[linux-audio-user] Question regarding the alsa's audio latency benchmark

[Anahata]

On Mon, Oct 06, 2003 at 09:47:27PM -0400, Ivica Bukvic wrote:
> At any rate, I was checking
> out the benchmark data and was wondering as to how did this
> person/software app get to the 0.73ms buffer fragment that is equal to
> 128bytes? In other words, what sampling rate was used?
> 
> 128 bytes in 44100Hz sampling rate = 3ms
> 128 bytes in 88200Hz sampling rate = 1.45ms
> 128 bytes in 176400Hz sampling rate = 0.725ms (this one being obviously
> closest, but at the same time, what kind of hardware supports this
> sampling rate

Are you counting bytes or 16 bit samples, or pairs of 16 bit samples
(stereo L and R) ?

i.e. a stereo signal at 44100Hz sampling rate 16 bit is 4 * 44100
= 176400 bytes/sec.

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Anahata
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