<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div>Not sure how you change the pipeline dynamically in the code to launch two different pipeline. </div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: times new roman,new york,times,serif; background-color: transparent; font-style: normal;"> </div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: times new roman,new york,times,serif; background-color: transparent; font-style: normal;">Both valve and identity plugin blocks the entire pipeline when drop properties are set during gst-launch as the pipeline elements will only be prerolled. </div><div> </div><div>Solution to stop the data passing in one side of the tee element which I can think of is - </div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: times new roman,new york,times,serif; background-color: transparent; font-style: normal;"> Launch the pipeline by creating individual elements by gst_element_factory_make and then </div> <div style="color: rgb(0, 0, 0); font-size: 16px; font-family: times new roman,new york,times,serif; background-color: transparent; font-style: normal;">*) Probe the data and drop the buffers as mentioned in </div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: times new roman,new york,times,serif; background-color: transparent; font-style: normal;">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/chapter-dataaccess.html </div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: times new roman,new york,times,serif; background-color: transparent; font-style: normal;"> </div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: times new roman,new york,times,serif; background-color: transparent; font-style: normal;"> </div> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div dir="ltr"> <hr size="1"> From: "gstreamer-devel-request@lists.freedesktop.org" <gstreamer-devel-request@lists.freedesktop.org> To: gstreamer-devel@lists.freedesktop.org Sent: Monday, 26 August 2013 6:33 PM Subject: gstreamer-devel Digest, Vol 31, Issue 80 </div> <div class="y_msg_container"> Send gstreamer-devel mailing list submissions to <a ymailto="mailto:gstreamer-devel@lists.freedesktop.org" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> To subscribe or unsubscribe via the World Wide Web, visit <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> or, via email, send a message with subject or body 'help' to <a ymailto="mailto:gstreamer-devel-request@lists.freedesktop.org" href="mailto:gstreamer-devel-request@lists.freedesktop.org">gstreamer-devel-request@lists.freedesktop.org</a> You can reach the person managing the list at <a ymailto="mailto:gstreamer-devel-owner@lists.freedesktop.org" href="mailto:gstreamer-devel-owner@lists.freedesktop.org">gstreamer-devel-owner@lists.freedesktop.org</a> When replying, please edit your Subject line so it is more specific than "Re: Contents of gstreamer-devel digest..." Today's Topics: 1. How to Stop/start recording using Valve element (Harish Sakhare) 2. Re: tcpclient failing to detect EOF/EOS? (Peter Maersk-Moller) 3. Problems with gstreamer on Raspberry PI (quali) 4. problem in linking filesrc and video converter (nikit) 5. Is there any proper element that is opposited of funnel? (???) 6. Re: appsrc and GMainloop problem (Deepak Jain) ---------------------------------------------------------------------- Message: 1 Date: Mon, 26 Aug 2013 04:49:52 -0700 (PDT) From: Harish Sakhare <<a ymailto="mailto:harish_sakhare@yahoo.com" href="mailto:harish_sakhare@yahoo.com">harish_sakhare@yahoo.com</a>> To: <a ymailto="mailto:gstreamer-devel@lists.freedesktop.org" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> Subject: How to Stop/start recording using Valve element Message-ID: <<a ymailto="mailto:1377517792915-4661728.post@n4.nabble.com" href="mailto:1377517792915-4661728.post@n4.nabble.com">1377517792915-4661728.post@n4.nabble.com</a>> Content-Type: text/plain; charset=us-ascii Hello All, I am sending a video stream using a C920 HD camera as an MP2TS stream. I am able to see video on android tablet