I've spent the last few days learning the web audio API and websockets,
so that we can have a browser control panel for DVS and Whitebox, and
use the mic and speaker through the browser. It turns out to not be that
hard, and once that's done you need no drivers, no GUI porting. All of
the necessary facilities are in Firefox, Chrome, Opera, Android, iOS.
There's even someone who has already written a nice waterfall in javascript.
WebRTC is not ready for embedded use unless you're embedding node.js.
The currently available C code pulls in much of Chrome. It's too big and
has a really steep learning curve. No doubt better embedded C code will
come along.
Websockets, on the other hand, have an excellent embedded C library in
libwebsockets, and both the C and javascript side are easy to use.
Finally, a session-based full-duplex connection in the browser. But you
lose some things that WebRTC would give you: the codecs, the variable
bandwidth, and the NAT traversal. On the other hand, you can code a much
easier connection scheme than WebRTC would require.
Astonishingly, the web audio API includes an FFT that runs in your
browser. And a nice biquad filter. There is a reasonably complete audio
processing graph. It is not inconceivable that a codec could be
implemented in javascript using some of these facilities.
For now, I'm not using a codec but just using downsampled audio in
narrower data than the native form.
Thanks
Bruce
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