I will check it late on my mac computer. for the moment I'm not on my computer, so I don't want to waste much time on it now as I think it will work on my computer.
For the moment I'm using your help file patch which is great. Your abstraction is so much more easy to use then from my patch I shared in the beginning.

what I'm trying to do is the following -
I will have 8 different sound files. Each sound file should be in a individual sf-play2~ object.
I will have data from 8 diffrent sensors I'm reading via arduino. Each sensor will report 0 or 1 (if sensor is pressed or not) alongside with analog data of numbers between 0 - 1023.
What I would like to do is that when I'm reading 1 the audio file will play, when reading 0 the audio file will pause . if 1 is held and the audio file arrived to it ends it will continue from the beginning. (loop 1) . I will also have a clocker to report how much time has elpased since sensor is 0. If more then x seconds the audio file will go back to beginnnig so next time sensor is held (1) it will start from the beginnnig.

here is my try with your abstraction.
the green part is to imitate the on off from the sensor. the blue is the analog data from sensor.
the orange part is my try to make a clocker that will start count when sensor is off. The yellow part is for small envelop to avoide clicks when audio is pause and resume.

How can I make the audio file go to the beginning when the orange part is reporting a bang (that time was elapsed) ?
Also - I'm not sure if the yellow section is working. Meaning if it is indeed making an envelope on every pause and resume ? Is that possible?

sfplay~example.pd

Thanks!

edit: the scale of the sensor should be inverted. 0 1023 0 1

edit 2: how can I make the sf-play2~ go back to the beginning of the audio file without starting it immediately?

edit3: how can I load the file to the sf-play2~ object without starting it? if I don't press "all" message it won't load the file. I would lik the file to be loaded to the beginnnig, but to start and play only when receiving data from sensor on off. for the moment if I'm not pressing all message the file won't play.

@jameslo said:

Honestly, I didn't know if that was @fintg's requirement,

It's certainly a reasonable guess. If the requirement instead were "I just played something cool; write the last 10 seconds to disk" you can do that without a circular buffer at all.

I was just surprised and annoyed that one can only access the delay line's internal buffer at audio rate (and was hoping that someone would prove me wrong).

Access to the internal buffer wouldn't be very useful without also knowing the record-head position. In that case delwrite~ would need an outlet for the current frame being written.

That would actually be a very nice feature request.

In SuperCollider as well, DelayN, DelayL and DelayC don't give you access to the internal buffer. But you can create your own buffer and write into it, with total control over phase, with BufWr -- and, because you control the write phase, you already know what it is. It's quite nice way to do it.

Basically the lack of ipoke~ in vanilla causes some headaches.

Look at the hoops I have to jump through! The extra memory I have to use!

I don't think there is any way to do this without using some extra memory.

In a circular buffer, you have:

|~~~~~~ new audio ~~~~~~|~~~~~~ old audio ~~~~~~|
                        ^ record head

When you write to disk, naturally you want the old audio earlier in the file. There are only two ways to do that. One is to write the "old" chunk without closing the file, and append the "new" chunk, and then close the file.

In SC, if I know the record head position, I'd do it like:

buf.write(path, "wav", "int24", startFrame: recHead, leaveOpen: true, completionMessage: { |buf|
    buf.writeMsg(path, "wav", "int24", numFrames: recHead, startFrame: 0, leaveOpen: false)
});

AFAICS Pd does not support this, so you're left with duplicating new after old data. (FWIW, though, there's plenty of memory in modern computers; I wouldn't lose sleep over this.)

Then there is the problem of synchronous vs asynchronous disk access. AFAICS Pd's disk access is synchronous, and because the control layer is triggered from the audio loop, slow disk access could cause audio dropouts. OS file system caching might reduce the risk of that, but you never know. Ross Bencina's article about real-time audio performance advises against time-unbounded operations in the audio thread.

SC's buffer read/write commands run in a lower priority thread; wrt audio, they are asynchronous. This is good for audio stability, but it means that, by the time you get around to writing, the record head has moved forward. So, even though I could do the two-part write easily, I'd get a few ms of new data at the start of the file. I think I would solve that by allocating an extra, say, 2 seconds and then just don't write the 2 seconds after the sampled-held recHead value: startFrame: recHead + (s.sampleRate * 2). (If it takes 2 seconds to write a 10 second audio file, then you have bigger problems than circular buffers.) Then the record head can move freely into that zone without affecting audio written to disk.

hjh

@KMETE

I realized, after posting that, that there are two ways of thinking about the "extra parameters" going into the synth clone, which for lack of a better term I could call "global" vs "local" parameters.

