Note: Please read FAQ before file an issue, see #2716
Description
There are no options for setting webrtc audio parameters. By default the sound is very bad, caused by echo cancelling and other webrtc opus based default settings. It is quite important for WHIP ingest or webrtc broadcast that a pure audio mode can be activated (Stereo, no further audio processing).
SRS Version: all
SRS Log:
n.a.
SRS Config:
n.a.
Replay
Ingest any audio/video codec doesn't matter what bitrate. Sound will always be bad in playback.
Description
There are no options for setting webrtc audio parameters. By default the sound is very bad, caused by echo cancelling and other webrtc opus based default settings. It is quite important for WHIP ingest or webrtc broadcast that a pure audio mode can be activated (Stereo, no further audio processing).
SRS Version: all
SRS Log:
n.a.
n.a.
Replay
Ingest any audio/video codec doesn't matter what bitrate. Sound will always be bad in playback.
Expect
Add some options in WHEP/RTC player. For example: &stereo=1 Like it's done in vdo.ninja https://docs.vdo.ninja/advanced-settings/audio-parameters
This is how it looks like on WHIP/WHEP side:
on WHIP side: a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 cbr=1;maxaveragebitrate=131072;minptime=10;sprop-stereo=1;stereo=1;useinbandfec=1
on WHEP side: a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1
No chance to set the custom parameters stereo=1 or sprop-stereo=1 on client side.