I'm currently trying to get realtime audio streaming over WebSockets to work. I'm encoding audio using libspeex on the server side (an iOS device) and decoding it again on the client using speex.js[1].
Transmitting Speex "frames" (where each frame is a 20ms chunk of audio) over WebSockets and decoding in JS works beautifully. However, I have a really hard time to queue up these 20ms AudioBuffer nodes perfectly, without producing any glitches or pops. I'm not sure its even possible with the current API.
What I would like to have is an AudioBuffer node where I can dynamically append chunks while its playing. Since all(?) source nodes are for one time only use, the browser could free the data that was already played again.
An AudioBuffer that allows modification of the data while it's playing would work too - you could just loop it and use it as a ring buffer. However, my understanding of the spec is, that modifying an AudioBuffer's data is explicitly prohibited once its playing.
[1] https://github.com/jpemartins/speex.js/