also I used "tee" element to record the video. To start stop video recording I am using "valve" element. The default property of valve=FALSE so I used it in my pipeline it works well the streaming as well as recording is done. But when I used valve = TRUE to stop recording then it stops streaming also. Streaming gets stuck just one frame is displayed on the screen. Here below is my working pipeline with recording:-- data->pipeline = gst_parse_launch("udpsrc caps=\"video/mpegts, systemstream=(boolean)true, packet-size=(int)188\" ! tsdemux ! h264parse ! tee name=myvid ! queue ! amcviddec-omxtiducati1videodecoder ! eglglessink myvid. ! queue ! filesink location=/storage/sdcard0/Movies/Test.mp4", &error); Here below is my pipeline where streaming gets STUCK:- data->pipeline = gst_parse_launch("udpsrc caps=\"video/mpegts, systemstream=(boolean)true, packet-size=(int)188\" ! tsdemux ! h264parse ! tee name=myvid ! queue ! amcviddec-omxtiducati1videodecoder ! eglglessink myvid. ! valve drop = true ! queue ! filesink location=/storage/sdcard0/Movies/Test.mp4", &error); So please any one guide how to use Valve to stop recording and how I used it in my pipeline. I would appreciate any help for my problem. -- View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/How-to-Stop-start-recording-using-Valve-element-tp4661728.html" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/How-to-Stop-start-recording-using-Valve-element-tp4661728.html</a> Sent from the GStreamer-devel mailing list archive at Nabble.com. ------------------------------ Message: 2 Date: Mon, 26 Aug 2013 14:46:12 +0200 From: Peter Maersk-Moller <<a ymailto="mailto:pmaersk@gmail.com" href="mailto:pmaersk@gmail.com">pmaersk@gmail.com</a>> To: Discussion of the development of and with GStreamer <<a ymailto="mailto:gstreamer-devel@lists.freedesktop.org" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a>> Subject: Re: tcpclient failing to detect EOF/EOS? Message-ID: <CAGf_jGmM14o5vVZOLW4rhTQPwNs5Bv2-Uexb=2Nxb6iar=<a ymailto="mailto:NMdw@mail.gmail.com" href="mailto:NMdw@mail.gmail.com">NMdw@mail.gmail.com</a>> Content-Type: text/plain; charset="iso-8859-1" Hi Timm-Phillipp See comments in-line. On Sun, Aug 25, 2013 at 2:14 PM, Tim-Philipp M?ller <<a ymailto="mailto:t.i.m@zen.co.uk" href="mailto:t.i.m@zen.co.uk">t.i.m@zen.co.uk</a>> wrote: > On Sat, 2013-08-24 at 23:59 +0200, Peter Maersk-Moller wrote: > Hi, > > Yes, the prerolling receiver needs a queue that in this case will make > > the pipeline change to playing after it has prerolled. However, that > > is mostly besides the point. > It is not besides the point, because without the pipeline prerolling EOS > propagation will not work. > Right. So what you are saying is this: While a pipeline is prerolling, it can NOT propagate EOS and later on you write that if a source detects EOF, it signals to the sink of the pipeline and then the sink may later signal EOS, when it enters Playing state. Then I conclude this: If GStreamer has no other way to handle EOF for input while prerolling, then GStreamer pipelines has a race condition and as such it can be impossible to determine the exact outcome of a particular scenario. Now I will not race to conclusion and race condtions can be many things and even okay if handled properly, but here the pipeline ends up doing nothing just sitting there and that somehow sounds pretty bad to me. Don't you think that is a problem? Somehow the sink needs to detect if it will NOT receive more data from upstream, and if the pipeline is still prerolled, the sink should send the EOS upstream. At least that that is the only way I know a pipeline will terminate. Am I wrong ? > Apparently tcpclientsrc fails to make the pipeline terminate upon > > EOS/EOF if it is in its prerolling state. > > It is not the source that terminates the pipeline on EOS. It pushes an > EOS even into the pipeline, which then needs to filter through to sinks, > which then post EOS messages on the bus, which the pipeline then > aggregates. > True, but problem still apply. > > > If