1 Nov
2003
1 Nov
'03
11:21 p.m.
On Sunday 02 November 2003 02.30, Ivica Bukvic wrote:
[...]
All that seems to be correct. However AFAIK, the
higher the
sampling rate the lesser amount of time do those 128 samples
encompass. Hence if you have double the sampling rate 88200Hz, then
128 samples ends up being only ~1.5ms. Of course you could go
really crazy and do 192000, in which case latencies would be
sub-millisecond.
But the sample rate *was* specified to 44.1 kHz in this case, wasn't
it...?
There is obviously still a question whether any
kernel on the face of earth would be able to provide soundcard with
data in time in order to avoid dropouts...
You can process one sample at a time at 44.1 kHz on most decent PCs
running RT-Linux, RTAI or some other "industrial strength" RTOS.
Those get the latencies practically all the way down to the hardware
limits, meaning worst case latencies of less than 10 microseconds on
well built Celeron or better systems. Also note that the x86 arch
sucks at this pretty much by design. Alpha, PPC and others can get
even lower latencies.
Of course, you can't use normal PCI busmaster DMA for that, since
there's a minimum burst size limit at 64 bytes or something. (Though
you could most probably get around that if the sound card is designed
for it, and/or does DMA with a fully programmable chip.) PIO or ISA
DMA works just fine, but that's generally only available on old ISA
sound cards.
//David Olofson - Programmer, Composer, Open Source Advocate
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