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Teerath Majumder

I am making a custom user interface for a project using jsui. To explain briefly what it does: The user can create containers and controls within each container. Properties of each control is stored in the parent container object, and all container objects are stored in an array named containers. What I want to do is embed the state of the user interface when the patch is saved so that when the patch is closed and reopened, the jsui object is initiated with its last saved state. To that end, I tried two methods following 

Both methods lead to Max crashing when reopening the patch. So I haven't been able to establish if either method is actually saving anything. Any ideas about what could be causing the crash, and if there's a better way to do what I'm trying to do?

jsui-causing-max-to-crash

jsui Causing Max to Crash

Valera Saenko

Hi, first time posting. I have decided to learn Max and the best way I know how to do it is to make a kick in an unorthodox (or rather the most orthodox) way. In the patch you can see that I'm trying to do the 909 kick with filters (as it was initially designed in the analogue world). The basic principle is you send an impulse into a resonant HP filter.  The cutoff/centre frequency gives you the pitch and the Q amount gives the decay. To keep the sync and set the envelope I am using phasor~ into ramp~ into shape~ into scale~. This gives me a smooth curve of values that send signal to filtercoeff~ "frequency" and do their magic for the biquad~. 

The problem is still somewhere around ramp/shape/scale (most likely the scale~) as I see that the peak values are different. When the sound is on you can hear the peak pitch go in a cycle (sin) rising several hits in a row and then lowering. What could be the reason for that?

It is unlikely to be biquad~ as this happens with all filters (svf, live, etc) although for sure I need to learn filter design in order to use one biquad~ instead of five as I'm doing right now. But that is for the next improvements.

Edit. After googling more about self oscillating filters I see that it is the filters' problem as they don't get enough energy, or there is a very sweet spot for Q I need to find to get it to oscillate on a consistent basis. Compression and overdrive should also help. Thanks for reading and hope you enjoyed this little panic moment I had. 

[Attachment: 909 kick.maxpat]
Hi, first time posting. I have decided to learn Max and the best way I know how to do it is to make a kick in...

is-my-filter-envelope-out-of-sync-filter-based-909-kick

Is my filter envelope out of sync? (Filter-based 909 kick)

Isaac Martin

I’m a beginner to intermediate Max user and currently working on my electronic music degree. My main project this semester is to create a tone matrix that dictates the order in which a sample is processed. For example, if I load a 16-beat break into the sampler, the system would generate a 16-tone row matrix, and then playback the sample according to that generated matrix.

So far, I’ve built the matrix engine using an Uzi and a series of zl objects. I’ve decided to try using FluCoMa for sample analysis, but right now, I’m feeling a bit overwhelmed by all the information. I’m not sure where to begin with integrating it into my project.

Any advice or guidance would be greatly appreciated!

Hi everyone,
I’m a beginner to intermediate Max user and currently working on my electronic music degree. My main project this semester is...

flucoma-for-sample-analyzation

FluCoMa for sample analyzation?

Juan Andrés  Ríos Santoyo

I'm currently trying to set up my Veikk A30 graphics tablet in Max/MSP, but I’ve run into some issues. I initially attempted to use the 

 object to detect input from the tablet, and while the Veikk A30 does appear in the device list, Max does not seem to register any input from it.

Has anyone successfully connected a Veikk tablet to Max/MSP? Are there any alternative methods to retrieve tablet input? Perhaps using 

Any guidance or suggestions would be greatly appreciated!

Hello everyone,

I'm currently trying to set up my Veikk A30 graphics tablet in Max/MSP, but I’ve run into some issues. I initially...

issues-configuring-veikk-a30-tablet-in-maxmsp

Issues Configuring Veikk A30 Tablet in Max/MSP

theophile gagnard

I am trying to make a device that let pass frequencies that are in a key, and shift the other frequency so they match the key. 

The first part is quite easy, but I can't find a way to shift frequencies of the bins properly. 

I don't know if if would be better to pitch shift or frequency shit.

I'm basically trying to understand the processing of Pitchmap or chroma.

thank you in advance and have a nice day !

