_alloc_context' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcfg.c:771: >> undefined reference to `avcodec_get_context_defaults' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegcfg.c:882: >> undefined reference to `av_free' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function >> `gst_ffmpegdemux_close': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:344: >> undefined reference to `av_close_input_file' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function >> `gst_ffmpeg_time_ff_to_gst': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: >> undefined reference to `av_rescale_q' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function >> `gst_ffmpegdemux_do_seek': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:442: >> undefined reference to `av_find_default_stream_index' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function >> `gst_ffmpeg_time_gst_to_ff': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: >> undefined reference to `av_rescale_q' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function >> `gst_ffmpegdemux_do_seek': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:468: >> undefined reference to `av_index_search_timestamp' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function >> `gst_ffmpeg_time_ff_to_gst': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: >> undefined reference to `av_rescale_q' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function >> `gst_ffmpegdemux_do_seek': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:488: >> undefined reference to `av_seek_frame' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function >> `gst_ffmpeg_time_ff_to_gst': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: >> undefined reference to `av_rescale_q' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: >> undefined reference to `av_rescale_q' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_loop': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1345: >> undefined reference to `av_read_frame' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function >> `gst_ffmpeg_time_ff_to_gst': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: >> undefined reference to `av_rescale_q' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:59: >> undefined reference to `av_rescale_q' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_loop': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1473: >> undefined reference to `av_free_packet' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_open': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1162: >> undefined reference to `av_open_input_file' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function `gst_ffmpegdemux_loop': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1436: >> undefined reference to `av_picture_copy' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1551: >> undefined reference to `av_free_packet' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1562: >> undefined reference to `av_free_packet' >> .libs/libgstffmpeg_la-gstffmpegdemux.o: In function >> `gst_ffmpegdemux_register': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:1865: >> undefined reference to `av_iformat_next' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdemux.c:2025: >> undefined reference to `av_iformat_next' >> .libs/libgstffmpeg_la-gstffmpegmux.o: In function >> `gst_ffmpegmux_collected': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:638: >> undefined reference to `url_fopen' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:645: >> undefined reference to `av_set_parameters' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:652: >> undefined reference to `av_write_header' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:662: >> undefined reference to `put_flush_packet' >> .libs/libgstffmpeg_la-gstffmpegmux.o: In function >> `gst_ffmpeg_time_gst_to_ff': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: >> undefined reference to `av_rescale_q' >> .libs/libgstffmpeg_la-gstffmpegmux.o: In function >> `gst_ffmpegmux_collected': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:754: >> undefined reference to `av_write_frame' >> .libs/libgstffmpeg_la-gstffmpegmux.o: In function >> `gst_ffmpeg_time_gst_to_ff': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegutils.h:74: >> undefined reference to `av_rescale_q' >> .libs/libgstffmpeg_la-gstffmpegmux.o: In function >> `gst_ffmpegmux_collected': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:735: >> undefined reference to `av_picture_copy' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:760: >> undefined reference to `av_write_trailer' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:762: >> undefined reference to `put_flush_packet' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:763: >> undefined reference to `url_fclose' >> .libs/libgstffmpeg_la-gstffmpegmux.o: In function >> `gst_ffmpegmux_change_state': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:801: >> undefined reference to `url_fclose' >> .libs/libgstffmpeg_la-gstffmpegmux.o: In function >> `gst_ffmpegmux_request_new_pad': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:453: >> undefined reference to `av_new_stream' >> .libs/libgstffmpeg_la-gstffmpegmux.o: In function `gst_ffmpegmux_register': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:877: >> undefined reference to `av_oformat_next' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegmux.c:951: >> undefined reference to `av_oformat_next' >> .libs/libgstffmpeg_la-gstffmpegdeinterlace.o: In function >> `gst_ffmpegdeinterlace_chain': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:190: >> undefined reference to `avpicture_deinterlace' >> .libs/libgstffmpeg_la-gstffmpegdeinterlace.o: In function >> `gst_ffmpegdeinterlace_sink_setcaps': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:124: >> undefined reference to `avcodec_alloc_context' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:136: >> undefined reference to `av_free' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:139: >> undefined reference to `avpicture_get_size' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:145: >> undefined reference to `gst_caps_set_value' >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegdeinterlace.c:130: >> undefined reference to `av_free' >> .libs/libgstffmpeg_la-gstffmpegaudioresample.o: In function >> `gst_ffmpegaudioresample_transform': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegaudioresample.c:278: >> undefined reference to `audio_resample' >> .libs/libgstffmpeg_la-gstffmpegaudioresample.o: In function >> `gst_ffmpegaudioresample_set_caps': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegaudioresample.c:251: >> undefined reference to `audio_resample_init' >> .libs/libgstffmpeg_la-gstffmpegaudioresample.o: In function >> `gst_ffmpegaudioresample_finalize': >> C:\MinGW\msys\1.0\home\Marko\gst-ffmpeg-0.10.11\ext\ffmpeg/gstffmpegaudioresample.c:154: >> undefined reference to `audio_resample_close' >> collect2: ld returned 1 exit status >> make[2]: *** [libgstffmpeg.la] Error 1 >> make[1]: *** [install-recursive] Error 1 >> make: *** [install-recursive] Error 1 >> >> >> >> ------------------------------------------------------------------------------ >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> Finally, a world-class log management solution at an even better price-free! >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> February 28th, so secure your free ArcSight Logger TODAY! >> http://p.sf.net/sfu/arcsight-sfd2d >> >> >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From styve.attia at gmail.com Mon Jan 31 11:48:36 2011 From: styve.attia at gmail.com (StyveA) Date: Mon, 31 Jan 2011 02:48:36 -0800 (PST) Subject: [gst-devel] Avi decoder Message-ID: <1296470916011-3248462.post@n4.nabble.com> Hello everyone ! I'm a newbie here and i'm reading the "Application Development Manual" which is really helpful. But here is my problem : I would like to play an Avi video file. I've seen an example in the manual for an ogg audio file, so i've started from here. But I can't find a working decoder for my avi file.. Anyone can help me plz? Here is my pipeline : pipeline = gst_pipeline_new ("video-player"); source = gst_element_factory_make ("filesrc", "file-source"); demuxer = gst_element_factory_make ("avidemux", "avi-demuxer"); decoder = gst_element_factory_make ("decodebin", "decoder"); conv = gst_element_factory_make ("videoconvert", "converter"); sink = gst_element_factory_make ("autovideosink", "video-output"); Thanks for anyone who can help me ! -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Avi-decoder-tp3248462p3248462.