Re: [music-dsp] Analog versus digital systems (Ezra Buchla)

2013-11-11 Thread Theo Verelst
Sure, and I didn't call you dumb. I'm not dumb for being left out the 
Russian space program, or wanting to write a book without a alpha-degree 
or what else.. It's a hard area, which I why I've tried to make a few 
noticeable remarks, to shake the particular subject up a bit, so to speak.


If you're serious, I don't mind summing a few things up with practical 
application, no problem. Might even be fun.


But not at this moment. Maybe tomorrow.

One main thing for now, there's an interesting matter about the Switched 
Filter, primarily at the moment to broaden the attention, and to make 
the payload of this message enter in a more general digital and signal 
processing domain for musical (and other) applications:


 - Most digital systems as we now know them are Time Quantized (usually 
equally spaced times steps with impulse-based samples (for the sake of 
the Reconstruction theorem)), and Amplitude Quantized (vertical steps, 
like 16 bits for instance), for the sake of being able to use digital 
computers.


 - As a interesting comparison (for the effect) there are also 
Time-Quantized, Amplitude-Continuous processing chips since at least the 
early (Time Division Multiplexing) digital phones, of which my switched 
filter is an example. Clear difference: no vertical resolution to create 
quantization noise, and it is possible to nowadays buy chips that have 
digitally programmable *analog* signal path chips that may benefit from 
such tricks, possibly applicable in modern digital: amplifiers.


 - Of course there are, at least since music in the 60s I think, 
machines which do amplitude quantization only, like a grungelize 
sound, applying a staircase distortion to the signal, but *without* 
sampling the signal in the time dimension.


 - For modern attempts as clean analog signal paths *with* time-varying 
filters one can make use of digitally programmable amplifiers, like the 
Crystal/Burr-Brown half-dB step, fast digital control amplifier chips (a 
project of mine here, at about 1/3th of the page, only if you're 
interested: http://www.theover.org/Audio/index2.html ), using these 
chips instead of Operational Trans-conductance Amplifiers, digitally 
controlled analog filters may be created. The switching technique could 
add more resolution to this solution, and requires fast digital control 
and an analysis of the effects of applying digital dithering of such kind.


 - Of course the original of many software implementations of digitally 
controlled State Varying Filters (besides the fully analog Moogs and 
others), from early computer-controlled analog signal path music 
synthesizers on, has all kinds of tuning built in the control path of 
the filters (and other components like oscillators and VCAs). So a part 
of the question I asked myself was: how is it that those analog filters 
sound good (or bad if that's the will of the sound programmer), given 
that a part of the control signal path comes from digital control. For 
instance: what are the properties of the DA convertor that changes a 
know-turn on the computer side of the instrument, into an electrical 
signal to drive the cut-off control line of the filter circuit.


Gr.

  T.V.
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[music-dsp] Analog versus digital systems

2013-11-10 Thread Theo Verelst


Hi all,

Of course I'm aware of it this work probably won't give m a (bit late) 
YUP existence in SanFrancisco or a well paid Berkeley professorship that 
I like, but at least I don't really run the risk of looking like a 
dumb-*ss when playing the unpaid professor a bit in this territory, and 
hopefully cut down some Non-Giant Redwood trees that appear to create 
more pollution than oxygen.


First, a repeat of what I've tried to communicate a number of times, as 
it were to discourage the idea of taking interesting mathematical truths 
from (lame or interesting) digital signal processing effects in music, 
let's first consider the theoretical basics that defy all ignoration:


(A)
  {Digital System} -- {Digital Sample stream} -- {Digital to Analog} 
-- {Analog signal}


, versus:

(B)
  {Analog System} -- {Analog Signal}

The main differences are on a short list:

