I am trying to write a pcm app which receives sound over the network and outputs it through the speakers. The network protocol is udp. From what I can tell, each packet is a single period with interleaved channels. My problem is that the period size appears to be below the minimum that my hardware will go. For example:
rate: 48000 Hz channels: 2 packet size: 512 bytes sample size: 16 bit
ok so far... (correct me if I'm wrong)
period size = (packet size * 8) / (channels * sample size) = (512 * 8) / (16 * 2) = 128
buffer size = (whatever i like...? latency/network stability...?)
When I try to set the period size exactly I get an error. If I set the period size near I get 192. How do I make this work? Should I reconfigure the network sound source? Do I put the packets together into larger periods? Can alsa do this for me?
cheers Tim
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