[music-dsp] Roland D-50 "Filter"

[Alen Koebel]

Re-posting - forgot the subject line!

-----------------------------------

My first post here. Be gentle. ;-)

I'm trying to find out how the D-50 (and D-550, D-10,
D-110, D-20 & MT32) simulates a resonant filter. I say
"simulates" not "models" because I doubt there is an
actual discretized model of a filter in the synth
engine. The evidence for this is (a) the fact that
samples (short though they were) cannot be processed
by the "filter" - it "processes" only pulse and saw-
tooth waveforms, (b) a block diagram in the D-50 manual
seen here http://homepage.mac.com/synth_seal/html/d110.html
shows pulse-width controlled waveforms being created,
then processed ("filtered") in some way to add resonance,
then multiplied (ring modulated) by a synchronous sine
wave to get a sawtooth-like waveform. The "filter" block
is where the magic happens. "Magic" is a relative term -
the filter sounds real enough to my ears (I have a D-10)
but doesn't have the silky-smooth quality we associate
with a well-modeled filter.

I've searched the music-dsp archives, as well as the
synth-diy archives and the electro-music.com forums,
but came up dry. General Iternet searches came up with
only one idea but I admit I don't fully understand the
description:

http://www.kvraudio.com/forum/viewtopic.php?p=3747093

Any pointers or ideas? Maybe the above idea is spot on
and I'm just not understanding it, so any help in that
direction would be appreciated too.

[Andy Farnell]

Roland advertised the D-series as using "Linear Arithmetic"
synthesis. To my knowledge this is an implementation of the
waveshaping methods. So what you hear as a filter is in fact
a sophisticated form of distortion.

On Wed, 27 Jan 2010 10:21:03 -0800 (PST)

Post by Alen Koebel
Re-posting - forgot the subject line!
-----------------------------------
My first post here. Be gentle. ;-)
I'm trying to find out how the D-50 (and D-550, D-10,
D-110, D-20 & MT32) simulates a resonant filter. I say
"simulates" not "models" because I doubt there is an
actual discretized model of a filter in the synth
engine. The evidence for this is (a) the fact that
samples (short though they were) cannot be processed
by the "filter" - it "processes" only pulse and saw-
tooth waveforms, (b) a block diagram in the D-50 manual
seen here http://homepage.mac.com/synth_seal/html/d110.html
shows pulse-width controlled waveforms being created,
then processed ("filtered") in some way to add resonance,
then multiplied (ring modulated) by a synchronous sine
wave to get a sawtooth-like waveform. The "filter" block
is where the magic happens. "Magic" is a relative term -
the filter sounds real enough to my ears (I have a D-10)
but doesn't have the silky-smooth quality we associate
with a well-modeled filter.
I've searched the music-dsp archives, as well as the
synth-diy archives and the electro-music.com forums,
but came up dry. General Iternet searches came up with
only one idea but I admit I don't fully understand the
http://www.kvraudio.com/forum/viewtopic.php?p=3747093
Any pointers or ideas? Maybe the above idea is spot on
and I'm just not understanding it, so any help in that
direction would be appreciated too.
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[sdiedrichsen]

This is a challenging statement. I work in this field for a pretty long time. But it never occured to me, that a waveshaping applied to a rectangular wave resulted in a totally different wave than a scaled version of a rectangular version. No wonder, considering that a rectangular wave has just two values (not looking at anti-aliasing efforts), which gives you just a set ot 2 other values after waveshaping. ;-)
Actually, the Roland Filter can be anything.
Theory #1: It's a true resonating filter, maybe a digital state variable filter.
Theory #2: The rectangular generator is a variation of the BLIT-concept. Instead of firing a set of sinc-impulses into a summer it uses impulse reponses of a resonant filter with subsequent integration. Such a table could grow quite big, since you have a 4D array (resonance, cutoff, sub-sample fraction, index), but it would explain the inconsistent sound of the D-50, especially in the low-end, due to shortened IRs.
Theory #3: It's a simple playback of sampled rectangular waveforms. Unlikely, since you have this symmetric PWM, which points to a rect wave being derived from the sine wave.

However, it was a clever way to put together a sound engine, which created a bunch of nice sounds, especially the combination with samples made it special.

Best,

Steffan

Post by Andy Farnell
Roland advertised the D-series as using "Linear Arithmetic"
synthesis. To my knowledge this is an implementation of the
waveshaping methods. So what you hear as a filter is in fact
a sophisticated form of distortion.

