> Ivica Bukvic wrote:
>
> ... stumbled across the following site:
>
> http://old.lwn.net/1999/0916/a/latency.html
>
> Admittedly, it's quite old but that, if anything speaks only in Linux's
> favor in terms of its pro-audio readiness. 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, especially in 1999 when this test was done?)
> 128 bytes in 192000Hz sampling rate = 0.3ms
>
> So what gives? It seems like it is some kind of a 176k-ish sampling rate
> that, AFAIK does not exist.
>
> Furthermore, my question is what app was used to produce those
> graphs/results and whether these latency tests take into account
> hardware latencies (i.e. DSP converters, PCI->CPU->PCI->output etc.), in
> other words, is this latency that is achievable with the following
> setup:
>
> Input->soundcard->cpu(with some kind of DSP)->soundcard->Output
>
> Your help on this matter is greatly appreciated!
>

This is from memory, IIRC:

What Benno's latency benchmarks measures is kernel scheduler latency, not
audio buffering latency as you pictured.

You can look at the 0.73ms scheduler latency value as the jitter for any
multimedia processing, which is the estimated time a process or thread
takes to be waken up by the kernel, for doing its next CPU time slice
(e.g. when it takes to read/write buffered audio).

Hope I'm not wrong or confusing you,
-- 
rncbc aka Rui Nuno Capela
[EMAIL PROTECTED]




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