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35 result(s) matching "audio-analysis-qlc", (0.03 seconds)

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

posted in patch~

NoxSiren - Modular synthesizer system <- [v15]

NoxSiren is a modular synthesizer system where the punishment of failure is the beginning of a new invention.

--DOWNLOAD-- NoxSiren for :

Pure Data :
NoxSiren v15.rar
NoxSiren v14.rar
Purr Data :
NoxSiren v15.rar
NoxSiren v14.rar

--DOWNLOAD-- ORCA for :

x64, OSX, Linux :
https://hundredrabbits.itch.io/orca

In order to connect NoxSiren system to ORCA system you also need a virtual loopback MIDI-ports:

--DOWNLOAD-- loopMIDI for :

Windows 7 up to Windows 10, 32 and 64 bit :
https://www.tobias-erichsen.de/software/loopmidi.html

#-= Cyber Notes [v15] =-#

added BORG-IMPLANT module.
introduction to special modules.
more system testing.

#-= Special Modules [v15] =-#

BORG-IMPLANT (connects ORCA MIDI system to NoxSiren system)

#-= Current Modules [v15] =-#

VCO (voltage-controlled-oscillator)
VCO2 (advance voltage-controlled-oscillator)
WAVEBANK (additive synthesis oscillator)
ADSR (Attack-Decay-Sustain-Release envelope)
C-ADSR (Curved Attack-Decay-Sustain-Release envelope)
CICADAS (128 steps-Euclidean rhythm generator)
CICADAS-2 (advance 128-steps polymorphic-Euclidean rhythm generator)
COMPRESSOR (lookahead mono compressor unit)
DUAL-COMPRESSOR (2-channel lookahead mono compressor unit)
STEREO-COMPRESSOR (lookahead stereo compressor unit)
MONO-KEYS (virtual 1-voice monophonic MIDI keyboard)
POLY-KEYS-2 (virtual 2-voice polyphonic MIDI keyboard)
POLY-KEYS-3 (virtual 3-voice polyphonic MIDI keyboard)
POLY-KEYS-4 (virtual 4-voice polyphonic MIDI keyboard)
POLY-KEYS-5 (virtual 5-voice polyphonic MIDI keyboard)
POLY-KEYS-6 (virtual 6-voice polyphonic MIDI keyboard)
BATTERY (simple manual triggered machine for drumming.)
REVERB (reverb unit with lowpass control)
STEREO-REVERB (stereo reverb unit with lowpass control)
RESIN (advanced rain effect/texture generator)
NOISE (generates black,brown,red and orange noise)
NOISE2 (generates yellow,blue,pink and white noise)
COBALT (6-stage polyrhythm generator)
SHAPER (basic shaper unit)
FOLDER (basic wave folding unit)
STEREO-FOLDER (stereo wave folding unit)
DUAL-FOLDER (advance wave folding unit)
POLARIZER (transform a signal into bi-polar, uni-polar, inverted or inverted uni-polar form)
CLOCK (generates a BPM clock signal for sequencing other modules)
CLOCKDIVIDER (a clock divider with even division of clock signal)
CLOCKDIVIDER2 (a clock divider with odd division of clock signal)
DELAY-UNIT (delay unit)
STEREO-DELAY (stereo delay unit)
CHORUS (chorus unit)
STEREO-CHORUS (stereo chorus unit)
SEQ (advance 16-step/trigger sequencer)
KICK (synthesize kick unit)
KICK2 (synthesize flavor of KICK module)
KICK3 (synthesize flavor of KICK module)
SNARE (synthesize snare unit)
CLAP (synthesize clap unit)
CYMBAL (synthesize cymbal unit)
RAND (RNG generator for other modules parameters)
FMOD (feedback modulation unit)
AM (amplitude modulation unit)
RM (ring modulation unit)
LFO (low-frequency-oscillator)
LFO2 (advance low-frequency-oscillator)
COMBINATOR (combine two waves)
COMBINATOR2 (combine three waves)
COMBINATOR3 (combine four waves)
STRING (Karplus-Strong string synthesis unit)
STRING2 (advance Karplus-Strong string synthesis unit)
DETUNER (parametric 4-channel detuner unit)
CRUSHER (basic audio resolution unit)
STEREO-CRUSHER (basic stereo audio resolution unit)
DUAL-CRUSHER (advance audio resolution unit)
FILTER (basic filter)
VCF (voltage-controlled-filter)
MAR (Moog-analog-resonant filter)
VCA (voltage-controlled-amplifier)
DUAL-VCA (advance voltage-controlled-amplifier)
FMUX (multiplexer with fast A/D internal envelope)
MMUX (multiplexer with medium A/D internal envelope)
SMUX (multiplexer with slow A/D internal envelope)
FDMX (demultiplexer with fast A/D internal envelope)
MDMX (demultiplexer with medium A/D internal envelope)
SDMX (demultiplexer with slow A/D internal envelope)
MIXER (mix 1-4 possible waves)
SCOPE (oscilloscope analyzer)
MASTER (fancy DAC~)
BOX (useless decorative module)

