[linux-audio-user] Re: [linux-audio-dev] jack_convolve-0.0.10,
mista.tapas at gmx.net
Mon Jun 27 10:33:13 EDT 2005
On Mon, 27 Jun 2005 16:06:54 +0200
fons adriaensen <fons.adriaensen at skynet.be> wrote:
> On Mon, Jun 27, 2005 at 01:57:34PM +0200, Florian Schmidt wrote:
> > Assume a samplerate of 96khz, then there's quite a bit of signal which
> > doesn't need to be processed since it's far out of the range of human
> > perception.
> This seems like a sensible idea, but one could wonder why in that case
> the sample frequency needs to be 96 kHz (*).
Well, the argument i often heard and which IMHO does make sense is that
when heavy processing is used the higher samplerate keeps many artefacts
out of the audible range for a longer time than with 48khz for example.
So, when the output of jack_convolve would be subject to additional
heavy processing it would probably make sense to use the whole 48khz
bandwidth. But in the case that the convolution is used as a send reverb
and the only additional stages of processing afterwards are mixing it
back into the sum and then maybe some dynamics, it would make sense to
only process i.e. the lower half of the spectrum (this would still leave
a bandwidth of 24khz at a samplerate of 96khz).
> It's probably possible to minimize the artefacts by using a gentle
> cutoff, e.g. a raised cosine from bin k1 to k2, k1 < k2.
I will try that in the next few days..
> (*) Recent experiments by prof. Angelo Farina (Univ. of Parma, Italy)
> suggest strongly that when the DA conversion is done properly, there is
> no audible difference between a sample rate of 48 kHz and any higher value.
> OTOH, he only found one type of DAC that was good enough in order to be
> completely free of audible artefacts at 48 kHz (by Apogee, and they are
> quite expensive).
Thanks for the info..
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