tdaeik

hi,
i want to find the highest, say, n peaks from a jit.matrix with magnitude values from a pfft~. is the zsa.freqpeak object from maxobjects.com currently the only freely available option for this? or is there something built into a max/msp/jiter object somewhere which i could use to achieve this? were i to do this myself would javascript be the best option to do this? what if i want to find the peaks from a single fft frame in real time, would java script be too slow for this?
cheers,,


peak-finder

Something like this, using [mxj buf.Op] might help.

hi zachary,
thanks for your response. this example may take me some days to digest, but thanks for more thought!


hi,
not that anyones interested but i came up with this solution, basically a column-wise traversal of the matrix, sorting each column and getting the highest n values from that. hopefully i can now do math operations on the highest values in the time i need. cheers,,


so, i noticed that when the values are all the same, [zl sort] still sorts them, always in the same way. why is this?


An easy way to reorder the peaks is to separate frequencies from the amplitude coming from zsa.freqpeak~ (using zl delace). Then you can simply sort the amplitudes (zl sort) and use the right outlet of zl sort in combination with zl lookup to reorder the frequency properly.


hi,
after reading the papers on your website i see that zsa.freqpeak~ implements the peak detection algorithm i was looking for, so it's just a matter of me to incorporate it into my little patch. cheers,,


Just out of curiosity, does the minimum number of peaks in the freqpeak object is always 4?

peak finder

noiseandrones

I am currently building a microtonal MIDI patch. So far, the tuning system works, but I have a problem.

MIDI is of course, note-based, and therefore microtonality is based on a pitch shift of a given note.

That's fine for widely spaced microtonal chords, but for close relationships - 110, 112, 115 Hz etc. it doesn't work and these pitches need to be sent to a different instance of a virtual instrument - I can't find a way of bending to instances of the same note at the same time.

So, what I'm looking to do is route duplicate MIDI note names to different instances of a virtual instrument, but I can't find an elegant way of doing it. 

I'm also attaching the basic microtonality 'engine'. The idea is to have microtonal playback based on a list of input frequencies, not based on tuning an instrument to a scale.

Hi there,
I am currently building a microtonal MIDI patch. 

list-processing-help

(Or, if anyone has a microtonal MIDI patch that can handle polyphonies on the same pitch class that I can reverse engineer to accept a frequency list, please do let me know! ;) 

use poly~ target based instead of midi note based.
All is explained in poly~ help file.



While not an answer to your specific question, this  video of how I did some microtonal exploration might be interesting and have some useful ideas. https://www.youtube.com/watch?v=2L7edwef-5k&t=9s&ab_channel=MusicwithLisp

Bascially I take route of going from midi into some non-midi way of communicating with instruments. I have previously made pitch bend mappers using the same tools though.

JPL Industries is an authorized wholesale vape and smoke distribution company that sells thousands of e-liquids and vaping devices based in New York.
Wholesale Vape Juice

Thanks so much for getting back - this is very helpful.

I opted for something like this in the end - it works pretty well.

I figured that I probably wouldn't have more than five close beating frequencies at the same time, so I set it up so that the lists are split into groups of 5, with the first in *each* group going to one instrument, the second of each to the second, and so on. As long as I'm careful not to have more than five close frequencies, it works!

List processing help

Loukas Perreault

Hi!
I know this is the kind of topic that a lot of posts in this forum answers, but I can't get the exact information i'm looking for.

I have three colls (could be more) in which I have symbols. 
Ex : 
Coll 1 : 
1, apple;
2, orange;
3, mango;
4, mulberry;

Coll 2 :
1, watermelon;
2, orange;
3, avocado;
4, apple;

Coll 3 :
1, orange;
2, strawberry;
3, papaya;
4, apple;

As you can see, "apple" and "orange" are in each of the colls.
I'd like to populate a new coll with only these two symbols.

I know there are some posts about removing the duplicates, but not so much about keeping the duplicates, and it's often about numbers, maybe I could do something with converting the symbols to ascii and itoa back when the duplicates are filtered but i can't seem to find the exact way of doing this...
Thanks in advance for your help!

Hi!
I know this is the kind of topic that a lot of posts in this forum answers, but I can't get the exact information i'm looking for.
I ... 

get-duplicate-data-from-coll

the zl/list object has modes for getting things in common between two lists.  It works with symbols or numbers.