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ensonic at hora-obscura.de Mon Jan 31 12:01:07 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 31 Jan 2011 13:01:07 +0200 Subject: [gst-devel] Avi decoder In-Reply-To: <1296470916011-3248462.post@n4.nabble.com> References: <1296470916011-3248462.post@n4.nabble.com> Message-ID: <4D469673.4030405@hora-obscura.de> On 31.01.2011 12:48, StyveA wrote: > Hello everyone ! > > I'm a newbie here and i'm reading the "Application Development Manual" which > is really helpful. > > But here is my problem : I would like to play an Avi video file. > > I've seen an example in the manual for an ogg audio file, so i've started > from here. > But I can't find a working decoder for my avi file.. > Anyone can help me plz? > > Here is my pipeline : > > pipeline = gst_pipeline_new ("video-player"); > source = gst_element_factory_make ("filesrc", "file-source"); > demuxer = gst_element_factory_make ("avidemux", "avi-demuxer"); > decoder = gst_element_factory_make ("decodebin", "decoder"); > conv = gst_element_factory_make ("videoconvert", "converter"); > sink = gst_element_factory_make ("autovideosink", "video-output"); > > Thanks for anyone who can help me ! decodebin would plug avidemux already. Just use playbin2 or filesrc ! decodebin2 ! ... stefan From t.i.m at zen.co.uk Mon Jan 31 12:08:18 2011 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 31 Jan 2011 11:08:18 +0000 Subject: [gst-devel] Avi decoder In-Reply-To: <1296470916011-3248462.post@n4.nabble.com> References: <1296470916011-3248462.post@n4.nabble.com> Message-ID: <1296472098.5356.1.camel@zingle> On Mon, 2011-01-31 at 02:48 -0800, StyveA wrote: > I'm a newbie here and i'm reading the "Application Development Manual" which > is really helpful. > > But here is my problem : I would like to play an Avi video file. Try gst-launch-0.10 -v playbin2 uri=file:///path/to/foo.avi > I've seen an example in the manual for an ogg audio file, so i've started > from here. > But I can't find a working decoder for my avi file.. > Anyone can help me plz? Hard to help if you don't say what format the video inside the AVI is in... but just use playbin2 (or uridecodebin/decodebin2). gst-ffmpeg provides video decoders for many common formats. Cheers -Tim From janne.grunau at collabora.co.uk Mon Jan 31 12:13:41 2011 From: janne.grunau at collabora.co.uk (Janne Grunau) Date: Mon, 31 Jan 2011 12:13:41 +0100 Subject: [gst-devel] wrong horizontal position of dvbsuboverlay if used behind videomixer? In-Reply-To: <1296302879878-3246076.post@n4.nabble.com> References: <1296302879878-3246076.post@n4.nabble.com> Message-ID: <4D469965.20609@collabora.co.uk> On 29.01.2011 13:07, Thomas Loewe wrote: > > since I use the videomixer to implement picture in picture the horizontal > positioning of subtitles via dvbsuboverlay is incorrect when putting it > behind the videomixer. > > For example: the video resolution is 720x576 in 16:9 and videomixer makes it > 1024x576 (768x576 in 4:3). Subtitles are shown on the left and not centered > because it uses 720 instead 1024. > > The position is of course correct when putting it before the videomixer, but > then the subtitles are overlayd by the pip. > > Is this a bug? Depends on the sample aspect ratio of the subtitles. In which case do the subtitles look correct? Can you make a sample available? Janne From giorgio9 at libero.it Mon Jan 31 14:18:23 2011 From: giorgio9 at libero.it (giorgino) Date: Mon, 31 Jan 2011 05:18:23 -0800 (PST) Subject: [gst-devel] v4l2loopback and gstreamer In-Reply-To: <4D4415B9.6090807@hora-obscura.de> References: <1296224217375-3244534.post@n4.nabble.com> <4D4415B9.6090807@hora-obscura.de> Message-ID: <1296479903222-3248701.post@n4.nabble.com> Hi and thanks for the answer. using the following pipe gst-launch -v v4l2src ! v4l2sink device=/dev/video1 and in another terminal gst-launch v4l2src device=/dev/video1 ! autovideosink it works although the quality is poor. But if I try to launch the second pipe as an application in another language (python, java) I don't see the video because I think the v4l2loopback driver had some limitations. I tried other ways without solve the isse. I repeat my issue is that to be able to use only a webcam and capture it from two different applications. Do you have any ideas? (I tried with udpsink, filesink, symbolic link, fifo without any success) G. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/v4l2loopback-and-gstreamer-tp3244534p3248701.