On Sun, Sep 26, 2010 at 11:40:08AM -0700, Niels Mayer wrote: > On Sun, Sep 26, 2010 at 4:12 AM, <fons@xxxxxxxxxxxxxxx> wrote: > > 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 > 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. _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
