Maar

A complete M4L blockhead here, hi. I have this audio to midi in realtime -plugin, which works almost perfectly, but it won't play two times the same note without my playing another note in between. So my guess is that I'd need to insert some sort of note-off behavior / note repeat rule / re-triggering box somewhere in MaxForLive editing tree. I have Ableton 9 / Max 7. This is the only plugin that I have been able to (almost) use. I'm an acoustic musician and need this for live performances.

I've tried contacting the developer, but since it's a free plugin, seems like he doesn't (understandably) wish to waste his time on it. Looks like I'm on my own. As far as I can understand, it's a very cleverly made plugin, except for this bug. So I thought maybe it's ok to ask here?

Like I said, I know nnnothing. Yet I feel that a more experienced brain would solve this in a minute. Anyone feeling like helping me out?


audio-to-midi-plugin-won't-play-same-note-twice-in-a-row-help-to-edit-with-m4l

Audio to midi plugin won't play same note twice in a row. Help to edit with M4L?

Lilian Ilodigwe

i'm having trouble getting a jit.multiple and its related object to show up at the right time. i have the bang set up so that the model itself is visible / enabled at a certain point in the song, and disabled before and after that part - but when the toggle for said part is triggered, nothing shows up , despite "enable" being toggled on. its only after the entire song is done that , when i disconnect the wire that goes from jit.multiple to jit.gl.mesh and attach to the model itself, then reattach it back to jit.gl.mesh again, that the multiple objects show up. i've checked my jit.gen to see if that's causing it, and no matter what values i change or remove there, the jit.multiple just fails to render until after the song is done and i move the wires a bit. is there something else i'm missing? could the mainmetro bang from the world be causing it to be coloured over? i've attached the edited more concise patcher version of my project below.

edit: it requires the corresponding .wav file from the audio content browser provided in max.

swirlingMultiples.maxpat

i'm having trouble getting a jit.multiple and its related object to show up at the right time. 

jit-multiple-jit-gl-modelmesh-animation-with-song-timing

Hi Lilian. Since your jit.gl.model that is feeding the jit.gl.mesh is @automatic 0, you need to send it a bang to tell it to output its geometry matrices to the jit.gl.mesh for rendering. You can do this either manually with a bang button, or automatically with a loadbang object (or in response to some other trigger in your patch).

that helped to fix the issue; i set that value to that model for something else since i duplicated it, but i didn't know that was what was blocking the render. thank you so much!

jit.multiple + jit.gl.model/mesh animation with song timing

Zétény Nagy

hey all! i could use the collective wisdom of the C74 forum.

i am experimenting with building a wavetable synth, where a single cycle of a waveform (sampled from an old analog synth at 192 kHz 24bit) is loaded into [wave~], which is driven by [phasor~]. however, this adds some aliasing to the mix; i can see frequency bands moving downwards on the spectrogram when i increase the frequency. if i put the oscillator in a [poly~] with upsampling, and @resampling set to 1, i get even worse aliasing somehow.

what am i missing? can't figure it out, even though it seems super simple.

attached is an image comparing the spectra of [cycle~], a single cycle of a sine wave loaded into [wave~], and various combinations of the [poly~] upsampling.

Image 2023. 03. 17. at 21.47.JPG

hey all! i could use the collective wisdom of the C74 forum.
i am experimenting with building a wavetable synth, where a single cycle of a ... 

aliasing-issues-phasor~-driving-wave~-exhibits-horrible-aliasing

update: it is probably something with my original waveform sample, as addig a phase-inverted sine reveals some "grit" (which i'd like to keep in there, that's why i sampled the old synth in the first place). any ideas how to prevent the aliasing from happening?

upsampling alone does not cause bandlimiting, the upsampling only makes sense in conjunction with a lowpassfilter. 
i dont use it myself, but the built-in poly~ anti-aliasing filters are only related to the downsampling at the output of poly~, it can not know at what frequency you run wave~ at and it can not remove the HF content in the buffer.

to prevent the wave~ object from folding frequencies in the sample at nyquist you would have to bandlimit it before you play it pitched up. 

or in othe words: this is only possible in discrete time.

a labourios but effective method is to bandlimit the sample to nyquist, and one time to nyquist/2, and one time to nyquist/4 and so on... and then read from these different buffers for each octave.

alternatively you could already record an individual sample for each octave, dont filter them at all, and only pitch them downwards. (this will also capture a few other things more from the original instrument compard to using only one sample)


try all this using a bunch of proper cos~ running at different frequencies (2.5k, 5k, 10k, 20k), then you can hear the effect of aliasing - and your attempts to supress it - best.

