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Simon Dawes

Am establishing communication between Max and a BLE-enabled ESP32-C3 with the following steps on a MacBook Pro.

Looking to automate everything as much as possible, though.

Is there any way to connect Max and the ESP32-C3 without having to use Audio Midi Setup (or any outside) application?

Hi all,

Am establishing communication between Max and a BLE-enabled ESP32-C3 with the following steps on a MacBook Pro.

Launching MIDI...

bluetoothble-can-max-connect-to-ble-devices-without-using-audio-midi-setup-mb

Bluetooth/BLE: can Max connect to BLE device without using Audio MIDI Setup (MBP)?

Jonatan Ewald

I just saw the release of the new iPad pro.

I guess many generations of iPad has been more powerful than the first computers Max MSP was run on.

Is there any development ongoing for launching a full version of Max for the iPad?

I like Mira a lot, but you cannot use java scripted graphics so GUI wise it is a bit limited and the big con is of course that if you want to bring your patch live, you need both a computer and an iPad (or just your computer).

I think it would be super nice to be able to run a full Max-patch as a dedicated app on an iPad. I wouldn't mind if development of the app was on my PC as long as it could run completely standalone with the full feature set including externals and java snippets on the iPad.

Is there any development ongoing for this?

I just saw the release of the new iPad pro.
https://www.apple.com/ipad-pro/ 
I guess many generations of iPad has been more powerful than th

full-version-of-max-for-ipad

Full version of Max for iPad?

Messinki

I'm using a dict object to store data about midi notes as an array of dictionaries:

            "Velocity" : 100,

            "Start" : 0.330751,

            "Start" : 0.66747,

 I found that I can append dictionaries to an array of dictionaries by sending the [dict] object two messages: first (append Notes *), which creates a new entry, and then (setparse Notes[5] Pitch: 46 Velocity: 127. Start: 0. End: 1).

This works, but I have to know the length of the current array before I append any new value. Does anyone know if there is a way to append a dictionary to the end of an array of dictionaries without having to know what index to set it at? If anyone knows how to do this with Max or Javascript, that would be amazing!

Let me know if you want me to explain more/better...

Hello!
I'm using a dict object to store data about midi notes as an array of dictionaries:
{
	"Notes" : [ 		{
			"Pitch" : 38,
			"Velocity"

appending-to-an-array-of-dictionaries

Appending to an array of dictionaries

Ernest

This is part of a series of tutorials on the Husserl3 design. All tutorials in this series:

This time it only took me a couple of hours to convert a polyphonic design into a multiphonic one. This tutorial describes how any number of voices can play different sounds set from any number of channels, with different parameters for each channel. I'll be describing how I'm running 16 MIDI channels with 128 parameters each on any of 32 poly~ voices, but the same technique generalizes to MC.gen~ with any number of voices and any number of parameters, within any available computational limitations. Also this tutorial contains a multiphonic voice allocator patch for MIDI and kslider notes.

While the technique is easy, this is an advanced tutorial that requires at least some acquaintance with Max buffers, pattrstorage, zl objects, gen codebox, Max polyphony, and javascript.

If you're wanting multiphony, really gen~ is the best option. It can do everything the msp~ audio processing objects can, and it is much faster. So first I got a polyphonic design working in a poly~ object in gen~., with all its params stored in pattrstorage.

That meant setting scripting names for all the panel controls, and using the same names for params in gen~, which was the majority of the effort. Using the same monicker for panel obects' scripting names and gen~ Params means for example you can send pattrstorage a 'dump' command and wire its output into gen~ to set Param values.

When I started on the naming thing, I didn't realize how much difference it would really make. Now I can use the same list of param names for javascript panel control, buffer peek/pokes, and midi i/o too. So it was worth the effort.

To convert the polyphonic design to multiphony, first I created a buffer~ called 'programs' with 16 channels, and one sample for each param value, as below. When the patch is closed, the freebang message saves the buffer's state, which is then loaded again when the patch is reopened. This snip also shows how I measure for CPU peaks. When the CPU peak is over 90%, I reduce the number of playing channels automatically.

