Hi all, I've had an interesting discussion with a professor and a distinguished member of the electroacoustic music community regarding audio latencies which made me realize that I did not understand the issue in its entirety. Hence, I looked around the net in order to educate myself. I soon 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! Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico
