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<p class="MsoNormal">I don’t know the protocol about YouTube Live. If it’s RTMP and has hand-shake. It’s possible Adobe Flash Player refused your connection since there’s some encrypted/signature data in hand-shake to verify the sever. It may only accept data
 from Adobe Flash Server.  Maybe you can try an old flash player version.<o:p></o:p></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> gstreamer-devel-bounces+feng.yuan=intel.com@lists.freedesktop.org [mailto:gstreamer-devel-bounces+feng.yuan=intel.com@lists.freedesktop.org]
<b>On Behalf Of </b>Peter Maersk-Moller<br>
<b>Sent:</b> Monday, May 06, 2013 5:18 AM<br>
<b>To:</b> Discussion of the development of and with GStreamer<br>
<b>Subject:</b> Re: Streaming to YouTube Live<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Hi Chuck.<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Thanks for your suggestion.<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">I don't see much else of interesting info from other elements than rtmpsink. Below is an extract of the debug=8  output (perhaps level=5 is max) .I've included a little bit of the pad capability listing too.<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">As you can see, rtmpsink gets to send about 92k of data before it quits with the message RTMP send error 104. It quits roughly 5 times out of 6 with that message and sometimes it just quits (without the RTMP
 send error).<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Suggestions are very welcome .Perhaps rtmpsink just doesn't work with YouTube Live for yet unknown reasons. However largely same setup works with UStream. And I have checked that the URL works with a Flash Live
 Media Encoder sending to YouTube. Perhaps some yet to me unknown mode is invoked by the Flash Live Media Encoder, a mode not yet supported by rtmpsink.<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Anyway, as written, suggestions are very welcome. It would be rather cool if we can get GStreamer and rtmpsink to work with YouTube Live.<o:p></o:p></p>
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<p class="MsoNormal">Best regards<o:p></o:p></p>
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<p class="MsoNormal">Peter<o:p></o:p></p>
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/GstPipeline:pipeline0/GstFaac:faac0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)1, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
 ><br>
/GstPipeline:pipeline0/GstFlvMux:mux.GstPad:audio: caps = audio/mpeg, mpegversion=(int)2, channels=(int)1, rate=(int)44100, stream-format=(string)raw, codec_data=(buffer)<br>
/GstPipeline:pipeline0/GstX264Enc:x264enc0.GstPad:src: caps = video/x-h264, width=(int)640, height=(int)360, framerate=(fraction)30/1, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)014d4029ffe10018674d4029eca05017fcb8088000000300bb9aca00078c18cb01000468ebecb2,
 stream-format=(string)avc, alignment=(string)au, level=(string)4.1, profile=(string)main<br>
/GstPipeline:pipeline0/GstX264Enc:x264enc0.GstPad:sink: caps = video/x-raw-yuv, level=(string)4.1, profile=(string)main, format=(fourcc)YV12, framerate=(fraction)30/1, width=(int)640, height=(int)360, color-matrix=(string)sdtv, chroma-site=(string)mpeg2<br>
/GstPipeline:pipeline0/GstCapsFilter:capsfilter1.GstPad:src: caps = video/x-h264, width=(int)640, height=(int)360, framerate=(fraction)30/1, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)014d4029ffe10018674d4029eca05017fcb8088000000300bb9aca00078c18cb01000468ebecb2,
 stream-format=(string)avc, alignment=(string)au, level=(string)4.1, profile=(string)main<br>
/GstPipeline:pipeline0/GstCapsFilter:capsfilter1.GstPad:sink: caps = video/x-h264, width=(int)640, height=(int)360, framerate=(fraction)30/1, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)014d4029ffe10018674d4029eca05017fcb8088000000300bb9aca00078c18cb01000468ebecb2,
 stream-format=(string)avc, alignment=(string)au, level=(string)4.1, profile=(string)main<br>
/GstPipeline:pipeline0/GstFlvMux:mux.GstPad:video: caps = video/x-h264, width=(int)640, height=(int)360, framerate=(fraction)30/1, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)014d4029ffe10018674d4029eca05017fcb8088000000300bb9aca00078c18cb01000468ebecb2,
 stream-format=(string)avc, alignment=(string)au, level=(string)4.1, profile=(string)main<br>
/GstPipeline:pipeline0/GstFlvMux:mux.GstPad:src: caps = video/x-flv, streamheader=(buffer)< 464c5601050000000900000000, 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,
 0900002c000000000000001700000000014d4029ffe10018674d4029eca05017fcb8088000000300bb9aca00078c18cb01000468ebecb200000037, 0800000200000000000000af000000000d ><br>
/GstPipeline:pipeline0/GstRTMPSink:rtmpsink0.GstPad:sink: caps = video/x-flv, streamheader=(buffer)< 464c5601050000000900000000, 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,
 0900002c000000000000001700000000014d4029ffe10018674d4029eca05017fcb8088000000300bb9aca00078c18cb01000468ebecb200000037, 0800000200000000000000af000000000d ><br>
Pipeline is PREROLLED ...<br>
Setting pipeline to PLAYING ...<br>
New clock: GstSystemClock<br>
0:00:00.812604837  4391      0x1e590f0 DEBUG               rtmpsink gstrtmpsink.c:235:gst_rtmp_sink_render:<rtmpsink0> Opened connection to rtmp://<a href="http://a.rtmp.youtube.com/live2/STREAMNAME_REMOVED">a.rtmp.youtube.com/live2/STREAMNAME_REMOVED</a><br>