Screenshot 2021-10-15 223311.jpg

The same buffer can hold multiphonic presets, but it's advisable to arrange them as sequences within the 16 channels, viz., "buffer~ programs -1 16 @samps 12800" for 100 multiphonic presets of 128 params. Max chokes with somewhere around more than 64 channels per buffer, I don't know the exact channel limit, but I tried 32,768 channels in one buffer, and Max was really unhappy about that.

For MIDI I/O and sending parameters to gen~, first I sent a dump message to pattrstorage to print all the script names and used the zl object to concatenate them into a messagebox. I edited the messagebox to create a list of all the param names.

So one can send name/value tuples around with send and receive objects, but each one adds a buffer delay and an additional message in the message queue, so I had painstakingly wired all my panel controls directly to gen~.

As I had hardwired it all, I dragged all the wires from the poly~ object input to an 'unpack' object's second input. Then here's the clever bit. I convert the param names into buffer index offsets with a 'zl sub' object, as shown below. I poke all the param name/value tuples into the buffer channel indices like this....this just pokes it into channel 1, I haven't made the panel control to select different channels yet.

Screenshot 2021-10-15 181801.jpg

The poly~ is very simple. It simply sends a thispoly message to the gen~ object on load. For simpler designs, the loadbang for thispoly~ can be inside the poly~ object, but if the gen~ code is more than about 500 lines it doesn't compile fast enough to receive the thispoly message before it fires. So now I send it a loadbang from the top-level patch.

Screenshot 2021-10-16 010341.jpg

The next stage of the conversion is to convert the gen~ object to read the 'programs' buffer instead of using params. When I got all the Param messages set up to work properly, I simply changed them all to History variables by changing 'Param' to History' in codebox.

Then each gen~ voice peeks the buffer object for its current channel and puts the values into the instantly created history variables. There's a reason to make them static History variables rather than simple variables: by interleaving the buffer peeks for each voice on a simple clock counting from 1 to 

, I was finally able to get the Buffer reads done for all 32 poly~ voices without causing a huge processor peak, and only adding a 0.6ms latency @48khz:

Screenshot 2021-10-15 182134.jpg

At first that may not look very efficient, but in fact Max processes audio I/O in chunks set by the DSP settings, so if it's set to 128, for example, all 128 buffer values for each voice are loaded into the cache at the same time. Programmatically, it would be cleaner if buffer peeks could load more than one value at a time, but as gen~ is compiled, the performance is quite good. the important thing is to keep all the peeks sequential. If you load the values from the buffers where you want them. rather than storing them as variables, the performance is terrible.

For MIDI output, I use a similar technique to the 'zl sub' example above with the output of the pattrstorage object. For MIDI input, a 'zl lookup' object uses the same list as 'zl sub' to create the name/value tuples, which may easily be sent to the panel objects with pattrhub if they are for the currently displayed channel, and which are simply poked into the 'programs' buffer to change playing voices.

Updating the panel to show different channels

So the big problem was how to display the panel controls for each channel. When I was first learning Max, I tried putting all the params for each channel in 'zl reg' objects each poly~ object. That REALLY didn't work. Then I tried making 16 separate panels for each MIDI channel. Updating them from pattrstorage drove me crazy. I tried doing dumps from pattrstorage for each channel and sending them as messages each time a voice switched to a new channel, and the coding for that drove me crazy too.

But I finally worked out to keep it entirely separate from pattrstorage, which doesn't even know there are separate channels, and simply saves and loads presets for whatever channel is currently displayed. When changing the panel controls for a different channel, I simply use a 'for' loop to peek the buffer and send 'set' messages to all the panel controls from JavaScript, using the same parameter list that stuffs index lookup values into the 'zl sub' object above. Piece of cake!

Screenshot 2021-10-15 201230.jpg

After I made a gen voice allocator, Cycling74 made a new voice allocator which may have fixed the problems with the original one. However, I still need to store the MIDI channel numbers with each note and send messages appropriately. So I amended my polyphonic voice allocator to support multiple channels in a matter of minutes. Here is the patch containing the voice allocator.

