<div dir="ltr"><br><div>Thanks guys I was missing a few elements, I tried with fakesink so long that it worked.</div><div><br></div><div>But now I have a different question I have to mux and transport audio and video combined.</div><div>I have to use MPEGTS, but when I use the MPEGTS muxer instead of mp4 an empty file is created and nothing happens,</div><div>i thought that MPEGTS container can have x264 and AAC.</div><div><br></div><div>My mp4mux pipeline is now like this:</div><div><div>OUT_FILE="stream.mp4"</div><div><br></div><div>LISTEN_AUDIO_PORT=1236</div><div>LISTEN_VIDEO_PORT=1234</div><div><br></div><div>ACAPS="\"application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)OPUS,payload=111\""</div><div>VCAPS="\"application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)VP8,framerate=30/1,payload=100\""</div><div><br></div><div>GST_PARAMS="-e -m -v --gst-debug=**:4 "</div><div><br></div><div>gst-launch-1.0 $GST_PARAMS \</div><div> udpsrc port=$LISTEN_VIDEO_PORT caps=$VCAPS ! rtpjitterbuffer latency=3000 do-retransmission=true ! queue ! rtpvp8depay ! vp8dec ! videoscale sharpen=1 method=2 ! videoconvert \</div><div>! queue ! x264enc bitrate=102400 subme=5 ! queue \</div><div>! muxer.video_0 \</div><div>udpsrc port=$LISTEN_AUDIO_PORT caps=$ACAPS ! rtpjitterbuffer latency=3000 do-retransmission=true ! queue ! rtpopusdepay ! opusdec ! audioconvert \</div><div>! queue ! lamemp3enc bitrate=64000 ! queue ! muxer.audio_0 \</div><div>mp4mux name=muxer streamable=true \</div><div>! filesink location=$OUT_FILE \</div></div><div><br></div><div><br></div><div><br></div><div>And if somebody could give me a hint how could i put this MP4 into a single udpsink? </div><div><br></div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Sep 12, 2016 at 6:16 PM, Olivier Crête <span dir="ltr"><<a href="mailto:olivier.crete@collabora.com" target="_blank">olivier.crete@collabora.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>Hi,</div><div><br></div><div>What version of GStreamer are you using? You may be using a very old version. This gst-launch-1.0 line works fine here on 1.8.x.</div><div><br></div><div>Olivier</div><div><br></div><div>On Sun, 2016-09-11 at 14:24 +0200, Miha Nedok wrote:</div><blockquote type="cite"><div dir="ltr"><div style="font-size:12.8px">I know that :)</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">but even if i ad rtpopusdepay i always get the same result.</div><div style="font-size:12.8px"><br></div><div><div><span style="font-size:12.8px">gst-launch-1.0 -vvvvv udpsrc port=1236 caps="application/x-rtp,media=<wbr>(string)audio,clock-rate=<wbr>48000,encoding-params=2,<wbr>encoding-name=(string)OPUS,<wbr>payload=111" ! rtpopusdepay ! opusdec ! audioconvert</span></div><div><span style="font-size:12.8px">Setting pipeline to PAUSED ...</span></div><div><span style="font-size:12.8px">Pipeline is live and does not need PREROLL ...</span></div><div><span style="font-size:12.8px">Setting pipeline to PLAYING ...</span></div><div><span style="font-size:12.8px">New clock: GstSystemClock</span></div><div><span style="font-size:12.8px">/GstPipeline:pipeline0/<wbr>GstUDPSrc:udpsrc0.GstPad:src: caps = "application/x-rtp\,\ media\=$string$audio\,\ clock-rate\=$int$48000\,\ encoding-params\=$int$2\,\ encoding-name\=$string$OPUS\<wbr>,\ payload\=$int$111"</span></div><div><span style="font-size:12.8px">/GstPipeline:pipeline0/<wbr>GstRTPOpusDepay:rtpopusdepay0.<wbr>GstPad:src: caps = "audio/x-opus\,\ channel-mapping-family\=$int\<wbr>)0"</span></div><div><span style="font-size:12.8px">/GstPipeline:pipeline0/<wbr>GstOpusDec:opusdec0.GstPad:<wbr>sink: caps = "audio/x-opus\,\ channel-mapping-family\=\(int\<wbr>)0"</span></div><div><span style="font-size:12.8px">/GstPipeline:pipeline0/<wbr>GstRTPOpusDepay:rtpopusdepay0.