[speex-dev] Sample rate inaccuracy

[Ico Doornekamp]

Hi all,

I'm having some problems with inaccurate sampling rates between sending
and receiving party. (Using a simple raw UDP protocol sending packets
from speex to speex)

Both PC's have their soundcard set to 8 khz, but the samplerate seems to
differ slightly between machines. The problem I have is that the sending
machine sometimes samples with a slightly lower rate than the player
machine. This causes the audio data to 'heap up' in network buffers etc,
and thus slowly increases the latency.

What is a common way to prevent this from happening ?

Thanks,

Ico 


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