On Wed, 2012-07-18 at 14:29 +0300, Ansis M?li?? wrote: > Due to clock drift, audio sent from one computer at, say, 48000 Hz will > arrive at the other computer at 47999Hz or 48003 Hz, and this value > constantly changes with ambient temperature. Eventually you get buffer > underruns and overruns unless you dynamically apply sample rate conversion > to keep the receiving buffer exactly half full. > > Do any of the PulseAudio components do this? Yes. For example, module-loopback and module-combine-sink do this. > My specific problem is that I get cracks and pops when streaming live audio > to module-simple-protocol-tcp from another machine. The simple protocol does not do any adaptive resampling, so it's not a very good solution for live audio. -- Tanu