Thanks for your help and the detailed answers, Roman!

I have checked out some more forum discussions regarding the topic and gathered some more information. (I have seen your name a lot in those threads, haha)

I have tried bandlimiting the sample with three methods; filtering frequencies above Nyquist using a brick-wall filter on the sample file itself, reducing the sample rate of the (already filtered) sample, and by "treating the symptom, not the cause": low-pass filtering the output inside of poly~ containing both phasor~ and wave~. The first two seem to be ineffective, the third option seems to help somewhat if I set the cutoff to 3 kHz-ish, which is uncomfortably low. Am I missing something here?

I have attached a patcher (MAIN_antialiasing_tester.maxpat) where you can try my wavetables and the methods I've tried so far, so you can see my results. Maybe you can spot what I'm not getting right. :)

As Roman says, upsampling using poly~ won't be very helpful here. If you are playing wavetable data back at different pitches, then you are moving through the wavetable's sample data at different speeds. This is a resampling problem, and the aliasing comes (in part) from the discrepancies between the current samplerate, and the rate at which you are moving through the wavetable data. 

There's a few ways to address this. One is to bake a set of waveforms, typically one per octave, each of which is already bandlimited for a certain playback rate, and blend between these according to the current desired pitch. This is like mipmapping in 3D graphics. You also need the math to identify which tables to blend between according to the current pitch, the table resolution, and the sample rate.  

The output can also  be affected by how you sample from the wavetable data itself -- i.e. what kind of interpolation you use. With no interpolation you will get another kind of aliasing. With other basic interpolations (e.g. linear, cubic) you may get an undesirable spectral footprint shaping to the waveform; using a better interpolator (e.g. sinc) with more data points (more samples from the wavetable) will result in a cleaner and more consistent waveform as pitch changes. 

There's a section in the gen~ book (https://cycling74.com/books/go) all about getting antialiased waveforms for arbitrary wavetable data (in 1D, 2D, and 3D), using both of the above methods.  The final example shows using a 16 point sinc interpolation for the interpolation filter, and a mipmap method blending between octaves that is dynamic, meaning that you don't have to pre-bake the octave tables at all.  

moving through the wavetable's sample data at different speeds. This is a resampling problem, and the aliasing comes (in part) from the discrepancies between the current samplerate, and the rate at which you are moving through the wavetable data.

Isn't the @resampling 1 attribute supposed to remedy this? I thought that's what it was designed for.  

I have tried baking a bandlimited waveform to test with, however, this seems to also exhibit some aliasing issues. Someone in the Max/MSP Facebook group pointed out that I might be missing a proper zero-crossing from one of the samples I have issues with (thanks Reaper), so I'll try padding it, to see if that helps. 
(edit: the lack of a proper zero-crossing was not the issue)

I'll try playing around with interpolation! Thanks for that idea!

I'll have to get that book soon! There seems to be a lot of good stuff in there.

Edit: If I do something crazy, like upsample the poly~ 32 times, put a low-pass filter in the poly~ on the output at SR*0.4, and use @resampling 1, the aliasing is a lot less noticeable. I know that this is a band-aid "solution" to the problem, just figured I'd try it for fun.

poly~ @resampling 1 will apply some antialiasing filters to suppress frequencies generated above the filtering frequency, in a kind of brute force way. It should work but if you have to upsample 64x then you have also effectively increased the CPU activity by 64x as well.    

What Roman & I are suggesting is different -- adjusting the bandlimiting & filtering to match the pitch at which you are playing the wavetable.  It is more complicated than just oversampling via poly~, but may be more accurate and efficient in the end.  

Also one last note on bandlimiting -- remember to use your ears more than your eyes! Even if you are seeing some aliasing visible in a spectrogram/sonogram, it doesn't mean it is audible -- and a little bit low amplitude aliasing up near 20kHz is not going to be a problem. 

in either cases consider what onepole~ does: it is only -6db per octave. so if you want to put it at 20kHz, you would proabably want 20 instances of it, not 1.  :)

(4-6 at 11kHz should be fine in practice.)


Aliasing issues - [phasor~] driving [wave~] exhibits unwanted aliasing

Diemo Schwarz

the subject and picture say it all...   nonsense output, but no error message

image.png

jit.buffer-from-char-bug.maxpat

the subject and picture say it all...   nonsense output, but no error message
 

 


bug-jit-buffer-from-char-matrix-produces-garbled-buffer-content

jit.buffer~ only supports float32 type. This is mentioned in the object's reference.

understood.  An error message would be nice, though, instead of nonsense output.

bug: jit.buffer from char matrix produces garbled buffer content

dnealelo

Hello Braintrust -- I keep getting this error popping up in my console when I'm running a patch with Mira in it:  

xebra_global: XEBListeningConnection_ListenProc: Error accepting connection: 670008 nodename nor servname provided, or not known

The connection appears to work fine, but the error pops up every several seconds, so it interferes with using my console to track output while I'm problem solving the patch.  I am running the latest versions of Max and Mira on OS X Yosemite 10.10.5 (it's a 2010 iMac, so I'm kind of done upgrading the OS) and iOS 9.3.5 on an iPad3.  

