I see. I have heard of the GstAppSrc but haven't found any documentation for it yet. It appears as if that component would be better suited to deal with my use case. <ul><li>What version of GStreamer contains this new GstAppSrc component? </li><li>Can anyone point me to an example showing how to use the GstAppSrc? </li><li>Does anyone have any suggestions as to what else I could try?</li><li>How difficult is it to make the fakesrc produce time-based data? I have so far stayed away from compiling GStreamer on my Windows machine, but I am willing to give OABuild a try. </li></ul> What I am trying to do basically is the following: In my application I have the encoded raw audio and video samples. I would like to feed them to a GStreamer pipeline for playback and I would prefer not having to write my own plugin. Are there any better ways to accomplish what I am trying to do? And if the only way is to write my own source plugin, can you give me any pointers? I am not looking for source code, just some hints that get me started quickly and on the right track. Thanks for all your help. Andreas <div class="gmail_quote">On Tue, May 6, 2008 at 9:33 AM, Wim Taymans <<a href="mailto:wim.taymans@gmail.com">wim.taymans@gmail.com</a>> wrote: <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> On Tue, 2008-05-06 at 09:05 -0500, Andreas Schuler wrote: You need to force fakesrc into producing time-based data (it produces byte based formats by default). Unfortunatly this is not possible yet. Wim <div><div></div><div class="Wj3C7c"> > I have compared log entries like the following to the samples I am > feeding the pipeline: > > "dshowaudiodec : chain (size 768)=> pts 0:00:00.128000000 stop > 0:00:00.320000000" > > This line is produced by the following statement (from > gstdshowaudiodec.c): > > GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec, "chain (size %d)=> > pts %" >       GST_TIME_FORMAT " stop %" GST_TIME_FORMAT, >       GST_BUFFER_SIZE (buffer), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP > (buffer)), >       GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer) + >           GST_BUFFER_DURATION (buffer))); > > Those timestamps are matching from what I can tell and therefore look > ok to me. > > > > It is possible that I am not setting all necessary data on the > GstBuffer in the fakesrc. The following fields get set at the moment: >       * GST_BUFFER_DATA >       * GST_BUFFER_SIZE >       * GST_BUFFER_TIMESTAMP >       * GST_BUFFER_DURATION >       * GST_BUFFER_FLAG_DISCONT (if present in source sample) >       * GST_BUFFER_FLAG_PREROLL (if present in source sample) > > It is also worth mentioning that my fakesrc has the "is-live" property > set to TRUE. > > Thanks, > > Andreas > > > On Tue, May 6, 2008 at 1:43 AM, Felipe Contreras > <<a href="mailto:felipe.contreras@gmail.com">felipe.contreras@gmail.com</a>> wrote: > >         On Tue, May 6, 2008 at 1:41 AM, Andreas Schuler >         <<a href="mailto:andreas@neokast.com">andreas@neokast.com</a>> wrote: >         > Hi all, >         > >         > I am having some trouble playing back Windows Media content. >         > >         > The setup is somewhat unique but necessary for my >         application: >         > >         > 2 fakesrc components, which are injecting encoded audio and >         video data >         > Sources are connected to capsfilters, on which I am setting >         the correct >         > input caps (see below) >         > Caps filters are connected to a multiqueue >         > Queue is connected to two decoding bins >         > Standard audio and video rendering tails (i.e. convert >         filters, rendering >         > sinks etc.) are connected to decoding bin source pads This >         setup actually >         > works fine for XVID/MP3 encoded content but I didn't have >         any luck getting >         > it to play with WM content. The caps I have for audio and >         video are as >         > follows: >         > >         > audio/x-wma, channels=(int)2, rate=(int)32000, >         wmaversion=(int)2, >         > bitrate=(int)32000, codec_data=(buffer)008800001f0000000000, >         > block_align=(int)768, depth=(int)16 >         >  video/x-wmv, width=(int)720, height=(int)496, >         framerate=(fraction)24/1, >         > wmvversion=(int)2, codec_data=(buffer)baf8ad00, >         format=(fourcc)WMV2 >         > >         > I have the same content in a regular WMV file, which is >         playing back fine >         > when using the playbin. I verified the caps with that setup >         (the only >         > difference appears to be the order of some of the fields and >         the video >         > framerate [the actual value is 23.97fps]): >         > >         > audio/x-wma, wmaversion=(int)2, bitrate=(int)32000, >         depth=(int)16, >         > rate=(int)32000, channels=(int)2, block_align=(int)768, >         > codec_data=(buffer)008800001f0000000000 >         > video/x-wmv, wmvversion=(int)2, framerate=(fraction)25/1, >         width=(int)720, >         > height=(int)496, codec_data=(buffer)baf8ad00, >         format=(fourcc)WMV2 >         > >         > >         > >         > When trying to play the content using the setup mention >         above I get a lot of >         > the following