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From giorgio9 at libero.it Mon Jan 31 16:00:24 2011 From: giorgio9 at libero.it (giorgino) Date: Mon, 31 Jan 2011 07:00:24 -0800 (PST) Subject: [gst-devel] gst-launch undecorated Message-ID: <1296486024374-3248896.post@n4.nabble.com> Hi all, I have some pipeline as for example gst-launch v4l2src ! autovideosink in which the result window is decorated (with the 3 button to iconize, to expand and close the windows). Is There a way to leave them? G. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/gst-launch-undecorated-tp3248896p3248896.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From arun.raghavan at collabora.co.uk Mon Jan 31 17:02:29 2011 From: arun.raghavan at collabora.co.uk (Arun Raghavan) Date: Mon, 31 Jan 2011 21:32:29 +0530 Subject: [gst-devel] Fighting rewinds / pulseaudio crash - update In-Reply-To: <4D46438A.40403@canonical.com> References: <4D46438A.40403@canonical.com> Message-ID: <1296489749.18322.10.camel@snowflake> On Mon, 2011-01-31 at 06:07 +0100, David Henningsson wrote: [...] > GStreamer #2: send larger packages by removing a strange limit to > gstreamer's "segment size". This fix has been posted to the > gstreamer-devel mailinglist, but not committed AFAIK. I'm attaching a > version of the patch which has proper git headers and against git head. This one makes sense - could you please put this on Bugzilla? Thanks, Arun From wim.taymans at gmail.com Mon Jan 31 17:16:40 2011 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 31 Jan 2011 17:16:40 +0100 Subject: [gst-devel] Fighting rewinds / pulseaudio crash - update In-Reply-To: <1296489749.18322.10.camel@snowflake> References: <4D46438A.40403@canonical.com> <1296489749.18322.10.camel@snowflake> Message-ID: Well, I just commited it, but for the future it would be better to attach patches to a bugreport. Wim On Mon, Jan 31, 2011 at 5:02 PM, Arun Raghavan wrote: > On Mon, 2011-01-31 at 06:07 +0100, David Henningsson wrote: > [...] >> GStreamer #2: send larger packages by removing a strange limit to >> gstreamer's "segment size". This fix has been posted to the >> gstreamer-devel mailinglist, but not committed AFAIK. I'm attaching a >> version of the patch which has proper git headers and against git head. > > This one makes sense - could you please put this on Bugzilla? > > Thanks, > Arun > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From gwunder at eoir.com Mon Jan 31 17:11:50 2011 From: gwunder at eoir.com (Greg Wunder) Date: Mon, 31 Jan 2011 11:11:50 -0500 Subject: [gst-devel] Disabling Timing Requirement in GStreamer Option? Message-ID: <451B4D021F581D4AB9B270C067B7062C0AD9F084@NEOEVS.eoir.com> Occasionally when running gst-launch I receive timing warnings which resemble the following: WARNING: from element /GstPipeline:pipeline0/GstGLImageSink:glimagesink0 A lot of buffers are being dropped. Additional debug info: ..\..\..\Source\gstreamer\libs\gst\base\gstbasesink.c(2689): gst_base_sink_is_too_late (): /GstPipeline:pipeline0/GstGLImageSink:glimagesink0: There may be a timestamping problem, or this computer is too slow. It appears that these warnings may be related to timing. Is there an option or easy way to turn off this timing requirement at runtime? -------------- next part -------------- An HTML attachment was scrubbed... URL: From hoyt6 at llnl.gov Mon Jan 31 19:15:20 2011 From: hoyt6 at llnl.gov (Hoyt, David) Date: Mon, 31 Jan 2011 10:15:20 -0800 Subject: [gst-devel] Gstreamer Winbuilds installation problem In-Reply-To: <686429.38343.qm@web121403.mail.ne1.yahoo.com> References: <686429.38343.qm@web121403.mail.ne1.yahoo.com> Message-ID: > Looking for a friendly multimedia software package under the windows 7 OS, I am trying to install Gstreamer Winbuilds. > Although I had few issues (e.g. iconv.h missing) I managed to install this package (or at least it seems so...). > Then, I compiled the sample "hellow world" program in your ducumentation and?linked it successfully with the required static libraries. > I am using the Code::Blocks IDE. The GST environment path was set to point to the GST Library. > Upon running the program, it crashed, displaying the attached error messages. > It seems that the missing DLL is not registered. An attempt to register it manually terminated with "error #193". I also re-installed both the > > SDK and OSSbuild, with no change. This should be filed as an issue under OSSBuild -- not in the gstreamer mailing list. Also, please describe the version you used. The errors you saw