  (1) The only way in which the two analog output signals of graph (A) 
and (B) are going to be (almost or perfectly) the same is when somehow 
the digital system creates very well made samples (which *CAN* come from 
simply playing back accurately sampled form of some frequency limited 
signal), and the Digital to Analog convertor is very high quality (or to 
achieve actual mathematical perfection: is a perfect reconstruction type)
  (2) The digital system implemented as a filtering of any kind of 
combination of FIR/IIR tap-connections is normally not coming close to 
making analog-equivalent signals, by far, unless it is big, and there 
explicit measures being taken (extremely high sampling frequencies and 
vertical resolution, tuning of the DA-convertors always-present 
transient behavior, medium long averaging effects control (hard problem) 
seriously long sinc-based integral corrections (computationally intensive)).
  (3) Reconstructing an analog signal from samples that isn't a 
retarded subset of all possible signals, will require a DA convertor 
design which has a serious signal delay, for all known normal and 
industrial Audio convertors. So to prevent some very measurable (by over 
see-ably simple traditional measurement techniques at the level of the 
THD of a very moderate transistor radio) distortion, serious measures 
would have to be taken, like outside the scope of this list. Even making 
sure those distortions don't become multi-fold ugly and even a potential 
danger to the hearing of the customers isn't easy (and thus far never 
has been discussed, even though these distortions are almost incredibly 
ugly, and host of unrealistic monitors have been invented which are 
supposed to smooth some of this over, apparently through lack of 
awareness of the impossibility to approach per-sample sinc functions by 
any resonance or other mechanical or switching amp trick).
  (4) It is quite possible to create a computer simulation of an 
electronics circuit, like a Moog filter, even with serious accuracy, and 
to state the output of such simulation in the form of a sequence of 
equidistant digital samples with accurate vertical quantization. Even 
this does not preclude you from having to take equal relevant care of 
the above, except for point (2).


There, that's a few New things, apparently for those not blessed with 
either the intelligence, means or geographical or time opportunity to 
follow a good EE university (or for most of this: bachelor level) 
Sophomore year equivalent.


Of course going a bit further in the better EE education (say second or 
third year of a serious education), you may want to practice yourself in 
creating computer models of interesting non-linear electronics circuits, 
and see if you computer simulations on the basis of these models and 
some form of circuit-to-signal strategy, be it based on the frequency 
domain or not, turn out to be accurate, and maybe invent some fun games 
with this, like a Virtual Prophet-5 that everybody can run on their 
home computer for free, or things equally thrilling and educational!


I had done some (extremely low budget) preparatory work because of my 
much longer standing personal interests for this (like owning various 
synthesizers and samplers with digital filters like the TG500 in the 
80s), see eg  http://theover.tripod.com/so1.html  and 
http://theover.tripod.com/switch.html  , written before the year 1999.


Ir. T. Verelst
http://www.theover.org/Synth

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Re: [music-dsp] Analog versus digital systems

2013-11-10 Thread douglas repetto


Theo, please stop with the insults.


On 11/10/13 9:55 AM, Theo Verelst wrote:

Of course I'm aware of it this work probably won't give m a (bit late)
YUP existence in SanFrancisco or a well paid Berkeley professorship that
I like, but at least I don't really run the risk of looking like a
dumb-*ss when playing the unpaid professor a bit in this territory, and
hopefully cut down some Non-Giant Redwood trees that appear to create
more pollution than oxygen.


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...repetto. http://music.columbia.edu/organism
... http://music.columbia.edu/~douglas


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Re: [music-dsp] Analog versus digital systems

2013-11-10 Thread Ezra Buchla
i don't want to feed the troll, but...

a) of course we all know about the sampling theorem and sync
interpolation. truly!

b) though i enjoy reading ASM synth code, i don't see anything here
that is interesting. what am i supposed to look at? a character
display? a phasor? an in-place filter or chromatic tuning table in
BASIC? confused.

c) why are you posting a schematic of a switched capacitor filter to
music-dsp? are you obliquely proposing a direct-modelling algorithm
for this kind of circuit? thats an interesting thought but i'd love to
see more elaboration, and what kind of advantage it would have over
FIR/IIR structures.

my degrees are in music, not EE, so actually i can often use a little
help connecting theoretical dots and i certainly admit that. but i
know how to hold the pen, and i can see the picture when it's done, so
to speak.

i would like to read your posts in the future, in case they are
useful, but i'm no longer giving them the benefit of the doubt if they
start off with vitiriol and end with ancient links of dubious
relevance.