the tcpserversink disconnects or the connection is disconnected > > for any other reason while the receiver pipeline with tcpclientsrc is > > prerolling, the pipeline may never enter playing state and > > subsequently the pipeline is not terminated even though its source had > > its conection terminated. or at least that is what I seem to > > experience. > > Does the sink not post an error messages if its connection is > disconnected by the peer? Shouldn't it? > Not sure if it does, but the sink at the end of the pipeline should somehow detects when it will never leave the prerolling state and terminate. Otherwise you have the race condition. And that is often pretty bad. Best regards Peter Maersk-Moller -------------- next part -------------- An HTML attachment was scrubbed... URL: <<a href="http://lists.freedesktop.org/archives/gstreamer-devel/attachments/20130826/f0455815/attachment-0001.html" target="_blank">http://lists.freedesktop.org/archives/gstreamer-devel/attachments/20130826/f0455815/attachment-0001.html</a>> ------------------------------ Message: 3 Date: Mon, 26 Aug 2013 05:40:30 -0700 (PDT) From: quali <<a ymailto="mailto:alex.pessoa@qualihouse.com.br" href="mailto:alex.pessoa@qualihouse.com.br">alex.pessoa@qualihouse.com.br</a>> To: <a ymailto="mailto:gstreamer-devel@lists.freedesktop.org" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> Subject: Problems with gstreamer on Raspberry PI Message-ID: <<a ymailto="mailto:1377520830522-4661729.post@n4.nabble.com" href="mailto:1377520830522-4661729.post@n4.nabble.com">1377520830522-4661729.post@n4.nabble.com</a>> Content-Type: text/plain; charset=us-ascii Hello, I developed an application using gstreamer to play songs in sequence on Raspberry PI, the problem is that sometimes the application does not play the sound, but throws no errors and seems to be working, just no sound. I also realized that sometimes even closing and opening the application again it still does not work, even resetting the operating system. I tested the same application on Ubuntu on a PC and it worked normally. Does the problem is the ARM architecture Rasperry PI that gstreamer does not support right? I am being pressed to solve this problem, can someone help me? -- View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/Problems-with-gstreamer-on-Raspberry-PI-tp4661729.html" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/Problems-with-gstreamer-on-Raspberry-PI-tp4661729.html</a> Sent from the GStreamer-devel mailing list archive at Nabble.com. ------------------------------ Message: 4 Date: Mon, 26 Aug 2013 05:41:08 -0700 (PDT) From: nikit <<a ymailto="mailto:nikitbiraj@gmail.com" href="mailto:nikitbiraj@gmail.com">nikitbiraj@gmail.com</a>> To: <a ymailto="mailto:gstreamer-devel@lists.freedesktop.org" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> Subject: problem in linking filesrc and video converter Message-ID: <<a ymailto="mailto:1377520868751-4661730.post@n4.nabble.com" href="mailto:1377520868751-4661730.post@n4.nabble.com">1377520868751-4661730.post@n4.nabble.com</a>> Content-Type: text/plain; charset=us-ascii Hi, I am working on encoding a raw file image to jpeg image in C Below is working structure in comand line to encode image : gst-launch filesrc location=/home/root/one.raw blocksize=608256 ! \ 'video/x-raw-yuv, format=(fourcc)I420,width=(int)704,height=(int)576,framerate=(fraction)0/1' ! ffmpegcolorspace ! 'video/x-raw-yuv,format=(fourcc)I420,width=(int)704,height=(int)576,,framerate=(fraction)0/1' ! vpuenc codec = 12 ! \ filesink location=/home/root/image.jpg Below are pipeline structure : app->src = gst_element_factory_make("filesrc", "mysrc"); app->encoder = gst_element_factory_make("vpuenc", "enc"); app->video_convert = gst_element_factory_make ("ffmpegcolorspace", "converter"); app->imagesink = gst_element_factory_make("filesink", "myvsink"); have created all elements, objects. g_object_set (app->src, "blocksize", BUFF_SIZE, NULL); where BUFF_SIZE is 608256 g_object_set (app->encoder, "codec", 