[Alen Koebel]

Post by Andy Farnell
Roland advertised the D-series as using "Linear
Arithmetic" synthesis. To my knowledge this is an
implementation of the waveshaping methods. So
what you hear as a filter is in fact a
sophisticated form of distortion. 

Thanks. That's what I guessed. It's the
"sophisticated" aspect that I want to find
out about in detail. Is there a Roland patent
on the exact method? I haven't been able to
find one in searches, but then maybe I'm not
using the right keywords.

[Victor NK-X-TODEL918ani]

With distortion methods you can emulate filters (see for instance the
article "Subtractive Synthesis without Filters") and also resonance
(see PAF and Modified FM). Casio Cz-series emulated resonance using a
heterodyne method (which is not too far away from PAF & ModFM).

Victor

Post by Alen Koebel

Post by Andy Farnell
Roland advertised the D-series as using "Linear
Arithmetic" synthesis. To my knowledge this is an
implementation of the waveshaping methods. So
what you hear as a filter is in fact a
sophisticated form of distortion.

Thanks. That's what I guessed. It's the
"sophisticated" aspect that I want to find
out about in detail. Is there a Roland patent
on the exact method? I haven't been able to
find one in searches, but then maybe I'm not
using the right keywords.
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[Alen Koebel]

Post by Victor NK-X-TODEL918ani
With distortion methods you can emulate filters (see for
instance the article "Subtractive Synthesis without
Filters") and also resonance (see PAF and Modified FM).

Hi Victor. I've read some of your papers, those I can find
openly on the Internet, at any rate (which excludes
most CMJ papers, of course). I can't say I full under-
stand all of the math, having been out of school far too
long (BSc, Physics, '83). These methods do seem to all
be related; maybe I'll just to have experiment myself
with some code to discover exactly how Roland did it.

Post by Victor NK-X-TODEL918ani
Casio Cz-series emulated resonance using a heterodyne
method (which is not too far away from PAF & ModFM).

I have read the Casio patent. Dry stuff but I understood
that better actually, since it's hardware oriented and
hardware design is what I used to do.

[Victor NK-X-TODEL918ani]

The resonance thing is relatively simple to understand. Simply
speaking you have
a modulator M and a sinusoid S and you multiply them together

M * S

to get the resonant sound. The modulator is a complex wave that should
ideally be
bandlimited or nearly bandlimited. It runs at the fundamental frequency.
The sinusoid is tuned to the centre of resonance. With PAF, for
instance, we have a bell-shaped modulator (a Gaussian shape).

To get things right, there should be some synchronisation between the
modulator and sinusoid phases so that they always start together. This
is the CZ trick too, they are syncd so that the when the modulator
'resets', the sinusoid is zero (it's in sine phase).

Also, to sweep smoothly, the trick (and this comes from PAF) is to use
2 sinusoids, tuned to adjacent harmonics of the fundamental around the
centre frequency. The output is then a linear interpolation of these
two carriers, controlled by the position of the centre frequency
relative to these two harmonics.

The two brilliant papers you should read about distortion are Moorer's
DSF paper (available from his site) and LeBrun's waveshaping paper
(not sure if it's freely available). But that 'Subtractive Synthesis
without Filters' (I think it's freely available) is a good overview.

Victor

Post by Alen Koebel

Post by Victor NK-X-TODEL918ani
Casio Cz-series emulated resonance using a heterodyne
method (which is not too far away from PAF & ModFM).

I have read the Casio patent. Dry stuff but I understood
that better actually, since it's hardware oriented and
hardware design is what I used to do.
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[Eric Brombaugh]

Post by Victor NK-X-TODEL918ani
The two brilliant papers you should read about distortion are Moorer's
DSF paper (available from his site) and LeBrun's waveshaping paper
(not sure if it's freely available). But that 'Subtractive Synthesis
without Filters' (I think it's freely available) is a good overview.

Moorer's DSF paper _should_ be here (along with some other good stuff):

http://www.jamminpower.com/main/articles.html

and actually appears to be, but downloading it results in a 0-length
file. Anyone have a copy?

'Subtractive Synthesis without Filters' is here:

http://www.eecs.berkeley.edu/~johnw/papers/audioAnecdotes.pdf

Eric

[Didier Dambrin]

Hi,

I've always tried to improve spectrograms the way we generally need them,
linear in octave scale, and I recently tried something that proved to work a
little.

