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Bad audio quality when using WebRTC publisher #1908
Hello, after meddling around with the WebRTC handshake i was able to force Opus to use the full band and respect a certain average bitrate. I used Chrome throughout the test.
I developed a patch (#1985), please test it by using this nightly release and let me know if it works:
Hello, after meddling around with the WebRTC handshake i was able to force Opus to use the full band and respect a certain average bitrate. I used Chrome throughout the test. I developed a patch (#1985), please test it by using this nightly release and let me know if it works:
Hi and thank you for your work.
I tested the enhancement in various scenarios. The quality is still very bad. It seems that echo cancelling option for webrtc is enabled by default. That causes music sound very bad. Here are some screenshots:
Perhaps it's neccessary to add some additional configuration options for audio publishing and playing.
I'll add an option for disabling audio effects too.
Meanwhile, please confirm that setting audio bitrate works as expected, you can check audio bitrate by opening about://webrtc-internals and checking tab "stats of outgoing stream, media=audio".
This seems to be solved. Sound is much better now. But there is a new issue with AV1 Video playback or encoding.
Have a look at the video. Every 2 seconds, there seems to be a short interrupt. Looks like he tries to scale or re-scale.
Also a problem on video: It does not playback on Chrome mobile devices, but AV1 is definitely supported (I tried in webm container with icecast server before). Perhaps the wrong AV1 encoding model is chosen by default. There are several ones, but not all supported by mobile devices.
Regarding the AV1 bug, open a new issue, specify whether it's a regression or not and provide replication details.
Furthermore, i've moved all codec selection-related code to client side, hence it should be really easy to hack the code and change things such as the AV1 profile.
This issue is being locked automatically because it has been closed for more than 6 months.
Please open a new issue in case you encounter a similar problem.
Activity
aler9 commentedon Jun 24, 2023
Hello, after meddling around with the WebRTC handshake i was able to force Opus to use the full band and respect a certain average bitrate. I used Chrome throughout the test.
I developed a patch (#1985), please test it by using this nightly release and let me know if it works:
[link removed]
webrtc: allow setting Opus bitrate (#1908)
zapf2000 commentedon Jun 25, 2023
Hi and thank you for your work.
I tested the enhancement in various scenarios. The quality is still very bad. It seems that echo cancelling option for webrtc is enabled by default. That causes music sound very bad. Here are some screenshots:
Perhaps it's neccessary to add some additional configuration options for audio publishing and playing.
aler9 commentedon Jun 25, 2023
I'll add an option for disabling audio effects too.
Meanwhile, please confirm that setting audio bitrate works as expected, you can check audio bitrate by opening about://webrtc-internals and checking tab "stats of outgoing stream, media=audio".
zapf2000 commentedon Jun 25, 2023
Yes, I can confirm that the bitrate is set correctly for outbound-rtp audio. Works fine. I think the bad quality is related to enabled effects.
aler9 commentedon Jun 26, 2023
i've added the option:
[link removed]
zapf2000 commentedon Jun 27, 2023
This seems to be solved. Sound is much better now. But there is a new issue with AV1 Video playback or encoding.
Have a look at the video. Every 2 seconds, there seems to be a short interrupt. Looks like he tries to scale or re-scale.
Also a problem on video: It does not playback on Chrome mobile devices, but AV1 is definitely supported (I tried in webm container with icecast server before). Perhaps the wrong AV1 encoding model is chosen by default. There are several ones, but not all supported by mobile devices.
2023-06-27_20-38-19.mp4
aler9 commentedon Jun 27, 2023
Regarding the AV1 bug, open a new issue, specify whether it's a regression or not and provide replication details.
Furthermore, i've moved all codec selection-related code to client side, hence it should be really easy to hack the code and change things such as the AV1 profile.
webrtc: allow setting Opus bitrate (#1908) (#1985)
webrtc: add option to disable audio effects (#1908)
webrtc: add option to disable audio effects (#1908) (#1989)
aler9 commentedon Jul 1, 2023
added in v0.23.7
github-actions commentedon Jan 3, 2024
This issue is being locked automatically because it has been closed for more than 6 months.
Please open a new issue in case you encounter a similar problem.