<wbr>GstPad:sink: caps = "application/x-rtp\,\ media\=\(string$audio\,\ clock-rate\=$int$48000\,\ encoding-params\=$int$2\,\ encoding-name\=$string$OPUS\<wbr>,\ payload\=$int$111"</span></div><div><span style="font-size:12.8px">/GstPipeline:pipeline0/<wbr>GstOpusDec:opusdec0.GstPad:<wbr>src: caps = "audio/x-raw\,\ format\=$string$S16LE\,\ layout\=$string$interleaved\<wbr>,\ rate\=$int$48000\,\ channels\=$int$2\,\ channel-mask\=$bitmask$<wbr>0x0000000000000003"</span></div><div><span style="font-size:12.8px">/GstPipeline:pipeline0/<wbr>GstAudioConvert:audioconvert0.<wbr>GstPad:src: caps = "audio/x-raw\,\ format\=$string$S16LE\,\ layout\=$string$interleaved\<wbr>,\ rate\=$int$48000\,\ channels\=$int$2\,\ channel-mask\=$bitmask$<wbr>0x0000000000000003"</span></div><div><span style="font-size:12.8px">/GstPipeline:pipeline0/<wbr>GstAudioConvert:audioconvert0.<wbr>GstPad:sink: caps = "audio/x-raw\,\ format\=$string$S16LE\,\ layout\=$string$interleaved\<wbr>,\ rate\=$int$48000\,\ channels\=$int$2\,\ channel-mask\=$bitmask$<wbr>0x0000000000000003"</span></div><div><span style="font-size:12.8px">ERROR: from element /GstPipeline:pipeline0/<wbr>GstUDPSrc:udpsrc0: Internal data flow error.</span></div><div><span style="font-size:12.8px">Additional debug info:</span></div><div><span style="font-size:12.8px">gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/<wbr>GstUDPSrc:udpsrc0:</span></div><div><span style="font-size:12.8px">streaming task paused, reason not-linked (-1)</span></div><div><span style="font-size:12.8px">Execution ended after 0:00:00.020207760</span></div><div><span style="font-size:12.8px">Setting pipeline to PAUSED ...</span></div><div><span style="font-size:12.8px">Setting pipeline to READY ...</span></div><div><span style="font-size:12.8px">Setting pipeline to NULL ...</span></div><div><span style="font-size:12.8px">Freeing pipeline ...</span></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Sun, Sep 11, 2016 at 2:11 PM, Sebastian Dröge <span dir="ltr"><<a href="mailto:sebastian@centricular.com" target="_blank">sebastian@centricular.com</a>></span> wrote:<br><blockquote type="cite">On So, 2016-09-11 at 04:17 -0700, MikeSI wrote:<br> > I cannot reeive opus via RTP. Even If i set payload it's always the<br> > same.<br> ><br> > gst-launch-1.0 -vvvvv udpsrc port=1236<br> > caps="application/x-rtp,media=<wbr>(string)audio,clock-<br> > rate=48000,encoding-params=2,e<wbr>ncoding-name=(string)OPUS"<br> <br> This is not a complete pipeline, you're missing at least the RTP<br> depayloader, possibly a decoder and converters, and a sink. E.g.<br> <br> gst-launch-1.0 -vvvvv udpsrc port=1236 caps="application/x-<br> rtp,media=(string)audio,clock-<wbr>rate=48000,encoding-params=2,e<wbr>ncoding-<br> name=(string)OPUS" ! rtpopusdepay ! opusdec ! fakesink<span class="HOEnZb"><font color="#888888"><br> <span><font color="#888888"><br> --<br> Sebastian Dröge, Centricular Ltd · <a href="http://www.centricular.com" rel="noreferrer" target="_blank">http://www.centricular.com</a></font></span><br>______________________________<wbr>_________________<br> gstreamer-devel mailing list<br> <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesk<wbr>top.org</a><br> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/<wbr>mailman/listinfo/gstreamer-dev<wbr>el</a><br> <br></font></span></blockquote></div><span class="HOEnZb"><font color="#888888"><br></font></span></div><span class="HOEnZb"><font color="#888888"> <pre>______________________________<wbr>_________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.<wbr>freedesktop.org</a> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">https://lists.freedesktop.org/<wbr>mailman/listinfo/gstreamer-<wbr>devel</a> </pre></font></span></blockquote><span class="HOEnZb"><font color="#888888"><div><span><pre><pre>-- <br></pre>Olivier Crête <a href="mailto:olivier.crete@collabora.com" target="_blank">olivier.crete@collabora.com</a> </pre></span></div></font></span></div><br>______________________________<wbr>_________________<br> gstreamer-devel mailing list<br> <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.<wbr>freedesktop.org</a><br> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/<wbr>mailman/listinfo/gstreamer-<wbr>devel</a><br> <br></blockquote></div><br></div>