Basically you'd take a huge FFT window that you'd need to get good detail at
the bottom of the spectrum, but in order to get more accuracy in time
instead of frequency as you go up, since you're gonna merge more & more
bands as you go up (to go from Hz to octave scale), you'd run an
opening/closing IIR lowpass before processing the FFT.
That is, you process one or more (could be from both directions to get
linear phase) IIRs that are opening linearly in octave until the middle of
the frame, then closing.


That said, it showed to be less useful than combining FFT's of different
lengths. Maybe using Wavelets would also be more useful, I never really
tried.
Reassigned spectrograms also seem to work well, but require to process
neighbors in order to compare phases.

[Danijel Domazet]

Hi list,
Here is a tough one :-).

I have a regular analysis/synthesis framework:

time in > hop size shift > window > DFT >
(spectrum modifications) >
iDFT > window >overlapp add > time out

I'd like to make this framework more robust to very harsh spectum
modifications by adding a zero-padding step on the analysis side, like so:

time in > hop size shift > window > zero padd > longer DFT >...

The problem here is the windowing of the zero padded stuff on a
synthesis side. iDFT will give more samples and these have to be taken
into account somehow. Just to overlap-add them isn't working because the
synthesis window isn't applied to zero-padded data, it is too short
since it has to be the same as the input window.

longer iDFT > window > ? > overlapp add > time out

How to apply the synthesis side windowing on the iDFT "tail" data
("tail" being the samples caused by zero padding the input)?

Thanks!

p.s.
To make it easier, lets say hop size iz 1/4 of the frame size, and zero
padding factor is 2 i.e. the number of zeros is equal to number of input
data.


Danijel Domazet
/CEO/
www.LittleEndian.com

[Lubomir I. Ivanov]

in my opinion, all time-frequency transformation will have some downsides,
and even if one day there is a close to perfect method, such may be quite
inefficient for a nowaday cpu.
alot is done by combining methods:

-----

here is an interesting read:
http://users.mct.open.ac.uk/tec69/TomC-DMRN.pdf

which is a proposal for a hybrid method in the lines of:
musical signal -> time-frequency transform -> frequency vector ->
spectrogram
in a time-frequency transform for a given time frame the signal is reduced
to frequency components (sinosoids) and then the relative strength of the
components is vectorized (matrix)

-------------
in the fashion of combining methods:
http://en.wikipedia.org/wiki/Gabor-Wigner_transform
again both the gabor and wigner transform have some issues. (the wigner
distribution function produces some pretty results at least)

not sure how to write integrals in an email?
but combining the two for signal x:
Dx(t, f) = Gx(t, f) * Wx(t, f)

----------
further extending the G-W transform read + some info on Fractional Fourier
Transform (doc file):
http://disp.ee.ntu.edu.tw/meeting/%E5%9C%8B%E9%8A%93/FrFT%20&%20LCT/Fractional%20Fourier%20Transform.doc

^ a nice read

----
i don't think there are much matlab / c code examples in the internet for
this though.


----- Original Message -----
From: "Didier Dambrin" <***@skynet.be>
To: "A discussion list for music-related DSP" <music-***@music.columbia.edu>
Sent: Tuesday, February 16, 2010 6:33 AM
Subject: [music-dsp] Spectrograms

Post by Didier Dambrin
Hi,
I've always tried to improve spectrograms the way we generally need them,
linear in octave scale, and I recently tried something that proved to work a
little.
Basically you'd take a huge FFT window that you'd need to get good detail at
the bottom of the spectrum, but in order to get more accuracy in time
instead of frequency as you go up, since you're gonna merge more & more
bands as you go up (to go from Hz to octave scale), you'd run an
opening/closing IIR lowpass before processing the FFT.
That is, you process one or more (could be from both directions to get
linear phase) IIRs that are opening linearly in octave until the middle of
the frame, then closing.
That said, it showed to be less useful than combining FFT's of different
lengths. Maybe using Wavelets would also be more useful, I never really
tried.
Reassigned spectrograms also seem to work well, but require to process
neighbors in order to compare phases.
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[Didier Dambrin]

