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Roman Thilenius

imagine a circular speaker setup with N speakers (for example 10)

now we are using a form of vbap to pan our audiomaterial.

mono source(s) -> vpab-like weighting/panning -> 10 channel format

while this works well for "normal" mixing approaches (say like controlling the azimuth using a computer mouse) we start to have a terrible problem as soon as we try to modulate the panning at high speeds.

(one which is not at all releated to the technology used, but who knows, maybe the solution is releated?)

lets say your material is a 40Hz sinewave and you rotate the virtual sound source with 10 Hz around your head.

it is clear what happens: every of the 10 speakers does now not even perform half a sine wave, but only small fragments of them.

while on the digital layer this is not a problem at all (you can distribute 1 signal on any number of channels, as fast as up to a per-sample speed modulation, then sum them together and get exactly the original again), but with reinforcement there are at least two problems which distort the audio the higher the modulation speed is:

2. the "fade out" of the amplitude modulation of speaker #7 is not exactly the opposite of the "fade in" of speaker #8 - you will hear a click, just as you do when you have a fast attack on a bass synth.

a very basic solution would be to do what they do in prologic formats: exclude the bass completely from panning and leave it mono.

but this only works well until a certain modulation speed ist reached - the plan is to modulate up to nyqist (speed of nyquist-per-360 degree, that is)

yes, one could modulate the crossover frequency according to the panner modulation speed. but then you quickly hit the border of how fast the filter can be modulated. :)


imagine a circular speaker setup with N speakers (for example 10)

now we are using a form of vbap to pan our audiomaterial.

mono source(s) -> vpab-like weighting/panning -> 10 channel format

while this works well for "normal" mixing approaches (say like controlling the azimuth using a computer mouse) we start to have a terrible problem as soon as we try to modulate the panning at high speeds.

(one which is not at all releated to the technology used, but who knows, maybe the solution is releated?)

lets say your material is a 40Hz sinewave and you rotate the virtual sound source with 10 Hz around your head.

it is clear what happens: every of the 10 speakers does now not even perform half a sine wave, but only small fragments of them.

while on the digital layer this is not a problem at all (you can distribute 1 signal on any number of channels, as fast as up to a per-sample speed modulation, then sum them together and get exactly the original again), but with reinforcement there are at least two problems which distort the audio the higher the modulation speed is:

1. there is no perfect hotspot

2. the "fade out" of the amplitude modulation of speaker #7 is not exactly the opposite of the "fade in" of speaker #8 - you will hear a click, just as you do when you have  a fast attack on a bass synth.

a very basic solution would be to do what they do in prologic formats: exclude the bass completely from panning and leave it mono.

but this only works well until a certain modulation speed ist reached - the plan is to modulate up to nyqist (speed of nyquist-per-360 degree, that is)

yes, one could modulate the crossover frequency according to the panner modulation speed. but then you quickly hit the border of how fast the filter can be modulated. :)

are there any other approaches to this?



imagine a circular speaker setup with N speakers (for example 10)

now we are using a form of vbap to pan our audiomaterial.

mono source(s

once-more-surroundvbap-for-synthesis

Hi Roman. Maybe a mechanical thing: the heavy paper of a bass loudspeaker needs time to return to a quiet state, before the next impulse. Thinking to micro-doppler, i.e. same level of 40Hz in every loudspeaker, modulating in some ms delay, this would create highest and lowest frequencies ? I dydn't test that, i shall try... Maybe it makes a real nice big drum ? I don't expect from a loudspeaker a big fidelity, it can make nice matieres.
curious
 michel z



it´s even worse. 

the AM distortion in the LF content equals normal HF content already on the digital layer, which will be accurately reproduced by the treble and HF speakers later.

even if you have 2 speakers with 10 centimeters distance to each other and 5 meters distance to the listener, the antiphase version of the squared LF content on the second speaker will not kill the sound from the first one.

the usual industry standards use a 120 hz 72 db filter and then recommend you to modulate the amplitude not faster than 16 Hz. but i am going much higher than that. (see patch. it is a bit less a problem when you have a broad vector like 7 out of 10 speakers - in this stereo example here it is on the extreme level.)