- ezra buchla ( in Berkeley CA )


On Sun, Nov 10, 2013 at 10:08 AM, douglas repetto
doug...@music.columbia.edu wrote:


 Theo, please stop with the insults.



 On 11/10/13 9:55 AM, Theo Verelst wrote:

 Of course I'm aware of it this work probably won't give m a (bit late)
 YUP existence in SanFrancisco or a well paid Berkeley professorship that
 I like, but at least I don't really run the risk of looking like a
 dumb-*ss when playing the unpaid professor a bit in this territory, and
 hopefully cut down some Non-Giant Redwood trees that appear to create
 more pollution than oxygen.


 --
 ... http://artbots.org
 .douglas.irving http://dorkbot.org
 .. http://music.columbia.edu/cmc/music-dsp
 ...repetto. http://music.columbia.edu/organism
 ... http://music.columbia.edu/~douglas



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Re: [music-dsp] Analog versus digital systems (Ezra Buchla)

2013-11-10 Thread Theo Verelst
...
a) of course we all know about the sampling theorem and sync
interpolation. truly!

b) though i enjoy reading ASM synth code, i don't see anything here
that is interesting.  ...

I'm sorry to say, but while of course I don't feel all too much of it, and of 
course that isn't a reason to use my free speech necessarily for placing a 
correction, but that little snippet is quite insulting, given the story thus 
far. So outside of rethoric, that is a technical/scientific insult of the first 
order that you're trying to force my direction.

I don't really take the insult, and am glad there's serious discussion, and 
people feeling inspired to share maxima code, etc. and apparently not 
overwhelmed or something, so they post about what interests them, so on the 
average, I am glad about the results.

I don't feel like scientifically defending the quotes I few simple quotes I 
posted.

As a serious remark about the content of many of the musical and signal 
processing subjects: it's a great idea to use well known *analog* synthesizer 
designs as the basis for (partial) digital simulation, which I though already 
before people like Dave Smith were writing award winning software to that 
effect, and which interested me long before the advent of a number of software 
companies that occupy themselves with the subject. As the suggestion is from 
some of my quotes, it would be good to have a potent, 64 bit circuit simulator 
which allows audio output, and explicit (parts with curves for parameter 
changes) or implicit (driven sources in the network, OTAs in replacement 
circuits, etc) time dependencies, possibilities for storing/continuing network 
states, and a choice of accuracy feedbacks that I've been hinting at, and which 
clearly isn't understood by most, which doesn't make me continue.

Also, I've suggested signal improvements, but they won't work without some 
fundamental changes to the most used algorithms, which I would prefer to be 
applied to some musical software, preferably Open Source. Those things are very 
audible, and it surprises me that people who may feel the need for improvements 
are so numb. Must be some limited musicians trying to rule the show, which in 
broader circles, which also can benefit from DSP for musical purposes is 
getting in demand. Just saying. Sounds like a simple statement of truth to me, 
and I think I'm qualified to judge that, so if you feel a bit humble, don't 
confuse that with feeling insulted. I do feel slandered, regulaly, and that 
*is8 a real issue for me, and the law.

T.V.


