Re: [music-dsp] Pointers for auto-classification of sounds?
On Tue, Jun 12, 2012 at 2:48 PM, douglas repetto doug...@music.columbia.edu wrote: I just tend to get over-critical of the claims people make about what they're doing. So much of the intersection between science/math/art/music is over-hyped that I've developed some sort of hype allergy. And because I'm a nerd I get all worked up about the imprecision of the language used to talk about these things in the arts. Sorry for being such a nerd, I'll stop spamming the list now!!! Hardly spam: The work, as Morton Feldman once said in a somewhat different context, rhapsodizes its own construction, exalts the intricacies of the structure through which it has acquired existence. Instead of revealing such properties as linear continuity, thematic and motivic development, formal cohesion, etc.-properties to a large extent jeopardized by the disappearance of a conventional syntax-the work reflects upon its own constitution. It is not surprising, then, that many composers establish systematic scaffoldings of great complexity on which to build their music, feats of dazzling virtuosity possessing considerable interest in their own right. The formal properties of the work become its true subject matter and thus a topic of primary interest. This explains, I think, why there is a tendency for each new work to have a system uniquely its own. This is perhaps overstated, yet if one thinks of a composer like Stockhausen, or even Ligeti or Xenakis, one sees that it touches upon an important facet of recent compositional thinking. The construction of the system has itself become an essential and inseparable component of the creative act. Robert Morgan, _On the Analysis of Recent Music_, 1977 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Pointers for auto-classification of sounds?
Well! An embarrassment of riches: enough for a graduate seminar at a non-free university... ;-) Thanks everyone who kindly contributed to this thread. There's a summer worth of insight and information here, so rather than make superficial comments, I'll return when I better understand my problem. Ross pretty much guessed my interest. Trying to see whether it's possible to automate the exploration of the parameter space of a synthesis algorithm. Imperative to something like that would be a procedure to analyze large quantities of sound data. Hence timbre classification. Of course, the process could be really simple: eliminating the settings that produced zero output, or immediately went into self-oscillation. But who knows where the line between too many to audition and convincing timbre classification lies? I did manage to find prior work in this area. Bill Tozier pointed to the Humies: http://www.genetic-programming.org/hc2011/combined.html One of the winners gave me the term EvoMusArt and some searching on SpringerLink led me to papers. I've collected some here if anyone's interested: http://vze26m98.net/music-dsp/ (I don't think any of these were behind a paywall.) I liked Colin Johnson's paper johnson-2006.pdf, though in general I found most of this work disappointing from a musical perspective. Best wishes and thanks again! Charles -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Pointers for auto-classification of sounds?
Another paper: Palle Dahlstedt's Creating and Exploring Huge Parameter Spaces: Interactive Evolution as a Tool for Sound Generation from the 2001 ICMC: http://quod.lib.umich.edu/i/icmc/bbp2372.2001.006/1?view=pdfsize=100 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Pointers for auto-classification of sounds?
Hahaha!! I had exactly the same experience in 2004 with a MSP patch I made. I finally generated one fantastic sound, and on the next compilation of my external, I couldn't recreate the sound with the same GP data. The MXO went the way of Metrowerks and the PPC... I do have a different thought now, and an interest to return to the process, but we'll see. On Mon, Jun 11, 2012 at 1:06 PM, douglas repetto doug...@music.columbia.edu wrote: I ended up with really simple things like is the signal non-zero? is there any variation in the amplitude? Yeah, these are pretty useful metrics, always worth keeping in mind. Best, Charles -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Pointers for auto-classification of sounds?
On Mon, Jun 11, 2012 at 2:32 PM, Phil Burk philb...@mobileer.com wrote: Wind was howling and rain was generating grains of sound before life even evolved. I agree with Phil here. I started with a single sine period and evolved stuff that would deform it: http://vze26m98.net/music-dsp/philthomson-01.png C. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Pointers for auto-classification of sounds?
