On Wed, 2014-01-01 at 13:56 +0000, Harry van Haaren wrote: > On Wed, Jan 1, 2014 at 1:21 PM, Joel Roth <joelz@xxxxxxxxx> wrote: > I was curious, if doubling the sample rate is a > practical way to reduce latency for live effects > processing. I would think it would reduce latency by half. > > > It would: you mention "practical", i'm not sure I'd call it that. > > > If one wanted to avoid the tradeoff of handling twice the > usual amount of audio data, > CPU load will go up, since there is 2x more of data to process, > > which also means every plugin / host has 2x more work to do. > > Adds up quickly if you're doing things like convolution reverbs > > or other CPU intense processing.. > > > > I was curious if ALSA sample > rate conversion, or some other clever hack could be used to > get low latency advantage of the high sample rate, while > actually dealing with 48k streams through JACK. > Theoretically possible I suppose, it seems like an awful lot of > > effort to get a few less ms latency.. > > > Latency below ~3ms isn't percievable at all IMO: most will agree. > > Why not run jack at 64 frames, 2 buffers? That'll achieve approx > > 3ms (on 44.1kHz and 48kHz).. which is fine for the purpose? > > > Perhaps I'm missing something, are you doing mulitple passes > trough the sound-card that you're adding its latency two or more > times? Filters of audio devices usually are optimized for 48 KHz usage, but on this list there often where discussions about this issue and IIRC 96 KHz are indeed used to lower latency for live usage. I recommend to search the archive. Happy New Year! Ralf _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
