[Live-devel] Stream to sound card hickups

[Ross Finlayson]

Gertjan,

Your code looks OK.  However, as you've discovered, the natural 
variablity (jitter) of incoming RTP packets means that you need to 
have sufficient buffering at the receiver end to handle this.

The simplest way to add buffer space - if you can - is using the 
operating system.  Perhaps there's some way to increase the buffer 
size used by the audio driver (e.g., using an "ioctl" of some sort, 
or perhaps some boot-time kernel configuration)??  Alternatively, if 
you have named pipes available, then you might be able to use one of 
those (in front of your audio device) to increase buffer space.

As a last resort, you could write your own buffering filter object - 
for your application - that would work with the "LIVE555 Streaming 
Media" libraries.  I would investigate OS-based solutions first, 
though.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/