NoxSiren integrated modules menu system.

posted in patch~

Crackled Audio from PD on Raspberry Pi 4

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?

posted in technical issues

vstplugin~ 0.2.0

mda Delay.vst.zip

[vstplugin~] v0.2.0
WARNING: on macOS, the VST GUI must run on the audio thread - use with care!
searching in '/Users/boonier/Library/Audio/Plug-Ins/VST' ...
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/BreakBeatCutter.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/Camomile.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/Euklid.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/FmClang.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/Micropolyphony.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/PhaserLFO.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/pvsBuffer.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/smGrain3.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/smHostInfo.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/smMetroTests.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/smModulatingDelays.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/smTemposcalFilePlayer.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/smTrigSeq.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/SoundwarpFilePlayer.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/SpectralDelay.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/SyncgrainFilePlayer.vst'... failed!
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/Vocoder.vst'... failed!
found 0 plugins
searching in '/Library/Audio/Plug-Ins/VST' ...
probing '/Library/Audio/Plug-Ins/VST/++bubbler.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/++delay.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/++flipper.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/++pitchdelay.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/ABL2x.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/BassStation.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/BassStationStereo.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Camomile.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Crystal.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Ctrlr.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Dexed.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Driftmaker.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/GTune.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Independence FX.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Independence.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/JACK-insert.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Lua Protoplug Fx.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Lua Protoplug Gen.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Ambience.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Bandisto.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda BeatBox.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Combo.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda De-ess.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Degrade.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Delay.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Detune.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Dither.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda DubDelay.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda DX10.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Dynamics.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda ePiano.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Image.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Leslie.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Limiter.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Looplex.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Loudness.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda MultiBand.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Overdrive.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Piano.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda RePsycho!.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda RezFilter.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda RingMod.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda RoundPan.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Shepard.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Splitter.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Stereo.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda SubBass.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda TestTone.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda ThruZero.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Tracker.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Transient.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda VocInput.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mda Vocoder.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mdaJX10.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/mdaTalkBox.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/ME80v2_3_Demo.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Metaplugin.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/MetapluginSynth.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Molot.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Nektarine.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Nektarine_32OUT.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Nithonat.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Obxd.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Ozone 8 Elements.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/PlogueBiduleVST.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/PlogueBiduleVST_16.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/PlogueBiduleVST_32.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/PlogueBiduleVST_64.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/PlogueBiduleVSTi.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/PlogueBiduleVSTi_16.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/PlogueBiduleVSTi_32.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/PlogueBiduleVSTi_64.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/sforzando.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Sonic Charge/Cyclone FX.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Sonic Charge/Cyclone.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Soundtoys/Devil-Loc.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Soundtoys/LittlePlate.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Soundtoys/LittleRadiator.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Soundtoys/SieQ.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/SPAN.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Spitter2.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Surge.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/Synth1.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/TAL-Chorus-LX.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/TAL-Reverb-2.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/TAL-Reverb-3.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/TAL-Reverb-4.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/TAL-Sampler.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/TX16Wx.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/u-he/Diva.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/u-he/Protoverb.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/u-he/Repro-1.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/u-he/Repro-5.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/u-he/Satin.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/u-he/TyrellN6.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/u-he/Zebra2.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/u-he/Zebralette.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/u-he/Zebrify.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/u-he/ZRev.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/UltraChannel.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/ValhallaFreqEcho.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/ValhallaRoom_x64.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/VCV-Bridge-fx.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/VCV-Bridge.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/WaveShell1-VST 10.0.vst'... failed!
found 0 plugins
searching in '/Users/boonier/Library/Audio/Plug-Ins/VST3' ...
found 0 plugins
searching in '/Library/Audio/Plug-Ins/VST3' ...
probing '/Library/Audio/Plug-Ins/VST3/TX16Wx.vst3'... error
couldn't init module

probing '/Library/Audio/Plug-Ins/VST3/WaveShell1-VST3 10.0.vst3'... error
factory doesn't have any plugin(s)

probing '/Library/Audio/Plug-Ins/VST3/Nektarine.vst3'... failed!
probing '/Library/Audio/Plug-Ins/VST3/Nektarine_32OUT.vst3'... failed!
probing '/Library/Audio/Plug-Ins/VST3/OP-X PRO-II.vst3'... failed!
probing '/Library/Audio/Plug-Ins/VST3/SPAN.vst3'... failed!
probing '/Library/Audio/Plug-Ins/VST3/Surge.vst3'... failed!
probing '/Library/Audio/Plug-Ins/VST3/Zebra2.vst3'...
[1/4] 'Zebrify' ... failed!
[2/4] 'ZRev' ... failed!
[3/4] 'Zebra2' ... failed!
[4/4] 'Zebralette' ... failed!
found 0 plugins
search done
print: search_done

posted in news

Scheduler queue object?