Can anyone direct me what to do to fix this error?


xebra-error-above-my-pay-grade

I have the same issue with Max 7.3.4 on my mid-2015 MBP, El Capitain, with iOS 11.1.2 on an iPad Air 2, running MIRA 1.2.1 - it's probably trivial, but it would be good to know...

Well, years and many Max versions later (currently 8.5.3) , I'm still getting this error: xebra_global: XEBListeningConnection_ListenProc: Error accepting connection: 670008 nodename nor servname provided, or not known

same :
xebra_global: XEBListeningConnection_ListenProc: Error accepting connection: 670008 nodename nor servname provided, or not known

Xebra error above my pay grade

Ru Gilfillan

I am using jit.world to simulate our solar system, so my gridshapes all have textures and there’s a cube map background. 

I have tried to give the planets streaks using both ‘erase color’ and adjusting the alpha, as well as jit.gl.pass smear.

Through trial and error I have achieved the effect I want, except the streaks don’t show when you first open the patch and switch on the world object. Once the patch is open, if you edit the jit.world object (without changing anything) causing it to reload, when it is switched back on the streaks are there.

Is there a way I can solve this so that the streaks appear when the patch is first opened?

jit-world-doesn’t-render-correctly-on-first-open-but-does-when-the-object-is-reloaded

a patch would help anyone to help here...

Jit.world doesn’t render correctly on first open, but does when the object is reloaded.

Sabina Grisi

I'm just trying to make a picture have a char of 30 30  using jit movie and jit.matrix. But of course the image by downloading it is really small. I know I can add another jit.matrix object with the dimensions I want, but I want the original ones for all the pictures. Not just the dims from one picture and then manually changing them after "reading" another one.

At the moment I have connected the jit.movie to a jit.matrixinfo and this one to a route dim. I've read that I can use jit.expr, and then connect that one to the last jit.matrix to change the char values, but I am stuck and I don't know how to proceed.

Hello there,
I'm just trying to make a picture have a char of 30 30  using jit movie and jit.matrix. 

trying-to-change-back-the-jit-matrix-to-the-original-dimention-of-a-jit-movie-after-changing-it-to-char-30-30

Ended up doing this for anyone who is interested:

{
	"boxes" : [ 		{
			"box" : 			{
				"maxclass" : "newobj",
				"text" : "receive original_dimensions",
				"id" : "obj-7",
				"patching_rect" : [ 441.0, 1113.983082234859467, 157.0, 22.0 ],
				"numinlets" : 0,
				"numoutlets" : 1,
				"outlettype" : [ "" ]
			}

}
, 		{
			"box" : 			{
				"maxclass" : "attrui",
				"fontname" : "Arial",
				"displaymode" : 3,
				"text_width" : 49.620026000000003,
				"fontface" : 0,
				"id" : "obj-8",
				"patching_rect" : [ 441.183670000000006, 1148.983082234859467, 134.0, 23.0 ],
				"numinlets" : 1,
				"fontsize" : 13.0,
				"numoutlets" : 1,
				"attr" : "dim",
				"parameter_enable" : 0,
				"outlettype" : [ "" ]
			}

}
, 		{
			"box" : 			{
				"maxclass" : "newobj",
				"text" : "jit.matrix 4 char 30 30",
				"id" : "obj-12",
				"patching_rect" : [ 363.0, 1196.552631497383118, 123.0, 22.0 ],
				"numinlets" : 1,
				"numoutlets" : 2,
				"outlettype" : [ "jit_matrix", "" ]
			}

}
, 		{
			"box" : 			{
				"maxclass" : "newobj",
				"text" : "send original_dimensions",
				"id" : "obj-60",
				"patching_rect" : [ 82.0, 343.0, 144.0, 22.0 ],
				"numinlets" : 1,
				"numoutlets" : 0
			}

}
, 		{
			"box" : 			{
				"maxclass" : "newobj",
				"text" : "route dim",
				"fontname" : "Arial",
				"id" : "obj-32",
				"patching_rect" : [ 82.0, 306.0, 63.0, 23.0 ],
				"numinlets" : 2,
				"fontsize" : 13.0,
				"numoutlets" : 2,
				"outlettype" : [ "", "" ]
			}

