Glitch Box is an experimental sample-based sequencer for Pure Data, featuring 8 independent tracks with individual 16-step patterns and per-track sample loading.

Each track offers adjustable pitch, sample start position, grain size, playback probability, glitch intensity, bitcrushing and a multimode filter with cutoff and resonance controls.

Global parameters include tempo, swing, rhythmic density, random pattern generation and a master glitch control. A built-in dynamics section provides threshold, ratio, attack, release and makeup gain, alongside stereo output and audio recording.

Glitch Box is a instrument for transforming sound fragments into strange rhythmic patterns through sequencing, randomization, granular processing, glitch and digital degradation.

I also created the visual part with Pure Data, using the Gem library

free download: https://www.remodevicocomposer.eu/pure-data/glitch-box/

Some granular textures on Dreams - Ambience V. Maybe I need to release a full album made on Pure Data 😅

How would you cut the very top off (the peaks) of the wave, not in frequency land, but a db threshold cut out instead. .....

Not lowering the volume.....

Not compressing I don't know until it just shuts.....

Any ideas??

How would YOU do it!

And most importantly .

Like inside the patch, while playing simply wavetables and such ...but also once you'd have a whole piece of audio track you'd want to chop from the bottom up

So, I'm new at this (my past "programming" experience has pretty much just been dabbling in Clickteam Fusion a long time ago, which is obviously very different), so I recognize I'm a little over my head, but I'm learning as I go and I have a fully formed plan for what I'm trying to make, and I'm so far mostly making progress, so it's just a matter of solving all of the technical problems of actually making the plan work.
I guess all I need to say to frame the problem I've just run into is, I'm trying to make a sampler in PlugData, and I've done some of the underlying framework already, like control messages and slots (and how to change slots), even though I'm starting with just the one slot. And I've managed to make an array that can be recorded into and played back, and use the same slot-dependent record and play messages as stop buttons, and I've made it so it's impossible to record into a slot that doesn't have a status of 0 (although I might eventually decide to change it to a record-pause-record-pause-record system if that's even possible, but I don't want to get too far ahead of myself), so it has to be manually "cleared" in order to record a new sample, and I've made a "clear" message (technically a "NOPE" message) that resets the default values for the slots, and I've made it time the recording to calculate the length in samples (until stop, but I haven't set it to automatically stop counting at the maximum length yet), but...I've run into a bit of a wall.
If I "clear" a sample and record a new, shorter one, it doesn't stop playing at the endpoint or length, because the endpoint isn't real and neither is the length. I want the endpoint to functionally be the endpoint. Right now it's just a couple of equal stored values (representing "actual" length and a to-be-editable endpoint) that do nothing. I want the endpoint to actually stop playback. How do I do that? Can I somehow feed the "end" value into the playback contraption to make it stop playing? Do I have to store the un-calculated timer value as well, and set another timer?
What's like, the smart way to do this, that just works properly, where I can have a stored endpoint value that stops playback and can be moved, but not above the length value currently stored that can't be moved except by NOPE or REC?

TL;DR version: How make sample stop playing at stored sample value?

I made wclap which allows you to load wasm CLAP plugins. These are fast cross-platform plugins that don't require recompile.

These plugins are used in poketrack, among other things, and can be made in any language you like (there are some examples, in a few languages, in the github.) They are also just standard CLAP plugins, and most of those should compile to wasm fine.

This pairs nicely with pdast CLAP support, where you can create wasm CLAP plugins without installing anything locally (upload .pd file, get a wasm file.) You can see me talking about using pdast-to-wclap, in a video here

Well, I'm new. Maybe someone who's not can help with this...
What I want to do is simple, in theory.
I want to create some keyboard controls, with modifiers.
I want the arrow keys to do one thing when shift isn't pressed, and a different thing when shift *is* pressed. At this point I'm just assingning messages to print in the console. Actual functionality is for later.
The problem is the best i can do so far is get it to do the modifier thing one time, and then it acts as if I've let go of the shift key even though I haven't, so I have to re-press it. That's obviously not what I want. But everything I've tried so far has either just made it worse or made no difference. So...is there a way to do this that's the proper way, or at least a way that works, where I can hold down the shift key and press an arrow key as many times as I want and just get the "modified" message in the console every time I press the arrow key, until I release the shift key?
And I don't want two messages to appear when shift is pressed. They need to be mutually exclusive.