0:00:00.812654112  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:241:gst_rtmp_sink_render:<rtmpsink0> Caching first buffer of size 13 for concatenation<br>
0:00:00.812694629  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:249:gst_rtmp_sink_render:<rtmpsink0> Joining 2nd buffer of size 260 to cached buf<br>
0:00:00.812722484  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 273 bytes to RTMP server<br>
0:00:00.812778812  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 59 bytes to RTMP server<br>
0:00:00.812813232  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 17 bytes to RTMP server<br>
0:00:00.812894371  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 166 bytes to RTMP server<br>
0:00:00.813448928  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 9302 bytes to RTMP server<br>
0:00:00.815407311  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 132 bytes to RTMP server<br>
0:00:00.815900534  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 130 bytes to RTMP server<br>
0:00:00.816342378  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 129 bytes to RTMP server<br>
0:00:00.816862090  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 136 bytes to RTMP server<br>
0:00:00.817290072  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 5841 bytes to RTMP server<br>
0:00:00.818967597  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 5849 bytes to RTMP server<br>
0:00:00.820808038  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 5858 bytes to RTMP server<br>
0:00:00.822764003  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 131 bytes to RTMP server<br>
0:00:00.823313489  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 132 bytes to RTMP server<br>
0:00:00.823740494  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 129 bytes to RTMP server<br>
0:00:00.824154593  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 133 bytes to RTMP server<br>
0:00:00.824576933  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 5753 bytes to RTMP server<br>
0:00:00.852905074  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 5846 bytes to RTMP server<br>
0:00:00.854825476  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 5741 bytes to RTMP server<br>
0:00:00.856572691  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 136 bytes to RTMP server<br>
0:00:00.857108289  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 136 bytes to RTMP server<br>
0:00:00.857549728  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 139 bytes to RTMP server<br>
0:00:00.857983149  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 5847 bytes to RTMP server<br>
0:00:00.859692554  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 5768 bytes to RTMP server<br>
0:00:00.861227906  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 139 bytes to RTMP server<br>
0:00:00.861721182  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 149 bytes to RTMP server<br>
0:00:00.862153646  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 146 bytes to RTMP server<br>
0:00:00.862577417  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 145 bytes to RTMP server<br>
0:00:00.863008600  4391      0x1e59050 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 5799 bytes to RTMP server<br>
0:00:00.864681394  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 5791 bytes to RTMP server<br>
0:00:00.872594409  4391      0x1e590f0 LOG                 rtmpsink gstrtmpsink.c:256:gst_rtmp_sink_render:<rtmpsink0> Sending 5753 bytes to RTMP server<br>
ERROR: WriteN, RTMP send error 104 (140 bytes)<br>
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<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
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<p class="MsoNormal">On Wed, Apr 24, 2013 at 3:50 PM, Chuck Crisler <<a href="mailto:ccrisler@mutualink.net" target="_blank">ccrisler@mutualink.net</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Turn on logging to get more info. Logging can deluge you, so start slowly until you get enough info to narrow down. Start with something like 'export GST_DEBUG=*:3', which will give you a lot of messages. Then
 pick out the very few (1-2-3) elements that are interesting and set a higher debugging level for them (and disabling logging for everything else). Something like 'export GST_DEBUG=element1:4,element2:4'<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Then post the logs if the answer isn't obvious.<br>
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<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
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<p class="MsoNormal">On Wed, Apr 24, 2013 at 8:15 AM, Peter Maersk-Moller <<a href="mailto:pmaersk@gmail.com" target="_blank">pmaersk@gmail.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Hi.<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">A little update on using GStreamer to generate a live stream for YouTube Live. Using the following information, the pipeline gets a little bit further.<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">     .... ! flvmux streamable=true name=mux ! rtmpsink location="$LOCATION"<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Here LOCATION is now defined as follows<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">    LOCATION="<span style="font-size:8.5pt;font-family:"Arial","sans-serif";color:#333333">Primary Server URL/Stream Name</span> live=1 flashver=FME/3.0\20(compatible;\20FMSc\201.0) "<br>
