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? My specific problem is that I get cracks and pops when streaming live audio to module-simple-protocol-tcp from another machine.
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