@ddw_music The time variance is because the event was triggered within the last audio block and arrives before the next block is processed. So as the beat clock is not a perfect division of the audio clock it has to skip to compensate.
You are right that event inaccuracy at that scale is not important, so Link is quite accurate enough for events.
But for the audio it is vital that the programs are in sync because our brains are wired to detect phase, but more importantly the program hasn't got a clue what to do with data that doesn't match its expectations.
It doesn't matter what the sample rate is.
The point is that it is the same for all the programs that are linked by the audio, and so they have to be in sync.
They are using that common distributed clock, with a natural sync every 64 samples, if that is the block size chosen for the audio..... so every 1.45ms roughly at 44.1 with a 64 sample buffer.
They all have to process those chunks and spit them out in sync.
It is not difficult for a modern computer to calculate time at every sample, or we would not be blessed with ASIO, DAW's and all. My DAW will show me the calculated time stamp of every sample within the 64 sample window.
Every block must pass or there would be dropouts somewhere.
The soundcard also uses the same clock as it counts it's way inexorably through the blocks.
The computer doesn't make mistakes as it counts through the blocks, unless another process blocks the calculations for too long because of a deferred procedure call. DPC latency is something to look out for when running audio applications and there are some tools available for analysis.
But the answer is no, in that there is no inaccuracy in a computer until you reach the limits of its bit depth.
David.

posted in technical issues

PD's scheduler, timing, control-rate, audio-rate, block-size, (sub)sample accuracy,

Hello,

this is going to be a long one.
After years of using PD, I am still confused about its' timing and schedueling.

I have collected many snippets from here and there about this topic,
-wich all together are really confusing to me.

I think it is very important to understand how timing works in detail for low-level programming …

(For example the number of heavy jittering sequencers in hard and software make me wonder what sequencers are made actually for ? lol )

This is a collection of my findings regarding this topic, a bit messy and with confused questions.

I hope we can shed some light on this.

a)
The first time, I had issues with the PD-scheduler vs. how I thought my patch should work is described here:
https://forum.pdpatchrepo.info/topic/11615/bang-bug-when-block-1-1-1-bang-on-every-sample

The answers where:

„
[...] it's just that messages actually only process every 64 samples at the least. You can get a bang every sample with [metro 1 1 samp] but it should be noted that most pd message objects only interact with each other at 64-sample boundaries, there are some that use the elapsed logical time to get times in between though (like vsnapshot~)
also this seems like a very inefficient way to do per-sample processing..
https://github.com/sebshader/shadylib http://www.openprocessing.org/user/29118
seb-harmonik.ar posted about a year ago , last edited by seb-harmonik.ar about a year ago 
• 1 
 whale-av 
@lacuna An excellent simple explanation from @seb-harmonik.ar.
Chapter 2.5 onwards for more info....... http://puredata.info/docs/manuals/pd/x2.htm
David.
“

There is written: http://puredata.info/docs/manuals/pd/x2.htm

„2.5. scheduling 
Pd uses 64-bit floating point numbers to represent time, providing sample accuracy and essentially never overflowing. Time appears to the user in milliseconds. 
2.5.1. audio and messages 
Audio and message processing are interleaved in Pd. Audio processing is scheduled every 64 samples at Pd's sample rate; at 44100 Hz. this gives a period of 1.45 milliseconds. You may turn DSP computation on and off by sending the "pd" object the messages "dsp 1" and "dsp 0." 
In the intervals between, delays might time out or external conditions might arise (incoming MIDI, mouse clicks, or whatnot). These may cause a cascade of depth-first message passing; each such message cascade is completely run out before the next message or DSP tick is computed. Messages are never passed to objects during a DSP tick; the ticks are atomic and parameter changes sent to different objects in any given message cascade take effect simultaneously. 
In the middle of a message cascade you may schedule another one at a delay of zero. This delayed cascade happens after the present cascade has finished, but at the same logical time. 
2.5.2. computation load 
The Pd scheduler maintains a (user-specified) lead on its computations; that is, it tries to keep ahead of real time by a small amount in order to be able to absorb unpredictable, momentary increases in computation time. This is specified using the "audiobuffer" or "frags" command line flags (see getting Pd to run ). 
If Pd gets late with respect to real time, gaps (either occasional or frequent) will appear in both the input and output audio streams. On the other hand, disk strewaming objects will work correctly, so that you may use Pd as a batch program with soundfile input and/or output. The "-nogui" and "-send" startup flags are provided to aid in doing this. 
Pd's "realtime" computations compete for CPU time with its own GUI, which runs as a separate process. A flow control mechanism will be provided someday to prevent this from causing trouble, but it is in any case wise to avoid having too much drawing going on while Pd is trying to make sound. If a subwindow is closed, Pd suspends sending the GUI update messages for it; but not so for miniaturized windows as of version 0.32. You should really close them when you aren't using them. 
2.5.3. determinism 
All message cascades that are scheduled (via "delay" and its relatives) to happen before a given audio tick will happen as scheduled regardless of whether Pd as a whole is running on time; in other words, calculation is never reordered for any real-time considerations. This is done in order to make Pd's operation deterministic. 
If a message cascade is started by an external event, a time tag is given it. These time tags are guaranteed to be consistent with the times at which timeouts are scheduled and DSP ticks are computed; i.e., time never decreases. (However, either Pd or a hardware driver may lie about the physical time an input arrives; this depends on the operating system.) "Timer" objects which meaure time intervals measure them in terms of the logical time stamps of the message cascades, so that timing a "delay" object always gives exactly the theoretical value. (There is, however, a "realtime" object that measures real time, with nondeterministic results.) 
If two message cascades are scheduled for the same logical time, they are carried out in the order they were scheduled. 
“