are because when you installed the SDK, you also installed the python bindings without having python installed. Reinstall without the python bindings and those errors will go away. As for why your program crashed, more info. is needed (can you provide a code sample?). OSSBuild doesn't provide static libraries, so those must be of your own making. If there's a problem w/ those, there's not much anyone can do. From ensonic at hora-obscura.de Mon Jan 31 20:17:41 2011 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 31 Jan 2011 21:17:41 +0200 Subject: [gst-devel] gst-launch undecorated In-Reply-To: <1296486024374-3248896.post@n4.nabble.com> References: <1296486024374-3248896.post@n4.nabble.com> Message-ID: <4D470AD5.5000802@hora-obscura.de> Am 31.01.2011 17:00, schrieb giorgino: > > > Hi all, I have some pipeline as for example > > gst-launch v4l2src ! autovideosink > > in which the result window is decorated (with the 3 button to iconize, to > expand and close the windows). > Is There a way to leave them? Yes, write an application and provide the window :) Stefan > > G. From lazyt at gmx.net Mon Jan 31 20:42:02 2011 From: lazyt at gmx.net (Thomas Loewe) Date: Mon, 31 Jan 2011 11:42:02 -0800 (PST) Subject: [gst-devel] wrong horizontal position of dvbsuboverlay if used behind videomixer? In-Reply-To: <4D469965.20609@collabora.co.uk> References: <1296302879878-3246076.post@n4.nabble.com> <4D469965.20609@collabora.co.uk> Message-ID: <1296502922837-3249517.post@n4.nabble.com> > Depends on the sample aspect ratio of the subtitles. In which case do the subtitles look correct? Only when putting dvbsuboverlay before videomixer. I haven't seen any channel yet (german tv on astra) which looks correct. > Can you make a sample available? Here's a short transportstream and the screenshot of the wrong position: http://ul.to/wegq5g Thanks, Thomas -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/wrong-horizontal-position-of-dvbsuboverlay-if-used-behind-videomixer-tp3246076p3249517.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From david.henningsson at canonical.com Mon Jan 31 21:04:29 2011 From: david.henningsson at canonical.com (David Henningsson) Date: Mon, 31 Jan 2011 21:04:29 +0100 Subject: [gst-devel] Fighting rewinds / pulseaudio crash - update In-Reply-To: References: <4D46438A.40403@canonical.com> <1296489749.18322.10.camel@snowflake> Message-ID: <4D4715CD.2000602@canonical.com> On 2011-01-31 17:16, Wim Taymans wrote: > Well, I just commited it, but for the future it would be better to > attach patches to a bugreport. Ok, thanks. Coming from alsa-devel, I'm used to getting mailinglist stuff being picked up and bug reports forgotten ;-) Anyway, after having read Arun's mail but not yours, I created a bug: https://bugzilla.gnome.org/show_bug.cgi?id=641072 ...feel free to close it the way you like. > > Wim > > On Mon, Jan 31, 2011 at 5:02 PM, Arun Raghavan > wrote: >> On Mon, 2011-01-31 at 06:07 +0100, David Henningsson wrote: >> [...] >>> GStreamer #2: send larger packages by removing a strange limit to >>> gstreamer's "segment size". This fix has been posted to the >>> gstreamer-devel mailinglist, but not committed AFAIK. I'm attaching a >>> version of the patch which has proper git headers and against git head. >> >> This one makes sense - could you please put this on Bugzilla? >> >> Thanks, >> Arun >> >> >> ------------------------------------------------------------------------------ >> Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! >> Finally, a world-class log management solution at an even better price-free! >> Download using promo code Free_Logger_4_Dev2Dev. Offer expires >> February 28th, so secure your free ArcSight Logger TODAY! >> http://p.sf.net/sfu/arcsight-sfd2d >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic From gibrovacco at gmail.com Mon Jan 31 22:01:50 2011 From: gibrovacco at gmail.com (Marco Ballesio) Date: Mon, 31 Jan 2011 23:01:50 +0200 Subject: [gst-devel] Disabling Timing Requirement in GStreamer Option? In-Reply-To: <451B4D021F581D4AB9B270C067B7062C0AD9F084@NEOEVS.eoir.com> References: <451B4D021F581D4AB9B270C067B7062C0AD9F084@NEOEVS.eoir.com> Message-ID: <1296507710.6141.5.camel@heck> Hi, Il giorno lun, 31/01/2011 alle 11.11 -0500, Greg Wunder ha scritto: > Occasionally when running gst-launch I receive timing warnings which > resemble the following: > > WARNING: from > element /GstPipeline:pipeline0/GstGLImageSink:glimagesink0 A lot of > buffers are being dropped. > Additional debug info: > ..