12, NULL); gst_bin_add_many(GST_BIN(app->pipeline), (GstElement*)app->src,app->video_convert,app->encoder, app->imagesink, NULL); filesrc location using : g_object_set(G_OBJECT(app->src), "location", "/home/root/one.raw",NULL); and, filesink location using : g_object_set(G_OBJECT(app->imagesink), "location", "/home/root/tr2.jpg",NULL); I am getting link failed when I try to link app->src and app->video_convert, and app->video_convert and app->encoder Below is C code for linking : link_ok = gst_element_link_filtered (app->src, app->video_convert, gst_caps_new_simple ("video/x-raw-yuv", "format", G_TYPE_STRING, "I420", "width",G_TYPE_INT, 704, "height",G_TYPE_INT, 576, //"framerate",GST_TYPE_FRACTION, 0,1, NULL)); and link_ok = gst_element_link_filtered (app->video_convert, app->encoder, gst_caps_new_simple ("video/x-raw-yuv", "format", G_TYPE_STRING, "I420", "width",G_TYPE_INT, 704, "height",G_TYPE_INT, 576, "framerate",GST_TYPE_FRACTION, 1,1, // "pixel-aspect-ratio",GST_TYPE_FRACTION,1,1, NULL)); Surprisingly the same caps works in command line but if I link through C its giving error. What am I doing wrong? -- View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/problem-in-linking-filesrc-and-video-converter-tp4661730.html" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/problem-in-linking-filesrc-and-video-converter-tp4661730.html</a> Sent from the GStreamer-devel mailing list archive at Nabble.com. ------------------------------ Message: 5 Date: Mon, 26 Aug 2013 21:44:23 +0900 From: ??? <<a ymailto="mailto:hoonh83.lee@gmail.com" href="mailto:hoonh83.lee@gmail.com">hoonh83.lee@gmail.com</a>> To: <a ymailto="mailto:gstreamer-devel@lists.freedesktop.org" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> Subject: Is there any proper element that is opposited of funnel? Message-ID: <CAE-EkXOOhHyYp216HBLj+<a ymailto="mailto:-FwCKDSf6sJyufVHq3J1XLEb7bD2g@mail.gmail.com" href="mailto:-FwCKDSf6sJyufVHq3J1XLEb7bD2g@mail.gmail.com">-FwCKDSf6sJyufVHq3J1XLEb7bD2g@mail.gmail.com</a>> Content-Type: text/plain; charset="iso-8859-1" Hello When playing multiple audio or video stream, I am using funnel instead of using input-selector and I am trying to use output-selector before linking fakesink. this is my approach. / ac3parse - capsfilter - decoder \ / fakesink demux - multiqueue - ac3parse - capsfilter - decoder - funnel - output-selector - fakesink \ ac3parse - capsfilter - decoder / \ fakesink \ h264parse - capsfilter - decoder - input-selector - fakesink Actually, I could not find a proper element that is opposited element of funnel. So I am not sure this approach is proper or not? And I am wondering that output-selector is a kind of opposite of funnel. Thus, Please let me know if you have correct approach or some advice. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <<a href="http://lists.freedesktop.org/archives/gstreamer-devel/attachments/20130826/bd200533/attachment-0001.html" target="_blank">http://lists.freedesktop.org/archives/gstreamer-devel/attachments/20130826/bd200533/attachment-0001.html</a>> ------------------------------ Message: 6 Date: Mon, 26 Aug 2013 18:33:26 +0530 From: Deepak Jain <<a ymailto="mailto:jain1982@gmail.com" href="mailto:jain1982@gmail.com">jain1982@gmail.com</a>> To: Discussion of the development of and with GStreamer <<a ymailto="mailto:gstreamer-devel@lists.freedesktop.org" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a>>, <a ymailto="mailto:t.i.m@zen.co.uk" href="mailto:t.i.m@zen.co.uk">t.i.m@zen.co.uk</a> Subject: Re: appsrc and GMainloop problem Message-ID: <CAJhXTGc4GAbMJrZJ=<a ymailto="mailto:h5xThTdvoeamY0emWZYCfyDjqDsunbj3Q@mail.gmail.com" href="mailto:h5xThTdvoeamY0emWZYCfyDjqDsunbj3Q@mail.gmail.com">h5xThTdvoeamY0emWZYCfyDjqDsunbj3Q@mail.gmail.com</a>> Content-Type: text/plain; charset="iso-8859-1" OK...i removed "codec_data" parameter from caps. But still I dont know why I am getting same no decoder context error like: 0:00:09.504370712 10014 0xb384f80 DEBUG ffmpeg gstffmpegcodecmap.c:2398:gst_ffmpeg_caps_with_codecid: have codec data of size 53 