[block~ smaller then 64] doesn't change the interval of message-control-domain-calculation?,
Only the size of the audio-samples calculated at once is decreased?
Is this the reason [block~] should always be … 128 64 32 16 8 4 2 1, nothing inbetween, because else it would mess with the calculation every 64 samples?
How do I know which messages are handeled inbetween smaller blocksizes the 64 and which are not?

How does [vline~] execute?
Does it calculate between sample 64 and 65 a ramp of samples with a delay beforehand, calculated in samples, too - running like a "stupid array" in audio-rate?

While sample 1-64 are running, PD does audio only?

[metro 1 1 samp]

How could I have known that? The helpfile doesn't mention this. EDIT: yes, it does.
(Offtopic: actually the whole forum is full of pd-vocabular-questions)
How is this calculation being done?
But you can „use“ the metro counts every 64 samples only, don't you?

Is the timing of [metro] exact? Will the milliseconds dialed in be on point or jittering with the 64 samples interval?
Even if it is exact the upcoming calculation will happen in that 64 sample frame!?

b )

There are [phasor~], [vphasor~] and [vphasor2~] … and [vsamphold~]
https://forum.pdpatchrepo.info/topic/10192/vphasor-and-vphasor2-subsample-accurate-phasors

“Ive been getting back into Pd lately and have been messing around with some granular stuff. A few years ago I posted a [vphasor.mmb~] abstraction that made the phase reset of [phasor~] sample-accurate using vanilla objects. Unfortunately, I'm finding that with pitch-synchronous granular synthesis, sample accuracy isn't accurate enough. There's still a little jitter that causes a little bit of noise. So I went ahead and made an external to fix this issue, and I know a lot of people have wanted this so I thought I'd share. 
[vphasor~] acts just like [phasor~], except the phase resets with subsample accuracy at the moment the message is sent. I think it's about as accurate as Pd will allow, though I don't pretend to be an expert C programmer or know Pd's api that well. But it seems to be about as accurate as [vline~]. (Actually, I've found that [vline~] starts its ramp a sample early, which is some unexpected behavior.)
[…]
“

c)

Later I discovered that PD has jittery Midi because it doesn't handle Midi at a higher priority then everything else (GUI, OSC, message-domain ect.)

EDIT:
Tryed roundtrip-midi-messages with -nogui flag:
still some jitter.
Didn't try -nosleep flag yet (see below)

d)

So I looked into the sources of PD:

scheduler with m_mainloop()
https://github.com/pure-data/pure-data/blob/master/src/m_sched.c

And found this paper

Scheduler explained (in German):

https://iaem.at/kurse/ss19/iaa/pdscheduler.pdf/view

wich explains the interleaving of control and audio domain as in the text of @seb-harmonik.ar with some drawings
plus the distinction between the two (control vs audio / realtime vs logical time / xruns vs burst batch processing).
And the "timestamping objects" listed below.

And the mainloop:

Loop
- messages (var.duration)
- dsp (rel.const.duration)
- sleep

With
[block~ 1 1 1]
calculations in the control-domain are done between every sample? But there is still a 64 sample interval somehow?
Why is [block~ 1 1 1] more expensive? The amount of data is the same!? Is this the overhead which makes the difference? Calling up operations ect.?