\..\..\Source\gstreamer\libs\gst\base\gstbasesink.c(2689): > gst_base_sink_is_too_late > (): /GstPipeline:pipeline0/GstGLImageSink:glimagesink0: > There may be a timestamping problem, or this computer is too slow. > > It appears that these warnings may be related to timing. > Is there an option or easy way to turn off this timing requirement at > runtime? you can disable check on too old timestamps by setting sync=false on the glimagesink. Alternatively, you can increase the value of its max-lateness property (try e.g. some hundreds of milliseconds). Both of the solutions have advantages and disadvantages. Regards > > > > ------------------------------------------------------------------------------ > Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! > Finally, a world-class log management solution at an even better price-free! > Download using promo code Free_Logger_4_Dev2Dev. Offer expires > February 28th, so secure your free ArcSight Logger TODAY! > http://p.sf.net/sfu/arcsight-sfd2d > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From styve.attia at gmail.com Mon Jan 31 12:13:28 2011 From: styve.attia at gmail.com (StyveA) Date: Mon, 31 Jan 2011 03:13:28 -0800 (PST) Subject: [gst-devel] Avi decoder In-Reply-To: <4D469673.4030405@hora-obscura.de> References: <1296470916011-3248462.post@n4.nabble.com> <4D469673.4030405@hora-obscura.de> Message-ID: <1296472408489-3248494.post@n4.nabble.com> Thanks for your reply Stefan =) Hum as I said I'm a newbie, so I don't understand everything yet ^^' If I'm correct, my pipeline should look like this : (?) pipeline = gst_pipeline_new ("video-player"); source = gst_element_factory_make ("filesrc", "file-source"); decoder = gst_element_factory_make ("decodebin2", "decoder"); conv = gst_element_factory_make ("videoconvert", "converter"); sink = gst_element_factory_make ("autovideosink", "video-output"); Am I correct? :o -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Avi-decoder-tp3248462p3248494.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From tomwg123 at gmail.com Mon Jan 31 21:10:46 2011 From: tomwg123 at gmail.com (Tom G) Date: Mon, 31 Jan 2011 12:10:46 -0800 Subject: [gst-devel] Why does adding ! audiorate screw up audio? Message-ID: Hello, I'm fairly new to g-streamer and have been unable to understand why the audio in my pipeline outputs poorly if I add ! audiorate. I can reproduce with a very simple gst-launch command line. gst-launch filesrc location=/tmp/test.flv ! decodebin ! audiorate ! autoaudiosink That produces audio output that has a constant clicking sound in the background. If I remove the ! audiorate element, then it sounds perfect. I started working with an existing pipeline that somebody else designed and am not even confident I should be using the audiorate element since the normal (non-debug) pipeline outputs to an mpegts. Why would it screw things up though? I would have expected the input audio to already be like a perfect stream and for it to output looking the same as on input. Any help is appreciated. -Tom From styve.attia at gmail.com Mon Jan 31 12:17:53 2011 From: styve.attia at gmail.com (StyveA) Date: Mon, 31 Jan 2011 03:17:53 -0800 (PST) Subject: [gst-devel] Avi decoder In-Reply-To: <1296472098.5356.1.camel@zingle> References: <1296470916011-3248462.post@n4.nabble.com> <1296472098.5356.1.camel@zingle> Message-ID: <1296472673350-3248503.post@n4.nabble.com> Thanks Stefan and Tim ! Hum as I said I'm a newbie, so I don't understand everything yet ^^' If I'm correct, my pipeline should look like this : (?) pipeline = gst_pipeline_new ("video-player"); source = gst_element_factory_make ("filesrc", "file-source"); decoder = gst_element_factory_make ("decodebin2", "decoder"); conv = gst_element_factory_make ("videoconvert", "converter"); sink = gst_element_factory_make ("autovideosink", "video-output"); Am I correct? :o Tim> I need to do this in code, I don't want to use a command. But thanks ^^ and for the format inside the AVI.. Oo.. I don't know this.. That's a common movie -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Avi-decoder-tp3248462p3248503.