Re: Javascript source client
Daniel James <[email protected]>
| Newsgroups | gmane.comp.audio.icecast.devel |
|---|---|
| Message-ID | <[email protected]> |
Hi Stephen, > So we have been advised from this thread > https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-18385702 > to not use http put as it is not in real-time, instead they are > suggesting the use of SDP, is that something that icecast supports? Or > does anyone have other ideas on this? The imminent Airtime 2.4.0 release has support for Opus, and it already has user authentication for input streams at specific times, based on a shared calendar. Plus it supports output to Icecast. So if you want an experience somewhere between Google Hangout and talk radio, that might be a good platform to build on: http://www.sourcefabric.org/en/airtime/ If that sounds promising, please mail me offlist and I can put you in touch with the dev team. The code is on GitHub (GNU GPLv3). Cheers! Daniel > On Sun 12 May 2013 01:51:31 AM CDT, Thomas Ruecker wrote: >> Hi, >> >> On 11 May 2013 15:32, Stephen Mahood <[email protected]> wrote: >>> Thank you for your interest in this, you description is as accurate as I >>> can see. >>> >>>> From my perspective your challenges will be to get the containers right. >>>> WebM for audio+video >>>> Ogg for audio >>>> >>>> Also (I'm not that familiar with webRTC) you might need to reencode >>>> to Opus and VP8 in some cases? >>> >>> here is the great news [...] >>> both Opus and VP8 are codecs are already supported in webrtc :) >> >> Yes, I'm aware of that, but I don't understand webRTC well enough to >> know if you can get away with a participant that only advertizes >> support for the above and doesn't support: >> http://www.webrtc.org/faq#TOC-What-is-the-iSAC-audio-codec- >> http://www.webrtc.org/faq#TOC-What-is-the-iLBC-audio-codec- >> and whatever else might still appear. >> >> That is nothing to worry about for a first proof of concept though. >> Start out with VP8 and Opus, worry about the rest later. >> >>> It seems to me the bigger challenge is getting a javascript source >>> client, the reason being is webrtc is written in javascript. >> >> Yes, as I said, I'd concentrate on getting an ogg and webm muxer >> working in Javascript. You might get away with something like >> emscripten and libogg for the start? Something proper would be >> preferred though. >> >> Sending the stream to icecast is just HTTP-PUT, so should be easy in Javascript. >> >>> I think there are amazing things that can be achieved with the webrtc >>> connection to icecast. It seems could be similar to google hangout >>> without the google dependency >> >> Yes, the possibilities are vast. Now go and make it something awesome! >> >> Cheers >> >> Thomas > _______________________________________________ > Icecast-dev mailing list > [email protected] > http://lists.xiph.org/mailman/listinfo/icecast-dev > -- Cheers! Daniel