0:00:09.504388937 10014 0xb384f80 DEBUG ffmpeg gstffmpegcodecmap.c:2088:gst_ffmpeg_caps_to_pixfmt: converting caps video/x-h264, width=(int)320, height=(int)240, parsed=(boolean)true, stream-format=(string)avc, alignment=(string)au, pixel-aspect-ratio=(fraction)4/3, codec_data=(buffer)014d400dffe10025674d400dec80a0fd8750808080a000007d200017701c00002b160002b079351807c50a448001000568ebec6c80 0:00:09.504452406 10014 0xb384f80 DEBUG ffmpeg gstffmpegcodecmap.c:2118:gst_ffmpeg_caps_to_pixfmt: setting pixel-aspect-ratio 3/4 = 0.750000 0:00:09.504476042 10014 0xb384f80 DEBUG ffmpeg gstffmpegdec.c:781:gst_ffmpegdec_setcaps:<mydecoder> forcing 25/1 framerate 0:00:09.504492294 10014 0xb384f80 DEBUG ffmpeg gstffmpegdec.c:793:gst_ffmpegdec_setcaps:<mydecoder> sink caps have pixel-aspect-ratio of 4:3 0:00:09.504509792 10014 0xb384f80 DEBUG ffmpeg gstffmpegdec.c:811:gst_ffmpegdec_setcaps:<mydecoder> Using framerate from codec 0:00:09.504522619 10014 0xb384f80 DEBUG ffmpeg gstffmpegdec.c:818:gst_ffmpegdec_setcaps:<mydecoder> trying to enable direct rendering 0:00:09.504537396 10014 0xb384f80 DEBUG ffmpeg gstffmpegdec.c:821:gst_ffmpegdec_setcaps:<mydecoder> disable direct rendering setup for H264 0:00:09.548427471 10014 0xb384f80 DEBUG ffmpeg gstffmpegdec.c:895:gst_ffmpegdec_setcaps:<mydecoder:sink> clipping to 320x240 0:00:09.548465407 10014 0xb384f80 DEBUG ffmpeg gstffmpegdec.c:904:gst_ffmpegdec_setcaps:<mydecoder:sink> final clipping to 320x240 0:00:09.548480556 10014 0xb384f80 DEBUG GST_CAPS gstpad.c:2741:gst_pad_set_caps:<mydecoder:sink> caps 0xb376a00 video/x-h264, width=(int)320, height=(int)240, parsed=(boolean)true, stream-format=(string)avc, alignment=(string)au, pixel-aspect-ratio=(fraction)4/3, codec_data=(buffer)014d400dffe10025674d400dec80a0fd8750808080a000007d200017701c00002b160002b079351807c50a448001000568ebec6c80 0:00:09.548699885 10014 0xb384f80 DEBUG ffmpeg gstffmpegdec.c:2549:gst_ffmpegdec_chain:<mydecoder> received DISCONT 0:00:09.548723745 10014 0xb384f80 ERROR ffmpeg gstffmpegdec.c:2341:gst_ffmpegdec_frame:<mydecoder> no codec context 0:00:09.549552644 10014 0xb384f80 DEBUG ffmpeg gstffmpegdec.c:2566:gst_ffmpegdec_chain:<mydecoder> waiting for keyframe 0:00:09.549607984 10014 0xb384f80 DEBUG ffmpeg gstffmpegdec.c:2571:gst_ffmpegdec_chain:<mydecoder> got keyframe 0:00:09.549626379 10014 0xb384f80 ERROR ffmpeg gstffmpegdec.c:2341:gst_ffmpegdec_frame:<mydecoder> no codec context As I can see from logs, everything is setting perfectly but still no codec context. Is it something related to DISCONT? Thanks for all your help Tim. Amy be I am very close..just a couple of steps away from actual playback:( On Mon, Aug 26, 2013 at 3:39 PM, Tim-Philipp M?ller <<a ymailto="mailto:t.i.m@zen.co.uk" href="mailto:t.i.m@zen.co.uk">t.i.m@zen.co.uk</a>> wrote: > On Mon, 2013-08-26 at 12:58 +0530, Deepak Jain wrote: > > > On appsrc, I am setting Caps as : > > > > GstBuffer* buf = gst_buffer_new_and_alloc(extradatalen); > > memcpy(GST_BUFFER_DATA(buf), extradata, extradatalen); > > GstCaps* caps = gst_caps_new_simple ("video/x-h264", > > "codec_data", GST_TYPE_BUFFER, buf, > > "stream-format", G_TYPE_STRING, "byte-stream", > > "width", G_TYPE_INT, <Width>, > > "height", G_TYPE_INT, <Height>, > > "pixel-aspect-ratio", GST_TYPE_FRACTION, <PixelAspectRatioX>, > > <PixelAspectRatio>, > > "framerate", GST_TYPE_FRACTION, <RateValue>, <RateScale>, > > NULL); > With H.264 in byte-stream format, you should not have codec_data in the > caps. Instead, the SPS/PPS should be sent as data in the stream (note: > in proper byte-stream format of course). > > The parser might just assume it's AVC when it sees codec_data. > > Cheers > -Tim > > > _______________________________________________ > gstreamer-devel mailing list > <a ymailto="mailto:gstreamer-devel@lists.freedesktop.org" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> > <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> > -------------- next part -------------- An HTML attachment was scrubbed... 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