Timing-relevant objects
from iemlib:

[...]
iem_blocksize~   blocksize of a window in samples
iem_samplerate~  samplerate of a window in Hertz 


------------------ t3~ - time-tagged-trigger --------------------
-- inputmessages allow a sample-accurate access to signalshape --
t3_sig~          time tagged trigger sig~
t3_line~         time tagged trigger line~


--------------- t3 - time-tagged-trigger  ---------------------
----------- a time-tag is prepended to each message -----------
----- so these objects allow a sample-accurate access to ------
---------- the signal-objects t3_sig~ and t3_line~ ------------
t3_bpe           time tagged trigger break point envelope
t3_delay         time tagged trigger delay
t3_metro         time tagged trigger metronom
t3_timer         time tagged trigger timer

[...]

What are different use-cases of [line~] [vline~] and [t3_line~]?
And of [phasor~] [vphasor~] and [vphasor2~]?

When should I use [block~ 1 1 1] and when shouldn't I?

[line~] starts at block boundaries defined with [block~] and ends in exact timing?
[vline~] starts the line within the block?
and [t3_line~]???? Are they some kind of interrupt? Shortcutting within sheduling???

c) again)

https://forum.pdpatchrepo.info/topic/1114/smooth-midi-clock-jitter/2

I read this in the html help for Pd: 
„
MIDI and sleepgrain
In Linux, if you ask for "pd -midioutdev 1" for instance, you get /dev/midi0 or /dev/midi00 (or even /dev/midi). "-midioutdev 45" would be /dev/midi44. In NT, device number 0 is the "MIDI mapper", which is the default MIDI device you selected from the control panel; counting from one, the device numbers are card numbers as listed by "pd -listdev."
The "sleepgrain" controls how long (in milliseconds) Pd sleeps between periods of computation. This is normally the audio buffer divided by 4, but no less than 0.1 and no more than 5. On most OSes, ingoing and outgoing MIDI is quantized to this value, so if you care about MIDI timing, reduce this to 1 or less.
„

Why is there the „sleep-time“ of PD? For energy-saving??????
This seems to slow down the whole process-chain?

Can I control this with a startup flag or from withing PD? Or only in the sources?

There is a startup-flag for loading a different scheduler, wich is not documented how to use.

e)

[pd~] helpfile says:

ATTENTION: DSP must be running in this process for the sub-process to run. This is because its clock is slaved to audio I/O it gets from us!

Doesn't [pd~] work within a Camomile plugin!?

How are things scheduled in Camomile? How is the communication with the DAW handled?

f)

and slightly off-topic:
There is a batch mode:
https://forum.pdpatchrepo.info/topic/11776/sigmund-fiddle-or-helmholtz-faster-than-realtime/9

EDIT:

g)
I didn't look into it, but there is:
https://grrrr.org/research/software/

clk – Syncable clocking objects for Pure Data and Max
This library implements a number of objects for highly precise and persistently stable timing, e.g. for the control of long-lasting sound installations or other complex time-related processes.

Sorry for the mess!
Could you please help me to sort things a bit? Mabye some real-world examples would help, too.

posted in technical issues

abl_link~ midi and audio sync setup

Hi Folks,

I thought I’d share this patch in the hopes that someone might be able to help improve upon it. I am by no means even semi competent with PD and jumped into this task without actually bothering to learn the basics of PD or RPi, but nevertheless here we are: maybe you can share a better implementation.

Mods/experienced folks, if I am sharing irrelevant/wrong/confusing info, mea culpa and please correct me.

I wanted to make a patch for PD in Raspberry Pi that would do 3 things:

Get the abl_link~ temp data over wifi
Create a midi clock output using a 5-pin midi adapter (I have one of the cheapo usb-to-midi cable things here)
-simultaneously create an audio pulse ‘clock’ output such as those used by volcas, Teenage Engineering Pocket operators, and the like (I am not sure if such an audio signal over a 3.5mm jack would be hot enough to be considered a CV pulse too, maybe you can help clear that up?)

As I say, after much struggles I have globbed something together that sort of does this.

A couple of things for newcomers like myself:

The abl_link~ object in the patch isn’t initially part of the standard pure data install as I write. I was able to use deken (ie the code that powers the ‘help/find externals’ bit of PD) to look for abl_link~. Search for it. At the time of writing there is a version for Arm7 devices like the Raspberry Pi 3 which was put together by the illustrious mzero with code from antlr. Go ahead and install the abl_link~ object. (Possibly you may have to uncheck the ‘hide foreign architectures’ box to get the arm7 version to show up. This is usually a safeguard to stop users from trying to install versions of externals that won’t work on their systems. So long as you see ‘arm7’ in the description it should hopefully be the one you want) PD will ask where you want to store the external, and I would just leave it at the default unless you have a special reason to do otherwise.

To get the patch to hook up to your preferred audio and midi outputs by default you may have to take certain steps. In my version of it I have deemed the built in audio and my cheapo USB midi output to be good enough for this task.