@Double_UG your approach seem to kind of do it! Thanks! The thing is that it puts every matching symbol on one coll line... But thank you for showing me the zl.sect! 

Here is the example in context using filenames. The filenames that are identical are in bold (the colls will contain a lot of entries) :
Coll 1 :
1, AC_SLATER_bass_loop_pitchy_af_01_128_E.wav;
2, AC_SLATER_drum_fill_akai_127.wav;
3, DD_RAA_77_drums_loops_soul_hat_groove_heavy.wav;
4, UMRU_bass_one_shot_metal_G.wav;
5, DECAP_kick_one_shot_big_bop.wav;

Coll 2 :
1, AC_SLATER_drum_fill_akai_127.wav;
2, DECAP_kick_one_shot_big_bop.wav;
3, MdL_kick_flap.wav;
4, UMRU_bass_one_shot_metal_G.wav;
5, osr_clap_one_shot_high_harsh_malt.wav;


Coll 3 :
1, osr_bass_loop_ring_128_Cm.wav;
2, UMRU_bass_one_shot_808_soft_C.wav;
3, DECAP_melodic_one_shot_dark_windchimes_one_shot_Amin.wav;
4, OAK_loop_ey_bruh_listen_drums_152_Bmaj.wav;
5, DECAP_kick_one_shot_big_bop.wav;
6, UMRU_bass_one_shot_metal_G.wav;

Coll final :
1, DECAP_kick_one_shot_big_bop.wav; 
2, UMRU_bass_one_shot_metal_G.wav;

your coll final should have 3 entries
1, UMRU_bass_one_shot_metal_G.wav;
2, DECAP_kick_one_shot_big_bop.wav;
3, AC_SLATER_drum_fill_akai_127.wav;

That's the thing, I only want the namefiles that are in all 3 colls. I will go check out what you sent, thanks so much for your reply

then you shoul have stated so, instead of duplicates.
If you now say only names that are contained in all 3 colls,
then use zl.sice 2 at output.
In theory, a singe coll could also have repeated names inside,
like :
1, AC_SLATER_bass_loop_pitchy_af_01_128_E.wav;
2, AC_SLATER_bass_loop_pitchy_af_01_128_E.wav;
17, AC_SLATER_bass_loop_pitchy_af_01_128_E.wav;

here is "tripple" detector


OMG @SOURCE AUDIO thank you sooo much!!! Yeah you are right my explanation was missing some key info. I will try and be more precise next time! That's really sweet of you that you took the time! Thank you again!

Get duplicate data from coll.

 I’m am trying to figure out where to begin in my patch! My idea is to be able to change the sample rate of all the audio and effects playing. What I am trying to replicate is something found on old Roland digital recorders that had a function called VariPitch. It would allow you to lower the playback speed of the audio (like a tape machine) and if digital effects were added then consequently lower that along with it too. I’m guessing from what I understand this works by changing the machines sample rate in real time. It could record at 32khz sample rate and then by lowering it, to say 1khz, play back slower. 

At first I found adstatus which doesn’t really do what I need but it would really interesting to control the global sample rate per 1hz incremental up or down. Then I came across poly~ and this seems like it would do what I need but I’m not sure. 

Hello!
 I’m am trying to figure out where to begin in my patch! My idea is to be able to change the sample rate of all the audio and effects

trying-to-change-sample-rate-playback-in-real-time

If anyone had a patch example to share or point to its appreciated!

I would start with groove~'s playback speed or ad-hoc driving of play~. You record a buffer~ at a given sample rate, then you play it back at a different rate, "simulating" a  different sampling rate. That's basically what VariPitch allows: live modification of playback rate.

I'd like to revive this thread. 

What I want to achieve is creating a master clock rate that changes the rate of the whole or part of the patch. Like the clock rate knobs on some pedals like Chase Bliss Mood or Alexander Pedals Marshmallow. Basically, slowing the whole patch. 


847-250-6868 Chicagoland veteran owned Junk Removal. Storage unit clean out, garage clean out,
donate furniture

Trying to change Sample rate playback in real time ??