}
, 		{
			"box" : 			{
				"maxclass" : "newobj",
				"text" : "jit.matrixinfo",
				"id" : "obj-3",
				"patching_rect" : [ 82.0, 272.0, 73.0, 22.0 ],
				"numinlets" : 1,
				"numoutlets" : 1,
				"outlettype" : [ "" ]
			}

}
, 		{
			"box" : 			{
				"maxclass" : "jit.pwindow",
				"presentation_rect" : [ 639.0, 194.355420351028442, 369.0, 333.0 ],
				"presentation" : 1,
				"id" : "obj-41",
				"patching_rect" : [ 363.0, 652.0, 297.0, 320.0 ],
				"numinlets" : 1,
				"sync" : 1,
				"numoutlets" : 2,
				"outlettype" : [ "jit_matrix", "" ]
			}

}
, 		{
			"box" : 			{
				"maxclass" : "newobj",
				"text" : "jit.movie",
				"id" : "obj-17",
				"patching_rect" : [ 28.0, 215.0, 53.0, 22.0 ],
				"numinlets" : 1,
				"numoutlets" : 2,
				"outlettype" : [ "jit_matrix", "" ]
			}

}
 ],
	"lines" : [ 		{
			"patchline" : 			{
				"source" : [ "obj-8", 0 ],
				"destination" : [ "obj-12", 0 ],
				"midpoints" : [ 450.683670000000006, 1185.0, 372.5, 1185.0 ]
			}

}
, 		{
			"patchline" : 			{
				"source" : [ "obj-7", 0 ],
				"destination" : [ "obj-8", 0 ],
				"midpoints" : [ 450.5, 1138.983082234859467, 450.683670000000006, 1138.983082234859467 ]
			}

}
, 		{
			"patchline" : 			{
				"source" : [ "obj-41", 0 ],
				"destination" : [ "obj-12", 0 ],
				"midpoints" : [ 372.5, 975.0, 372.5, 975.0 ]
			}

}
, 		{
			"patchline" : 			{
				"source" : [ "obj-32", 0 ],
				"destination" : [ "obj-60", 0 ],
				"midpoints" : [ 91.5, 330.0, 91.5, 330.0 ]
			}

}
, 		{
			"patchline" : 			{
				"source" : [ "obj-3", 0 ],
				"destination" : [ "obj-32", 0 ],
				"midpoints" : [ 91.5, 297.0, 91.5, 297.0 ]
			}

}
, 		{
			"patchline" : 			{
				"source" : [ "obj-17", 0 ],
				"destination" : [ "obj-3", 0 ],
				"midpoints" : [ 37.5, 258.0, 91.5, 258.0 ],
				"order" : 2
			}

}
 ],
	"appversion" : 	{
		"major" : 8,
		"minor" : 5,
		"revision" : 3,
		"architecture" : "x64",
		"modernui" : 1
	}
,
	"classnamespace" : "box"
}

Trying to change back the jit.matrix to the original dimention of a jit.movie after changing it to char 30 30

Fred Voisin

Hello, 
I'm wondering how far in Max8.3 one can use the embed sqlite thru max JS. I encouter several issues when trying to use it :
- the sqlite's old doc  seems to be obsolete since the method 'open' makes Max8 crash as soon as it mentions the ram-based optional boolean arg ;
-after opening a file.db (with no error nor warning) , exec a statement such as "SELECT * FROM table;" returns the error: "sqlite_exec: 1 (no such table: atable)" - even if writing the full path "Macintosh HD:/Users/me/path/to/data.db" ;
- the example in the (old) tutorial using "sqlite.getversion()"  tells that sqlite.getversion is not a function ;
- and the statement sqlite.exec("SELECT sqlite_version() AS version;", res) returns an empty JS object with no error nor warning.

If older posts in the Forum mention these issues here and there, they still have no answer (see my previous post today).
I experienced that [node.script] object makes sqlite3 working, but I would prefer, if possible, to use only Max own ressources and use Max-JS instead of node.js.
Anyone have a recent experience with ?

Hello, 
I'm wondering how far in Max8.3 one can use the embed sqlite thru max JS. 

sqlite-use-from-max-js

Magic mushrooms or Setas Magicas have been and continue to be used in indigenous New World cultures in medicine, religious, divinatory, or spiritual. They may be depicted in Stone Age rock art in Africa and Europe, but are most famously represented in the Pre-Colum.
retiro psilocibina españa

Yes this tutorial is out of date and has some incorrect information.

