Hi all again, I think I found a reasonable solution, after reading around page 74 of this document:
Introduction to Sound Processing by Davide Rocchesso http://profs.sci.univr.it/~rocchess/SP/sp.pdf ----8<---- begin quote In practice, once we have constructed a lossless FDN prototype, we must insert attenuation coefficients and filters in the feedback loop. For instance, following the indications of Jot [45], we can cascade every delay line with a gain g_i = a^m_i // m_i is delay line length in samples This corresponds to replacing D(z) with D(z/a) in (42). With this choice of the attenuation coefficients, all the poles are contracted by the same factor a. As a consequence, all the modes decay with the same rate, and the reverberation time (defined for a level attenuation of 60dB) is given by Td = -3 Ts / log a // Ts is 1/samplerate ----8<---- end quote That gives a different attenuation for each delay, but such that the "decay per sample" is constant for all of the delay lines, which makes the reverb time calculations much easier! gain_i = 10^(-3 * delay_i / reverbTime) where delay_i and reverbTime are measured in the same unit (eg: ms). Claude Miller Puckette wrote: > hi all, > > I don't think anyone knows the answer to this. Traditionally, ever since > Schroeder's reverberator, I think people have used delay times within a ratio > of 1.5:1 of each other so that any old mean works OK. > > cheers > Miller > > On Fri, Aug 29, 2008 at 10:07:30PM +0100, Claude Heiland-Allen wrote: >> Hi all, >> >> Referring to Miller's book [1], and having experimented with various >> delay times, I'm wondering what the "average" delay time used in the >> text is. If all the delay times are close to equal, then using the >> arithmetic mean as "average" gives me a reasonably accurate >> reverberation time calculation. But the more they differ the worse the >> result is (comparing with measurement against my implementation). >> >> [1] http://crca.ucsd.edu/~msp/techniques/latest/book-html/node111.html >> >> Intuitively it seems that the sound recirculates more often (and is thus >> attenuated more) through the shorter delay lines, but this is obviously >> not taken into account with arithmetic mean. I'm thinking something >> like harmonic mean might be better (but I tried it and it wasn't a huge >> improvement, nor was geometric mean). >> >> Any clarification would be enlightening, thanks, >> >> >> Claude >> -- >> http://claudiusmaximus.goto10.org >> >> _______________________________________________ >> [email protected] mailing list >> UNSUBSCRIBE and account-management -> >> http://lists.puredata.info/listinfo/pd-list _______________________________________________ [email protected] mailing list UNSUBSCRIBE and account-management -> http://lists.puredata.info/listinfo/pd-list