Hello, 
I am trying to make a device that let pass frequencies that are in a key, and shift the other frequency so they match the key. 
The...

pfft-shift-specific-bins-frequency-to-match

Pfft - shift specific bins frequency to match 

The Harlekin

!!! It looks more complicated then it probably is !!! PLEASE, give me a. lift !!! Hi, 

i've been looking up this for quite some time now but it seems to me that after all these years no one has come up with a solution on how to transfare a file (a.k. kick drum sample) via max/maxforlive to the elektron machinedrum sps-1. I collected quite  some stuff from the forum here but can't get my head around how to put it together. The most difficult part it seems is the 8to7 bit conversion. It would be outragous if any of you who sees the forrest and not just the trees as I do and gets this solved. Of course there is a solution but as neither max is nor am I c++,  I can't translate this to its javascript equival, which would be one solutuion taken from the 

const int SAMPLES_PER_PACK = 40; // only 40 16bit samples can fit in the packed 120 7bit bytes
    size_t datapos = 0;
    for (int packetSeq = 0; datapos < data.size(); packetSeq++)
    {
        buffer[0] = 0xf0;
        buffer[1] = 0x7e;
        buffer[2] = sysExChannel;
        buffer[3] = 0x02;
        buffer[4] = packetSeq % 128;

        unsigned char *b = &buffer[5];
        for (int t = 0; t < SAMPLES_PER_PACK; t++)
        {
            int s = 0;

            if (datapos < data.size())
                s = data[datapos++] + 0x8000;
            else
                s = 0;

            (*(b++)) = (s >> 9) & 0x7f;
            (*(b++)) = (s >> 2) & 0x7f;
            (*(b++)) = s & 0x3;   
        }

        // calc checksum
        unsigned char ll = 0;
        for (int t = 1; t < 125; t++)
            ll ^= buffer[t];

        ll &= 0x7f;

        buffer[125] = ll;
        buffer[126] = 0xf7;

        l = write(buffer, 127);
        if (l != 127)
        {
            return false;
        }

        if (progress)
        {
                progress(((float)datapos) / data.size());
        }
    }

I also found this script, which from my understanding will/must covers a core function in this task if there is conversion involved ( credits for this go to who ever took on this challenge already ):

autowatch = 1;

////////////////////////////////////////////////////////////////////////////
// Converts 8 bytes of "packed MS bit" sysex into the original 7 bytes
//
// Input (8 bytes):
//  7  6  5  4  3  2  1  0 bits
// 00 G7 F7 E7 D7 C7 B7 A7
// 00 A6 A5 A4 A3 A2 A1 A0
// 00 B6 B5 B4 B3 B2 B1 B0
// 00 C6 C5 C4 C3 C2 C1 C0
// 00 D6 D5 D4 D3 D2 D1 D0
// 00 E6 E5 E4 E3 E2 E1 E0
// 00 F6 F5 F4 F3 F2 F1 F0
// 00 G6 G5 G4 G3 G2 G1 G0
//
// Output (7 bytes):
//  7  6  5  4  3  2  1  0
// A7 A6 A5 A4 A3 A2 A1 A0
// B7 B6 B5 B4 B3 B2 B1 B0
// C7 C6 C5 C4 C3 C2 C1 C0
// D7 D6 D5 D4 D3 D2 D1 D0
// E7 E6 E5 E4 E3 E2 E1 E0
// F7 F6 F5 F4 F3 F2 F1 F0
// G7 G6 G5 G4 G3 G2 G1 G0
////////////////////////////////////////////////////////////////////////////

function printHexArray(arr, linelen) {
  var str = "";
  if (!linelen) linelen = 7;
  for (var i = 0; i < arr.length; i++) {
    if (i % linelen == 0) str += "\n"
    str += "0x" + ("0" + arr[i].toString(16)).slice(-2) + " ";
  }
  return str;
}

// assumed that you've already skipped over any header bytes, removed any
// footer bytes, etc. This is just processing the raw data.

// this is MIDI 7-bit data
var dataBuffer = [ 0x7F, 0x01, 0x02, 0x04, 0x08, 0x10, 0x20, 0x40,
                   0x55, 0x01, 0x02, 0x04, 0x08, 0x10, 0x20, 0x40,
                   0x2A, 0x01, 0x02, 0x04, 0x08, 0x10, 0x20, 0x40 ];

// this is the expected output of the algorithm applied to the above data
var expectedOutput = [ 0x81, 0x82, 0x84, 0x88, 0x90, 0xA0, 0xC0,
                       0x81, 0x02, 0x84, 0x08, 0x90, 0x20, 0xC0,
                       0x01, 0x82, 0x04, 0x88, 0x10, 0xA0, 0x40 ];

post("original:" + printHexArray(dataBuffer, 8) + "\n\n");

var outputBuffer = [];
var count = 0;
var highBits = 0;

// this is the conversion algorithm
for (var i = 0; i < dataBuffer.length; i++) {
  var row = i % 8; // relative row in this group of 8 bytes
  if (!row) { // first row
    highBits = dataBuffer[i];
  }
  else {
    var highBit = highBits & (1 << (row - 1));
    highBit <<= (8 - row); // shift it to the 7th bit
    outputBuffer[count++] = dataBuffer[i] | highBit;
  }
}

post("unpacked:" + printHexArray(outputBuffer) + "\n\n");
post("expected:" + printHexArray(expectedOutput) + "\n\n");