Local parameters are "per-note" -- they are set at the time the note begins, and they hold their value for the duration of the note. The next note (which might overlap) could have completely different values.

Global parameters affect all notes simultaneously. This is the usual way that, for instance, filter frequency works. You twist the filter cutoff knob and all notes' cutoff changes simultaneously (even though each note could have its own filter cutoff envelope).

At minimum, pitch (note number or frequency) and velocity (on/off trigger) should be per-note. Other parameters could be global or local (and you can mix those styles).

As far as I can see, the best way is:

• All local parameters get packed into a list, along with the clone instance number (which is supplied by [poly]). The "local" list goes into the leftmost input, and only the leftmost.
• "Global" parameters add one or more inlets to the right. Note carefully in the [clone] help file that control inlets need to tag their values for specific instances. If we're using these as global parameters, then the tag should be all. Signal inlets are always global.

IMO it's best to be consistent about the handling of these input types, to avoid introducing bugs into your patch.

Example with filter cutoff and envelope parameters treated globally:

Example with filter cutoff and envelope parameters treated locally:

(Note: The example patches use the ELSE external library, and the global one uses cyclone.)

23-0104-poly.zip

Hope this helps. IMO polysynth is a basic use case for any audio programming environment; if what I've posted here can't be easily figured out from the documentation, then it suggests a gap in the help. (One reason why I worked it out is that I teach a class in Pd; if the students ask "how do I make a poly synth?" then I'd better have an answer at the ready. So I tried a few ways and came up with this as a basic template that Just Works... then, just do it that way.)

hjh

@svanya said:

What I need to do is, for example:

Take soundfiles, aka. tabplay~'s of varying sizes/lengths and shrink them all so they play over the same amount of time.

I think cyclone/play~ is the easiest way to do this. Its "start" message is in the format:

• Time within the array where you want to start playing (ms);
• Time within the array where you want to stop playing (ms);
• Duration to play.

The playback rate is then automatically calculated from (end - start) / dur.

The problem with cyclone buffer~ / play~ is that (AFAICS) they don't handle the situation where the audio file being loaded is at a different sample rate from the soundcard. (Maybe they do, but if so, I don't see it.)

If you're sure that the sound files will match the DSP sample rate, then, no problem -- if one array contains 750 ms of audio and you want it to play in 250 ms, then:

[start 0 750 250(
|
[cyclone/play~ arrayname]

If you're not sure, then one solution would be some abstractions in my library (https://github.com/jamshark70/hjh-abs) that provide [value] variables with information about sound files loaded into arrays. [value xxxscale] is file's sample rate / system sample rate (btw DSP should be on before loading... I should fix that later).

(Edit: You may need to have "-lib cyclone" in the [declare] too.)

22-0810-sf-same-dur-play.pd

hjh

@eulphean said:

I'm using pure data on a raspberry pi 3+ for a project. I have bluetooth configured properly on it. I can send regular audio to bluetooth speaker connected, but the preferences -> audio in PureData doesn't show the bluetooth speaker. It only shows internal audio card or if I have a USB audio card, that shows too. But no bluetooth device.

How do I configure that? Do I need to use jack or something to route audio to bluetooth for Pd?

My experience with Bluetooth audio in Linux has been:

• JACK has zero tolerance for the audio driver ever being late -- expect crashes or system lockups if you try to route audio from JACK to Bluetooth. That is, just don't.

• PulseAudio's support for BT audio is pretty good -- the "regular audio" that you spoke of. Audio production apps typically bypass PulseAudio, in which case BT audio may simply not be supported for them. That is, I expect you'd hit the same problem with SuperCollider, Audacity, Ardour, VCV Rack etc etc etc.

I'm not aware of a solution... That's not to say that there absolutely isn't one, but the Linux audio space is not unified as it is in Mac so audio device support may not be universal.

hjh

I've been trying to understand some of the basic principles of filter design lately, and here is a result of my modest first attempts in that field. It's a patch demonstrating the effect of different pole and zero locations by (graphically) calculating the corresponding frequency responses.

pole-zero-diagram.pd

To avoid confusion, I should probably say some things about how the patch actually works ...

Most importantly, I decided to only use complex filters and to arrange them in conjugate pairs to get real valued outputs at every stage within the cascade of elementary filters. Shortcuts as used in [vcf~] (a one-pole complex filter) aren't possible in this kind of implementation. To get the result corresponding to its left outlet (= real output), you would have to place a (real) zero right in the middle between the two (complex conjugate) poles etc.