looks interesting, thanks

Post by Lubomir I. Ivanov
in my opinion, all time-frequency transformation will have some downsides,
and even if one day there is a close to perfect method, such may be quite
inefficient for a nowaday cpu.
-----
http://users.mct.open.ac.uk/tec69/TomC-DMRN.pdf
musical signal -> time-frequency transform -> frequency vector ->
spectrogram
in a time-frequency transform for a given time frame the signal is reduced
to frequency components (sinosoids) and then the relative strength of the
components is vectorized (matrix)
-------------
http://en.wikipedia.org/wiki/Gabor-Wigner_transform
again both the gabor and wigner transform have some issues. (the wigner
distribution function produces some pretty results at least)
not sure how to write integrals in an email?
Dx(t, f) = Gx(t, f) * Wx(t, f)
----------
further extending the G-W transform read + some info on Fractional Fourier
http://disp.ee.ntu.edu.tw/meeting/%E5%9C%8B%E9%8A%93/FrFT%20&%20LCT/Fractional%20Fourier%20Transform.doc
^ a nice read
----
i don't think there are much matlab / c code examples in the internet for
this though.
----- Original Message -----
To: "A discussion list for music-related DSP"
Sent: Tuesday, February 16, 2010 6:33 AM
Subject: [music-dsp] Spectrograms

Post by Didier Dambrin
Hi,
I've always tried to improve spectrograms the way we generally need them,
linear in octave scale, and I recently tried something that proved to
work
a
little.
Basically you'd take a huge FFT window that you'd need to get good detail at
the bottom of the spectrum, but in order to get more accuracy in time
instead of frequency as you go up, since you're gonna merge more & more
bands as you go up (to go from Hz to octave scale), you'd run an
opening/closing IIR lowpass before processing the FFT.
That is, you process one or more (could be from both directions to get
linear phase) IIRs that are opening linearly in octave until the middle of
the frame, then closing.
That said, it showed to be less useful than combining FFT's of different
lengths. Maybe using Wavelets would also be more useful, I never really
tried.
Reassigned spectrograms also seem to work well, but require to process
neighbors in order to compare phases.
--
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dsp links
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[Pascal Lemoine]

Post by Didier Dambrin
I've always tried to improve spectrograms the way we generally need them,
linear in octave scale, and I recently tried something that proved to work a
little.

AFAICT you should be interested in "Warped DFT". Look for :
"Warped Discrete-Fourier Transform: Theory and Applications" -
Anamitra Makur & Sanjit K. Mitra - IEEE TRANSACTIONS ON CIRCUITS AND
SYSTEMS - I: FUNDAMENTAL THEORY AND APPLICATIONS, VOL. 48, NO. 9,
SEPTEMBER 2001
"Design and Computation of Warped Time-Frequency Transforms" - Thesis
by Salvatore Caporale - UNIVERSITÀ DI BOLOGNA

You could also envision your problem differently and, provided that
you don't need realtime implementation, look into sparse
decompositions from a redundant dictionary. For example Matching
Pursuit:
http://en.wikipedia.org/wiki/Matching_pursuit
Performing a pursuit from a dictionary of Gabor atoms yields a T/F
representation composed of analytic "grains" ordered by decreasing
energy (the approximation is carried out in the L2 sense).
This is an adaptive method and as such, you won't be "locked" in the
harmonic frequencies inherent with the DFT. It's a modern way to
perform an offline analysis, in comparison to
spectograms/periodograms.

Wavelets are appropriate for 2D signals but for T/F analysis of a 1D
signal they're no good, because wavelets have generally bad T/F
localization. You'd better stay away from them for what you're
after...

just my 2 cents,
Pascal

[Alen Koebel]

Post by Victor NK-X-TODEL918ani
The resonance thing is relatively simple to
understand. Simply speaking you have a
modulator M and a sinusoid S and you multiply
them together M * S to get the resonant
sound.

How do you control the amount of resonance?
On the D-series synthesizers, there are 30
or so settings of resonance from none to
maximum. Is this perhaps just the amplitude
of the resonant M*S component relative to
the amplitude of a non-resonant "base"
waveform summed with it?

Post by Victor NK-X-TODEL918ani
The modulator is a complex wave that
should ideally be bandlimited or nearly
bandlimited. It runs at the fundamental
frequency. The sinusoid is tuned to the
centre of resonance. With PAF, for instance,
we have a bell-shaped modulator (a Gaussian
shape).

I suppose you can still get useful results
even with non-ideal envelopes? A sign this
might be the case is Yamaha's implementation
of formants in their FSR-1 where the sine is
modulated by cos^N where N = 1 to 8 (IIRC),
which they call "skirt" and cleverly achieve
by shifting the word read out from the sine
table, which actually stores the logarithm of
sine (same trick as in the DX7 and successive
models to avoid multiplications). I've never
heard an FSR-1 myself but I haven't read of
anyone complaining it sounds bad.

