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ThomasA

I know we can access soundfonts (.sf2 files) in Max through the vst~ object. But is there anybody that has worked more "directly" with a soundfont? For example built a soundfont player from scratch in Max? Is such a thing at all possible?

Hi!
I know we can access soundfonts (.sf2 files) in Max through the vst~ object. But is there anybody that has worked more "directly" with a soundfont? For example built a soundfont player from scratch in Max? Is such a thing at all possible?

Hi!
I know we can access soundfonts (.sf2 files) in Max through the vst~ object. But is there anybody that has worked more "directly" with a

using-soundfonts-in-maxmsp

There is the fluidsynth~ object.
Check this thread.

Yes, I've seen that.
Unfortunately as far as I can see only 64-bit on Mac. I'm on 64-bit Windows.

WOW!
Installed Max 7 32-bit and tried out the fluidsynth~ object. It can do exactly what I was after – namely microtuning a soundfont.

Too bad fluidsynth~ is not available as 64-bit. I really do not want to work in 32-bit Max. Anybody that has an idea how to get a 64-bit version?


vst~ object and your favorite VST plugin capable to play soundfonts?

ah.... sorry... didn’t noticed, you dont want VST

No, can not microtune individual notes through vst~ ... as far as I can understand at least.


https://github.com/FluidSynth/fluidsynth/releases/tag/v2.2.1 - but, please correct me if I' wrong, there is already a 64bit Windows available...

https://cycling74.com/forums/the-inevitable-drive-to-64bit-litter-power-and-fluidsynth 

hilarious, ircam even blocks archive.org. but volkers link works.

That's all well and good, but can I make this work with Max 8 on a PC with Windows 10 and 64-bit? If so please show me how to install what and where.

The version of fluidsynth~ for Max 7, 32 bit is exactly what I need.  Could you tell me how/from where you obtained this version?

Thank you very much, Source Audio.  I look forward to seeing how this works! :-)

Using soundfonts in MaxMSP?

billyanok

Is there any way to rename the max console to "my app console" for a standalone app?

Is there any way to rename the max console to "my app console" for a standalone app?
Thx, 
Bill

rename-max-console

Max Console is in the executable code .
Rename it but keep number of letters.
For menu item Max Console, edit maxinterface.json.
Similar in windows version


Hello Source Audio,
"Max Console is in the executable code . Rename it but keep number of letters."
In the .mxf file?
Thx, B

Executable file on mac is in app/Contents/MacOS


Just stumbled on this old thread, valuable info that I never knew, thanks @Source Audio!

You are welcome !
I guess you want to customise own apps as much as possible...
I allways decouple location of Settings file on Windows, because they usually land in 
/user/appData/Roaming/Cycling '74/my-app
One can replace Cycling '74 (11 chars) using own "company" id
from version to version replacement  is located differently, but searching for
hex 
43 00 79 00 63 00 6C 00 69 00 6E 00 67 00 20 00 27 00 37 00 34

will get one to right location (one with P.r.e.f.e.r.e.n.c.e.s)   (or S.e.t.t.i.n.g.s  in Max 8)

Name of the preferences folder itself should be left as is, that's job of standalone inspector


I mod the apps using mac, but any editor like HxD would work too on windows
One reason to do so ... I often prepare settings for my apps and 
extract them to my location, if cycling one were used, my installer could overwrite
settings of other apps, like max itself.



Source Audio,
Doesn't appear to work in 8..x executable files?

Max Console doesn't seem to show up in search field?
I'm sure I'm missing something, maybe the Universal Binary?
Thx, 
B


Source Audio, 
I used -
https://hexed.it/
It did the trick.
Thx, 
B

Rename Max console?

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

leafcutter

LeafyTracker - An open, versatile 10 lane tracker / data sequencer with an elegant and user friendly interface by Leafcutter John. Featuring:

Comprehensive examples in the  'Tracker V1 8 5' patch.

Multiple methods of note and chord creation.

Various functions to manipulate columns & rows.

Conditional data (can be turned on and off).

Cut, copy, paste, replace, cut & shift, paste & shift, etc...

All data can be easily saved and recalled.

Rain simulator and Drum machine examples.

Download here:  https://tinyurl.com/ybulhznr 

 LeafyTracker - An open, versatile 10 lane tracker / data sequencer with an elegant and user friendly interface by Leafcutter John. Featurin

-sharing-leafytracker-an-open-versatile-10-lane-tracker-data-sequencer

Nice one! 
I haven't tried it yet, but I always enjoy your work and learn huge amounts by taking your patches apart and poking around inside. 
Many thanks.

Good luck inside this one - Let me know how you get along!

I am blown away, this is amazing work J, great stuff!

This is pretty exciting stuff! I've dreamt of having a basic tracker in Max - it really ought to be a built in object, kinda like the offspring of [coll] and [detonate] ;P

Unfortunately, I've had a few issues that are making it hard to use:
- occasionally the GUI is totally blank, and i have to reload the patch
- there's some bugginess when linking with [textedit], where either that object or the tracker doesn't display data as its being typed
- sometimes the output lags behind the row marker
Perhaps it's just my laptop though, 2011 MBP (maxed out).

Feature requests:
- customize colors
- horizontal markers, a la [live.grid]

MaxMSP Forum