It also contains velocity shaping described at the end.

Screenshot 2021-10-15 230256.jpg

The gen voice allocator resets to the number of voices set by a negative value on the first input, or a note when a positive value is received on the first input. That simply avoided making it 'hot.' It also turns notes off on the attached kslider object when the available voice count is exceeded. the 'lastv' signal is used to update the display indicators from the last played voice on the currently visible channel, in javascript.

Buffer pch(), vel(), age(), chn();    // voice state buffers
History numv(0), vcnt(32), lastv(32), gcnt(1);
x, aoff = 0; 
chan = in3;
if (in1 < 0){                 // RESETS 
    for (i=1; i<=32; i+=1){
        vel.poke(0, i); 
        pch.poke(0, i);
        age.poke(i, i);
        chn.poke(0, i); 
    }
    numv     = 0;
    vcnt     = neg(in1);
    lastv     = vcnt;
}else if (in2 >0) {        // NOTE ON
    for(i=1;i<=vcnt; i+=1){
        x = age.peek(i);
        if(x == vcnt){
            if(numv==vcnt){
                if(vel.peek(i)>0){
                    out4 = pch.peek(i); 
                    //kb overflow update
                }
            }else{
                numv +=1;
                out4= 0;
            }
            chn.poke(in3, i);
            age.poke(1, i);
            lastv = i;    // last voice on
            out7 = in2;    // velocity
            gcnt += 1;
            out6 = in3;
            out3 = gcnt;    //gate out 0-1
            out2 = in1;     //fc out, Hz
            out1 = i;
            pch.poke(in1,i); 
            vel.poke(out7,i);
        }else{
            age.poke(x+1,i);
        }
    }
    p0  = in1;
} else if (in2 ==0){                    //NOTE OFF
    for(i=1;i<=vcnt; i+=1){
        if(pch.peek(i) ==in1 && chn.peek==in2){
            aoff = age.peek(i);
            pch.poke(0, i);
            vel.poke(0, i);
            chn.poke(0, i);
            for(j=1;j<=vcnt; j+=1){
                x = age.peek(j);
                if(x>aoff && x<=numv){
                    age.poke(x-1, j);
                }
            }
            age.poke(numv, i);
            numv -=1;        
            out6 = in3;
            out4 = 0;     //kb fix
            out3 = 0;     //vel 0
            out2 = in1;     //fr off
            out1 = i;    //voice off
            break;
        }
    }
}
out5 = lastv;

The result has no performance problem. On my 4Ghz 6700k i7, which admittedly is still a pretty good cpu, it only adds 2% to peek all 128 params into gen~ with it running at 48kHz with 256/64 dsp buffer sizes and 32 channels. It was interleaving the buffer reads across clock cycles for each poly~ voice that really made the difference, because otherwise that 2% becomes 64%. It just means a 0.6ms latency between param updates on channels, which actually is faster than the params are typically updated in gen~, which I'm told happens every 256 clock cycles, although I don't know if that's actually dependent on the DSP settings.

Screenshot 2021-10-15 210049.jpg

I hope you enjoyed the tutorial. Happy patching to you all )

This is part of a series of tutorials on the Husserl3 design. All tutorials in this series:
Designing a good LFO in gen~ Codebox:  https://c

implementing-multiphony-in-max

Husserl tutorial series (5). Implementing Multiphony in Max

👽!t W∆s ∆lienz!👽

First off, THANK YOU IMMENSELY TO VOLKER BÖHM!!!!!!!!!!!!!!!! 

https://github.com/v7b1/max-thirdParty_externals

Volker has been raining down gifts like a god for decades! they are ALL worth a long hard look, not to mention the maestro's music: 

https://vboehm.net/sound-music/current-projects/

 electronic and electroacoustic works i've ever heard!!  🙌

so wow, we've had threadz that didn't seem to go anywhere with the actual patching:

MaxMSP Forum