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

trainvids

hi all -
i have a patch that's running at about 25% CPU. when i open sub patches (or send "front" message to a thispatcher in a bpatcher). i get short glitches/crackles in the msp stream when the window opens, and then the dsp seems to operate as normal. is there a way to deprioritize window/gpu so that audio is always more important? something like redraw queue throttle in prefs?
appreciate your thoughts! 

hi all -
i have a patch that's running at about 25% CPU. when i open sub patches (or send "front" message to a thispatcher in a bpatcher). 

dsp-glitches-when-switching-windows-opening-subpatchers-redraw-queue-throttle

dsp glitches when switching windows / opening subpatchers (redraw queue throttle?)

cyusim

I'm having issues saving my patches on Max 6. The patches or projects are being saved, but not where I'm telling Max to do it. This is happening every time I save a new patch/project, and sometimes saving an existing patch.
Usually the patches ended up being saved in Applications --> Max 6 folder (mac).
I made sure that the folder in which I want my patches to be saved is a declared path under "File Preferences".

max-6-patches-wont-be-saved-in-specified-location

I'm having this problem now! Did you ever manage to sort it?


You need to provide an absolute path. Not by changing file preferences.

e.g. write /my/folder/filename.txt to put the file filename.txt in the folder /my/folder


Sorry I see that you're referring to patches. You'll need to post a much clearer description of what you're experiencing and expecting. Step by step.


I'm having the same problem. It also won't let me set the default project save folders to the folder i want - which is a folder within My Documents. I'm wondering if the two are related. It seems everything sames in an msp tutorial patch folder, which is the first place I accidentally saved something. When I move patches out of that folder, it just moves an "alias" unless I trick it by sending it to the trash then moving it back to a different folder from there. 

The problem that is most frustrating, which I also assume is related, is that I can't access saved patches as new objects, and even copying and pasting them from another file they come up as a grey box instead of white, can't be accessed, and don't do anything. I created a new patches folder and set it to that in Max, but it doesn't seem to be recognizing that folder. I'm on OS X 10.8.2, a retina MacBook pro. I'm very new to Max. Thanks for your help!


A perfect description of the problem, I'm handling right now! did anybody found a solution and can share the answer here?

Never happened to me with 10.8.2 or .3. Are you talking about patches or projects ? This still isn't clear to me.


ah, thanks for the hint "project folder". using them - what make sense - solve this problem! I'm quite new to max and did't know how files are organized.


has anyone found the solution to this? I'm having the same issue


Also waiting for an answer! It's quite confusing.


I am having the same problem. I've been using max for 6 months bug-free, but when i installed max on a new SSD this problem began happening right away. I'm not concerned about a default save location, I'm concerned with Save/SaveAs->Selecting a location->saving->My patch is now the name i chose through saving->but it's not located in the directory i chose, it is in SSD/Applications/Max6/"filename".maxpat, which I assume is the default directory.

I have a lot of very different projects, I want to be able to organize them my way, not have them all lumped into the default folder. Can someone please offer a solution to this problem? 

AHA! Well, I solved my problem, at least. The folder I was trying to save to was called "MAX/MSP" The "/" was the culprit. So I guess that Max is rather smart in that it automatically reroutes the file to a default folder in case it cannot save to an invalid directory. At the very least this, helps from someone losing their work, I guess.

I realise that this is quite an old thread, but I just started having the same problem with Max 7.0.3 ..... patches are being saved inside a folder in the max application, rather than the specified folder. Did any of you figure out definitively what was causing this? It is becoming quite infuriating!


did you confirm you don't have a "/" in any of the folder names of your specified path?


Yes, there is definitely no '/' in the path...... in face, I was trying to save a patch into a folder which already existed; the problem only started when I installed v7.0.3.


Any movement on this? I'm finding the same problem. Thanks.


I was getting all kinds of problems, like patchers not updating , files in recent patchers not loading, having to select replace file every time I saved a patch, all caused by having the name "kits/trig" for the folder I was saving the patches into.  Get rid of the "/" and it all works normally.


Just bumping this thread and chiming in "me too!"


I still had the same problem with Max 7.2.3. When I removed the "/" out of my folder it worked. I suppose this isn't completely fixed yet? 

Thanks anyways for the answers. It saved me from going crazy!