The patch allows to use up to six (pairs of) poles or zeros, which may be either real or complex. If less poles or zeros are used, the coefficients of the unused filters are simply set to zero (just as the conjugate poles or zeros are set to zero for the real elementary filters).

By the way: In an earlier version of the patch, I also graphed the phase response of the filter, but I dropped that very soon because I thought this wouldn't be too helpful anyway.

Now for the use: The key control options are explained in the patch itself. Apart from that, you can click and drag the poles and zeros or specify each of its five parameters using the number boxes (values are rounded according to the graphic resolution of one hundredth).

The radii of all poles an zeros are clipped to guarantee stability.

Well, that's about it, I guess.

Bug reports or hints at unexpected behaviour are, as always, very welcome!

Ganymede.7z (includes its own limited set of samples)

Background:

Ganymede was created to test a bet I made with myself:

that I could boil down drum sequencing to a single knob (i.e. instead of writing a pattern).

As far as I am concerned, I won the bet.

The trick is...

Instead of using a knob to turn, for example, up or down a metro, you use it to turn up or down the modulus of a counter, ie. counter[1..16]>[mod X]>[sel 0]>play the sample. If you do this then add an offset control, then where the beat occurs changes in Real-Time.

But you'll have to decide for yourself whether I won the bet. .

(note: I have posted a few demos using it in various stages of its' carnation recently in the Output section of the Forum and intend to share a few more, now that I have posted this.)

Remember, Ganymede is an instrument, i.e. Not an editor.

It is intended to be "played" or...allowed to play by itself.

(aside: specifically designed to be played with an 8-channel, usb, midi, mixer controller and mouse, for instance an Akai Midimix or Novation LaunchPad XL.)

So it does Not save patterns nor do you "write" patterns.

Instead, you can play it and save the audio~ output to a wave file (for use later as a loop, song, etc.)

Jumping straight to The Chase...

How to use it:

REQUIRES:
moonlib, zexy, list-abs, hcs, cyclone, tof, freeverb~ and iemlib

THE 7 SECTIONS:

• GLOBAL:
  • to set parameters for all 8 tracks, exs. pick the samples directory from a tof/pmenu or OPEN_IND_DIR (open an independent directory) (see below "Samples"for more detail)
  • randomizing parameters, random all. randomize all every 10*seconds, maximum number of bars when randomizing bars, CLR the randomizer check boxes
  • PLAY, L(imited) or I(nfinite) counter, if L then number of bars to play before resetting counter, bpm(menu)
  • MSTVOL
  • transport/recording (on REC files are automatically saved to ./ganymede/recordings with datestamp filename, the output is zexy limited to 98 and the volume controls the boost into the limiter)

• PLAYHEADS:
  • indicating where the track is "beating"
  • blank=no beat and black-to-red where redder implies greater env~ rms

• MODULAE:
  • for information only to show the relative values of the selected modulators

• WEIGHTS:
  • sent to [list-wrandom] when randomizing the When, Accent, and Offset modulators
  • to use click READ_ARRAYS, adjust as desired, click WRITE, uncheck READ ARRAYS
  • EVEN=unweighted, RND for random, and 0-7 for preset shapes

• PRESETS:
  • ...self explanatory

• PER TRACK ACCORDION:

  • 8 sections, 1 per track
  • each open-closable with the left most bang/track
  • opening one track closes the previously opened track
  • includes main (always shown)
    • with knobs for the sample (with 300ms debounce)
    • knobs for the modulators (When, Accent, and Offset) [1..16]
    • toggles if you want that parameter to be randomized after X bars
  • and when opened, 5 optional effects
    • adsr, vcf, delayfb, distortion, and reverb
    • D-W=dry-wet
    • 2 parameters per effect

• ALL:
  when ON. sets the values for all of the tracks to the same value; reverts to the original values when turned OFF

MIDI:

CC 7=MASTER VOLUME

The other controls exposed to midi are the first four knobs of the accordion/main-gui. In other words, the Sample, When, Accent, and Offset knobs of each track. And the MUTE and SOLO of each track.

Control is based on a midimap file (./midimaps/midimap-default.txt).

So if it is easier to just edit that file to your controller, then just make a backup of it and edit as you need. In other words, midi-learn and changing midimap files is not supported.