// just for fun, let's pack it back up
var packedBuffer = [];
var MSidx = 0;
count = 0;

for (var i = 0; i < outputBuffer.length; i++) {
  var row = i % 7; // relative row in this group of 7 bytes
  if (!row) {
    MSidx = count; // cache the position of the MSB byte
    packedBuffer[count++] = 0;
  }
  var highBit = outputBuffer[i] & 0x80;
  highBit >>= 7 - row; // position this bit in the MSB byte
  packedBuffer[MSidx] |= highBit;
  packedBuffer[count++] = outputBuffer[i] & 0x7F;
}

post("repacked:" + printHexArray(packedBuffer, 8) + "\n");

Here you'll see right at the first page at the bottom why I think that the above conversion somehow is needed.Here is as far as I got and If anyone out there could give me the missing link it would be one big of a favour I can not even describe:

// inlets and outlets
inlets = 1;
outlets = 2;

autowatch = 1;
 
var fileQueue = [];
var buffer = new Buffer("sampleBuffer")
var midiPort = 1; // Adjust to your MIDI output port for Machinedrum
var bufferSize = 120; // MIDI SDS chunk size
var scaleFactor = 2047; // 12-bit conversion factor
var currentFile = "";
var totalFrames = 0;
var processedFrames = 0;
var bfSR = [];

function loadBatch() {
	post("jsarguments.length: " +jsarguments.length + "\n", "argument 0: " + jsarguments[0] + "\n" );
	outlet(1, "jsarguments: " + jsarguments);
    if (jsarguments.length < 1) {
        post("No files provided.\n");
        return;
    }

    fileQueue = [];
    for (var i = 0; i < jsarguments.length; i++) {
        fileQueue.push(jsarguments[i]);
    }

    post("Loaded batch: ", fileQueue.length, " files.\n");
    processNextFile();
}

function processNextFile() {
    if (fileQueue.length === 0) {
        post("Batch processing complete.\n");
        outlet(1, "progress", "done");
        return;
    }

    currentFile = fileQueue.shift();
    post("Processing file: ", currentFile, "\n");
    outlet(1, "progress", "loading", currentFile);
	buffer.send("replace",currentFile);
    
	if (buffer.framecount() > 0) {
        totalFrames = buffer.framecount();
        processedFrames = 0;
        resampleTo44100Mono();
    } else {
        post("Error loading file: ", currentFile, "\n");
        processNextFile(); // Move to next file even if failed
    }
}

function resampleTo44100Mono() {
    var sr = buffer.samplerate();
    var channels = buffer.channelcount();

    if (sr !== 44100 || channels > 1) {
        post("Resampling to 44.1kHz and converting to mono...\n");
        outlet(1, "progress", "resampling");

        var tempBuffer = new Buffer("convertedBuffer");
		tempBuffer.send("sizeinsamps",totalFrames,1);

        for (var i = 0; i < totalFrames; i++) {
            var left = buffer.peek(1, i);
            var right = channels > 1 ? buffer.peek(2, i) : left;
            var monoSample = (left + right) / 2;
            tempBuffer.poke(1, i, monoSample);
        }

        tempBuffer.send('sr',44100);
		tempBuffer.send("bang");
        convertTo12bit(tempBuffer);
    } else {
        convertTo12bit(buffer);
    }
}

function convertTo12bit(buffer) {
	var frameCount = buffer.framecount();
    var convertedBuffer = new Buffer("converted12bit");
    convertedBuffer.send("sizeinsamps",frameCount, 1);

    for (var i = 0; i < frameCount; i++) {
        var sample = buffer.peek(1, i);
        var convertedSample = Math.round(sample * scaleFactor) / scaleFactor;
        convertedBuffer.poke(1, i, convertedSample);

        // Update progress every 1000 samples
        if (i % 1000 === 0) {
            processedFrames = i;
            updateProgress();
        }
    }
	convertedBuffer.send("bang");
    post("Conversion to 12-bit done.\n");
    outlet(1, "progress", "converting", "done");
    sendSample(convertedBuffer, frameCount);
}

function sendSample(buffer, frameCount) {
    var header = [0xF0, 0x00, 0x3C, 0x02, 0x00]; // Elektron SDS Header
	var kit = [0x52, 0x04, 0x01];
	var smplTransHead = [0xF0,0x7E,0x00,0x01];
	var smplTransInit = [0x00,0x00,0x10,0x14,0x31,0x01,0x16,0x4A,0x00,0x00,0x00,0x00,0x15,0x4A,0x00,0x7F,0xF7]
    var footer = [0xF7]; // End of SysEx

    processedFrames = 0;
    totalFrames = frameCount;
	post("Conversion to totalFrames: " + totalFrames + "\n");
    for (var i = 0; i < frameCount; i += bufferSize) {
        var chunk = [];

        for (var j = 0; j < bufferSize && (i + j) < frameCount; j++) {
            chunk.push(Math.floor(buffer.peek(1, i + j) * 127)); // MIDI 7-bit conversion
			post(chunk);
        }
		
        var sysExData = header.concat(chunk, footer);
        
		outlet(0, sysExData);