The biggest issue I'm seeing is that it is no longer possible to open a database file, if that database doesn't already exist. The workaround would be to create an empty file, and then pass that file path to the database open message. You can easily do this with the "write" message to an empty text object.

The other issue is the tutorial uses a relative path for the database file, and I'm not sure if that ever worked. The fix here is to pass absolute paths (or supported Max paths like Desktop:/mydbfile).

(and yes, getversion is not a supported message)

We'll ticket these issues, but I would proceed with the workarounds for now. Fortunately, it looks like the rest of the database functionality is still working as expected. 

here's the modified tutorial patch (use with the existing tutorial JS file)


Thank's Rob for your help which makes my patch to work.
In effect, it seems that there is an issue with the form of the pathname, which could make Max crash if not in the good form and if the  ram-based arg of open method is 1 or 0 (but no crash if no ram-based arg).
Cheers.



Thanks so much. I have my answear, I resolve this issue 

sqlite use from max-JS

Jonathan Hallstrom

This is driving me crazy... I'm running Max 8.0.3 on a macPro (ashtray) under Mojave and am unable to get an input signal.    I've tried a simple patch that just has an ezadc~ going into a meter~ object - no signal.  I've checked Max's input routing and the audio Input is mapped to Channel 1. There's no problem getting input in LogicPro, WavePad and Zoom, and there's no problem with output from Max; it's just input. I've tried multiple audio interfaces and I've tried restarting my Mac and none of it helped.

Anyone have any idea what's up?  I've never had this problem before.

This is driving me crazy... I'm running Max 8.0.3 on a macPro (ashtray) under Mojave and am unable to get an input signal.    I've tried a s

no-input-signal-in-max-but-everywhere-else!

Double-check the choice of audio interface in 
Max -> Options -> Audio Status


First thing I did.  Core Audio on, audio device correct

Solved it!  In case anyone else encounters this:  In desperation, I tried reinstalling Max and when I launched the new copy, it gave me a message saying that the mic input had not been authorized under Security and Privacy/Privacy system settings.  How access could have gotten turned off for the prior install, I have no idea, but (re)enabling it solved the problem.

THANK YOU, Jonathan, for posting your solution. This was driving me crazy and I never would have figured out what the problem was.

No input signal in Max (but everywhere else!)

Eric T.

Hey everybody I'm somewhat new to M4L and I know that my Topic seems really dumb. You might be thinking "Just use the Right or Left signal coming from "Plugin~", but that just isn't going to work for me.

I'm turning one of my Effect Racks into a M4L device so I can make it take up less space, and possibly be lighter on CPU (idk if it will lmao).

Basically its a stereo device that takes a stereo signal, drops it down to mono, then inverts the right channel only, delays it by a few MS, then removes the mid channel. 

Signal -> Mono -> Invert Right -> Delay -> Side Only

But I cannot for the life of me get how to turn the mono signal I get from "+~" into a Left-Right pair. 

Literally all I need is a way to turn the mono signal coming out of "+~" into a Left-Right pair, but because the SEO of M4L documentation is completely garbage, I can't find any solutions. 

I'm sure there is a simple way to do this but I don't have a good grasp on M4L logic.

EDIT: TLDR: I'm trying to replicate an Audio Effect Rack with M4L to improve on efficiency.
Below is a diagram describing the Audio Effect Rack in question. 


--Guide--
============================================================================================
>[MSEQ(x)] - Linear Phase M/S EQ with an Brickwall HP at 22khs on the "mid" channel and a 
              Brickwall LP at 0hz on the "mid" channel, so only the Side channel remains, it
              has no mono image and nulls completely when I click the mono button on my 
              interface.
>[1-3ms delay] - This is a 0 Feedback Delay with a time range that only goes from 1ms to 3ms              
>[Big Mono L] - This is the sum of [Device In L] and [Device In R], but panned hard Left
>[Big Mono R] - This is the sum of [Device In L] and [Device In R], but panned hard Right
============================================================================================


--Effect Rack Diagram--
==========================================================================================================                  
[Device In L]→┐        ┌→[Big Mono L]→─────────────→[1-3ms delay]→┐                     ┌→[Device Out L]
              ├→[Mono]→┤                                           ╞═→[MSEQ(x)]═→[-6db]═→╡
[Device In R]→┘        └→[Big Mono R]→[Phase Invert]→[1-3ms delay]→┘                     └→[Device Out R]

[Device In L&R]→──→[Mono]→──→[Device Out L&R]                                                     
==========================================================================================================

Hey everybody I'm somewhat new to M4L and I know that my Topic seems really dumb. 

cant-separate-mono-signal-into-lr!-problem-is-not-as-simple-as-you-think!

your question about +~ to stereo pair is easilly answered
if one ignores that there is nothing stereophonic,
but simply a mono signal sent to 2 outputs.