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From styve.attia at gmail.com Mon Jan 31 13:54:08 2011 From: styve.attia at gmail.com (StyveA) Date: Mon, 31 Jan 2011 04:54:08 -0800 (PST) Subject: [gst-devel] Avi decoder In-Reply-To: <4D469673.4030405@hora-obscura.de> References: <1296470916011-3248462.post@n4.nabble.com> <4D469673.4030405@hora-obscura.de> Message-ID: <1296478448979-3248645.post@n4.nabble.com> Thanks Stefan and Tim ! Hum as I said I'm a newbie, so I don't understand everything yet ^^' > decodebin would plug avidemux already. Just use playbin2 or filesrc ! > decodebin2 ! ... > > stefan If I'm correct, my pipeline should look like this : (?) pipeline = gst_pipeline_new ("video-player"); source = gst_element_factory_make ("filesrc", "file-source"); decoder = gst_element_factory_make ("decodebin2", "decoder"); conv = gst_element_factory_make ("videoconvert", "converter"); sink = gst_element_factory_make ("autovideosink", "video-output"); Am I correct? :o And for Tim : The problem is that I need to do it by a code not a command =/ -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Avi-decoder-tp3248462p3248645.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From amartin at vicomtech.org Mon Jan 31 14:06:28 2011 From: amartin at vicomtech.org (amartin) Date: Mon, 31 Jan 2011 05:06:28 -0800 (PST) Subject: [gst-devel] Compositing and GStreamer In-Reply-To: References: <1290934705.2457.5.camel@deumeu> Message-ID: <1296479188999-3248673.post@n4.nabble.com> Dear Timothy and Edward, Please, would you mind posting also the code of the callback functionon PriorityChange that includes the gst_ghost_pad_get_target call? g_object_connect(oper, "signal::input-priority-changed", onPriorityChange, mixer, NULL ); I am trying to accomplish the callback function but i don't achieve it (It is never called). I try with several alternatives from the following code: static void onPriorityChange (GstElement *element, GstPad *pad, int newpriority, // Based on pitivi code gnl.connect("input-priority-changed", ... videoInputPriorityChangedCb implementation in track.py gpointer data) { GstPad *sinkpad; GstPad *ghostpad; printf("PRIORITY %d \n", newpriority); // class with the data GstManager *gstManager = (GstManager *) data; // data is a program object with a lot of data including the mixer // ghostpad = gst_element_get_static_pad (gstManager->mixer, "sink"); ghostpad = GST_GHOST_PAD (pad); sinkpad = gst_ghost_pad_get_target (GST_GHOST_PAD (ghostpad)); // Some code to retrieve the proper xpos and zpos from gstManager according to newpriority g_object_set(sinkpad,"zorder",newpriority,NULL); g_object_set(sinkpad,"xpos",data->left,NULL); g_object_set(sinkpad,"ypos",data->top,NULL); gst_object_unref (sinkpad); } Thank you in advance, Angel -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Compositing-and-GStreamer-tp3059218p3248673.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From styve.attia at gmail.com Mon Jan 31 14:52:23 2011 From: styve.attia at gmail.com (StyveA) Date: Mon, 31 Jan 2011 05:52:23 -0800 (PST) Subject: [gst-devel] Avi decoder In-Reply-To: <1296470916011-3248462.post@n4.nabble.com> References: <1296470916011-3248462.post@n4.nabble.com> Message-ID: <1296481943230-3248759.post@n4.nabble.com> I tried this code : play = gst_element_factory_make ("playbin", "play"); g_object_set (G_OBJECT (play), "uri", argv[1], NULL); bus = gst_pipeline_get_bus (GST_PIPELINE (play)); gst_bus_add_watch (bus, my_bus_callback, loop); gst_object_unref (bus); gst_element_set_state (play, GST_STATE_PLAYING); /* now run */ g_main_loop_run (loop); /* also clean up */ gst_element_set_state (play, GST_STATE_NULL); gst_object_unref (GST_OBJECT (play)); But I get the following error : error : URI not valid. don't understand =/ -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Avi-decoder-tp3248462p3248759.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From amartin at vicomtech.org Mon Jan 31 14:58:50 2011 From: amartin at vicomtech.org (amartin) Date: Mon, 31 Jan 2011 05:58:50 -0800 (PST) Subject: [gst-devel] Video transitions using gnloperation In-Reply-To: <4D2366CF.6020009@brooks.nu> References: <4D22BCF2.90006@brooks.nu> <1294144782.23475.5.camel@localhost> <4D235254.3040103@brooks.nu> <4D235F67.2010200@brooks.nu> <4D2366CF.6020009@brooks.nu> Message-ID: <1296482330871-3248771.post@n4.nabble.com> Dear Lane, I suggest you to take a look at GStreamer Editing Services 0.10.1 approach http://gstreamer.freedesktop.org/releases/gst-editing-services/0.10.1.html It is more or less the same solution described in: http://notes.brooks.nu/gstreamer-video-crossfade-example/ It works fine for me. Best Regards, Angel -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Video-transitions-using-gnloperation-tp3173228p3248771.