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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

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

I'm following up on a thread started by Stephen a couple months ago about building a JavaScript source client using webrtc. The first step suggested was to figure out how to mux the audio and video. After I posted a feature request on the webrtc experiment js library, we seem to have a solution: https://github.com/muaz-khan/WebRTC-Experiment/issues/28#issuecomment-20791759 Based on the last

Thanks so much for the comprehensive response Romain. It sounds like you are working hard getting this working with audio streams and it's a lot more complicated than I had originally hoped. I'm still curious about the proper syntax of the http put command though (and I'm sure others would be interested as well). Is there any documentation on the parameters to send? I also had a

Hi, On 07/23/2013 07:44 PM, Jamie McClelland wrote: > I'm following up on a thread started by Stephen a couple months ago > about building a JavaScript source client using webrtc. > > The first step suggested was to figure out how to mux the audio and > video. After I posted a feature request on the webrtc experiment js > library, we seem to have a solution: >

This project appears to be a fork of my Canvas Streaming Example project for Facebook: https://github.com/fbsamples/Canvas-Streaming-Example <https://l.facebook.com/l.php?u=https%3A%2F%2Fgithub.com%2Ffbsamples%2FCanvas-Streaming-Example&h=ATMcDwZSpl5j0Yw_EH8uhG4JciG9nHmZRTEYJBxZVjYA4B3UI3T5_3gYZFMANr4Lkon7gUmRpcsnKkiHZ6AWeU6JKoP4rrYXu4FSJbOfdEHTn2YNkya0wOZM1i1XVD7x1Q> I would recommend

Five years after initially posting to this list[0], I finally completed as browser-based video source client iceast. The code is here: https://gitlab.com/jamie/icecream As Romain Beauxis responded to my initial email, webrtc was unsuitable and websockets was the way to go. Thanks for the help! jamie 0. http://lists.xiph.org/pipermail/icecast-dev/2013-July/002223.html -- May First/People Link

Hi Guys, 2012/7/20 juantello <juantello1234 at gmail.com>: > > Hello. > > Icecast client for android > http://droidtools.sourceforge.net/content/icecast-client-android > http://droidtools.sourceforge.net/content/libogglibvorbis-and-libshout-libraries-android I just finished compiling an android version of our reshaped shine encoder [1] there:

Hi all, 2012/8/21 Romain Beauxis <toots at rastageeks.org>: > Hi Guys, > > 2012/7/20 juantello <juantello1234 at gmail.com>: >> >> Hello. >> >> Icecast client for android >> http://droidtools.sourceforge.net/content/icecast-client-android >> http://droidtools.sourceforge.net/content/libogglibvorbis-and-libshout-libraries-android > > I

Hi just saw this today :a real icecast client on android ! Seems to have been published on november 2012 : https://play.google.com/store/apps/details?id=sa.broadcastmyself <-----------------------------------------------------------------------------------------------------------> web perso : http://memeteau.org <xmpp%3Afreechelmi at jabber.fr> 2012/8/24 Romain Beauxis

Hi list, The Savonet team is proud to announce a new release of its programmable audio stream generator, liquidsoap 0.3.0. Liquidsoap is a simple ruby-like script language allowing one to build audio stream sources from various elementary sources, source combinators and audio outputs. It is mainly intended to be used as an icecast client for internet radios through the use of the shout output

Hi, On Sun, Sep 7, 2008 at 12:30 AM, Dick Trump <dtrump1 at triadav.com> wrote: > kosnickx wrote: >> But there is one thing : while falling back from one mount point to >> another it takes about 10 to 15 seconds. Does this happen to you too? Is >> there any way to reduce this to minimum or to 0 if possible. I've been wanting to advise you to give liquidsoap a try

Hi list, I'm the main developper of liquidsoap, the streamer from the Savonet project (http://savonet.sf.net). This streamer mainly uses shout2 as an output plugin, but also alsa, raw data over rtp, etc. And many outputs can be used in the same instance of liquidsoap. For that reason, we don't use directly sync and delay from libshout, but have a centralized scheduler, which should avoid

In my experience liquidsoap is a lot easier to compile from source on Debian or Ubuntu. Both platforms also have packages in the repos. To experiment it might be easier to spin up a small local VM, or there's always a digital oceans droplet, or EC2 micro instance. On Jan 7, 2015 4:24 PM, "Steve Matzura" <sm at noisynotes.com> wrote: > Looks like a lot of tedium. I'll

Hey everyone, I am new to the dev list here, but my question specific about any development towards webrtc integration. Let me explain, a couple colleagues and I are currently working on our webrtc build at live.mayfirst.org site useing nodejs. We are currently looking into seeing if there is any development of a javascript source client? This would be used to send the webrtc room to a public

I could, but I have this big old Fedora system just waiting for me to get LS working on it, so I think I need to concentrate on making that happen. If it's hard, difficult, tedious, inconvenient, then that's what it is. I want the cue sheet capability some way somehow, so I'm just going to have to knuckle down and buckle down and do it. Right? On Wed, 7 Jan 2015 17:21:27 -0800, you

Hey all, 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? ~stephen On Sun 12 May 2013 01:51:31 AM CDT, Thomas Ruecker wrote: > Hi, > > On 11

Looks like a lot of tedium. I'll give it a go on Red Hat, but not before a full backup first. On Wed, 7 Jan 2015 11:26:13 -0800, you wrote: >Hello, > >Fedora and CentOS don't have any prebuilt packages in the standard repos. > >Here are a few guides from the Liquidsoap community, specific to CentOS. >http://puck.in/2013/12/install-liquidsoap-1-1-on-centos-6-4/

Hello, Fedora and CentOS don't have any prebuilt packages in the standard repos. Here are a few guides from the Liquidsoap community, specific to CentOS. http://puck.in/2013/12/install-liquidsoap-1-1-on-centos-6-4/ http://sourceforge.net/p/savonet/mailman/message/31330094/ On Wed, Jan 7, 2015 at 10:05 AM, Steve Matzura <sm at noisynotes.com> wrote: > Looks like it's what I

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