This is driving me nuts...
This fails... gst-launch-1.0 \ webmmux name=mux streamable=true \ ! filesink location=new.webm \ rtpbin name=rtpbin \ udpsrc address=127.0.0.1 port=30000 caps="application/x-rtp, media=video, clock-rate=90000, encoding-name=VP8, payload=96" ! rtpbin.recv_rtp_sink_0 \ udpsrc address=127.0.0.1 port=30001 caps="application/x-rtcp" ! rtpbin.recv_rtcp_sink_0 \ udpsrc address=127.0.0.1 port=30002 caps="application/x-rtp, media=audio, clock-rate=48000, encoding-params=2, payload=120, channels=2, encoding-name=OPUS, caps=audio/x-opus" ! rtpbin.recv_rtp_sink_1 \ udpsrc address=127.0.0.1 port=30003 caps="application/x-rtcp" ! rtpbin.recv_rtcp_sink_1 \ rtpbin.recv_rtp_src_0_2220_96 \ ! rtpvp8depay \ ! queue \ ! mux.video_0 rtpbin.recv_rtp_src_1_1110_120 \ ! rtpopusdepay \ ! queue \ ! mux.audio_0 I always get this error message.. ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc2: Internal data stream error. but if I sink udpsrc2 to fakesink, it will work. I've tried about every caps combination on the OPUS side I can think of, cannot figure out why this continues to fail. Here in the sending side..and this works flawlessly... AUDIO_SSRC0=1110 AUDIO_PT=120 VIDEO_SSRC_BROADCAST_0=2220 VIDEO_PT=96 videoBroadcastRtpPort=30000 videoBroadcastRtcpPort=30001 audioBroadcastRtpPort=30002 audioBroadcastRtcpPort=30003 transportIp=127.0.0.1 gst-launch-1.0 \ rtpbin name=rtpbin do-retransmission=true \ rtpfunnel name=video ! rtpbin.send_rtp_sink_0 \ rtpfunnel name=audio ! rtpbin.send_rtp_sink_4 \ rtpbin.send_rtp_src_0 ! udpsink name=rtpudpsink0 host=${transportIp} port=${videoBroadcastRtpPort} sync=true \ rtpbin.send_rtcp_src_0 ! udpsink name=rtcpudpsink0 host=${transportIp} port=${videoBroadcastRtcpPort} sync=false async=false \ rtpbin.send_rtp_src_4 ! udpsink host=${transportIp} port=${audioBroadcastRtpPort} sync=true \ rtpbin.send_rtcp_src_4 ! udpsink host=${transportIp} port=${audioBroadcastRtcpPort} sync=false async=false \ filesrc location=${MEDIA_FILE} ! qtdemux name=demux \ demux.video_0 \ ! tee name=v \ v. \ ! queue \ ! decodebin \ ! videoconvert \ ! videoscale \ ! video/x-raw,width=1280,height=720 \ ! timeoverlay \ ! textoverlay text="720p@4" valignment=top halignment=right \ ! vp8enc target-bitrate=2000000 deadline=1 cpu-used=4 \ ! rtpvp8pay pt=${VIDEO_PT} ssrc=${VIDEO_SSRC_BROADCAST_0} picture-id-mode=2 mtu=1180 \ ! clocksync sync=true \ ! video. \ demux.audio_0 \ ! tee name=a \ a. \ ! queue \ ! decodebin \ ! audioresample \ ! audioconvert \ ! opusenc bitrate=96000 \ ! rtpopuspay pt=${AUDIO_PT} ssrc=${AUDIO_SSRC0} mtu=1180 \ ! clocksync sync=true \ ! audio. I can take out everything, it's always the udpsrc failing on the opus audio over rtp. -- Sent from: http://gstreamer-devel.966125.n4.nabble.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
We did sort this out.
The issue was the webmmux, it required the insertion of *opusparse* after we came out of the rtp payload, but before the muxer. -- Sent from: http://gstreamer-devel.966125.n4.nabble.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Free forum by Nabble | Edit this page |