web-audio-api
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Modifying incoming MediaStream
I am implementing a "relay" server using Node, that connects to several peers using WebRTC and then forwards each peer's audio stream to all other peers. I was following this architecture in order to remove the need for every single pair of users to maintain their own WebRTC connection, and also to be able to move the audio processing load to the back-end (instead of doing it on the client browsers).
However, since this library does not support MediaStreamSourceNode yet, I wanted to ask:
- if there are plans to implement this in the near future;
- if there is any alternative approach to change the gain of incoming streams (for example, in triggering some event, reduce the gain to 50%).
Thank you.