[As part of my troubleshooting process I ended up installing amidiauto which is linked to here: https://community.blokas.io/t/script-for-launching-pd-patch-with-midi-without-aconnect/1010/2
I undertook several installations in support of amidiauto which may be helping my system to see and link up my USB midi and PD, but nothing worked until I took the step in the following paragraph about startup flags in PD. (It may also be that I did not need to put in amidiauto at all. Maybe I’ll try that on another card to see if it simplifies the process. I’m saying you might want to try it without amidiauto first to see).]

Midi: - (ALSA is the onboard audio and midi solution that is part of Raspbian). To have PD use ALSA midi at the start I made the following setting in the preferences/startup dialog - within that window there is a section (initially blank) for startup flags. Here you can set instructions for PD to take note of when it starts up. I put in -alsamidi to tell it that alsamidi will be my preferred midi output. (I also took the step of going to file/preferences/midi settings, then ‘apply’ and ‘ok’ to confirm the Alsa midi ports that showed up. Then I went back to file/preferences/save all preferences. This seems to have (fingers crossed) saved the connection to my USB midi output.

Audio: I used the terminal and sudo raspi-config to set my audio out to the internal sound card (advanced options/audio/3.5mm jack). Since I had a fairly unused installation of PD I’d never asked it to do anything but work with the system defaults so getting audio out was fairly simple.

[nb I initially stuck this patch together on my Mac where everything worked pretty trouble free in terms of audio and midi selection]

About the patch. Obviously it is sort of horrible but there it is. It is a combination of stuff I cribbed from the demo example of abl_link~ in the example, and two example patches created by users NoDSP and jpg in this forum post https://forum.pdpatchrepo.info/topic/9545/generate-midi-clock-messages-from-pd/2
As well as some basic synthesis to make the bip bip noises I learned from LWMs youtube channel
https://www.youtube.com/channel/UCw5MbnaoDDuRPFsqaQpU9ig

Any and all errors and bad practice are mine alone.
The patch has some comments in it that doubtless expose my own lack of understanding more than anything. Undoubtedly many users can do a better job than I can.

Some observations on limitations/screwups of the patch:

If you disconnect from the stream for a bit, it will attempt to catch up. There will be a massive flurry of notes and/or audio bips as it plays all the intervening notes.
It doesn’t seem to be too fussy about where in the bar it is getting started (It will be "on" the beat but sometimes the ‘1’will be the ‘2’ etc. This is okay if I’m using internal sequencers from scratch (in the volca, say) but not if there is an existing pattern that I am trying to have come in 'on the 1'.
My solution to more detailed subdivision of bars was to make a big old list of numbers up to 32 so that abl_link~ can count up to more than 4. There’s probably a better solution for this. If you find that you need even more subdivisions because you are making some sort of inhumanly manic speed gabba, add even yet more numbers and connections.

I haven’t tested this much. And since it’s taken me the better part of 18 months to do this at all, I’m really not your guy to make it work any better. I’m posting here so that wiser souls can do a better job and maybe share what I think has the potential to be a useful midi sync tool.

I plan to revisit https://community.blokas.io/t/script-for-launching-pd-patch-with-midi-without-aconnect/1010/3
for some pointers on setting this up to launch the patch at startup to give me a small, portable midi Link sync device for 5-pin and audio-pulse clocked devices.

This is my first ever bit of quasi productive input to any technical community (mostly I just hang around asking dumb questions… So be kind and please use your giant brains to make it better) I look forward to spending some time learning the basics now. link-sync.pd

posted in patch~

Web Audio Conference 2019 - 2nd Call for Submissions & Keynotes

Apologies for cross-postings

Fifth Annual Web Audio Conference - 2nd Call for Submissions

https://www.ntnu.edu/wac2019

The fifth Web Audio Conference (WAC) will be held 4-6 December, 2019 at the Norwegian University of Science and Technology (NTNU) in Trondheim, Norway. WAC is an international conference dedicated to web audio technologies and applications. The conference addresses academic research, artistic research, development, design, evaluation and standards concerned with emerging audio-related web technologies such as Web Audio API, Web RTC, WebSockets and Javascript. The conference welcomes web developers, music technologists, computer musicians, application designers, industry engineers, R&D scientists, academic researchers, artists, students and people interested in the fields of web development, music technology, computer music, audio applications and web standards. The previous Web Audio Conferences were held in 2015 at IRCAM and Mozilla in Paris, in 2016 at Georgia Tech in Atlanta, in 2017 at the Centre for Digital Music, Queen Mary University of London in London, and in 2018 at TU Berlin in Berlin.

The internet has become much more than a simple storage and delivery network for audio files, as modern web browsers on desktop and mobile devices bring new user experiences and interaction opportunities. New and emerging web technologies and standards now allow applications to create and manipulate sound in real-time at near-native speeds, enabling the creation of a new generation of web-based applications that mimic the capabilities of desktop software while leveraging unique opportunities afforded by the web in areas such as social collaboration, user experience, cloud computing, and portability. The Web Audio Conference focuses on innovative work by artists, researchers, students, and engineers in industry and academia, highlighting new standards, tools, APIs, and practices as well as innovative web audio applications for musical performance, education, research, collaboration, and production, with an emphasis on bringing more diversity into audio.