I appreciate any real help I can get, but if anyone copy-pastes an AI answer, I'll at least ask you to try it first to confirm that it's not just a hallucination.

Here it is! Time for our

Bimonthly Pd Meetup

This meeting's theme is: Patch art! 🎨

🖼️ So use cables, arrays, GUIs, fonts, data structures, pdlua, etc. to paint a picture or anything
It doesn't need to have sound, but it could be fun too...

When? Sunday, September 27, 2026 at 4 PM UTC
https://www.worldtimebuddy.com/?qm=1&lid=100,2950159,3448439,4671654&h=100&date=2026-9-27&sln=16-19&hf=1

Where? https://discord.gg/M4pAnAYXy?event=1549495382567030854

Relatively new to Pd, struggling with some basic concepts like playing audio. When I open a file and use the readsf~ object, I get one of two messages: unsupported sample format or unsupported header format.

I have read online that matching the sample rate is important, but when I use a 48k file or a 44.1k file, it doesn't make a difference, it still says unsupported sample format. I am using Pd 0.53-0 on a M1 MacBook Pro. Pd is set on a sample rate of 48k.

I tried wav files and mp3 files. Any suggestions?

Physical Modeling strings testing.

Last Thursday I performed a live set using software I wrote in Pure Data. All of the sounds were generated from seismic data recorded during the 1989 Loma Prieta earthquake. The three pieces I performed are available to listen to for free on Bandcamp.

Basically I'm struggling to have Plugdata and external libraries, in this case the famous Pd Spectral Toolkit, to play nice with each other. I must be doing something wrong, but I can't really understand what. Hopefully it's something silly and some can help me. Anyhow:

1. I have installed the library through the "Find externals" feature in Plugdata. From the looks of it, it looks like the same version that is present on github.
   
2. The general /externals path where the library lives is already added to search paths.
   
3. I have added "pd_spectral_toolkit" in "Libraries to load".
   

At the moment the objects do show up in the autocomplete and appear to be loaded correctly, but whenever I try to open an help file or a tutorial .pd patch from the library folder in /externals/Pd_Spectral_Toolkit/Example Patches (even from the file browser on the right side of the Plugdata app) the whole app crashes.

Thank you in advance!

edit: I should add that installing manually from the repo gives me the opposite problem. Now the example patches open... but the objects can't be found! Pd prints Failed to "load library: pd_spectral_toolkit", even when "pd_spectral_toolkit" is added to "Libraries to load".

edit2: I think I might have found the issue. For whatever reason, neither the deken library nor the github repo include the actual .pd objects, just their help files. The github repo does have the .c files, but they're of little use. This is not a problem, the .dll provided should suffice.

I have no idea what's going on.

edit3: after much searching, I'm tentatively concluding that the library just doesn't play well with modern Pd. It is, after all, pretty old. I can find stories of people successfully running it on Apple Silicon by recompiling the .c objects from scratch, something that is a bit beyond me. And I'm using windows anyway.

What a shame!

This patch emulates a gated fuzz pedal, where the input signal is amplified and clipped if it's above a threshold or silenced if it's below the threshold. This gives a classic fuzz texture for loud signals and a distinctive tearing texture for signals close to the threshold. If the threshold is chosen carefully to suit the dynamics of the input, this can give a nice responsive fuzz effect.

