[linux-audio-user] Some remarks about USB audio and low latency

Playback and capture streams can never be exactly synchronized -- even if both use the same sample rate, packet sizes are not guaranteed to be the same. This means that it's possible to use smaller buffers with Jack by using only one direction (-P or -C instead of -d), if only one direction is needed. Due to synchronization issues with the host controller, stopping and restarting a stream will take several milliseconds. This means that it is often preferrable to use Jack's --softmode option to reduce the length of the gap when an underrun happens. The snd-usb-audio driver in CVS, or in ALSA 1.0.10rc1 (to be released Real Soon Now(TM)) has several improvements: - Maximum additional latency of captured data reduced to 1 ms (this was a bug in the earlier version). - Consistent handling of USB packets that go over the buffer boundary. Sample rates that aren't a multiple of 1000 now work just as well as sample rates that are. This affects programs (like Jack) that don't use a buffer size that is a multiple of rate/1000, too. - The nr_packs parameter can now be changed after the module has been loaded, e.g.: echo 1 > /sys/module/snd_usb_audio/parameters/nr_packs This change will take effect when a stream is started the next time. Regards, Clemens