yes, or you leave it as is and be happy about the special effect it creates. :)

#P window setfont "Sans Serif" 9.;
#P window linecount 1;
#P comment 304 545 347 9109513 how to make it dynamic?;
#P comment 444 71 169 9109513 arbitrary panning modulation;
#P comment 231 97 64 9109513 music;
#P window linecount 2;
#P comment 304 489 347 9109513 traditional method: create LFE channel \, 72 db crossover \, 120Hz. then dont use panning / amplitude modulations over ~16 Hz;
#P window linecount 1;
#P comment 297 466 212 9109513 but speakers? the AM leads to distortion.;
#P newex 377 154 43 9109513 *~ 0.5;
#P newex 382 124 43 9109513 +~ 1.;
#P user scope~ 358 245 488 375 64 3 16 -1. 1. 0 0. 0 0. 255 255 255 85 85 85 0;
#P user scope~ 8 180 138 310 64 3 16 -1. 1. 0 0. 0 0. 255 255 255 85 85 85 0;
#P newex 373 183 43 9109513 !-~ 1.;
#P flonum 430 44 57 9 0.13 2.01 3 139 0 0 0 221 221 221 222 222 222 0 0 0;
#P newex 389 94 82 9109513 cycle~ 0.13;
#P user scope~ 155 486 285 616 64 3 16 -1. 1. 0 0. 0 0. 255 255 255 85 85 85 0;
#P user scope~ 155 357 285 487 64 3 16 -1. 1. 0 0. 0 0. 255 255 255 85 85 85 0;
#P toggle 110 36 15 0;
#P newex 310 208 43 9109513 *~ 1.;
#P newex 220 207 43 9109513 *~ 1.;
#P newex 77 65 82 9109513 dac~;
#P newex 235 125 82 9109513 cycle~ 1.;
#P comment 372 389 164 9109513 digital summing = no problem;
#P connect 9 0 8 0;
#P connect 8 0 13 0;
#P connect 13 0 14 0;
#P connect 14 0 3 1;
#P connect 14 0 10 0;
#P connect 3 0 6 0;
#P connect 3 0 12 0;
#P connect 4 0 7 0;
#P connect 4 0 12 0;
#P connect 10 0 4 1;
#P connect 1 0 11 0;
#P connect 1 0 3 0;
#P connect 1 0 4 0;
#P connect 5 0 2 0;
#P window clipboard copycount 20;

once more : surround/vbap for synthesis

chapelier fou

I'm curious to try gamepads with Max/MSP 8.6.0, seems to be fun. I'm not a gamer at all so I literally don't know anything about these controllers. Are they those used for the PS4 ? Does the [gamepad] object only work via USB connection, or is it ok with Bluetooth ? Any recommendation ? Thanks ! 

Hi there, 

I'm curious to try gamepads with Max/MSP 8.6.0, seems to be fun. I'm not a gamer at all so I literally don't know anything...

gamepads

Gamepads ?

we probably all know the 25 years old ircam vowels table, but i am in need of more and different ones. does anyone know of a data set which is different or eventually also more complete than the ircam one?

for example i am currently using the ircam tenor data for the systhesis of a falsett voice - but i dont really like the result. i would also like to have different sets for "male", "female" and "children". the tenor and mezzosopran range, for example, applies in real life for both genders.

the ircam data is probably already made from averaging a bunch of different recordings. but the more data i had, the more it would be possible to create an artificial ruleset of how different voices sound like and how the relate to each other (so that you can describe transitions between them)

if you know what a colorspace is - i am looking to create a vocal formant space...

the same request i had for resonance models. it seems impossible to find some onb the net.

what i would like to have is something like 10 or so different acoustic guitar body resonance models and stuff. i know that i could start using impulse-response sample collections of such recordings and process resonance models from there, but this is a lot of work and the time value will still be a matter of trial and error (with no reference recording to check against)

we probably all know the 25 years old ircam vowels table, but i am in need of more and different ones. does anyone know of a data set which is different or eventually also more complete than the ircam one? 

the same request i had for resonance models.  it seems impossible to find some onb the net.

we probably all know the 25 years old ircam vowels table, but i am in need of more and different ones. does anyone know of a data set which 

where-to-find-formant-tables-and-resonance-models

btw, what about consonants? :) ssss-shhhh-hhhh ?

in an hypothetic, ideal case, the "vocal format space" would be multidimesional:

 - maybe even registers? (those could be created artificially maybe)

i almost said "black, white" but this is probably bullshit from a technical standpoint.

ok i guess we can file this under "question too complicated".