Keynote Speakers

We are pleased to announce our two keynote speakers: Rebekah Wilson (independent researcher, technologist, composer, co-founder and technology director for Chicago’s Source Elements) and Norbert Schnell (professor of Music Design at the Digital Media Faculty at the Furtwangen University).
More info available at: https://www.ntnu.edu/wac2019/keynotes

Theme and Topics

The theme for the fifth edition of the Web Audio Conference is Diversity in Web Audio. We particularly encourage submissions focusing on inclusive computing, cultural computing, postcolonial computing, and collaborative and participatory interfaces across the web in the context of generation, production, distribution, consumption and delivery of audio material that especially promote diversity and inclusion.

Further areas of interest include:

Web Audio API, Web MIDI, Web RTC and other existing or emerging web standards for audio and music.
Development tools, practices, and strategies of web audio applications.
Innovative audio-based web applications.
Web-based music composition, production, delivery, and experience.
Client-side audio engines and audio processing/rendering (real-time or non real-time).
Cloud/HPC for music production and live performances.
Audio data and metadata formats and network delivery.
Server-side audio processing and client access.
Frameworks for audio synthesis, processing, and transformation.
Web-based audio visualization and/or sonification.
Multimedia integration.
Web-based live coding and collaborative environments for audio and music generation.
Web standards and use of standards within audio-based web projects.
Hardware and tangible interfaces and human-computer interaction in web applications.
Codecs and standards for remote audio transmission.
Any other innovative work related to web audio that does not fall into the above categories.

Submission Tracks

We welcome submissions in the following tracks: papers, talks, posters, demos, performances, and artworks. All submissions will be single-blind peer reviewed. The conference proceedings, which will include both papers (for papers and posters) and extended abstracts (for talks, demos, performances, and artworks), will be published open-access online with Creative Commons attribution, and with an ISSN number. A selection of the best papers, as determined by a specialized jury, will be offered the opportunity to publish an extended version at the Journal of Audio Engineering Society.

Papers: Submit a 4-6 page paper to be given as an oral presentation.

Talks: Submit a 1-2 page extended abstract to be given as an oral presentation.

Posters: Submit a 2-4 page paper to be presented at a poster session.

Demos: Submit a work to be presented at a hands-on demo session. Demo submissions should consist of a 1-2 page extended abstract including diagrams or images, and a complete list of technical requirements (including anything expected to be provided by the conference organizers).

Performances: Submit a performance making creative use of web-based audio applications. Performances can include elements such as audience device participation and collaboration, web-based interfaces, Web MIDI, WebSockets, and/or other imaginative approaches to web technology. Submissions must include a title, a 1-2 page description of the performance, links to audio/video/image documentation of the work, a complete list of technical requirements (including anything expected to be provided by conference organizers), and names and one-paragraph biographies of all performers.

Artworks: Submit a sonic web artwork or interactive application which makes significant use of web audio standards such as Web Audio API or Web MIDI in conjunction with other technologies such as HTML5 graphics, WebGL, and Virtual Reality frameworks. Works must be suitable for presentation on a computer kiosk with headphones. They will be featured at the conference venue throughout the conference and on the conference web site. Submissions must include a title, 1-2 page description of the work, a link to access the work, and names and one-paragraph biographies of the authors.

Tutorials: If you are interested in running a tutorial session at the conference, please contact the organizers directly.

Important Dates

March 26, 2019: Open call for submissions starts.

June 16, 2019: Submissions deadline.

September 2, 2019: Notification of acceptances and rejections.

September 15, 2019: Early-bird registration deadline.

October 6, 2019: Camera ready submission and presenter registration deadline.

December 4-6, 2019: The conference.

At least one author of each accepted submission must register for and attend the conference in order to present their work. A limited number of diversity tickets will be available.

Templates and Submission System

Templates and information about the submission system are available on the official conference website: https://www.ntnu.edu/wac2019

Best wishes,
The WAC 2019 Committee

posted in news

Vanilla struct multislider with jump on click and drag

i updated the original post. The current version is now stable even without the virtual mouse click.

The problem was, that to drag the whole plotted array, the click has to go to the invisible grabbing area of the element, which represents the variable y. In the first version, when this area was missed, the click would not grab the element and dragging wouldn't work. The virtual mouse click solved this, because it would always click at the center of the element where the grabbing area is located.

The current version doesn't need the virtual click anymore, because it moves the whole struct on click, so that the grabbing area of the element is moved to where the mouse is. After it got the click, it is moved back to the original position.

