<div class="gmail_quote">On Fri, Jul 31, 2009 at 3:05 PM, <<a href="mailto:gstreamer-devel-request@lists.sourceforge.net">gstreamer-devel-request@lists.sourceforge.net</a>> wrote: <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> Send gstreamer-devel mailing list submissions to <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> To subscribe or unsubscribe via the World Wide Web, visit <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> or, via email, send a message with subject or body 'help' to <a href="mailto:gstreamer-devel-request@lists.sourceforge.net">gstreamer-devel-request@lists.sourceforge.net</a> You can reach the person managing the list at <a href="mailto:gstreamer-devel-owner@lists.sourceforge.net">gstreamer-devel-owner@lists.sourceforge.net</a> When replying, please edit your Subject line so it is more specific than "Re: Contents of gstreamer-devel digest..." Today's Topics: 1. audio video synchornization problem (sreerenj b) 2. New pre-releases of core and base (Jan Schmidt) 3. Re: audio video synchornization problem (AJAY GAUTAM) 4. Re: Pausing bins within a pipeline (Tim-Philipp M?ller) 5. Gstreamer-generated mpg2ts not read with vlc (Albert Costa) 6. Enable h264 and mpeg4 encoder in ubuntu (Nguyen Thanh Trung) ---------------------------------------------------------------------- Message: 1 Date: Fri, 31 Jul 2009 12:39:20 +0530 From: sreerenj b <<a href="mailto:bsreerenj@gmail.com">bsreerenj@gmail.com</a>> Subject: [gst-devel] audio video synchornization problem To: <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <<a href="mailto:c3376b410907310009p2cc26838w217e17a81331ee3@mail.gmail.com">c3376b410907310009p2cc26838w217e17a81331ee3@mail.gmail.com</a>> Content-Type: text/plain; charset="iso-8859-1" Hi, I am getting no soud in the recorder video.Getting only the video for the following pipeline. gst-launch -e rtspsrc location="rtsp:// <a href="http://root:carinov1@10.0.0.100/axis-media/media.amp" target="_blank">root:carinov1@10.0.0.100/axis-media/media.amp</a>" name=rtsp ! queue ! rtph264depay ! video/x-h264, width=320, height=240, framerate=30/1 ! ffdec_h264 ! videorate ! ffenc_mpeg4 ! avimux name=avi ! filesink location=s.avi rtsp. ! queue ! rtpmp4gdepay ! aacparse ! avi. Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock ^CCaught interrupt -- handling interrupt. Interrupt: Stopping pipeline ... EOS on shutdown enabled -- Forcing EOS on the pipeline Waiting for EOS... WARNING: from element /GstPipeline:pipeline0/GstAviMux:avi: No or invalid input audio, AVI stream will be corrupt. Additional debug info: gstavimux.c(1611): gst_avi_mux_stop_file (): /GstPipeline:pipeline0/GstAviMux:avi Got EOS from object "/GstPipeline:pipeline0". EOS received - stopping pipeline... Execution ended after 9806220692 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... ^C But for the following pipeline i got both audio and video.But they are not synchronized!!! first hearing the sound and then the video.(video is lacking behind the audio). gst-launch -e rtspsrc location="rtsp:// <a href="http://root:carinov1@10.0.0.100/axis-media/media.amp" target="_blank">root:carinov1@10.0.0.100/axis-media/media.amp</a>" name=rtsp ! queue ! rtph264depay ! video/x-h264, width=320, height=240, framerate=30/1 ! ffdec_h264 ! videorate ! ffenc_mpeg4 ! avimux name=avi ! filesink location=s.avi rtsp. ! queue ! rtpmp4gdepay ! faad ! audioconvert ! audioresample ! avi. -------------- next part -------------- An HTML attachment was scrubbed... ------------------------------ Message: 2 Date: Fri, 31 Jul 2009 08:28:17 +0100 From: Jan Schmidt <<a href="mailto:thaytan@noraisin.net">thaytan@noraisin.net</a>> Subject: [gst-devel] New pre-releases of core and base To: Discussion of the development of GStreamer <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <1249025297.2653.5.camel@fancy.localdomain> Content-Type: text/plain Hi all, I uploaded new pre-release tarballs of core and base last night, and forgot to send a mail about it. The final release was supposed to be last night, but there were a few bugs that are important enough to warrant a bit more testing. Expect the releases next week, Monday or Tuesday as I get time, and then the Good/Bad freeze starting next Friday (get your module moves arranged!) Current tarballs are: <a href="http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.23.5.tar.bz2" target="_blank">http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.23.5.tar.bz2</a> <a href="http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.23.5.tar.bz2" target="_blank">http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.23.5.tar.bz2</a> and <a