[sdiy] Additive Synthesis - Wendy would say it works...

[harrybissell]

Yknow Batz I get some great ideas from you :^P

while reading your post, It made me think of a graphics program I use
called "the Gimp"  (which is a sort of photoshop clone).  People have made some

amazing image processing and transforming software in the form of macros or
functions.  Our music waveforms could probably have the same kinds of things,
like 'gaussian blur' or 'unsharp mask' etc...

in an open system, people could add a bunch of these and manipulate sound like
we manipulate video.  Some look good, others (like the classic bbd ;^) suck.

The tough part is that there are probably less people working with synthesis
than
with photo manipulation.  So lets get to it !!!

H^) harry

Batz Goodfortune wrote:

> Y-ellow All.
>
> At 07:09 AM 3/22/05 -0600, Bill Felton wrote:
> >I'm a bit surprised no one has mentioned Kyma yet -- it can do live
> >analysis/resynthesis, with real-time spectral modification.  Pretty
> >rich set of additive and quasi-additive approaches and some interesting
> >presentations of controls.
>
> Actually I did in passing as it happens.
>
> Ed asked (off list) why the K5K is close but no banana. And while I'm
> addressing that, it also sieges nicely with the GUI vs complexity thing. So
> I'll kill them both with a tiny drop of vitriol. Strong stuff that vitriol
> by all accounts.
>
> You'll have to forgive my lack of intimate knowledge of the K5K for the
> sake of argument. It's been a hell of a long time since I used one so I may
> have numbers and names wrong. Feel free to nit pick if you wish.
>
> Kawai actually released a small bit of software which tried to do Fourier
> re-synthesis but it wasn't very good. Actually useful but way short of the
> mark. Mainly because of the lack of control over the partials. There are
> something like 100 oscillators per voice. Each one has it's own envelope
> generator but they are way to simple. Being only ADSR types. Each partial
> has a fixed relationship to each other which means that if harmonic content
> doesn't fall neatly into this relationship then bad luck. And from memory,
> there is no control over individual partials from LFO sources or other more
> universal EGs etc. Although, I think there's some kind of grouping which is
> mildly useful.
>
> At best, the Kawai editor lets you grab some characteristics of the sound
> you're trying to translate but a great deal of post work is needed to make
> it sound anything but "THIN" I'm not saying this isn't useful because
> within this complex structure you wouldn't dream of starting a patch from
> scratch. (I know, I spent all day once trying to program a few patches into
> the thing.) But however useful it is, re-synthesis it aint.
>
> There are lots of ways to break down sound into useful byte-sized chunks.
> MP3 for example, uses a psycho-acoustic method. Obviously the goal of MP3
> is quite different and doesn't lend it self to synthesis. Although I've
> often wondered why could be done with that kind of psycho-acoustic modeling
> synthesis wise. I'll leave that for another day however.
>
> MP4 uses a whole bag of tricks to get massive reduction in audio size and
> one of the main tricks is re-synthesis. And here, it's aim isn't all that
> different except that it's designed for streaming. The language of MP4
> audio is called SAOLC. (pronounced sale sea) It's based on a cross between
> CSOUND and a MOD/TRACKER player. I'm too tired to go into this and I don't
> want to sound like I'm big-noting myself but some 5 years before SAOLC was
> even embryonic, I worked for a sound card company and we were developing
> something remarkably similar. The point being that a MOD/TRACKER system is
> yet another way to break down audio into partials and resynthesisze it
> later. And you get remarkable quality for a comparatively tiny amount of
> stored data. The reason I'm covering this background is that it should be
> noted that rather high quality playback can be achieved from layers of
> rather low quality stored data.
>
> To effectively use additive re-synthesis, there needs to be lots of
> partials. Each one needs more than you're average ADSR but less than a full
> 16 bit wave form. However the EG needs to resemble something like a wave
> form. It would be unipolar however and could be quite low in word size. Say
> 8 bit. And maybe as few as 256 samples. You could perhaps draw the envelope
> on-screen with a light pen like the old Fairlite.
>
> Using the K5K as a model, that would require about 25K of amplitude data
> per patch. It would be derived initially at least, by a fourier transform
> from an original sound. I also suspect that you could do with having some
> pitch variation of the individual oscillators in a voice as well. So that
> the harmonic relationships are not static. Just how un-static they need to
> be I'm not sure at this point. It would also depend on how frugal you have
> to be with your "Computational resources." It would be nice to have a
> similar EG arrangement for the pitch relationship of each partial similar
> to the amplitude relationship. However it may turn out that you can get
> away with much less.
>
> So at the base of the system you have a resynthesis engine that can take
> any recorded sound and resynthesize it with a high degree of faithfulness.
> You could then mess with each of the 100 odd partial envelopes and tweak
> them painstakingly but there's another way.
>
> You take a layered approach. And this is something Kawai did get right at a
> fundamental level with the K5K. You could start with one big, throbbing red
> knob on the front panel. With "Phat" at one end and "Suck" at the other. if
> you like. Like an Equalizer reduced to a single knob, you could have it max
> out all the odd harmonics at the "Phat" end and kill them off at the "Suck"
> end. Underneath that you might have other knobs assigned to groupings of
> parials that could shift them in amplitude or pitch. Underneath that you
> could have a series of universal EGs that do much the same thing but with a
> transient. Underneath that you might have a bunch of LFOs which can be
> assigned to groups of partials. All the way down to control of each
> partial's envelope it self. And of course, a bunch of universal EGs to
> control the over-all transient of the voice etc.
>
> The degree of control you use depends on how much tweaking you think the
> sound needs to achieve what you're after. You may find that making it suck
> a bit more gives you what you want. That dulcet BBD tone for example? On
> the other hand, you may be trying to introduce the bowing effect of a
> violin sound. In which case you might have to adjust a universal pitch EG
> to give you the proper pitch transient. You may find that the upper
> harmonics of the sound need squashing over time so you could apply a
> tweezing envelope to partials 50 thru 100. Perhaps a parameter that allows
> you to increase the effectiveness of this tweezing EG to progressively
> higher partials. All without ever having to have touched the original sets
> of EG data.
>
> Since it's mainly all about the control of the amplitude of a harmonic
> series, your GUI can simply be innovative ways to tweeze the original EG
> data. From the very simple to the very complex. The point being that if
> you're re-synthesis engine does the business in the first place, then it's
> up to you as to how complex or simple you tweeze it.
>
> In fact, what the hell, you could make that whole process modular.
>
> Anyway, I'm too fried to do a summery right now but I think I've addressed
> the issues asked of me. Except to say that. yes, Re-synthesis is too
> complex at a stick and rudder level perhaps but we're way beyond that these
> days. I would hope anyway.
>
> Hope this helps.
> Be absolutely Icebox.
>
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