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Re: [music-dsp] Analog versus digital systems (Ezra Buchla)

2013-11-10 Thread Ezra Buchla
i'm sorry mr. vereslt, i do apologize for insulting you. to be honest,
i made a mistake with the e-mail and thought this was still under that
other thread. and so i tried to read your code, [ which if course is
very interesting and informative to me (since i am not nearly at your
level as an engineer), it shows some fundamental aspects of
high-resolution sample reconstruction rendered in a highly-efficient
way ] - but it is not so directly relevant to coefficient changing
within the filter, is it?

if it is, can you put the relationship a little more plainly so that
dummies like me could understand?

for example, would you bandlimit an SVF coefficient change signal
using sinc or something? perhaps by offline-processing it and using
buffer playback?

is this stuff only relevant at 64 bits (high vertical resolution)?
what if we obtain much greater processing efficiency at 32?

switched capacitors is interesting to me too because i have been
thinking a lot about tunable sampling rates. this schematic itself is
old news though, is it the precise behavior of these integrators while
changing the clock rate that you wand to point out?

you see, i really respect your intelligence and erudition on this
stuff, so i woud love to hear more explanation, just without being so
angry. i have of course read the code on your website before now, but
still need to know more specifically what part you refer to.

i mean to try and starve troll in all of us, i am glad that your
pursue correctness aggressively, it is a benefit for all. and so
telling you that i'm from a certain area of the world that you make
fun of, and so is dave smith for that matter, and that it is easier to
pay attention when the tone is more civil. thank you for reminding me
of it as well.

sincerely, and thanks,
ezra b

On Sun, Nov 10, 2013 at 3:31 PM, Theo Verelst theo...@telfort.nl wrote:
 ...
a) of course we all know about the sampling theorem and sync
interpolation. truly!

b) though i enjoy reading ASM synth code, i don't see anything here
that is interesting.  ...

 I'm sorry to say, but while of course I don't feel all too much of it, and of 
 course that isn't a reason to use my free speech necessarily for placing a 
 correction, but that little snippet is quite insulting, given the story thus 
 far. So outside of rethoric, that is a technical/scientific insult of the 
 first order that you're trying to force my direction.

 I don't really take the insult, and am glad there's serious discussion, and 
 people feeling inspired to share maxima code, etc. and apparently not 
 overwhelmed or something, so they post about what interests them, so on the 
 average, I am glad about the results.

 I don't feel like scientifically defending the quotes I few simple quotes I 
 posted.

 As a serious remark about the content of many of the musical and signal 
 processing subjects: it's a great idea to use well known *analog* synthesizer 
 designs as the basis for (partial) digital simulation, which I though already 
 before people like Dave Smith were writing award winning software to that 
 effect, and which interested me long before the advent of a number of 
 software companies that occupy themselves with the subject. As the suggestion 
 is from some of my quotes, it would be good to have a potent, 64 bit circuit 
 simulator which allows audio output, and explicit (parts with curves for 
 parameter changes) or implicit (driven sources in the network, OTAs in 
 replacement circuits, etc) time dependencies, possibilities for 
 storing/continuing network states, and a choice of accuracy feedbacks that 
 I've been hinting at, and which clearly isn't understood by most, which 
 doesn't make me continue.

 Also, I've suggested signal improvements, but they won't work without some 
 fundamental changes to the most used algorithms, which I would prefer to be 
 applied to some musical software, preferably Open Source. Those things are 
 very audible, and it surprises me that people who may feel the need for 
 improvements are so numb. Must be some limited musicians trying to rule the 
 show, which in broader circles, which also can benefit from DSP for musical 
 purposes is getting in demand. Just saying. Sounds like a simple statement of 
 truth to me, and I think I'm qualified to judge that, so if you feel a bit 
 humble, don't confuse that with feeling insulted. I do feel slandered, 
 regulaly, and that *is8 a real issue for me, and the law.

 T.V.