Not sure this helps. But I was having a similar problem. I specified where I wanted to save a patcher, saving window appeared, I click save but when I go to specified folder there's nothing there... I found that when I saved each patcher as a project first and save it to my default Max 8/projects folder I could open each patcher from there. The project file would open with it and I wouldn't have any issues saving it and locating it again...Seems that saving things as a project first rather than a patch was the solution for me! Hope it helps

Max 6: Patches won't be saved in specified location

mkap7

I'm having trouble with something that used to work fine for me - saving a patcher as a text file.   (Basic, I know.)  The online documentation says to open a patcher, go to Save As in the file menu, and select the Max text format ( .mxt).  But I do not see a Max text format among the choices.  Any idea why not?
Thanks,
-mk


Hi folks,
I'm having trouble with something that used to work fine for me - saving a patcher as a text file. 

saving-a-patcher-as-text-not-working

Save it as a .maxpat file, which should be plain text.

Hmmm.....ok, and I know you can open any .maxpat as a text file......but then what is this Max text format (.mxt)?  


I am not sure, but my suspicion is that it is a holdover from the days when some filesystems only allowed you to have three-character  extensions. Just guessing, though.

Edit: Looking around a bit more it appears that .mxt is a pre-JSON text format. Maybe some who was around before version 5 (I believe) will have a more enlightened answer, however.

why would you need to save a patch in Max Text format ?

I guess there isn't much need for that if the maxpat file itself is just a text file.  (I didn't realize that.)  
I went into the Max help stacks looking for how to share patches, and got onto a page about saving things as a "Max text" file (.mxt).....so that's what I tried to do.  I see now that it's not necessary - just open as text, cut and paste.

saving a patcher as text not working

Samu

I'm trying to access an address whose index could change but the same operation needs to be performed based on the last tag in the OSC address. 

With o.select for example, /*/x  would select both /1/x and /2/x but not /1/y. 

This behavior is desired, but instead within the context of o.expr.codebox. How do I reference a variable incoming address in the o.expr language? 

I'm trying to access an address whose index could change but the same operation needs to be performed based on the last tag in the OSC ... 

odot-library-referring-to-dynamic-input-address-in-o-expr-codebox

Don't know if you already solved your problem but you can find bellow an example of how I am doing this with o.expr.codebox
I hope it will be helpful ^^

odot library -- referring to dynamic input address in o.expr.codebox

seejayjames

Hello everyone! Been a long time. Great to see how far Max has progressed!

I’m interested in using barcode scanners to generate audio/MIDI/etc and am wondering what others have found in this area. Seen a few interesting videos with a Japanese group doing this, but it seems like the scanners have been modified, which I’d like to avoid---just use what they naturally output and let Max do the rest.

From what I’ve read, most scanners send a numeric text string as keystrokes, so it should be pretty simple to get access via [key] or other text objects. That’s OK to get single strings of unique numbers to play back samples etc. But I’d like to make it more of an instrument versus a sample trigger. Specifically:

-- Long unique numbers aren’t needed. Rather, I’d like a smaller set of simpler numbers that I can translate to frequencies, instruments, or other parameters. So do the scanners output shorter numbers if they read fewer lines? Or maybe the full number is padded with zeroes, which would be easy to strip off.

-- If the scanners output shorter numbers/padded with zeroes when fewer bars are read, that means distance to the barcode would be one parameter of playing the instrument (and would be ideal). The result could be actualized in many different ways, like octave shift of the same note, etc. What I don’t know is how far away the scanners can reliably (or even somewhat reliably) read codes.

-- I need maybe up to 16 scanners at once, and they will be wireless. The idea is a collaborative space with hundreds of codes all over the walls that multiple people can play. Maybe I’ll have barcode T shirts too, heh. If it’s only keystrokes it will be impossible to parse which scanner sent which data. (While not absolutely critical, it would be much preferred to know---each scanner could use different timbres, octaves, scales, etc from the same codes, and they could always be set to “unison parameters” if desired.) It looks like some scanners can use serial ports---if so, the data would already be separated by port and I’d just use multiple [serial] objects. Any experience with this?

-- I’m thinking that most of the time the scanners would be on continuously so the data is a stream. What kind of rates are possible?

