Hi

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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