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Re: [music-dsp] Analog versus digital systems (Ezra Buchla)

2013-11-10 Thread Ezra Buchla
or, oh! would you suggest that we stick to an analog design like the
switched capacitors? but there are still lots of reasons to try and
keep everything on DSP in terms of part count in a device.

again, i think of this space as very informal discussion, s please no
hard feelings,

ezra b

On Sun, Nov 10, 2013 at 4:06 PM, Ezra Buchla ezra.buc...@gmail.com wrote:
 i'm sorry mr. vereslt, i do apologize for insulting you. to be honest,
 i made a mistake with the e-mail and thought this was still under that
 other thread. and so i tried to read your code, [ which if course is
 very interesting and informative to me (since i am not nearly at your
 level as an engineer), it shows some fundamental aspects of
 high-resolution sample reconstruction rendered in a highly-efficient
 way ] - but it is not so directly relevant to coefficient changing
 within the filter, is it?

 if it is, can you put the relationship a little more plainly so that
 dummies like me could understand?

 for example, would you bandlimit an SVF coefficient change signal
 using sinc or something? perhaps by offline-processing it and using
 buffer playback?

 is this stuff only relevant at 64 bits (high vertical resolution)?
 what if we obtain much greater processing efficiency at 32?

 switched capacitors is interesting to me too because i have been
 thinking a lot about tunable sampling rates. this schematic itself is
 old news though, is it the precise behavior of these integrators while
 changing the clock rate that you wand to point out?

 you see, i really respect your intelligence and erudition on this
 stuff, so i woud love to hear more explanation, just without being so
 angry. i have of course read the code on your website before now, but
 still need to know more specifically what part you refer to.

 i mean to try and starve troll in all of us, i am glad that your
 pursue correctness aggressively, it is a benefit for all. and so
 telling you that i'm from a certain area of the world that you make
 fun of, and so is dave smith for that matter, and that it is easier to
 pay attention when the tone is more civil. thank you for reminding me
 of it as well.

 sincerely, and thanks,
 ezra b

 On Sun, Nov 10, 2013 at 3:31 PM, Theo Verelst theo...@telfort.nl wrote:
 ...
a) of course we all know about the sampling theorem and sync
interpolation. truly!

b) though i enjoy reading ASM synth code, i don't see anything here
that is interesting.  ...

 I'm sorry to say, but while of course I don't feel all too much of it, and 
 of course that isn't a reason to use my free speech necessarily for placing 
 a correction, but that little snippet is quite insulting, given the story 
 thus far. So outside of rethoric, that is a technical/scientific insult of 
 the first order that you're trying to force my direction.

 I don't really take the insult, and am glad there's serious discussion, and 
 people feeling inspired to share maxima code, etc. and apparently not 
 overwhelmed or something, so they post about what interests them, so on the 
 average, I am glad about the results.

 I don't feel like scientifically defending the quotes I few simple quotes I 
 posted.

 As a serious remark about the content of many of the musical and signal 
 processing subjects: it's a great idea to use well known *analog* 
 synthesizer designs as the basis for (partial) digital simulation, which I 
 though already before people like Dave Smith were writing award winning 
 software to that effect, and which interested me long before the advent of a 
 number of software companies that occupy themselves with the subject. As the 
 suggestion is from some of my quotes, it would be good to have a potent, 64 
 bit circuit simulator which allows audio output, and explicit (parts with 
 curves for parameter changes) or implicit (driven sources in the network, 
 OTAs in replacement circuits, etc) time dependencies, possibilities for 
 storing/continuing network states, and a choice of accuracy feedbacks that 
 I've been hinting at, and which clearly isn't understood by most, which 
 doesn't make me continue.

 Also, I've suggested signal improvements, but they won't work without some 
 fundamental changes to the most used algorithms, which I would prefer to be 
 applied to some musical software, preferably Open Source. Those things are 
 very audible, and it surprises me that people who may feel the need for 
 improvements are so numb. Must be some limited musicians trying to rule the 
 show, which in broader circles, which also can benefit from DSP for musical 
 purposes is getting in demand. Just saying. Sounds like a simple statement 
 of truth to me, and I think I'm qualified to judge that, so if you feel a 
 bit humble, don't confuse that with feeling insulted. I do feel slandered, 
 regulaly, and that *is8 a real issue for me, and the law.

 T.V.


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