[LAU] Buffering, soundcard clocks, synchronization, streaming

[David Olofson]

On Thu, Nov 6, 2014 at 12:35 PM, Ken Restivo <ken at restivo.org> wrote:
[...]
> Now, I've been told that the buffer problem is due to clock synchronization on soundcards. Some soundcards on't stream exactly at 44.1khz, they're like 44.0Khz, and that causes a buffer that gradually declines in size, as it's been explained to me.
[...]

Indeed; a pair of sound cards running at (practically) identical
sample rates is going to take something much more accurate than
crystal oscillators. Or dumb luck. :-) This is something any streaming
software needs to deal with (sample drop/insert, or resampling, based
on buffer status), so there's no valid excuse for this happening.
Sounds like a bug.


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//David Olofson - Consultant, Developer, Artist, Open Source Advocate

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