[Icecast] Low-bitrate audio encode/stream application

Ralph Giles giles at xiph.org
Wed Dec 22 21:51:28 UTC 2004

On Wed, Dec 22, 2004 at 02:28:18PM -0700, david feldman wrote:

> I am totally new to icecast, and would appreciate a pointer in the right 
> direction. My application is to encode audio (at the linux server) at a low 
> bit rate, say ~16 kbps, for transmission to a decoder at another location. 
> The audio source will be bandlimited (basically voice grade circuit such as 
> you'd find in a telephone application.) The decoder end (connected via a 
> TCP/IP session) would be a Windows CE 3.0 machine which may be limited to 
> MP3 decode format.

> I've looked through what documentation I could find on 
> icecast and haven't seen anything that tells me how to characterize the 
> audio encoding rules (how do you pick 16 kbps, 32 kbps, etc., stereo, mono, 
> etc.?) Anyway, a pointer to the correct documentation resource would be 
> greatly appreciated.

There are actually two pieces to this. Icecast itself is a server. It 
reflects streams from so-called 'source clients' to listening clients. 
It's in the source client that you set the encoding parameters. Ices is 
probably the best thing to use on linux, but there are others. See 
http://icecast.org/ices.php and http://icecast.org/3rdparty.php for 
links and associated howtos.

BTW, if you're actually doing voice over your very-low-bandwidth 
connection, you might consider our speex codec. It's designed 
specifically for speech and will in general do a much better job than 
vorbis or mp3 below 32 kbps. http://speex.org/


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