Post by Victor NK-X-TODEL918ani
To get things right, there should be some
synchronisation between the modulator and
sinusoid phases so that they always start
together. This is the CZ trick too, they
are syncd so that the when the modulator 
'resets', the sinusoid is zero (it's in
sine phase).

Another thing about the CZ is that it uses
just linear segments to form the modulator
envelope, which I assume was just easier to
do in a custom IC and avoided another table
look-up. Casio's patent also describes
modulation by cosine, which judging by the
spectra in the patent is "better" than the
other modulator shapes they ended up using).

The downside of Casio ultra-simple approach
seems to be, no control over the amount of
resonance, only its frequency.

Post by Victor NK-X-TODEL918ani
The two brilliant papers you should read about
distortion are Moorer's DSF paper (available from
his site) and LeBrun's waveshaping paper (not
sure if it's freely available).

I have read the Moorer DSF paper but the LeBrun
paper, as old as it now is, does not seem to be
freely available (I would enjoy being wrong!).
I may have to physically visit my alma mater
to get inside the Great JSTOR Firewall and
download that paper and a few dozen other
CMJ and AES papers that look very promising.

Post by Victor NK-X-TODEL918ani
But that 'Subtractive Synthesis without Filters'
(I think it's freely available) is a good
overview.

Read that one too, but perhaps I missed
something important. Equation (4) in that
paper looked a little overly complicated
given the result, which is just a low-pass
filtered square wave with no resonance.
And implementing on a fixed-point DSP
or an FPGA (Hi Scott!!) would seem non-
trivial. Whatever Roland was doing, my
instincts are that it was relatively simple.

My feeling is that distortion still has
huge potential for new sounds, and
possibly even for imitative synthesis of
acoustic instruments. Sure, we've had
commercial products in the past based
on FM/PM and phase distortion from
Yamaha, Casio and a few others (e.g.,
the DK Synergy, which in truth implemented
FM more often than additive in its voices)
and now in many VST soft-synths. But
technology-related limits on past
implementations have tainted the modern
view of how useful it is. Plus, for
researchers physical modeling still seems
to be the siren.

[Victor NK-X-TODEL918ani]

It depends on what 'amount of resonance' exactly is. If it's the
bandwidth, then narrowing the modulator shape (which makes it pulse-
like) would widen the resonance region. But if you are mixing two
signals, yes, it could be something like what you suggest.

As for the FSR-1 (I have not looked closely at the patent, but thanks
for remind me), I guess by multiplying cosines together you will get a
pulse-like wave, which gets narrower with higher powers (and thus with
wider resonant region).

The main things about the modulator shape are, I think: to be able to
control it (to change the bandwidth) and to make sure any
discontinuities in the carrier are smoothed out.

There is definitely lots of potential here. I find it useful to use a
computer music language to test and play with these things (in my case
I use Csound, but PD and SC3 will also do the trick)

Victor

Post by Alen Koebel
How do you control the amount of resonance?
On the D-series synthesizers, there are 30
or so settings of resonance from none to
maximum. Is this perhaps just the amplitude
of the resonant M*S component relative to
the amplitude of a non-resonant "base"
waveform summed with it?

[Risto Holopainen]

Agreed. I think FM and similar nonlinear techniques can benefit from
putting some memory into them, for instance in the form of applying a
filter to the modulating wave. This can be a way to accommodate for how an
acoustic instrument reacts to changing pitches. The transition is colored
by resonances in the instrument.

And as to Moorer's paper, I just found a copy. Should I post it as an
attachment to the list?


Risto Holopainen
Department of Musicology
University of Oslo

Post by Alen Koebel
My feeling is that distortion still has
huge potential for new sounds, and
possibly even for imitative synthesis of
acoustic instruments. Sure, we've had
commercial products in the past based
on FM/PM and phase distortion from
Yamaha, Casio and a few others (e.g.,
the DK Synergy, which in truth implemented
FM more often than additive in its voices)
and now in many VST soft-synths. But
technology-related limits on past
implementations have tainted the modern
view of how useful it is. Plus, for
researchers physical modeling still seems
to be the siren.

[Eric Brombaugh]

Post by Risto Holopainen
And as to Moorer's paper, I just found a copy. Should I post it as an
attachment to the list?

Several folks have responded to me off-list and I've got a copy now so
don't post on my account.

