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
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
Pd-list@iem.at mailing list UNSUBSCRIBE and account-management -> http://lists.puredata.info/listinfo/pd-list
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
Pd-list@iem.at mailing list UNSUBSCRIBE and account-management -> http://lists.puredata.info/listinfo/pd-list
-
Claude Heiland-Allen -
Miller Puckette