Webrtc-stats and the PlaybackRate

[Stefano Mas]

We are developing a tool that allows multiple co-located stages and participants to perform together. We are using WebRTC for live media. The audio quality for the tool is crucial, as we want to transmit music over it. We managed to disable the noise suppression, echo cancellation, etc., that is more oriented to video calls. However, there is an issue that we don't know how to handle:

We assume that to manage a stable jitter buffer, the playback rate changes. That's again great for the video calls but not suitable for music. Is there a way to control, enable, disable that to ensure a stable playback rate? We are working with Chrome (GStreamer/Chrome).

Many thanks in advance for your help.

[Ben Weekes]

Hi Stefano, this is an interesting use case which I can help you build. Singing together suffers from latency but your use case may be fine. Please contact me on b...@agora.io to discuss.

[davidsw...@gmail.com]

The "music" audio hint of  https://www.w3.org/TR/mst-content-hint/#audio-content-hints should help, but I've found that Safari sent no audio at all when I tried it (although other browsers worked).

[Stefano Masneri]

Thanks David, I am gonna give it a try (we are using Chrome)

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Stefano

[Lorenzo Miniero]

Hi Stefano,

just FYI, I made a presentation on this topic a few months ago:

https://fosdem.org/2021/schedule/event/webrtc_musicians/

Since then I created an ugly prototype as well:

https://www.slideshare.net/LorenzoMiniero/jamrtc-wonder-webrtc-unconference

https://github.com/lminiero/jamrtc

Probably not an exact answer to what you asked, but there may be helpful info for you in there anyway.

Lorenzo

[Stefano Masneri]

Thanks a lot Lorenzo! This looks very interesting. I don´t know how I may have missed this presentation.

Best,

Stefano

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Stefano