If you post your patch with decriptions about what the goal is,
I am sure it would be easy to do.



i was about to post the same answer last night and then deleted it; the description of his DSP algo does not make any sense and  it is probably more a question about the usual ableton quirks in regard to "stereo channels".

Maybe it is simply bad explained, because language or whatever,
but as you say , this :
Signal -> Mono -> Invert Right -> Delay -> Side Only

can't exist, there is no Right after Mono conversion


the holy grail of side signals of a mono source.

Yeah I know its a weird issue. But I have a working version in an effect rack that works really well at stereoizing narrow signals. 

"can't exist, there is no Right after Mono conversion"

I disagree. For one, the effect rack that I use almost daily uses this algo and works as intended, so I know without a shadow of a doubt that this is possible. Secondly, There can still be a left and a right in a mono signal, so long as the mono signal is running in a stereo pair of channels, ie; mono in the sense that the two stereo channels are identical, moreso than there just being 1 channel. 

What I want is Signal A collapsed to mono, then I want that mono signal panned Hard Left and Hard Right, so that I can invert the phase of the Right side of Signal A. When the 2 Hard Panned elements recombine, It creates a signal that is entirely within the side channel, ie; there is no similarities between the left and right pair. 

In Ableton, If you take a signal, mono it, then you can still do things to the left and right channel separately, and that's what I'm doing with the effect rack I'm trying to convert into a M4L device. 

 I knew that people would misunderstand my problem, which is why I tried to clarify how non-elementary this issue actually is. 

All I need to know how to do is take a stereo pair from the "device in section", mono that stereo pair, then turn that mono signal into two identical signals that are panned hard left and right, THEN all I need to do is invert the phase of the Hard Panned Right channel.

I know that its possible, and as of right now I actually have it working, but I'm using a BEEP add-on thing to pan the mono signal left and right, and I'd really just like to have a simpler way to do it. 

I don't care what live routings into or out of racks or whatever is.
We talk plain signal paths.
You asked to mix stereo signal into mono, and simply route it to 2 destinations, inverting one of them.

if you mix them to mono later, you'll hear nothing.

Eric T. wrote: "There can still be a left and a right in a mono signal, so long as the mono signal is running in a stereo pair of channels"

Well, no. 
We have a slight difference in terminology. 
Let's find the same language in the audio dialect:

Mono is one channel. There is no left, right, mid or side.

Two indentical Monos in two channels we like to call Big Mono. Through loudspeakers this can sound stereo-like because of speaker differences and room resonances .

By duplicating a mono channel and manipulating one or both sides in a certain way, you can get Stereo. There are various methods for this. Stereo must have two channels in a relationship to each other that creates spatiality. Otherwise it is not called that.

After manipulating one or both channels to an extreme extend, you get Dual-Channel, which means two independent signals, one left, one right.


The whole audio world makes stereo out of mono with delay, phase, level and frequency. If I understand you right, you just have to do what Source Audio suggests. Duplicate the mono signal and manipulate each channel to your heart's content.

Anyway, maybe I'm missing something. I do not understand what this has to do with Ableton Live. It's just channels, right? I don't know Ableton, but assume it has channels and can split/duplicate by sends and busses and pan them. You don't need Max to do this, but it could make sense in your application to deliver the signals already prepared to the Ableton DAW.

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It looks as Eric tries to create stereo widener of some sort.
that is done by L+R & L-R mix with gain adjustments and
other toys being phase shifts, delays, crossovers etc
to make existing differences in stereo signal more prominent
or to fake mono signal into stereo.


But whatever one does, if one cares about mono compatibility, one has to insert that mono mix preview button somewhere, to avoid
stereo -> mono disaster at the end.


Hey Source Audio, This effect is totally mono compatible, because the processed chain runs through a "side only" effect before returning with the unprocessed mono signal. 

The effect rack version maintains a completely clean mono signal, and so does my current implementation of the M4L device.

Trust me, I'm quite a stickler for mono compatibility. And this plugin ensures a wide stereo image (more than 6db quieter than the mono signal ofc), that leaves a completely in-tact mono image no matter the processing choices. 

If you want, I can send you a video that shows that the effect rack leaves the mono image completely unaffected. There is no phase cancellation.