messages from gstsegment.c (702): >         > >         > CRITICAL **: file E:\devel-release\src_releases\gstreamer >         \gst\gstsegment.c: >         > line 702: assertion `segment->format == format' failed >         > >         > Enabling the debug output reveals the following from the >         dshow audio >         > decoder: >         > >         > dshowaudiodec : received new segment from 0:00:00.000000000 >         to >         > 99:99:99.999999999 >         > dshowaudiodec : chain (size 768)=> pts 0:00:00.000000000 >         stop >         > 0:00:00.192000000 >         >  dshowaudiodec : this buffer has a DISCONT flag >         (0:00:00.000000000), >         > flushing >         > dshowaudiodec : chain (size 768)=> pts 0:00:00.128000000 >         stop >         > 0:00:00.320000000 >         > dshowaudiodec : buffer is out of segment, start >         0:00:00.128000000 stop >         > 0:00:00.256000000 >         >  dshowaudiodec : buffer is out of segment, start >         0:00:00.256000000 stop >         > 0:00:00.320000000 >         > dshowaudiodec : chain (size 768)=> pts 0:00:00.320000000 >         stop >         > 0:00:00.512000000 >         > dshowaudiodec : buffer is out of segment, start >         0:00:00.320000000 stop >         > 0:00:00.448000000 >         >  dshowaudiodec : buffer is out of segment, start >         0:00:00.448000000 stop >         > 0:00:00.512000000 >         > dshowaudiodec : chain (size 768)=> pts 0:00:00.512000000 >         stop >         > 0:00:00.704000000 >         > dshowaudiodec : buffer is out of segment, start >         0:00:00.512000000 stop >         > 0:00:00.640000000 >         >  dshowaudiodec : buffer is out of segment, start >         0:00:00.640000000 stop >         > 0:00:00.704000000 >         > dshowaudiodec : chain (size 768)=> pts 0:00:00.704000000 >         stop >         > 0:00:00.896000000 >         > dshowaudiodec : buffer is out of segment, start >         0:00:00.704000000 stop >         > 0:00:00.832000000 >         >  dshowaudiodec : buffer is out of segment, start >         0:00:00.832000000 stop >         > 0:00:00.896000000 >         > >         > The video decoder is showing a similar picture: >         > >         > dshowvideodec : new segment received => >         start=0:00:00.000000000 >         > stop=99:99:99.999999999 >         >  dshowvideodec : chain (size 1749)=> pts 0:00:00.009000000 >         stop >         > 0:00:00.010000000 >         > dshowvideodec : this buffer has a DISCONT flag >         (0:00:00.009000000), flushing >         > dshowvideodec : buffer is out of segment, start >         0:00:00.009000000 stop >         > 0:00:00.010000000 >         >  dshowvideodec : chain (size 2972)=> pts 0:00:02.411000000 >         stop >         > 0:00:02.412000000 >         > dshowvideodec : buffer is out of segment, start >         0:00:02.411000000 stop >         > 0:00:02.412000000 >         > dshowvideodec : chain (size 3735)=> pts 0:00:02.445000000 >         stop >         > 0:00:02.446000000 >         >  dshowvideodec : buffer is out of segment, start >         0:00:02.445000000 stop >         > 0:00:02.446000000 >         > dshowvideodec : chain (size 3500)=> pts 0:00:02.478000000 >         stop >         > 0:00:02.479000000 >         > dshowvideodec : buffer is out of segment, start >         0:00:02.478000000 stop >         > 0:00:02.479000000 >         >  dshowvideodec : chain (size 5141)=> pts 0:00:02.545000000 >         stop >         > 0:00:02.546000000 >         > dshowvideodec : buffer is out of segment, start >         0:00:02.545000000 stop >         > 0:00:02.546000000 >         > dshowvideodec : chain (size 7849)=> pts 0:00:02.578000000 >         stop >         > 0:00:02.579000000 >         >  dshowvideodec : buffer is out of segment, start >         0:00:02.578000000 stop >         > 0:00:02.579000000 >         > dshowvideodec : chain (size 5101)=> pts 0:00:02.611000000 >         stop >         > 0:00:02.612000000 >         > dshowvideodec : buffer is out of segment, start >         0:00:02.611000000 stop >         > 0:00:02.612000000 >         >  dshowvideodec : chain (size 7712)=> pts 0:00:02.645000000 >         stop >         > 0:00:02.646000000 >         > dshowvideodec : buffer is out of segment, start >         0:00:02.645000000 stop >         > 0:00:02.646000000 >         > >         > >         > I understand that the problem appears to be that the >         "format" in GstSegment >         > is not of type GST_FORMAT_TIME when "gst_segment_clip" is >         called in >         > gstdshowaudiocodec.c. The question is why. Can anyone point >         me in the right >         > direction on this one? > > >         What about the timestamps? Are those ok? > >         -- >         Felipe Contreras </div></div>> ------------------------------------------------------------------------- > This SF.net email is sponsored by the 2008 JavaOne(SM) Conference > Don't miss this year's exciting event. There's still time to save $100. > Use priority code J8TL2D2. > <a href="http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone" target="_blank">http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone</a> > _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> </blockquote></div>