Bruce,

Thanks for this info regarding websockets. Very interesting.

Regarding codecs, it is my understanding that all major browsers
support Opus by now but I admit that I do not have any practical
experience how that works for real-time audio streaming outside
WebRTC. For client/server written in C, opus is really great though.

Alex


On Thu, Apr 23, 2015 at 5:25 AM, Bruce Perens <[email protected]> wrote:
>
> 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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