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From amartin at vicomtech.org Mon Jan 31 15:22:50 2011 From: amartin at vicomtech.org (amartin) Date: Mon, 31 Jan 2011 06:22:50 -0800 (PST) Subject: [gst-devel] Compositing and GStreamer In-Reply-To: References: <1290934705.2457.5.camel@deumeu> Message-ID: <1296483770754-3248816.post@n4.nabble.com> Dear Timothy and Edward, Please, would you mind posting also the code of the callback functionon PriorityChange that includes the gst_ghost_pad_get_target call? g_object_connect(oper, "signal::input-priority-changed", onPriorityChange, mixer, NULL ); I am trying to accomplish the callback function but i don't achieve it (It is never called). I've tried with several alternatives from the following code: static void onPriorityChange (GstElement *element, GstPad *pad, guint newpriority, // Based on gst-inpect gnloperation signal description gpointer data) { GstPad *sinkpad; GstPad *ghostpad; printf("PRIORITY %d \n", newpriority); // class with the data GstManager *gstManager = (GstManager *) data; // data is a program object with a lot of data including the mixer // ghostpad = gst_element_get_static_pad (gstManager->mixer, "sink"); ghostpad = GST_GHOST_PAD (pad); sinkpad = gst_ghost_pad_get_target (GST_GHOST_PAD (ghostpad)); // Some code to retrieve the proper xpos and zpos from gstManager according to newpriority g_object_set(sinkpad,"zorder",newpriority,NULL); g_object_set(sinkpad,"xpos",gstManager->left,NULL); g_object_set(sinkpad,"ypos",gstManager->top,NULL); gst_object_unref (sinkpad); } Thank you in advance, Angel -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Compositing-and-GStreamer-tp3059218p3248816.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From sholzer at my-gekko.com Mon Jan 31 16:05:22 2011 From: sholzer at my-gekko.com (gekko) Date: Mon, 31 Jan 2011 07:05:22 -0800 (PST) Subject: [gst-devel] Avi decoder In-Reply-To: <1296481943230-3248759.post@n4.nabble.com> References: <1296470916011-3248462.post@n4.nabble.com> <1296481943230-3248759.post@n4.nabble.com> Message-ID: <1296486322254-3248913.post@n4.nabble.com> the argv[1] has to look like "file:///home/myvideos/video.avi" -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Avi-decoder-tp3248462p3248913.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From styve.attia at gmail.com Mon Jan 31 16:28:06 2011 From: styve.attia at gmail.com (StyveA) Date: Mon, 31 Jan 2011 07:28:06 -0800 (PST) Subject: [gst-devel] Avi decoder In-Reply-To: <1296486322254-3248913.post@n4.nabble.com> References: <1296470916011-3248462.post@n4.nabble.com> <1296481943230-3248759.post@n4.nabble.com> <1296486322254-3248913.post@n4.nabble.com> Message-ID: <1296487686678-3248953.post@n4.nabble.com> gekko wrote: > > the argv[1] has to look like "file:///home/myvideos/video.avi" > Ah thanks that works fine but not in the framebuffer =/ Do you know what could work in the framebuffer? Decodebin maybe? I absolutly need to read my video in the framebuffer cause I need to use DirectFB which use it.. =/ -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Avi-decoder-tp3248462p3248953.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From styve.attia at gmail.com Mon Jan 31 18:34:45 2011 From: styve.attia at gmail.com (StyveA) Date: Mon, 31 Jan 2011 09:34:45 -0800 (PST) Subject: [gst-devel] Avi decoder In-Reply-To: <1296487686678-3248953.post@n4.nabble.com> References: <1296470916011-3248462.post@n4.nabble.com> <1296481943230-3248759.post@n4.nabble.com> <1296486322254-3248913.post@n4.nabble.com> <1296487686678-3248953.post@n4.nabble.com> Message-ID: <1296495285120-3249260.post@n4.nabble.com> Okay I've had a look to older threads and it seems that I need to use dfbvideosink (which I need..). So I've tried something : pipeline = gst_pipeline_new ("video-player"); source = gst_element_factory_make ("filesrc", "file-source"); decoder = gst_element_factory_make ("decodebin", "decoder"); sink = gst_element_factory_make ("dfbvideosink", "video-output"); But I can't play the video... It stays on a black screen when I run it.. Do I have missed something? a converter maybe? an avidemux? Thanks. -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Avi-decoder-tp3248462p3249260.html Sent from the GStreamer-devel mailing list archive at Nabble.com.