The default midimap is:

By track

CCs

---TRACK---	---SAMPLE---	---WHEN---	---ACCENT---	--- OFFSET---
0	16	17	18	19
1	20	21	22	23
2	24	25	26	27
3	28	29	30	31
4	46	47	48	49
5	50	51	52	53
6	54	55	56	57
7	58	59	60	61

NOTEs

---TRACK---	---MUTE---	---SOLO---
0	1	3
1	4	6
2	7	9
3	10	12
4	13	15
5	16	18
6	19	21
7	22	24

SAMPLES:

Ganymede looks for samples in its ./samples directory by subdirectory.

It generates a tof/pmenu from the directories in ./samples.

Once a directory is selected, it then searches for ./**/.wav (wavs within 1-deep subdirectories) and then ./*.wav (wavs within that main "kit" directory).

I have uploaded my collection of samples (that I gathered from https://archive.org/details/old-school-sample-cds-collection-01, Attribution-Non Commercial-Share Alike 4.0 International Creative Commons License, 90's Old School Sample CDs Collection by CyberYoukai) to the following link on my Google Drive:

https://drive.google.com/file/d/1SQmrLqhACOXXSmaEf0Iz-PiO7kTkYzO0/view?usp=sharing

It is a large 617 Mb .7z file, including two directories: by-instrument with 141 instruments and by-kit with 135 kits. The file names and directory structure have all been laid out according to Ganymede's needs, ex. no spaces, etc.

My suggestion to you is unpack the file into your Path so they are also available for all of your other patches.

MAKING KITS:

I found Kits are best made by adding directories in a "custom-kits" folder to your sampls directory and just adding files, but most especially shortcuts/symlinks to all the files or directories you want to include in the kit into that folder, ex. in a "bongs&congs" folder add shortcuts to those instument folders. Then, create a symnlink to "bongs&congs" in your ganymede/samples directory.

Note: if you want to experiment with kits on-the-fly (while the patch is on) just remember to click the REFRESH bang to get a new tof/pmenu of available kits from your latest ./samples directory.

If you want more freedom than a dynamic menu, you can use the OPEN_IND(depedent)_DIR bang to open any folder. But do bear in mind, Ganymede may not see all the wavs in that folder.

---

AFTERWARD/NOTES

• the [hcs/folder_list] [tof/pmenu] can only hold (the first) 64 directories in the ./samples directory

• the use of 1/16th notes (counter-interval) is completely arbitrary. However, that value (in the [pd global_metro] subpatch...at the noted hradio) is exposed and I will probably incorporate being able to change it in a future version)

• rem: one of the beauties of this technique is: If you don't like the beat,rhythm, etc., you need only click ALL to get an entirely new beat or any of the other randomizers to re-randomize it OR let if do that by itself on AUTO until you like it, then just take it off AUTO.

• One fun thing to do, is let it morph, with some set of toggles and bars selected, and just keep an ear out for the Really choice ones and record those or step in to "play" it, i.e. tweak the effects and parameters. It throws...rolls...a lot of them.

• Another thing to play around with is the notion of Limited (bumpy) or Infinite(flat) sequences in conjunction with the number of bars. Since when and where the modulator triggers is contegent on when it resets.

• Designed, as I said before, to be played, esp. once it gets rolling, it allows you to focus on the production (instead of writing beats) by controlling the ALL and Individual effects and parameters.

• Note: if you really like the beat Don't forget to turn off the randomizers. CLEAR for instance works well. However you can't get the back the toggle values after they're cleared. (possible feature in next version)

• The default.txt preset loads on loadbang. So if you want to save your state, then just click PRESETS>SAVE.

• [folder_list] throws error messages if it can't find things, ex. when you're not using subdirectories in your kit. No need to worry about it. It just does that.

---

POSTSCRIPT

If you need any help, more explanation, advise, or have opinions or insight as to how I can make it better, I would love to hear from you.

I think that's >=95% of what I need to tell you.

If I think of anything else, I'll add it below.

Peace thru Music.
Love thru Pure Data.

-s
,

According to block~ help, if you bang [switch~] it runs one block of DSP computations, which is useful for performing computations that are more easily expressed as audio processing. Something I read (which I can't find now) left me with the impression that it runs faster than normal audio computations, i.e. as if it were in control domain. Here are some tests that confirm it, I think: switch~ bang how fast.pd

The key to this test is that all of the bangs sequenced by [t b b b b] run in the same gap between audio block computations. When [switch~] is banged, [osc~] fills array1, but you can see that element 63 of array1 changes after [switch~] is banged. Furthermore, no logical time has elapsed. So it appears that one block of audio processing has occurred between normal audio blocks. [bang~] outputs when that accelerated audio block processing is complete.

