| > Can anyone point me in the direction of technical books or papers on
> the subject of morphing audio signals?
Not to reopen this can of worms, but I have seen very little serious
litterature on audio morphing. There was considerable debate a while
back about this, on this list and elsewhere. One can can effectively
argue that morphing two audio signals is very similar to simply mixing
them. Interpolating two fft analysis files on a bin-to-bin basis over a
period of time would, in theory, be close to the well-known analogous
process in computer graphics/animation, but in my experience, the
results of this process in audio is almost identical to conventional
linear mixing. Which is to say that other more complex operations
(stretching, compressing, transposing etc.) on a pair of
frequency-domain data blocks have much more interesting sonic potential.
The term "morphing" is the problem here.
THe concept behind a popular synthesizer (Morpheus) a few years back was
not truly about morphing as much as multi-pole filtering with complex
control functions of filter poles, in a manner close to linear
predictive applications.
--
________________________________________________________
Jean Piche
Universite de Montreal
http://mistral.ere.umontreal.ca/~pichej
http://www.musique.umontreal.ca/Org/CompoElectro/CEC/
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Date: Fri, 28 Feb 1997 08:45:57 -0800
From: Erik Spjut
Subject: Re: Morpheus Filters
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At 10:29 PM -0800 2/27/97, Hans Mikelson wrote:
>Hi All,
>
>It would be nice to have access in Csound to a set of filter
>coefficients which could be used to morph from one type of filter
>to another as you can on the Emu Morpheus and Ultraproteus
>synthesizers. Their ad states that they use a 14 pole filter which
>would amount to Yn, Yn-1, Yn-2,...Yn-14.
>
>One could start messing around with 6 pole filters. That would
>enable things like formant filters with three peaks. One could set
>up a series of vowels with these.
To amplify Jean Piche's comment:
Csound already has tools that are in some ways superior to the Morpheus.
If one concatenates two different sounds in one soundfile, uses lpanal
to analyze the file and create two data frames, then one can use an lpread/
lpreson pair (along with balance) to interpolate between the two. Filters
in the 6 to 14 pole range are easy and usually quite stable. Above twenty
poles you have to make certain that the poles are all inside the unit circle
(or really increase the k-rate and trust in balance) but you can do 40 poles
if you realy want to. If the original soundfile had three sounds, you could
"morph" among three sounds and so forth.
-----------------------------------------------------------------------------
Erik Spjut (rhymes with cute) - Associate Professor of Engineering and/or
Associate Director for Engineering Computing in the Engineering Design Center
Harvey Mudd College, Claremont, CA 91711 USA
Erik_Spjut@hmc.edu Ph & Voice mail (909) 607-3890 Fax (909) 621-8967
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From: Mike Berry
Cc: csound
Subject: Re: Morphing
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There is also what is called Mutation by Larry Polansky
(implemented in SoundHack) which is based on mixing portions of a source's
FFT with portions of a target's FFT, in a variety of different ways. This
comes the closest I've heard to "morphing," i.e. giving a sound
characteristics of another sound.
Mike Berry
mikeb@mills.edu
http://www.mills.edu/PEOPLE/gr.pages/mikeb.public.html/mikeb.homepage.html
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From: Peter Kearton
Organization: Psychology Dept, Surrey Univ. U.K.
To: csound@maths.ex.ac.uk
Date: Fri, 28 Feb 1997 17:38:14 GMT
Subject: constant power density
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Hi all,
I've been attempting to run a simulation of Fletcher's band-
widening experiment where the threshold amplitude of a pure tone is
measured as a function of the a bandwidth of a bandpassed noise
masker centered at the pure tone frequency (2000Hz in this case).
It's essential for the experiment that noise power density is held
constant for different bandwidths of noise masker. I have been using
the Butterworth filter opcode to filter 'rand' generated white
noise, and in this basic form the max amp in a segment (as indicated
by CSound) increases with bandwith. I've then tried the approach
of using 'balance' to match the filtered noise samples to a reference
RMS value, and/or multiplying the instr 'out' signal by a fraction
that causes all max amp values for each bandwidth to be approximately
equal. I need some more info. or reasurrance that this
approach is indeed producing a more or less constant power density
across frequencies in the noise bands. When I look at the wave files
with a digital frequency analyser (Wave SE) the noise power density
seems to decrease with bandwidth, eventhough all the max amp values
are now the same. Is this just an artifact of the processing in the
frequency analyser, have I got decreasing noise power density, or
both ?
Any help appreciated......
Thanks in anticipation,
Pete Kearton
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Subject: LMJ Ed-in-Chief opening
NOTE
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