Thanks,

Eric

[Ross Bencina]

Post by Alen Koebel
I'm trying to find out how the D-50 (and D-550, D-10,
D-110, D-20 & MT32) simulates a resonant filter.

Any chance you could post some sampled waveforms for the different settings,
say for a low frequency tone with different cutoff frequencies and resonance
amounts...

Even if not, if it's a time domain process you'll probably gain some
intuitive understanding of what's going on by examining the waveforms or
watching them on a real-time scope while changing the parameters

Ross.

[Alen Koebel]

Post by sdiedrichsen

Post by Andy Farnell
Roland advertised the D-series as using "Linear
Arithmetic" synthesis. To my knowledge this is
an implementation of the waveshaping methods.
So what you hear as a filter is in fact
a sophisticated form of distortion. 

This is a challenging statement.

If by challenging you mean interesting, I agree. Ever
since I got a D-10 I have been intrigued by how Roland
managed to do what they did "way" back then. The D-50,
which was the first model with "LA Synthesis" (a
marketing term if I ever heard one) came out in 1987,
only four years after the DX7. AFAIK it was the first
keyboard synth to implement a digital low-pass resonant
"filter" that emulated what an analog subtractive synth
did, although we question whether it was a real filter
as we now implement them.

I was a bit surprised to find that apparently few people
have been interested in how Roland did it. I not only
want to find out, but hope to promote some discussion.

Post by sdiedrichsen
Theory #1: It's a true resonating filter, maybe a digital
state variable filter.

I think this unlikely because of when the D-50 was released
(long before synths with digital filters we know for certain
to be real) and because of the filter's inability to
process samples. Why limit it such if the filter is real?
(I suppose if it sounded really bad on the samples for some
reason they might have chosen to prevent that.)

Post by sdiedrichsen
Theory #2: The rectangular generator is a variation of the
BLIT-concept. Instead of firing a set of sinc-impulses into
a summer it uses impulse reponses of a resonant filter with
subsequent integration. Such a table could grow quite big,
since you have a 4D array (resonance, cutoff, sub-sample
fraction, index), but it would explain the inconsistent
sound of the D-50, especially in  the low-end, due to
shortened IRs.

I was thinking along these lines too. How about an MBLEP
related technique? Table size would still be an issue,
I suppose, but other aspects of implementation might
be easier, just as it seems to be with MBLEP (going by
what I read in papers - I've never done either myself).

Post by sdiedrichsen
Theory #3: It's a simple playback of sampled rectangular
waveforms. Unlikely, since you have this symmetric PWM,
which points to a rect wave being derived from the sine
wave.

Agreed.

Post by sdiedrichsen
However, it was a clever way to put together a sound
engine, which created a bunch of nice sounds,
especially the combination with samples made it special.

Yes. Some of the D-50 sounds are still sonically special
to this day. AFAIK, there is no VST implementation.
Short of using a real D-50, the only way to get the
same sounds (with the same control, so sampling a D-50
doesn't really count) is to buy the VC-1 D-50 option
for Roland's V-Synth.

[Alen Koebel]

Post by sdiedrichsen
Actually, the Roland Filter can be anything.

Here are some comments from user reviews of the D-50,
on Harmony Central that might provide some insight:

"The filter has a very unique feeling (reminds somehow
of FM synthesis feedback loop)..."

"Excellent PWM - very clean, not really digital sounding."

"Warm sound, filter is pretty good (see remark above,
though), resonant."

"The filters don't have enough resonance,...I don't find
the sounds that expressive, partly because of the not-
so-very-good filters."

And here's a few concerning D-5/D-10/D-20/D-110 models,
which along with the MT-32 all shared a simpler
(cheaper!) implementation of "LAS":

"The "analog" section has all what analog needs. But it
sounds a little too "clean"."

"Filter section - Low Pass filter with resonance. I don't
know how many decibels the filters are, but they don't
exactly 'bite'..."

"The Oscillators always acts strangely which can be a
problem. (might be a design error - don't know?)
EXAMPLE: This means that they randomly switch upside
down. If you make a dual-partial sound with two
simultanious SAW waveforms they sometimes 'kinda'
phase out each other."

[Scott Gravenhorst]

Post by Risto Holopainen
And as to Moorer's paper, I just found a copy. Should I post it as an
attachment to the list?

I'm interested... (having a D-110)


-- ScottG
________________________________________________________________________
-- Scott Gravenhorst
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