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Unable to build gst-meet #1
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Hey @daimoc, thanks for checking it out. Right now it's using a patched The new gstreamer is needed for RTP header extensions, for transport-cc. Unfortunately most distributions don't carry the 1.19.x branch since it's unstable. If you have nix, you can run |
I'm having the same issue. I have cloned https://gitlab.com/xmpp-rs/xmpp-rs to parent directory and:
Where is the source code of xmpp-parsers-gst-meet? |
Ah, right. The released version of gst-meet (
It's on crates.io (all Rust crates are distributed as source code) but it looks like we need to add it to this repository for now, to make building from source easier. I'll take care of that soon. |
Hi @jbg , Regards, |
For development we're using the nix env, but it looks like there has been a GStreamer release since I last worked on this, and that branch was merged and deleted as part of the release. I will update the nix env for the GStreamer changes and update this issue once done. Cheers |
I've updated |
BTW, building should get significantly easier after the gstreamer 1.20 stable release (in the next few weeks, apparently) since distro packages for these newer gstreamer versions will become available. |
I didn't have luck with nix-shell (log here), but building GStreamer 1.19.2 and installing it system-wide did the job. Here's my script for this, found somewhere online and modified, can't find the original one anymore. Thanks for explaining how to use the |
Hi @teowoz thank you for the Gstreamer 1.19.2 build script it helps me to run gst-meet on my Ubuntu server (I was unable to a have a working gstreamer with nix-shell ). But I still have issue with the audio stream. A more detailed error log with GST_DEBUG=4,webrtc:7 : 0:00:01.235794920 2435 0x7f22f0009e30 WARN GST_CAPS gstpad.c:5758:pre_eventfunc_check:capsfilter0:sink caps application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)multiopus, num_streams=(string)4, coupled_streams=(string)2, channel_mapping=(string)"0,4,1,2,3,5", encoding-params=(string)6, sprop-maxcapturerate=(string)48000, payload=(int)111, extmap-1=(string)< "", urn:ietf:params:rtp-hdrext:ssrc-audio-level, "vad=on" >, extmap-5=(string)http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, ssrc=(uint)1417847294, timestamp-offset=(uint)1466924858, seqnum-offset=(uint)23684 not accepted |
Hi @daimoc! Can you provide the full
It should be 2-channel Opus, this looks like 6-channel (2x2 + 2, like front stereo, rear stereo, centre and sub). You can solve it by adding another gstreamer element to your send pipeline, like |
Hi, you're right I didn' see that the audio of my BigBuckBunny video test file is in 5.1.
But now I getting this error : 0:00:00.922767420 11945 0x7fc6c4007ed0 WARN GST_CAPS gstpad.c:5758:pre_eventfunc_check:capsfilter1:sink caps application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)OPUS, sprop-stereo=(string)1, encoding-params=(string)2, sprop-maxcapturerate=(string)48000, payload=(int)111, extmap-1=(string)< "", urn:ietf:params:rtp-hdrext:ssrc-audio-level, "vad=on" >, extmap-5=(string)http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, ssrc=(uint)3111025154, timestamp-offset=(uint)3583645051, seqnum-offset=(uint)20999 not accepted |
Can you provide the full Also, if it's not too much trouble, can you set |
Thanks for you reply, WARN GST_CAPS gstpad.c:5758:pre_eventfunc_check:capsfilter1:sink caps application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)OPUS, sprop-stereo=(string)1, encoding-params=(string)2, sprop-maxcapturerate=(string)48000, payload=(int)111, extmap-1=(string)< "", urn:ietf:params:rtp-hdrext:ssrc-audio-level, "vad=on" >, extmap-5=(string)http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, ssrc=(uint)3111025154, timestamp-offset=(uint)3583645051, seqnum-offset=(uint)20999 not accepted I don't know if extmap-1=(string)< "", urn:ietf:params:rtp-hdrext:ssrc-audio-level, "vad=on" > is valid in caps. If I comment this audio_caps
Now I get audio and video on my conference. |
Hi you made a great job with gst-meet,
I made same tests few weeks ago but since you add dependency to xmpp-rs/xmpp-parsers and switch to gstreamer 1.19 it is more difficult to build gst-meet.
Could you update the readme to explain a little more all the build process to have a working gst-meet ?
Regards,
Damien.
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