7 Oct
2003
7 Oct
'03
12:22 a.m.
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?
its based, IIRC, on the frequency of the RTC interrupt, which is
always in power-of-2 Hz.
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
It isn't measuring that kind of latency. Benno wrote a custom program
to measure the ability to satisfy audio interface requirements.
Benno's test program is measuring how rapidly after an audio interface
interrupt it is possible to service the card. this is the "bottom
line" for latency performance - if you can't service the interrupt
quickly enough, nothing else matters. what his test was showing that
we can, from user space, service the card, although his test was based
on using the RTC, which for some hardware will allow you to get even
lower latencies than using the interface's own interrupt. its a hack,
though, and i wouldn't encourage people to do it.
--p