I guess so... though I'd be interested to know more about this. 

The only lead I have is from the book Designing Sound by Andy Farnell - here he builds a simple vocal tract model in PD using a list of vowel formants. I think I convereted it to Max at some stage and posted it here, though I now can't seem to find it.

I'm sure I could dig it out if you're interested, but the data was limited (similar to this - if that's the Ircam one you refer to). However, he references The spectral database at Center for Spoken Language Understanding, Oregon University - compiled by Tim Carmell. Perhaps a lead?

i´ve seen this before but didnt use it. it adds 3 additional vocals to what ircam offers - but otoh it lacks gain coefficients.

of course you can use your ears and match these 2 databases together somehow ... :)

Heads up for necroposting!
Not for exactly spoken sounds, but there’s a Diphone Studio in the IRCAM forum software set that can create resonant models from arbitrary sounds. I used them with the CNMAT’s [resonators~] object.


Jumping in years later while doing some grad research on formants for Max/PD programming (we're using the Farnell book as well).

Not sure if this is what was requested but I found these pages with good info & some formant plots over vocal ranges that seemed decent and may give more data options to work with:

https://www.classes.cs.uchicago.edu/archive/1999/spring/CS295/Computing_Resources/Csound/CsManual3.48b1.HTML/Appendices/table3.html

https://home.cc.umanitoba.ca/~krussll/phonetics/acoustic/formants.html

http://ec-concord.ied.edu.hk/phonetics_and_phonology/wordpress/learning_website/chapter_2_vowels_new.htm

Another little bump.
I found this funny/simple/rather realistic/very illustrative voice generator called Pink trombone. It's implemented in javascript and if you reveal the page' source code you can see the whole stuff, well organized, with original scientific publication sources. And it's kinda CC-BY licensed.




i ended up mixing the two existing data tables back in the days by freely (i.e. by ear) interpolating the data.

of course for the three given applications i drive with it (biquad, reson, and paf) different settings are needed in order to get a nice result. so now i have a "formant lab" which only uses the table data as start preset, then lets me create new filters.

however, human voice is interesting, but not all.

ready-made tables for musical instruments would also be great to have.

(my gamma version of diphone for OS9 is somehow out of order)


where to find formant tables and resonance models?

Dekkuran

I have a dictionary of adjectives in a .txt file and I am trying to read them into my Max patch and then display one at random but I haven't been able to find a way to do this in max. Is it possible? If so, how is it done? Thanks in advance!

how-to-get-max-to-read-in-a-txt-file-and-display-the-result

Here's an example copied from the helpfile:


Hi 
Had you find the solution of your question 
If yes please share that with me 
Thanks in advance,

Did you look at the [text] help file as LSKA recommended? A "read" message into a [text] object will open a text file in Max.

How to get Max to read in a .txt file and display the result?

lysdexic

Adding all links to examples using Google Mediapipe in Max here. Will update these as the collection grows. [updated 2023]

https://github.com/robtherich/jweb-mediapipe

https://github.com/lysdexic-audio/jweb-pose-landmarker

https://github.com/lysdexic-audio/jweb-hands-landmarker

https://github.com/lysdexic-audio/jweb-hands-gesture-recognizer

https://github.com/lysdexic-audio/jweb-facemesh

https://github.com/lysdexic-audio/jweb-face-landmarker

https://github.com/lysdexic-audio/jweb-object-detection

https://github.com/lysdexic-audio/n4m-handpose 

https://github.com/lysdexic-audio/n4m-facemesh

MaxMSP Forum