<div dir="ltr"><div><div><div><div><div>Hi Sebastian. </div>Video works fine alone (also works when MP3 audio is included). </div>Audio alone works for MP3. Audio never gets to play when using AAC. So it does not work for AAC. </div>Audio alone scripts are included for you to verify. With rtpbin, they can now hardly be stripped further. </div>Player script: #!/bin/bash port_base=14100 port_audio_rtp=$(($port_base+2)) port_audio_rtcp=$(($port_base+3)) video_caps='application/x-rtp,media=video,payload=96,clock-rate=90000,encoding-name=H264' if [ X$1 = Xmp3 ] ; then audio_caps='application/x-rtp,media=audio,payload=14,clock-rate=90000,encoding-name=MPA' else audio_caps='application/x-rtp,media=audio,payload=96,clock-rate=44100,encoding-name=MP4A-LATM' fi gst-launch-1.0 -v rtpbin name=rtpbin buffer-mode=slave \ udpsrc caps=$audio_caps port=$port_audio_rtp !\ rtpbin.recv_rtp_sink_1 \ rtpbin. !\ decoder2. \ udpsrc port=$port_audio_rtcp !\ rtpbin.recv_rtcp_sink_1 \ decodebin name=decoder2 ! audioconvert !\ audioresample ! autoaudiosink </div>Encoder script: <div>#!/bin/bash port_base=14000 port_audio_rtp=$(($port_base+2)) port_audio_rtcp=$(($port_base+3)) host=127.0.0.1 AUDSRC="audiotestsrc is-live=1" channels=2 rate=44100 if [ $1 = "mp3" ] ; then AUDIOENCODER="lamemp3enc bitrate=128 cbr=1" AUDIOPAY="rtpmpapay" AUDIOPARSE=queue AUDIOFORMATOUT="audio/mpeg,mpegversion=1" else AUDIOENCODER="faac bitrate=128000" AUDIOPAY="rtpmp4apay" AUDIOPARSE=aacparse AUDIOFORMATOUT="audio/mpeg,mpegversion=4,stream-format=raw" fi AUDIOFORMAT="audio/x-raw,format=S16LE,layout=interleaved,rate=$rate,channels=$channels" gst-launch-1.0 -v rtpbin name=rtpbin \ $AUDSRC !\ queue !\ $AUDIOFORMAT !\ audioparse rate=$rate channels=$channels !\ audioconvert !\ $AUDIOENCODER !\ $AUDIOPARSE !\ $AUDIOFORMATOUT !\ $AUDIOPAY !\ rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 !\ udpsink host=$host port=$port_audio_rtp \ rtpbin.send_rtcp_src_1 !\ udpsink host=$host port=$port_audio_rtcp sync=false async=false </div></div><div class="gmail_extra"> <div class="gmail_quote">On Tue, May 10, 2016 at 12:54 PM, Sebastian Dröge <<a href="mailto:sebastian@centricular.com" target="_blank">sebastian@centricular.com</a>> wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">On Di, 2016-05-10 at 12:47 +0200, Peter Maersk-Moller wrote: > Hi. > > I have a bit of a problem getting rtpbin to work reliably with AAC. > Works fine with MP3. The problem can be reproduced with following two > scripts. The player part fails in varying ways when using AAC, but > works as said fine with MP3. When using BOTH the scripts with the > argument 'mp3' it will use MP3, otherwise they will use AAC. Using > GStreamer 1.8.1. Note the player does not send stats back to the > encoder/sender, but that ought to be okay. > > When using AAC, the player script fails in one of the following ways: > A video window pops up showing the first frame, but no more frames > are shown and no audio is coming out. > [...] This looks like some general (gst-launch caused) linking confusion. Can you simplify your scripts to only do audio? I would guess that the problem with AAC disappears then. Can you confirm? And I assume the pipelines also works fine if you remove the audio parts all together and only have video? -- Sebastian Dröge, Centricular Ltd · <a href="http://www.centricular.com" rel="noreferrer" target="_blank">http://www.centricular.com</a> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote></div> </div>