Yannick "Modah" Gouez
ymg at stea.mn
Sat Aug 3 00:56:09 PDT 2013
Following up on this topic ( sorry if this starts a new thread but I just
joined the ml ),
I do no understand why it is not possible to use the audio stream from
webRTC's getUserMedia and then send it over a websocket ?
It seems that the webRTC implementation can natively encode in ogg format
in stereo from any interface ( according to
Why wouldnt it be suitable ?
In the other, I assume, like Jamie, that there is a way to send it over a
PUT request. Isnt it ?
> Hi Jamie,
> The webRTC API does not sound suitable for source->server streaming
> for many reason. For instance, the peer-to-peer connection requires
> input from both end and seems quite unfeasible to implement in a
> server. Likewise, codecs are completely abstracted and much more.
> In reality, webRTC is an API to acheive full-duplex conversations a-la
> skype and not for streaming.
> For these reasons, we at liquidsoap have been working on implementing
> a simple websocket protocol for sending source streams from a browser
> to a server. The protocol is documented and implemented there:
> We also have a pull request on liquidsoap that implements the protocol
> and should be merged fairly soon:
> The bottlenecks right now are the availability of the Web Audio API,
> which is only partially implemented in firefox and the encoding speed.
> for mp3 encoding. However, only firefox seems to show suitable
> performances for mp3 encoding, using the libshine build and thanks to
> its asm.js support.
> On the other hand, only chrome implements the adequate Web Audio API,
> but is too slow to encode :-o
> All in all, if we keep forging, it is very likely that once mozilla
> finishes implementing the web audio API, we should have a function
> browser source client using firefox. And Chrome when their asm.js
> perfs improve as well.
> 2013/7/23 Jamie McClelland <jm at mayfirst.org <http://lists.xiph.org/mailman/listinfo/icecast-dev>>:
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