-- Am planning to use printouts for the barcodes. However, am wondering if the scanners can read barcodes on screens, and if so, how well. This would allow installation of some screens in the space with dynamic codes.

I realize that most answers will probably start with “Why don’t you get one and try it?!” Hehe. Absolutely…that’s the plan. But am curious if there’s any experience with all of this that could steer me in the right direction. Or away from wrong directions!

using-barcode-scanner-data-for-audiomidi

Update: Got the scanner! Works great and the documentation is solid. Great wireless distance. It’s this one:

First got it working with [key], that was quick and easy, using the CR ASCII value (13) to know where the end is. It sends this so that you can enter values into a spreadsheet and it will auto-advance to the next cell down. Then with some configuration (in the User Manual there are a bunch of barcodes to tweak settings) I was able to get it through [serial], so this will allow differentiating multiple devices on different ports. I set the polling to 50ms because there doesn’t seem to be a way to know that something’s come in---when you set it to USB/serial, the key-based input (which does output immediately) is disabled. If there’s a setting that would allow [serial] to know something’s come in, please advise, because that would be more efficient. Note that this particular scanner only supports serial input on Windows.

I’m subtracting 48 from the ASCII values so they map to the integers of the code itself, grouping them with [zl group], and looking for the CR (13) to bang out the collected list. Then I use [match] for the full code. Works great, will probably just have a subpatch with a whole slew of [match] objects for my notes and parameters...I don’t see that any other approach with [coll] etc would be any more streamlined. Of course, am open to suggestions.

There’s a bit of latency but not too much...it’s certainly usable, especially if the tones are more fluid instead of percussive. I may use audio instead of MIDI because there’s more latency with the MIDI output (which comes from the OS). Or maybe I’ll try a combination, lots of possibilities.

Was hoping there could be a way to tell how far the scanner is away from the code (by reading only parts of it as I described in the earlier post), but it doesn’t seem possible. The scanner needs to see the beginning and end lines to complete the read and to send the data, so it’s all or nothing.

The scanner can indeed be set to “Continuous” with a special barcode. This will be very helpful as a performance option. In this mode, players can avoid accidentally triggering music codes if they turn the scanner sideways, cover the beam with their hand, get far enough away that it can’t read them, etc.

On-screen barcodes don’t seem to work. There may be a contrast setting that would do it, but I haven’t gotten the black-on-white version to read at all.

Anyway, just wanted to post this if anyone’s interested. With a bunch of scanners and codes, plus some carefully-chosen musical parameters, this could be really fun. If and when I make an installation I’ll post a video. Thanks everyone!

Hi James, good to know you're well and still maxing.  

Using barcode scanner data for audio/MIDI

Ben Bracken

We are happy to announce the release of Max 8.5.3 and RNBO 1.1.0. This update includes 20+ new features and 80+ bug fixes.

As always, you can download the update here:  
https://cycling74.com/downloads

For a list of all Max 8.5.3 and RNBO 1.1.0 improvements, check out the change log below, or here: 
https://cycling74.com/releases/max/8.5.3 
https://cycling74.com/releases/rnbo/1.1.0 

umenu: @highlight color used in dropdown highlight when @applycolors is 1

Audio <-> Scheduler timing: improved accuracy

buffer~: ‘writeraw’ file dialog is suppressed when file name is given

cascade~: works when input list contains integers

crosspatch: inlabel / outlabel have text editor in Inspector

crosspatch: presets/pattrstorage/snapshots connections cleared before restore

GLCore: color attr of jit.gl.node can override color of attached objects

GLCore: support for distinct face drawing modes

GLCore: jit.gl.material support for two-sided lighting

hi: fixed crash with some controllers on disconnect (Mac)

jit.gl.skybox / jit.gl.environment: drawn after fullscreen jit.window (Win)

