Re: [MMM] Open-source/free production tool for micro and macro-tonal consonance

Carl Lumma <ekin@...> wrote: Hi,

>Sounds interesting, but it looks like there are no docs at all
>available for this. What are the command line parameters?
Good point. There is no "command line" so to speak, even though it is a command prompt program. Run it and it will ask you how you want to process the file: select (2), it will take a while to process before it gives a list of processing info. After it does that type a number and press enter and the program will be finished.

I will release full documentation on a web site in the near future (within the next month).

> Just use it on any individual instrument tracks
>Must they be monophonic?

Actually they must be stereo: the program exploits binaural properties of hearing to increase consonance and uses stereo input and output files.

> Take, for example, a wave file guitar track in 19tet and a piano
>track in mean-tone.
> Run them both through the program individually and mix them
>together (for example, with Audacity open-source mixing software).
> Enter "2" (production mode) at the "encode, decode, or production
>mode" prompt.
> The result .wav sound files will be consonant since this is since
>the software aligns the sound to a common harmonic schema.

>Uh... ok. Do you have a real description of what you're doing
>somewhere?
______________________________________________

What I'm doing...sorry about this I will try and restate:
Piano track (stereo format, meantone wave file) = 1.wav
Guitar track (stereo format, 19Tet wave file) = 2.wav
1. Rename 1.wav copyray.wav in the same directory as the program.
2. Run the program using option 2 (the program will ask you to select an option after about 30 seconds initialization/start-up time).
3. Rename the processed/output file from "newcleaned.wav" to "11.wav"

4. Rename 2.wav copyray.wav in the same directory as the program.
5. Run the program again using option 2 (the program will ask you to select an option after about 30 seconds initialization/start-up time).
6. Rename the processed/output file from "newcleaned.wav" to "12.wav"
7. Mix the two processed wave tracks "11.wav" and "12.wav" files any way you please ______________________________________________
The "basic" reason it works is it categorizes each instrument part's base harmonic and overtones and retunes the overtones to match a fairly flexible structure based on tonotopic theory (as opposed to "auto-tuning", which forces base harmonics , and not overtones, to match an unflexible, often emotionally compromised scale).

The reason why recording in different tunings can work together is each peak
harmonic/overtone is pushed up or down in frequency to align with the schema. So, for example, amplitude peaks in the guitar track at 956hz and the piano track at 1023hz may both be added to produce a peak at 1010hz, the nearest point in the schema and their phases rotated accordingly to maintain most of the original track's non-noisy brightness.

I wish I had a research paper to show for this.

But, truth is, I'm no PHD and there is a good chance several of you could explain the nuts and bolts of the elements of the theories I use better than I can. I'm simply a hobbyist trying to take combine the best of several non-patented ideas I've discovered online into a single program.

However, some related ideas/theories to this program are tonotopic theory, beating, phase vocoders, dissonance equations, binaural vs. monoaural dissonance perception, and pitch-shifting. Searching for papers on those can hopefully help give you more details, many of my ideas also are related to insights of DSP expert and professor Bill Sethares.

However, in the same way synthesizers are often best used by playing them and not necessarily knowing every mathematical feature of every LFO,PWM, compressor, etc. the best way to take advantage of this program is to run it and see (hear) the results.
Hope this helps, let me know if you still have any questions and thank you for your interest and suggestions.

-Michael

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[Non-text portions of this message have been removed]

>>Sounds interesting, but it looks like there are no docs at all
>>available for this. What are the command line parameters?
>
>Good point. There is no "command line" so to speak, even though it is
>a command prompt program. Run it and it will ask you how you want to
>process the file: select (2), it will take a while to process before
>it gives a list of processing info. After it does that type a number
>and press enter and the program will be finished.
>
>I will release full documentation on a web site in the near future
>(within the next month).

I guess I should say, Thanks for the early notice!

For me personally, I am happy to wait until the docs arrive (I am
already curious to find out what "1" does...).

>> Just use it on any individual instrument tracks
>>Must they be monophonic?
>
> Actually they must be stereo: the program exploits binaural
>properties of hearing to increase consonance and uses stereo input and
>output files.

Hmm, interesting. I wonder how that would work?

> > Take, for example, a wave file guitar track in 19tet and a piano
> >track in mean-tone.
> > Run them both through the program individually and mix them
> >together (for example, with Audacity open-source mixing software).
> > Enter "2" (production mode) at the "encode, decode, or production
> >mode" prompt.
> > The result .wav sound files will be consonant since this is since
> >the software aligns the sound to a common harmonic schema.
>
>>Uh... ok. Do you have a real description of what you're doing
> >somewhere?
> ______________________________________________
>
>What I'm doing...sorry about this I will try and restate:
>Piano track (stereo format, meantone wave file) = 1.wav
>Guitar track (stereo format, 19Tet wave file) = 2.wav
>1. Rename 1.wav copyray.wav in the same directory as the program.
>2. Run the program using option 2 (the program will ask you to select
>an option after about 30 seconds initialization/start-up time).
>3. Rename the processed/output file from "newcleaned.wav" to "11.wav"
>4. Rename 2.wav copyray.wav in the same directory as the program.
> 5. Run the program again using option 2 (the program will ask you to
>select an option after about 30 seconds initialization/start-up time).
>6. Rename the processed/output file from "newcleaned.wav" to "12.wav"
>7. Mix the two processed wave tracks "11.wav" and "12.wav" files any
>way you please ______________________________________________
>The "basic" reason it works is it categorizes each instrument part's
>base harmonic and overtones and retunes the overtones to match a
>fairly flexible structure based on tonotopic theory (as opposed to
>"auto-tuning", which forces base harmonics, and not overtones, to
>match an unflexible, often emotionally compromised scale).

Sounds like some form of adaptive tuning. Are you familiar with
Bill Sethares' stuff (music/book)?

My only concern is resynthesis errors -- or are you just pitch-
shifting bands of audio or something?

And where does the binaural part come in?

Just put your effort into the docs if you prefer -- I can wait to
have my questions answered there if it's better for you.

>The reason why recording in different tunings can work together is each peak
>harmonic/overtone is pushed up or down in frequency to align with the
>schema. So, for example, amplitude peaks in the guitar track at 956hz
>and the piano track at 1023hz may both be added to produce a peak at
>1010hz, the nearest point in the schema and their phases rotated
>accordingly to maintain most of the original track's non-noisy brightness.

Hmm... added...

>But, truth is, I'm no PHD and there is a good chance several of you
>could explain the nuts and bolts of the elements of the theories I use
>better than I can. I'm simply a hobbyist trying to take combine the
>best of several non-patented ideas I've discovered online into a
>single program.

Cool!

>many of my ideas also are related to insights of DSP expert
>and professor Bill Sethares.

Oh, disregard the above. And sorry for not reading before replying.

> However, in the same way synthesizers are often best used by
>playing them and not necessarily knowing every mathematical feature of
>every LFO,PWM, compressor, etc. the best way to take advantage of
>this program is to run it and see (hear) the results.
>Hope this helps, let me know if you still have any questions and thank
>you for your interest and suggestions.
>
>-Michael

Thanks, I'm looking forward to it!

-Carl