This next test takes things further and bangs [switch~] 10 times at control rate. Still, no logical time elapses, and [bang~] only outputs when all 10 bangs of [switch~] are complete. [rzero_rev~ 0] is just an arcane way of delaying by one sample, so this patch rotates the contents of array1 10 samples to the right. switch~ bang how fast2.pd

(There are better ways to rotate a table than this, but I just needed something to test with. Plus I never pass up a chance to use [rzero_rev~ 0] )

Finally, I've seen some code that sends a 1 to [switch~] and then sends 0 after one block of processing. In this test you can see that one block of audio is processed in one block of logical time, i.e. the normal way. switch~ bang how fast3.pd

But that second test suggests how you could embed arbitrary offline audio processing in a patch that's not being run with Pd's -batch flag or fast-forwarded with the fast-forward message introduced in Pd 0.51-1. Maybe it's an answer to two questions I've seen posted here: Offline analysis on a song and Insant pitch shift. Here's a patch that writes 20s of 440 Hz to a file as fast as possible (adapted from @solipp's patch for the first topic). You just compute how many blocks you need and bang away. write440File.zip

Here's another that computes the real FFT of an audio file as fast as possible: loadFFT.zip

But as with any control rate processing, if you try to do too much this way, Pd will fall behind in normal audio processing and stutter (e.g. listen to the output while running that last patch on a >1 minute file). So no free lunch, just a little subsidy.

Just Another (Drum) Sequencer...SortOf, codename: Virgo

virgo.zip

REQUIRES: zexy, moonlib, tof (as of Pd 0.50.2, all of which are in deken) and hcs (which comes by default with Pd 0.50.2 and is in deken (for extended))

Special Features

• Unique playhead per row; each with their own metro (beat)
• Up to 8 Volume states-per-beat (by clicking multiple times on the bang; where an rms=1 is divide among the states (2 states:0=rms=0(black), 1=rms=1(red); 3 states:rms=[0|0.5|1])
• Design approach: using creation arguments to alias abstractions, so subsequently they are referred to by their creation arguments, ex. in [KITS sample] sample is referred to as [$1]; which is how they are listed below)

(notes: what I learned experimenting with this design approach, I will share as a separate post. Currently, it does not include cut-copy-paste (of regions of the pattern)). I good way to start trying it out is clicking the "R" to get a random kit and a random pattern).

virgo:[virgo/PROJECT KITS PATTERNS]