        // Update progress every chunk
        processedFrames = i;
        updateProgress();
    }

    post("Sample sent to Machinedrum.\n");
    outlet(1, "progress", "sending", "done");
    processNextFile(); // Move to next file after sending
}

function updateProgress() {
    var percentage = Math.floor((processedFrames / totalFrames) * 100);
    outlet(1, "progress", "processing", percentage);
}

// For debugging: print any received integers to the post window
function msg_float(v) {
    post("Received float: " + v + "\n");
}

// Max for Live Inputs
/*
function anything() {
	post("jsarguments: " + jsarguments + "\n");
    var args = arrayfromargs(messagename, arguments);
	post("messagename: " + messagename + "\n", "arguments: " + args + "\n");
    if (args[0] === "batch") {
		post("Found batch\n");
        jsarguments = args;
		post("!! jsarguments: " + jsarguments + "\n", "!! jsarguments 0: " + jsarguments[0] + "\n", "!! jsarguments 1: " + jsarguments[1] + "\n", "!! jsarguments 2: " + jsarguments[2] + "\n");
        loadBatch();
    }
}
*/

function list(){
	var a = arrayfromargs(messagename, arguments);
	post('args',a);
	post("Buffer: ", bfn, "| Value: ", bfv,"\n");
	bfSR[bfn] = bfv;
}

function anything() {

	var a = arrayfromargs(messagename,arguments);
	if (a[0] === "dropFile") {
		outlet(1,"Preparing file: ",a[1]);
		fileQueue.push(a[1]);
		post("Loaded batch: ",a[1], fileQueue.length, " files.\n");
    	processNextFile();
	} else if(a[0] === "smplBf"){
		bfSR[a[1]] = a[2];
		post("Recived SR:",a[1],"|", a[2], ".\n");
		
	};
		

}

Buffer.prototype.samplerate = function(){
    return 1000*this.framecount()/this.length()
}

Sysex dump sample (a.k. Kick oneshot  file) to MD-SPS1, Machinedrum

sysex-dump-sample-ak-kick-oneshot-file-to-md-sps1-machinedrum

Mariano Sibilia

I’m working on this patch (below) where I want to randomise the current values of a multislider using an integer to set the deviation percentage (0% = no deviation, 100% = full-range deviation). The multislider’s range is variable (min 0, max adjustable). The patch I’ve attached is working okay, but the issue is that the values tend to drop to 0 and stop moving. I think it might be a float vs. integer issue...

Deviation works fine when the multislider range is set to 

 (floating-point), but if I change the range, I need to rescale the deviation percentage. For example, if the multislider 

, 100% deviation currently creates values from 

 (and say i have value 5 to randomise 100% the other values are <5, instead of ±5).

I also tried another way using prepend range to random, but noticed that with other max versions i get the message that "random doesn't understand range"...

Any ideas on how to make the deviation percentage proportional to the variable range 

Hey everyone,

I’m working on this patch (below) where I want to randomise the current values of a multislider using an integer to set the...

random-value-deviation-factor

Random Value + Deviation Factor

ddgg

 example in the max-sdk repository, I can see that there is the possibility to create 2 externals using the 

 file to compile both versions from the same project folder?

Hello everybody!

Looking at the limi~ example in the max-sdk repository, I can see that there is the possibility to create 2 externals...

compiling-an-external-and-its-mcs-version

compiling an external and its mcs* version

Sander Schuurman

Hello, I want to create a delay/looper that has a switch to reverse the buffer but can also keep recording on top of it.

I tried multiple ways, but the disconnection of the groove playhead to the record head because of the inability to record in reverse (by my knowledge?) is a problem that I don't know how to fix.

I had an idea where I have a 30second buffer~ with a record~ and a groove~ just loop-recording, and that I made smaller buffers with grooves for the delay-lines, and then I can reverse the smaller buffers. But than again the play and record head go out of sync.

Anyone has a pointer of how I can achieve this with Max for Live?

(or do I need gen~ to control play and record heads on a sample base level?)

Hello, I want to create a delay/looper that has a switch to reverse the buffer but can also keep recording on top of it.

I tried multiple...

Misc Forum