"Thomas B. Rücker"
thomas at ruecker.fi
Sat Aug 3 01:08:28 PDT 2013
On 08/03/2013 07:56 AM, Yannick "Modah" Gouez wrote:
> Following up on this topic ( sorry if this starts a new thread but I
> just joined the ml ),
No problem, but _please_ do not post HTML to mailing lists, thanks.
> 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 ?
That seems to refer to Romain's statement.
I can't comment on that as I don't follow his view.
> It seems that the webRTC implementation can natively encode in ogg
> format in stereo from any interface ( according to
> https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC ).
> Why wouldnt it be suitable ?
Please note that there is a important difference between *container* and
Ogg is the container.
Opus and Vorbis are codecs commonly found in Ogg containers.
Just to make sure this is clear.
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