The drawback is, that it is not always accurate while dragging (the virtual click solution has the same problem). The further you click away from the center of a slider, the more inaccurate it becomes. i am still looking for a solution for this.

i also made another version, which uses pixel dragging instead of the plotted array drawing feature:

multislider-pixel.pd

Now this way it is always accurate, but because it is processed through Pd and not directly working on the data structure, it will miss some sliders if you move the mouse fast. This is the effect you also get if you use an external to get the mouse position and set the sliders from there.

posted in patch~

Looking for PD guru...private proj

Cool...excellent in fact. Will start with 1 issue and go 1 at a time...

Main behaviour is a single click/double click/long hold. ie For midi sustain, a quick single click enables a filter which allows only note on values to play midi notes (for sustained atmoshpheres played from a midi guitar). Anything longer than a quick click is a standard momentary sustain pedal. A double click resets the pedal action and sends an 'all notes off' to clear everything.
I was planning to use this in a browser so I used an active X timer so it would be async to the rest of the script. Hope I have added enough comments to explain;

Insert Code Here
Case 64
    'Case CC 64 = sustain message from floorboard
    'Dat1 = Midi message type+channel
    'Dat2 = message value
    
    'default action is send cc64 to instrument out ie Dat2 = 127 and begin sustain regardless of note off filter/sustain action

    If Dat2>0 Then mox.OutputMidiMsg InstOut, &hB0, 64, 127
    'SC is prefix: SingleClick
    'Check if its the first trigger in timer sequence and timer has been fired
    If SCTiming = 1 then
        'Test how many pedal downs have occured during timer period
        If Dat1 = SCOldDat1 THEN
            SCClkCnt = SCClkCnt+1
        End If
    Else
        'Waiting for Single Click Pedal Up
        If WaitForSCPU=1 Then
            If (SCOldDat1=Dat1 AND Dat2=0) Then
                'Long Hold Action - cc64 sustain off
                mox.OutputMidiMsg InstOut, &hB0, 64, 0
                'Clear wait flag
                WaitForSCPU = 0
            End If
        Else
            'fresh trigger
            'sc and dc are single/double click actions that are executed as strings according to timer
            sc="FilterMGNoteOff = FilterMGNoteOff XOR True : mox.OutputMidiMsg InstOut, &hB0, 64, 0"
            ' PUREDATA NOTE: couldnt see any XOR functions?
            dc="mox.OutputMidiMsg InstOut, &hB0, &h7B, &h7F : FilterMGNoteOff = False : mox.OutputMidiMsg InstOut, &hB0, 64, 0"
            'SCTiming is a flag to indicate timer has fired
            SCTiming = 1
            SCTimer.enabled = true
            SCTimer.Duration = 400
            'set the click count to 1 (ie pedal down of 127 has ocurred)
            SCClkCnt = 1
        End If
    End If
    'Single Click previous value used for comparison/click count
    SCOldDat1 = Dat1

Insert Code Here

And here is the timer sink/sub;

Insert Code Here
Sub SC_Timer
    SCTiming=0
    'Do Timer results here
    If (SCClkCnt=3 OR SCClkCnt=4) Then
        'DC Behaviour
        Execute(dc)
    Else
        If SCClkCnt=2 Then
            'msgbox "sc"
            'SC Behaviour
            Execute(sc)
        Else
            If SCClkCnt=1 Then
                'Wait for pedal up
                'msgbox "pedal up"
                WaitForSCPU=1
            End If
        End If
    End If
    'Reset ClkCnt
    SCClkCnt=0
    SCTimer.enabled = false
End Sub

Hope that helps explain

The above example is a bit parametric in that it feeds strings to the sub. This type of module is used throughout the script so Im totally open to simplifying it/modding to make it better in PD...which, from what I have seen, looks like it should be able to do.

If I can get all the previous code to translate, would like to add a control panel showing the fcb with the the current status/values of footswitches on screen, or even just just plug a tablet on the front of the controller for the future.

Many thanks
Mark

posted in technical issues

Advanced Search

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

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

Special Features

virgo:[virgo/PROJECT KITS PATTERNS]

NoxSiren - Modular synthesizer system <- [v15]

Crackled Audio from PD on Raspberry Pi 4

vstplugin~ 0.2.0

Scheduler queue object?

PD's scheduler, timing, control-rate, audio-rate, block-size, (sub)sample accuracy,

Hello,

*I think it is very important to understand how timing works in detail for low-level programming … *

abl_link~ midi and audio sync setup

Web Audio Conference 2019 - 2nd Call for Submissions & Keynotes

Fifth Annual Web Audio Conference - 2nd Call for Submissions

Keynote Speakers

Theme and Topics

Submission Tracks

Important Dates

Templates and Submission System

Vanilla struct multislider with jump on click and drag

Looking for PD guru...private proj

I think it is very important to understand how timing works in detail for low-level programming …