On Mon, Jun 11, 2012 at 2:35 PM, douglas repetto doug...@music.columbia.edu wrote: Let's hear it! OK. Perhaps haven't stood the test of time as well as your sound, but it came out of nowhere, and I loved the appearance of the waveform: http://vze26m98.net/music-dsp/chp%20g%2002.png http://vze26m98.net/music-dsp/chp%20g%2002.aif and another: http://vze26m98.net/music-dsp/my0-1.png http://vze26m98.net/music-dsp/my0-1.wav -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Pointers for auto-classification of sounds?
On Fri, Jun 8, 2012 at 1:56 PM, Thomas Young thomas.yo...@rebellion.co.uk wrote: You haven't really explained which aspect of the timbre you want to use to organise the sounds Thanks. Very useful question. I'll pursue it. Bon ouikend! Charles -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
[music-dsp] [OT] Rebecca Kim contact info?
Hi all- It occurred to me that someone here might know how I can contact Rebecca Kim. I like to speak to her about the work she did on this presentation of _Bohor_, the electro-acoustic work by Xenakis: http://www.music.columbia.edu/masterpieces/notes/xenakis/index.html If anyone has any leads, please contact me offline. Thanks! Charles -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Boulez
On Feb 25, 2012, at 6:34 AM, Andy Farnell wrote: And whereas I do agree with Pierre Boulez here, maybe it is misguided to turn to reductionism and simplicity for their own sake. It may be equally hopeless to embark on a quest for authenticity this way. Hi Andy- I should apologize for hastily listing the publication date of the book. The book collects his Darmstadt lectures from 1954-56, so it comes from a much earlier time. I don't think Boulez would have changed his mind on things though. Sounds like you come from a much more Schaefferian era. Isn't the point not to take sides, but recognize the tension? Cultures that are busily exploring harmonic relations, haven't simultaneously plunged deep into the world of rhythm. Music is just too big a subject, and some of its properties exist in a dialectical relation to others. Although we all enjoy a sweet dessert, we don't put sugar in everything. (Unless you're the Nestle company!) My point was that the checkpoint raised by callbacks feeding a sample buffer may come from resistances outside the technical world. Boulez sees timbre as the enemy of harmony. Could very well be that the callback is the result of a cultural outlook, and not the result of engineering design… Best, Charles -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
[music-dsp] Bandlimiting, Aliasing and Reconstructed Signals
Happy Holidays Everyone! I wanted to ask a question provoked by reading the SuperCollider Users' mailing list, which had me thinking I didn't understand the underlying concepts that were being discussed. I went back and looked at stuff like Hamming's _Digital Filters_ and Oppenheim's _Signals and Systems_, although I'm greatly challenged by the math most of the time. Also, the discussions of sampling theory are just that: the authors are happy to shift back and forth from A-D and D-A to clarify a general theory, but as I'm interested in SYNTHESIS, taking away the finer points of signal reconstruction isn't always easy. Here's my limit case: let's assume some typical laptop with CD-quality sound generation capability with a sample rate of 44.1khz and sample size of 16 bits. I create a sinusoidal waveform on the computer with a period of 4,410hz. I choose to create this waveform by feeding 4,410 divisions of the unit circle into a sine function. In other words, I calculate a unique value for each sample of the period at the sample rate of the laptop's D-A converter. As this waveform is sent out the DAC, I assume it's subjected to a zero order hold of approximately 0.023 milliseconds. The DAC may also do it's own filtering of the signal before going out to a set of speakers. My questions are: 1) Is the synthesized signal aliased? If so, how can we anti-alias it? 2) Is the signal band-limited? If not, do we want it to be, and how do we do it? I'd also ask the same question about a similarly synthesized square wave. That may seems a bit simple, but there was the assertion on the SC-list that smoothing (I think that was the word) helped a loudspeaker figure out where it needed to be at a given point in time. I understand generally the point the poster was making, but isn't this a slippery slope? Not all speakers are designed with the same frequency response, so unless we tailor waveform synthesis to the specific characteristics of a loudspeaker, aren't we in danger of smoothing either too much or too little? Also, a square wave is a square wave: it has sharp transitions. What timbral or spectral components of a square wave are intrinsic to its waveform, and what is introduced by a particular DAC and speaker combination? Or in other words, is the acoustic result of a synthesized square wave its resultant output, or is it something that sounds good? Comments, links and/or laughter welcome. Thanks, Charles -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