Hey Peter, Thanks for the info bomb. Yeah sorry I'm not an expert on all the nomenclature, but you really hit on what exactly I'm trying to do. "Big Mono" is the goal here; to take a stereo pair, make the stereo pair mono, and then use that mono signal to make a "Big Mono" pair, to which I invert the phase of one of those "Big Mono" signals, then recombine it with the original mono. The thing is, the two "Big Mono" signals have to be hard panned left and right. One signal hard panned left, and one signal hard panned right (The device also has a slider to preserve the sources side channel, and there is a delay, but I'm not worried about that in this context)

Precisely, I don't need max to do this. As I've said before, I already have a working version of this in an Ableton Audio Effect Rack, but I want to use Max because I want to customize the aesthetics of it, and there's a chance that its actually lighter on my CPU than the Audio Effect Rack.

Below is a poor diagram to describe what exactly I want to do. 
Again, all I want to do is mono a stereo pair, (turn 2 signals into one signal), then convert it *back* to a stereo pair of Identical signals, one panned hard left, one panned hard right. Thats it. 

The issue is that I cannot for the life of me find a way to simply make a "big mono" pair of identical signals panned hard left and right, AFTER making them mono.

And as I said previously, I do have the current version working using a BEEP addon that allows me to pan the single mono channel hard left and right, but I want a simpler implementation of it. 

For the device to work as I intend it to, I need the source signals to be collapsed to mono BEFORE any of the processing. 

Again though, thanks for the help, I hope the diagram helps elaborate exactly what I need help with. 


                    This is the step I'm having issues with
                       ↓
[Device In L]→┐        ┌→[Big Mono L]→─────────────→[Device Out L]
              ├→[Mono]→┤                             
[Device In R]→┘   │    └→[Big Mono R]→[Phase Invert]→[Device Out R]
                  └→─────────────→─────────────→──→[Device Out L&R]

In your scheme above, we don't care about <Device In L/R> because their sum is mono and ALL stereo information got lost (really!). Now we have a mono source and you wrote that correctly in your drawing. Then you duplicate this signal, pan the two hard left/right and invert one phase. This gives a phasing signal on two speakers and in an ideal world there would be silence when you combine the two equally to mono again.

Then you seem to mix the mono source to the half-inverted Big Mono pair. Maybe it's just running in parallel channels. However, the audible impression of <Big Mono L> gets louder because the source adds to it. The right channel changes the sound because it's invertion is partially compensated by the original mono signal. This method gives a tunable phasing effect. Technical shortcomings depending on the DAW can result in interesting side effects.

The result may be or may not be mono compatible, controlled by the level relationship. Depending on the levels, the soundfield can also tilt to one side (which could give better mono compatibility :-) and yes, here Max joins the game. It is relatively easy in Max to modulate one channel and automatically hold the balance if wanted. Would be tedious with DAW automation.

I would stay in Max for that and probably add some vst's here and there. If the mix in the other DAW doesn't quite match, you can always go back to Max and change a few numbers.

let´s leave aside suggestions how to change his DSP: connecting a one channel signal to a 2 channel output is done like shown in fig. 1 by source audio.

Hey Peter, thanks for the reply! A few things real quick:

1] "This gives a phasing signal on two speakers and in an ideal world there  would be silence when you combine the two equally to mono again." 

This is totally correct, but like I said, I'm actually delaying these signals (equally between the Big Mono L and R) by a few ms. (1ms to 3ms max), so there is a noticeable phasing when you change the delay amount, but somehow it only affects the side, and when I collapse down to mono, there is no phasing.  

2]"...<Big Mono L> gets louder because the source adds to it..."

Yeah I actually counter this with a few things. Firstly, in the Effect Rack that I'm basing this M4L device on: the Big Mono L&R actually go through a Linear Phase M/S EQ that removes all mids, therefore only the interactions between the Big Mono L and Inverted Big Mono R are being preserved. If I didn't run this Linear Phase M/S eq, then the un-effected Big Mono L would in fact add to the source signal, creating loudness issues like you said. Secondly, I'm reducing this chain by -6 db and delaying it by a very small amount. All these things work together to make a relatively "natural" Side Channel that doesn't have any effect on the mono image. 

The Linear M/S EQ, and Delay are removed from the diagram to remove unnecessary detail and focus on what my issue really is; separating the Big Mono L&R from the Sum of the incoming stereo pair, then inverting the R. 


Its surprising to say that this effect genuinely doesn't affect the mono image at all, and I can actually prove it from a null test (if you want, I could provide a link to a clip of me demonstrating this).

Let me elaborate:

Lets say we have incoming stereo signals XL and XR
If we run XL and XR into my effect, we can call the out coming stereo pair YL and YR
If we sum XL and XR we get the Big Mono signal I'll call XM
If we do the same to YL and YR, we'll get the Big Mono signal I'll call YM
If you were to invert the phase of XM and sum it with YM, You would get a -infinity null
Therefore XM and YM are mathematically indistinguishable, despite the fact that YL and YR are not identical to their corrosponding pairs XL and XR.