jit.movie (AVF): fixed output ready/failure notifications

jit.movie (AVF): improved reading and disposing of files

line~ / curve~: fixed stop after immediate output

live.adsr~: fixed decay stage when driven by a signal

live.step: parameter saved when set by dictionary input

Math fuzzy float comparision: misc improvements

Max for Live editor: tour suppressed when launched from Live

Max for Live editor: update checking suppressed when launched from Live

mc.function: fixed focus issues while changing points and values are output

mc.snowphasor~: reduced minimum allowable rate / interval to 0.1

MIDI: removed CC#0-31 restriction for 14-bit CC messages

mtr: fixed immediate output when events are inserted

Packages: Package fonts are loaded on first run

Patcher Continuous Zoom: works in RNBO patcher / edited Max for Live device

path: don’t return ‘/’ if a ‘boot path’ fails to conform

pattrstorage: fixed ‘renumber’ message

playlist~: fixed harmless timestretching errors

plugsync~: fixed samplecount outlet when used in amxd~

Project / amxd~ / Gen: fixed crash when saving after ‘open original’ from bpatcher

sflist~ / sfplay~: fixed crash printing cues

snapshot~: bang message is sample accurate

Soundfile: improved mp3 loading on Windows

Standalone: fixed crashing when jit.window/pwindow is present

subdiv~: pattern cleared correctly (fixed no pattern behavior)

textedit: fixed value sync issues when setting contents via scheduler thread

themecolor: fixed notification for dynamic color human-readable names

VIZZIE: fixed bang output when texture-output enabled

vst~: hide JUCE 'MIDI CC %d|%d' parameters by default

xctlin: first message not skipped with new channel

Cycling '74 Forums

Max 8.5.3 / RNBO 1.1.0 Released

We are happy to announce the release of Max 8.5.3 and RNBO 1.1.0.

[sharing] Looping Gestures with mtr

Hello Andrew, I recently created a problem report about the MTR new version, but maybe I should have done it ...

Max 7.3.4 Released

Thank you both for replying. I didn't put any 32bit object but it didn't work. I asked to cycling74 then they told me that they will not ...

peak finder

Hi Emmanuel, Just out of curiosity, does the minimum number of peaks in the freqpeak object is always 4? Thank you

List processing help

Hi all, Thanks so much for getting back - this is very helpful. I opted for something like this in the end - it works pretty well. I ...

Get duplicate data from coll.

OMG @SOURCE AUDIO thank you sooo much!!! Yeah you are right my explanation was missing some key info.

Trying to change Sample rate playback in real time ??

I'd like to revive this thread.

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

in either cases consider what onepole~ does: it is only -6db per octave.

dsp glitches when switching windows / opening subpatchers (redraw queue throttle?)

ill try that in a few places!

Max 6: Patches won't be saved in specified location

Not sure this helps. But I was having a similar problem.

saving a patcher as text not working

I guess there isn't much need for that if the maxpat file itself is just a text file.

odot library -- referring to dynamic input address in o.expr.codebox

Hello SAMU, Don't know if you already solved your problem but you can find bellow an example of how I am doing this with o.expr.codebox I ...

Using barcode scanner data for audio/MIDI

Hi James, good to know you're well and still maxing.

Cycling '74 Forums

.css-lq0bf2{font-size:18px;line-height:24px;font-weight:700;color:var(--theme-ui-colors-accent);-webkit-text-decoration:none;text-decoration:none;}.css-lq0bf2:visited{color:var(--theme-ui-colors-dark30);}.css-lq0bf2:hover{color:var(--theme-ui-colors-accent);}Max 8.5.3 / RNBO 1.1.0 Released

We are happy to announce the release of Max 8.5.3 and RNBO 1.1.0.

Hello Andrew, I recently created a problem report about the MTR new version, but maybe I should have done it ...

Thank you both for replying. I didn't put any 32bit object but it didn't work. I asked to cycling74 then they told me that they will not ...

Hi Emmanuel, Just out of curiosity, does the minimum number of peaks in the freqpeak object is always 4? Thank you

Hi all, Thanks so much for getting back - this is very helpful. I opted for something like this in the end - it works pretty well. I ...

OMG @SOURCE AUDIO thank you sooo much!!! Yeah you are right my explanation was missing some key info.

I'd like to revive this thread.

in either cases consider what onepole~ does: it is only -6db per octave.

ill try that in a few places!

Not sure this helps. But I was having a similar problem.

I guess there isn't much need for that if the maxpat file itself is just a text file.

Hello SAMU, Don't know if you already solved your problem but you can find bellow an example of how I am doing this with o.expr.codebox I ...

Hi James, good to know you're well and still maxing.

Max 8.5.3 / RNBO 1.1.0 Released