• PROJECT[KITS PATTERNS]
  • $1:[KITS sample]
    • GUI
      • K: openpanel to load a previously saved *.txt (text object) kit of samples; on loadbang the default.txt kit is loaded
      • S: save the current set of samples to the most recently opened *.txt (kit) preset
      • SA: saveas a *.txt of the current set of samples
      • D: foldererpanel a sample directory to load the first (alphabetically) 16 samples into the 16 slots
      • RD: load a random kit from the [text samples] object where the samples where previously loaded via the "SAMPLES" bang on the right
      • U: undo; return to the previously opened or saved *.txt kit, so not the previously randomized
      • MASTER: master gain
      • (recorder~: of the total audio~ out)
        • record
        • ||: pause; either recording or play;
        • play: output is combined with the sequencer output just before MASTER out to [dac~]
      • SAMPLES: folderpanel to load a (recursive) directory of samples for generating random kits
    • ABSTRACTIONS
      • $1: sample
        • bang: openpanel to locate and load a sample for a track
        • canvas: filename of the opened sample; filenames are indexed in alignment with track indices in the PATTERNS section
  • $2:[PATTERNS row]
    • GUI
      • P: openpanel to load a previously saved *.txt (pattern) preset file; on loadbang the default.txt pattern is loaded; the preset file includes the beat, pattern, and effect settings for the row
      • S: save the current pattern to the most recently opened pattern .txt
      • SA: save as (self-explanatory)
      • states: the number of possible states [2..8] of each beat;
      • %: weight; chance of a beat being randomized; not chance of what it will result in; ex. 100% implies all beats are randomized ; random beats result in a value)gain) between 1 and states-1
      • PLAY(reset): play the pattern from "start" or on stop reset all playheads to start
      • start: which beat to start the playheads on
      • length: how many beats to play [+/-32]; if negative the playheads will play in reverse/from right to left
      • bpm: beats-per-minute
      • rate: to change the rate of play (ie metro times) by the listed factor for all playheads
      • R: randomize the total pattern (incl period and beats, but not the effect settings; beats of 1/32 are not included in the possibilities)
      • CL: clear, set all beats to "0", i.e. off
      • U: undo random; return to the previously opened or saved preset, ie. not the previous random one
      • M: mute all tracks; the playheads continue moving but audio does not come out of any track
      • ||:pause all playheads; play will resume from that location when un-paused
      • per: period; if 0=randomizes the period, >0 sets the period to be used for all beats
      • Edit Mode
        • Check the [E] to enter edit mode (to cut, copy, or paste selected regions of the pattern)
          • Entering edit mode will pause the playing of the pattern
          • Play, if doing so beforehand, will resume on leavng edit mode
          • The top-left most beat of the pattern grid will be selected when first entering edit mode
        • Single-click a beat to select the top-left corner of the region you wish to cut or copy
        • Double-click a beat to select the bottom-right corner
          • You may not double-click a beat "less than" the single-clicked (top-left) beat and vice-versa
        • Click [CL] to clear your selection (i.e. start over)
        • The selected region will turn to dark colors
        • If only one beat is selected it will be the only one darkened
        • Click the operation (bang) you wish to perform, either cut [CU] or copy [CP]
          • Then, hold down the CTRL key and click the top-left corner of where you want to paste the region
          • The clicked cell will turn white
          • And click [P] to paste the region
        • Cut and copied regions may both be pasted multiple times
          • The difference being, cutting sets the values (gains) for the originating region to "0"
        • Click [UN] to undo either the cut, copy, or paste operation
          • Undoing cut will return the gains from 0s to their original value
      • (effect settings applied to all tracks)
        • co: vcf-cutoff
        • Q: vcf-q
        • del: delay-time
        • fb: delay-feedback
        • dist: distortion
        • reverb
        • gn: gain
    • ABSTRACTIONS
      • $1: [row (idx) b8] (()=a property not an abstraction)
        • GUI
          • (index): aligns with the track number in the KITS section
          • R: randomize the row; same as above, but for the row
          • C: clear the row, i.e. set all beats to 0
          • U: undo the randomize; return to the originally opened one, ie. not the previous random one
          • M: mute the row, so no audio plays, but the playhead continues to play
          • S: solo the row
          • (beat): unit of the beat(period); implying metro length (as calculated with the various other parameters);1/32,1/16,1/8, etc.
          • (pattern): the pattern for the row; single-click on a beat from 0 to 8 times to increment the gain of that beat as a fraction of 1 rms, where resulting rms=value/states; black is rms=0; if all beats for a row =0 (are black) then the switch for that track is turned off; double-click it to decrement it
          • (effects-per-row): same as above, but per-row, ex. first column is vcf-cutoff, second is vcf-q, etc.
        • ABSTRACTIONS
          • $1: b8 (properties:row column)
            • 8-state bang: black, red, orange, yellow, green, light-blue, blue, purple; representing a fraction of rms(gain) for the beat

Credits: The included drum samples are from: https://www.musicradar.com/news/sampleradar-494-free-essential-drum-kit-samples

p.s. Though I began working on cut-copy-paste, it began to pose a Huge challenge, so backed off, in order to query the community as to 1) its utility in the current state (w/o that) and 2) just how important including it really is.

p.p.s. Please, report any inconsistencies (between the instructions as listed and what it does) and/or bugs you may find, and I will try to get an update posted as soon as enough of those have collect.

Love and Peace through sharing,
Scott

I'm running a Raspberry Pi 4 with "Raspberry Pi OS (32-bit) Lite [August 2020 / 2020-08-20 / Kernel 5.4]" headless and the latest version of Purr-Data, set up the Raspberry using this guide , did the Alsa no audio (glitch) issue Pi 4

fix to get audio working through the 3.5 mm jack.

I'm interfacing the Raspberry Pi through the macOS terminal and with VNC Viewer through ethernet.

I get the following error as soon a starting Purr-Data,

error: audio I/O stuck... closing audio 

I open up a patch and turn off and turn on the DSP signals and get the following error

error: audio I/O stuck... closing audio error: audio I/O dropout 

Even though if I play sample .wav or bang a sinewave, the audio does get played but it gets glitchy as in the audio sound is crackled.

has anyone experienced it before? any knowledge on how to overcome it?