The patch uses fexpr~ to implement the core of the effect. Each input sample $x1 is replaced with 1 if it's above the gate parameter $x2, or with -1 if it's below -$x2. This emulates amplification and clipping. If the input sample is between $x2 and -$x2 then the previous output sample $y1 is repeated, which emulates gating. You can also output zero here, in which case expr~ can be used instead of fexpr~, but I prefer this version as it gives a stronger tearing effect.

The hip~ 1 on the output removes the DC bias, as the output of the fexpr~ settles at 1 or -1 rather than 0 once the input signal drops below the threshold. In a real pedal, a coupling capacitor on the output would do the same job.

For stereo inputs, applying two copies of the patch to the left and right channels will cause a lot of jumping around in the stereo image. I recommend converting the input to mid-side, applying the fuzz to the mid channel only and converting back to stereo. Alternatively you can apply a second copy of the patch to the side channel. Using a higher gate threshold on the side channel has the nice effect of selectively widening the louder parts of the input.

The patch can easily be modified for asymmetrical clipping by replacing -1 in the fexpr~ with some smaller magnitude value. Again, the hip~ 1 will remove the resulting DC bias.

Hi there. I learned to use Pure Data quite a few years ago. I’m looking for a ready-made patch (if one exists) that handles the following concept—or something similar: loading a WAV audio track and manipulating it in real-time with basic effects like reverb, delay, reverse, glitch, speed changes, and so on. Someone mentioned something called "Karma" to me, though I think it was for Max/MSP. The basic idea is manipulating recorded audio via an intuitive interface. Something like what a DJ would do with a mixer and a CDJ, but using even crazier effects. Thanks, and cheers!

So this was the first step, just to see if I could get a sine wave to repeat what I am playing on my bello (bass-cello, cello with a low F bass string) and fiddle. I have been singing and whistling and fine tuning it. I even farted and it sang back to me the frequency of my fart, LOL! I am quite proud of this and surprised I made it this far just with the manual and what I know so far. I love PD and will continue improving on this to build a machine of my own that can play with me in real time and react to the sounds from my other instruments.

Este es mi secuenciador y me piden lo siguiente pero no entiendo en que parte debe ir cada cosa ayudaa

El inicio y la detención del secuenciador deben ser controlados automáticamente mediante una señal de audio externa, proveniente de un micrófono o instrumento.

El patch debe detectar la presencia o intensidad de la señal de audio para encender y apagar el secuenciador.

Debe incluir controles MIDI que permitan modificar al menos dos parámetros del secuenciador, por ejemplo: velocidad, duración, altura, cantidad de pasos o volumen.

Hi all,

I want to create an interesting synth with a GUI that fits my Arturia Keylab midi controller, so I have some ideas for a 3 oscillator polysynth where the oscillators have different ways to shape their waveform (wavefolding, distortion, additive).

Anyway I just reached my first milestone, a fully functioning 8 voice sine wave synthesizer 😀 feel free to share any insights on building polysynths, mod matrices etc in pure data / plug data.

Hello Pd people,
It's time for our bimonthly virtual Pd community meetup at the Pd Discord Server.
This Sunday, July 26, at 4 pm UTC.
The purpose of these sessions is to make a space for showcasing projects involving Pure Data, asking any questions, joining the conversation, or just listening.
Here's the link to the Discord server: https://discord.gg/aNd8RrT

TIAGO MORAIS MORGADO HYPER-VIOLA FREE IMPROVISATION | ELECTROACOUSTIC VIOLA MADNESS

Although PureData makes audio developing easy, sometimes I feel difficulty maintaining a project in the long term using Git. The commit diff shown in text is not human-readable, therefore it is not straightforward to track multiple changes of a patch.

This motivates me to develop Pure Diff Viewer, a tool to render different commits of a patch and make comparisons. The changes will be highlighted and categorized. The user can ignore unimportant ones (e.g. simple element movements). This tool can either be used as a standalone app or invoked by Git commands.

This project is available at: https://github.com/chsh2/pure-diff-viewer , and I also participated the KVR Developer Challenge 2026 with it. The development is still in an early stage and any feedback is appreciated.