Hello PD-list,
I'm still working on my mp3cast~ project :-)
The final problem (I hope) is that I experience quite considerable latency on the client-side of around 10-12 seconds. The application is supposed to be real-time so this latency is quite a problem.
A colleague suggested that it could be Icecast using a fixed size buffer. I had a look through the Icecast source code but couldn't find anything hopeful. So I put a message on the Icecast forum and was told to try reducing the burst-on-connect size to 0. I did this and it didn't seem make any difference.
Of course, buffering could be the problem, but there are four places where buffering could occur: 1) in the client media streamer; 2) in the streaming server output; 3) in the streaming server source/input; and 4) in the source encoding (in this case, mp3cast~).
I've made some attempts at checking cases 2 and 3. 4 is problematic in that I can't control what a user's client does. (I can provide documentation, though).
So I'm trying to check case 4. Google returns four or five hits for "mp3cast~ buffering", one of which is one my previous posts to this list!
Does anyone know how mp3cast~ buffering is controlled and if it is configurable either by a setting or source code hack?
Cheers, Richard