linuxaudio.org

On Sun, Sep 26, 2010 at 11:40:08AM -0700, Niels Mayer wrote:
> On Sun, Sep 26, 2010 at 4:12 AM,   wrote:

quoted text
> > The reason is that step (5), multiplication in the

> > frequency domain, is equivalent to convolution in the

> > time domain. And the convolution of two signals of N

> > samples (T1 and T2) has lenght 2*N-1. This means that

> > this result is wrapped around in step (6) since the

> > inverse FFT produces only N samples.

>

> Doesn't this also apply to basic digital "mixing" of audio signals --

No.
> and appears to be the fundamental difference in both sound and

quoted text
> algorithm between "analog summing bus", where each digital output in

> the mix is sent to an independent DAC, getting full 24 bit resolution

> per and full samplerate per signal, and then summing that in the

> analog domain. This effectively multiplies the bitrates of all the

> sources and is the equivalent to the "convolution of C-number of

> signals of N samples (T1 and T2) has length C*N-1"

Don't know where you get this, but I assume it's not the result

of your own intellectual efforts and so I can just say 'bullshit'

(the pure liquid variety actually).
I won't even comment on the rest, as I'd have to be at least

as rude :-)
Ciao,
--

FA
There are three of them, and Alleline.
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