And its because of this fact that I use it so frequently on narrow or mono signals, to give me a quick way to stereoize things without causing mono compatibility issues. But because I use it so frequently, and because it uses 3rd party VST's that arent optimized for my particular use case, I thought it would be best to hand-build a device so I could optimize it. 

BTW: I really just wanna say thank you for your help this whole time, its been a huge help.

Hey Roman, I really appreciate the input, but I fear that you might not really be understanding what exactly I'm doing. Everything you're saying is 100% correct, but I think you misunderstand what exactly I need to do. 

I understand that summing the incoming stereo pair, and connecting the sum to the two outputs will create a big mono signal where there is two identical signals playing on the left and right channel. But if you take a look at my diagrams, there's actually a few more steps involved. 

I'm going to put a more detailed version of the Effect Rack  that I'm trying to rebuild in M4L, as to alleviate confusion. 


--Guide--
============================================================================================
>[MSEQ(x)] - Linear Phase M/S EQ with an Brickwall HP at 22khs on the "mid" channel and a 
              Brickwall LP at 0hz on the "mid" channel, so only the Side channel remains, it
              has no mono image and nulls completely when I click the mono button on my 
              interface.
>[1-3ms delay] - This is a 0 Feedback Delay with a time range that only goes from 1ms to 3ms              
>[Big Mono L] - This is the sum of [Device In L] and [Device In R], but panned hard Left
>[Big Mono R] - This is the sum of [Device In L] and [Device In R], but panned hard Right
============================================================================================


--Effect Rack Diagram--
==========================================================================================================                  
[Device In L]→┐        ┌→[Big Mono L]→─────────────→[1-3ms delay]→┐                     ┌→[Device Out L]
              ├→[Mono]→┤                                           ╞═→[MSEQ(x)]═→[-6db]═→╡
[Device In R]→┘        └→[Big Mono R]→[Phase Invert]→[1-3ms delay]→┘                     └→[Device Out R]

[Device In L&R]→──→[Mono]→──→[Device Out L&R]                                                     
==========================================================================================================


...and what happens if you connect [*~ 0.5] to both inlets of [plugout~] ? what does not work with it?

Well Roman, what does not work is that it doesn't do what I'm trying to do. 
What you're describing is not what I'm describing. 

Sum [plugin L~] and [plugin R~], lets call this "Signal A"

Take "Signal A" and hard pan it Left, (Which we will call "Signal A_L")

Take "Signal A" and hard pan it Right, (Which we will call "Signal A_R")

Then we delay both "Signal A_R" and "Signal A_L" by 1 to 3ms

Then we run both "Signal A_R" and "Signal A_L" through a Linear Phase M/S EQ that removes the mid channel. We can call the outputs of this EQ;  "Signal B_L" and "Signal B_R"

"Signal B_L" and "Signal B_R" are strictly different, if we were to sum these two channels, they would null completely because they do not have a mono image. 

Then we reduce "Signal B_L" and "Signal B_R" by 6db and send both to [plugout L~] and [plugout R~] respectively

We also route "Signal A" directly to both [plugout L~] and [plugout R~]


Connecting [*~ 0.5] to both inlets of [plugout~] is just going to halve the volume of any signal reaching [plugout L~] and [plugout R~], but this is unnecessary because the Pair "Signal B_L" and "Signal B_R" are equally loud as [plugin L~] and [plugin R~] because my effect creates a side channel that is almost identical to the mid channel without phase cancellation, so all I need to do is turn  "Signal B_L" and "Signal B_R" down by 6db. 

This does result in an increase in gain if the source signal is mono, but this is because my effect leaves the mono signal completely un-effected, and only ADDS a side channel that fits it. Therefore since all I'm doing is ADDING a side image to the stereo pair, it makes sense that the gain will increase. 


"we don't care about <Device In L/R> because their sum is mono and ALL stereo information got lost (really!)"

This isn't necessarily true in my case because of the diagram I've provided. I run these two channels through a Linear Phase M/S EQ that removes the Mid channel, therefore, if I don't hard pan the Sum of [plugin L~] and [plugin R~], I will get no audio out of the Linear M/S EQ because the resulting image will actually just be a mono "print", (for lack of a better term), of the side channel. 

I need to pan the two elements before the Linear M/S EQ so that I can remove the Mid channel, so that the two output signals of the EQ will only contain "side" information, thus leaving the "Mid" pass-through channel entirely unnefected. (Which is the whole point of me making this effect).

I'm going to go ahead and repost the diagram just so its extra clear what exactly I'm trying to recreate in M4L.

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