On Wed, Jan 01, 2014 at 03:21:29AM -1000, Joel Roth 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. At least by half, and if the antialiasing filters in the AD/DA converters are designed for it, much better. When using small period sizes (32 or less), the delay in these filters (which are usually linear phase) will be the largest contribution to latency. This delay is roughy inversely proportional to the transition band of the filter. If you don't use the full bandwidth that 96 kHz offers (up to 48 kHz) it could be much less than half the value at 48 kHz. This is why 'digital snakes' connecting stage and FOH mixer use 96 kHz. > If one wanted to avoid the tradeoff of handling twice the > usual amount of audio data, 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. That won't work. Resampling requires the same type of filters, when you resample to/from 48 kHz you'll get the same delay as an AD/DA operating at that frequency. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