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<span style="font-size:8.5pt;font-family:"Arial","sans-serif";color:#333333">Primary Server URL</span> and<span style="font-size:8.5pt;font-family:"Arial","sans-serif";color:#333333"> Stream Name</span> is copied from the settings from the live event created
 for the YouTube account.<o:p></o:p></p>
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<p class="MsoNormal">Nevertheless the script now fails with the following message after apparently setting up communication with the YouTube servers.<o:p></o:p></p>
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<p class="MsoNormal"><br>
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   Pipeline is PREROLLED ...<br>
   Setting pipeline to PLAYING ...<br>
   New clock: GstSystemClock<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">   ERROR: WriteN, RTMP send error 104 (126 bytes)<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Anybody know what that is?<o:p></o:p></p>
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<p class="MsoNormal">Best regards<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On Mon, Apr 22, 2013 at 11:57 PM, Peter Maersk-Moller <<a href="mailto:pmaersk@gmail.com" target="_blank">pmaersk@gmail.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Hi.<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">I was wondering if anybody has been successful streaming to the YouTube Live streaming servers using GStreamer?<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">I get the following error message which I believe indicates that YouTube doesn't accept my data:<br>
<br>
  Pipeline is PREROLLED ...<br>
  Setting pipeline to PLAYING ...<br>
  New clock: GstSystemClock<br>
  ERROR: RTMP_ReadPacket, failed to read RTMP packet header<br>
  ERROR: from element /GstPipeline:pipeline0/GstRTMPSink:rtmpsink0: Could not   open resource for writing. 
<br>
  Additional debug info:<br>
  gstrtmpsink.c(228): gst_rtmp_sink_render (): /GstPipeline:pipeline0  /GstRTMPSink:rtmpsink0:<br>
  Could not connect to RTMP stream "rtmp://<a href="http://a.rtmp.youtube.com/videolive?........" target="_blank">a.rtmp.youtube.com/videolive?........</a>..<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">I am using the following script that works great for UStream, but not for YouTube LIve. Anybody got any good suggestion?<o:p></o:p></p>
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<p class="MsoNormal">Best regards<o:p></o:p></p>
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<p class="MsoNormal">Peter<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><br>
================<br>
#!/bin/bash<br>
if [ $# != 1 ] ; then exit ; fi<br>
AUDIOFORMAT='audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)1'<br>
case $1 in<br>
  you|youtube)<br>
        width=1280<br>
        height=720<br>
        framerate='30/1'<br>
        videoencoder="x264enc bitrate=2000 profile=main bframes=2"<br>
        videomode='video/x-h264, width=(int)1280, height=(int)720, framerate=(fraction)25/1, pixel-aspect-ratio=(fraction)1/1, profile=(string)main'<br>
        videomode='video/x-h264, profile=main'<br>
        audioencoder="faac bitrate=64000 profile=2"<br>
        location="rtmp://<a href="http://a.rtmp.youtube.com/videolive?signature=REMOVED_FOR" target="_blank">a.rtmp.youtube.com/videolive?signature=REMOVED_FOR</a> OBVIOUS_REASONS flashver=FME/3.0\20(compatible;\20FMSc\201.0)"<br>
<br>
        ;;<br>
  ustream)<br>
        width=1024<br>
        height=576<br>
        framerate='25/1'<br>
        videoencoder="x264enc bitrate=800 bframes=0"<br>
        videomode='queue'<br>
        audioencoder="lame mode=3 bitrate=64"<br>
        location="rtmp://<a href="http://SOMETHING.fme.ustream.tv/ustreamVideo/SOMETHING/MORESOMETHING" target="_blank">SOMETHING.fme.ustream.tv/ustreamVideo/SOMETHING/MORESOMETHING</a>' flashver=FME/3.0\20(compatible;\20FMSc\201.0)'"<br>
        ;;<br>
  *)    echo 'Use youtube or ustream'<br>
        exit<br>
esac<br>
<br>
gst-launch -v videotestsrc !\<br>
        'video/x-raw-yuv, framerate='$framerate', width='$width', height='$height !\<br>
        ffmpegcolorspace                !\<br>
        $videoencoder                   !\<br>
        "$videomode"                    !\<br>
        mux. audiotestsrc               !\<br>
        $AUDIOFORMAT                    !\<br>
        $audioencoder                   !\<br>
        flvmux streamable=true name=mux !\<br>
        rtmpsink location="$location"<br>
<br>
==================<br>
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<o:p></o:p></p>
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<p class="MsoNormal"> <o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">_______________________________________________<br>
gstreamer-devel mailing list<br>
<a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a><br>
<a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a><o:p></o:p></p>
</blockquote>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><br>
_______________________________________________<br>
gstreamer-devel mailing list<br>
<a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a><br>
<a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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