href="http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.15.3.tar.bz2" target="_blank">http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.15.3.tar.bz2</a> The changes are: * Flushing and locking fixes in CollectPads, which affects adder, and all muxers. * Hide the details of the new GstStreamConsistency testsuite helper * Make adder reset properly on state change from READY * reset alsasrc on state change properly to avoid crashes. * rename the GType of the stream-selector pads in playbin so they don't clash * Remove a bogus assert in the audiofilter base class. Happy testing! J. -- Jan Schmidt <<a href="mailto:thaytan@noraisin.net">thaytan@noraisin.net</a>> ------------------------------ Message: 3 Date: Fri, 31 Jul 2009 13:19:06 +0530 From: AJAY GAUTAM <<a href="mailto:ajaygautam1981@gmail.com">ajaygautam1981@gmail.com</a>> Subject: Re: [gst-devel] audio video synchornization problem To: Discussion of the development of GStreamer <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:9e93fd980907310049p22501104lb10d1bf062de8869@mail.gmail.com">9e93fd980907310049p22501104lb10d1bf062de8869@mail.gmail.com</a>> Content-Type: text/plain; charset="iso-8859-1" Try this: gst-launch -e rtspsrc location="rtsp:// <a href="http://root:carinov1@10.0.0.100/axis-media/media.amp" target="_blank">root:carinov1@10.0.0.100/axis-media/media.amp</a>" name=rtsp ! queue ! rtph264depay ! video/x-h264, width=320, height=240, framerate=30/1 ! ffdec_h264 ! videorate ! ffenc_mpeg4 ! avimux name=avi ! filesink location=s.avi rtsp. ! queue ! rtpmp4gdepay ! faad ! audioconvert ! audioresample ! avi ! sync=true </blockquote><div> Hi, gst-launch -e rtspsrc location="rtsp://<a href="http://root:carinov1@10.0.0.100/axis-media/media.amp">root:carinov1@10.0.0.100/axis-media/media.amp</a>" name=rtsp ! queue ! rtph264depay ! video/x-h264, width=320, height=240, framerate=30/1 ! ffdec_h264 ! videorate ! ffenc_mpeg4 ! avimux name=avi ! filesink location=sre.avi rtsp. ! queue ! rtpmp4gdepay ! faad ! audioconvert ! audioresample quality=10 ! faac ! avi. sync=true I tried this,but now the audio is lacking behind the video.! </div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> On Fri, Jul 31, 2009 at 12:39 PM, sreerenj b <<a href="mailto:bsreerenj@gmail.com">bsreerenj@gmail.com</a>> wrote: > > > Hi, I am getting no soud in the recorder video.Getting only the video for > the following pipeline. > > gst-launch -e rtspsrc location="rtsp:// > <a href="http://root:carinov1@10.0.0.100/axis-media/media.amp" target="_blank">root:carinov1@10.0.0.100/axis-media/media.amp</a>" name=rtsp ! queue ! > rtph264depay ! video/x-h264, width=320, height=240, framerate=30/1 ! > ffdec_h264 ! videorate ! ffenc_mpeg4 ! avimux name=avi ! filesink > location=s.avi rtsp. ! queue ! rtpmp4gdepay ! aacparse ! avi. > > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > ^CCaught interrupt -- handling interrupt. > Interrupt: Stopping pipeline ... > EOS on shutdown enabled -- Forcing EOS on the pipeline > Waiting for EOS... > WARNING: from element /GstPipeline:pipeline0/GstAviMux:avi: No or invalid > input audio, AVI stream will be corrupt. > Additional debug info: > gstavimux.c(1611): gst_avi_mux_stop_file (): > /GstPipeline:pipeline0/GstAviMux:avi > Got EOS from object "/GstPipeline:pipeline0". > EOS received - stopping pipeline... > Execution ended after 9806220692 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > ^C > > > > > > But for the following pipeline i got both audio and video.But they are not > synchronized!!! first hearing the sound and then the video.(video is lacking > behind the audio). > > gst-launch -e rtspsrc location="rtsp:// > <a href="http://root:carinov1@10.0.0.100/axis-media/media.amp" target="_blank">root:carinov1@10.0.0.100/axis-media/media.amp</a>" name=rtsp ! queue ! > rtph264depay ! video/x-h264, width=320, height=240, framerate=30/1 ! > ffdec_h264 ! videorate ! ffenc_mpeg4 ! avimux name=avi ! filesink > location=s.avi rtsp. ! queue ! rtpmp4gdepay ! faad ! audioconvert ! > audioresample ! avi. > > > > > > > > > > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. <a href="http://p.sf.net/sfu/bobj-july" target="_blank">http://p.sf.net/sfu/bobj-july</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> > > -- Thanx & Regards Ajay Gautam +91-9717785580 -------------- next part -------------- An HTML attachment was scrubbed... ------------------------------ Message: 4 Date: Fri, 31 Jul 2009 10:10:59 +0100 From: Tim-Philipp M?ller <<a href="mailto:t.i.m@zen.co.uk">t.i.m@zen.co.uk</a>> Subject: Re: [gst-devel] Pausing bins within a pipeline To: <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <1249031459.5051.5.camel@zingle> Content-Type: text/plain On Thu, 2009-07-30 at 22:18 -0700, Oliver Yu wrote: Hi, > I'm trying to control multiple videos behind a videomixer and having > some trouble. Each video is contained within a Bin with a GhostPad > and the Bin is connected to the videomixer. I kick off the containing > Pipeline with set_state(gst.STATE_PLAYING). When I try to set_state > on individual Bins to gst.STATE_PAUSED, nothing happens and everything > keeps on playing. Elements in the middle of a pipeline usually behave the same in PAUSED or PLAYING state. Sinks will block when set to PAUSED state though, which will at some point block upstream data flow as well (e.g. when queues fill up). > If I try to do the same on the filesrc within the > Bin, there is still no response. The same applies to non-live sources. > - Is is possible to pause individual Bins within a pipeline? If so, > how? You can block pads to block data flow at certain points in a pipeline. You'll need to make sure that won't lead to other parts of the pipeline 'drying up' (e.g. muxers, videomixer, adder, those kind of elements). > - Is it possible to start certain bins in a paused state when starting > the pipeline? - Also, is there a way to enforce the order of sinks in > the videomixer? I want to be able to control the stacking order of > the videos. Videomixer (sink) pads have "xpos", "ypos", "alpha" and "zorder" properties which you can set. I think "zorder" takes care of stacking order. Cheers -Tim ------------------------------ Message: 5 Date: Fri, 31 Jul 2009 09:28:39 +0000 (GMT) From: Albert Costa <<a href="mailto:costa_albert@yahoo.fr">costa_albert@yahoo.fr</a>> Subject: [gst-devel] Gstreamer-generated mpg2ts not read with vlc To: gstreamer <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:80664.16979.qm@web28411.mail.ukl.yahoo.com">80664.16979.qm@web28411.mail.ukl.yahoo.com</a>> Content-Type: text/plain; charset="utf-8" Hi All, I am producing some mpeg2 ts files with gstreamer that I would like to see with VLC (first with local files, then using streaming). I have following pipeline: gst-launch ksvideosrc ! ffmpegcolorspace ! ffenc_mpeg2video ! ffmux_mpegts ! filesink location=myfile.mpg I'm able to read the file in gstreamer using filesrc ! ffdemux_mpegts ! ffdec_mpeg2video ! ffmpegcolorspace ! directdrawsink I'm able to read the file in windows media player. But... VLC can't display the file. It gets the good width&height, framerate, and displays a frame with good dimensions, but the content is just all black. Is there something in the encoding or muxing that is not standard in gstreamer plugins (I'm running gstreamer winbuilds v0.10.4 on XP) ? Regards, Al -------------- next part -------------- An HTML attachment was scrubbed... ------------------------------ Message: 6 Date: Fri, 31 Jul 2009 02:35:39 -0700 (PDT) From: Nguyen Thanh Trung <<a href="mailto:trungnt_hut@yahoo.com">trungnt_hut@yahoo.com</a>> Subject: [gst-devel] Enable h264 and mpeg4 encoder in ubuntu To: <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <<a href="mailto:687191.95202.qm@web63107.mail.re1.yahoo.com">687191.95202.qm@web63107.mail.re1.yahoo.com</a>> Content-Type: text/plain; charset="utf-8" Hello, May be this's not right place to ask this question. But it'll be great if any one can help. Here's my problem: I need a program to encode video using h264 and mpeg4 codec in gstreamer, but seem gstreamer in ubuntu lacks of these encoder. I've searched google, tried some packages and tried to build the package from source code, too. With gst-ffmpeg source code, mpeg4 encoder is enabled by default but not h264, and although I tried some parameters to enabled h264 encoder, but all failed. So, is there any 1 know already built package(s) with these encoder or how to enabled h264 encoder to build it from gst-ffmpeg source code ? Thanks and best regards. trungnt &quot;T?t h?n, tho?ng g?n h?n, nhanh h?n -Tr?i nghi?m Yahoo! Mail m?i h?m nay! <a href="http://vn.mail.yahoo.com" target="_blank">http://vn.mail.yahoo.com</a>&quot; -------------- next part -------------- An HTML attachment was scrubbed... ------------------------------ ------------------------------------------------------------------------------ Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. <a href="http://p.sf.net/sfu/bobj-july" target="_blank">http://p.sf.net/sfu/bobj-july</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> End of gstreamer-devel Digest, Vol 38, Issue 74 *********************************************** </blockquote></div> -- Sreerenj B Software engineer,Carinov Networks Pvt Ltd <a href="mailto:bsreerenj@gmail.com">bsreerenj@gmail.com</a> mob: +91 9739469496