Previously been using an older MacBook Pro with OSX 10.10, and PurrData, all good.

New MacBook Pro M4 Max, and PurrData v2.19.3 from here... (2.20 was reported as damaged by the OS)...

https://github.com/agraef/purr-data/releases

PurrData v2.19.3 and v2.19.2 work fine, but there's no audio coming in from ~adc.

Opened a DAW, Reaper, set the input channels correctly, and there's audio coming in - to my new MacBook Po M4 Max. So the audio interface - a MOTU, is fine, and set up correctly ...it's just PurrData - there's nothing coming in from ~adc (I tried hooking up all input channels ~adc 1 2 3 4 5 6 7 8, and still nothing). Otherwise midi flows back and forth ok, and audio is sent out to the audio interface (via ~dac), but ~adc is dead ...and just to reiterate Reaper hears the input on those channels fine - but PurrData doesn't.

The audio properties in PurrData are set up fine, the audio interface selected, and multiple channels - so all good there (and I've using PD for many many years).

Anyone come across this - what to do?

This is a continuation to the issue I wanted to solved in this topic. It just went to different places so I though I will open a new topic to this problem I'm facing.

I have a pd patch that doing some audio playback reading some files from buffer. I'm running it on my Pi4.

I wanted it to start on boot every time and to be able to reset itself if crashing for some reason.
I was suggested to use that service script:

[Unit]
Description=My PureData service

[Service]
Type=simple
LimitNOFILE=1000000
ExecStart=/usr/bin/puredata -nogui  -open /home/pi/mypatch.pd
WorkingDirectory=/home/pi
User=pi
Group=pi
Restart=always
# Restart service after 10 seconds if service crashes
RestartSec=10

[Install]
WantedBy=multi-user.target

The above was working great using the built in 3.5mm audio jack.
I then bought UGREEN USB audio interface as I was facing with some poor audio quality at the output.

I set the audio preference in PD to choose the USB Audio Interface as the output.

When I boot the Pi I'm getting this error from the service (see picture)

If I'm typing sudo systemctl restart my_puredata.service the PD patch is back to work just fine. No Alsa error, but on the initial boot it is not working.

Any idea why this happen when using the USB AudioInterface? anything I can do in order to make it work?

So If to conclude:

When I start the same pd patch using the same service script but without a USB audio interface is working just fine.

When I start the same pd patch with the USB audio interface but using the autostart file:

sudo nano /etc/xdg/lxsession/LXDE-pi/autostart

Is also working just fine.

But the combination of the USB audio interface and the service script is just not working.

Thanks for any help.

It still does not work.

I added my-guitar-rig/my-guitar-rig~ to a window and it generated a lot of errors

I am running version 0.52.1 of pd

Hopefully this is helpful. The errors I got are:

 z~ 64
... couldn't create
 limiter~ 98 1
... couldn't create
io pair already connected
delay(wavey)(v).pd 39 0 40 0 (snapshot~->gatom) connection failed
 tof/pmenu 1 1 black white red
... couldn't create
 tof/pmenu 1 1 black white red
... couldn't create
 z~ 64
... couldn't create
 limiter~ 98 1
... couldn't create
io pair already connected
delay(wavey)(v).pd 39 0 40 0 (snapshot~->gatom) connection failed
 tof/pmenu 1 1 black white red
... couldn't create
 tof/pmenu 1 1 black white red
... couldn't create
 z~ 64
... couldn't create
 limiter~ 98 1
... couldn't create
io pair already connected
delay(wavey)(v).pd 39 0 40 0 (snapshot~->gatom) connection failed
 tof/pmenu 1 1 black white red
... couldn't create
 tof/pmenu 1 1 black white red
... couldn't create
 z~ 64
... couldn't create
 limiter~ 98 1
... couldn't create
 date
... couldn't create
 time
... couldn't create
 z~ 64
... couldn't create
 limiter~ 98 1
... couldn't create
 mknob 42 0 0 1 0 0 empty empty ratio:1.5:1 -2 -6 0 10 -262144 -1 -1 20175 1
... couldn't create
 tof/pmenu 1 1 black white red
... couldn't create
pd-float: rounding to 2048 points
warning: fx3000-in-2: multiply defined
warning: fx3000-in-2: multiply defined
warning: fx3000-in-1: multiply defined
warning: fx3000-in-1: multiply defined
warning: fx3000-in-0: multiply defined
warning: fx3000-in-0: multiply defined
pd-float: rounding to 2048 points
pd-float: rounding to 2048 points
warning: fx3000-in-2: multiply defined
warning: fx3000-in-2: multiply defined
warning: fx3000-in-1: multiply defined
warning: fx3000-in-1: multiply defined
warning: fx3000-in-0: multiply defined
warning: fx3000-in-0: multiply defined
pd-float: rounding to 2048 points
pd-float: rounding to 2048 points
warning: fx3000-in-2: multiply defined
warning: fx3000-in-2: multiply defined
warning: fx3000-in-1: multiply defined
warning: fx3000-in-1: multiply defined
warning: fx3000-in-0: multiply defined
warning: fx3000-in-0: multiply defined
pd-float: rounding to 2048 points

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

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.

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

warning: D: multiply defined
warning: D: multiply defined
warning: C: multiply defined
warning: C: multiply defined
warning: B: multiply defined
warning: B: multiply defined
warning: A: multiply defined
warning: A: multiply defined
warning: D: multiply defined
warning: D: multiply defined
warning: C: multiply defined
warning: C: multiply defined
warning: B: multiply defined
warning: B: multiply defined
warning: A: multiply defined
warning: A: multiply defined

I was not troubled by this so far operating the patch of this, but still wondering what this is? Can this be resolved? Could this be causing a problem in the future?

Allows you to play various game music formats, including:

AY - ZX Spectrum/Amstrad CPC
GBS - Nintendo Game Boy
GYM - Sega Genesis/Mega Drive
HES - NEC TurboGrafx-16/PC Engine
KSS - MSX Home Computer/other Z80 systems (doesn't support FM sound)
NSF/NSFE - Nintendo NES/Famicom (with VRC 6, Namco 106, and FME-7 sound)
SAP - Atari systems using POKEY sound chip
SPC - Super Nintendo/Super Famicom
VGM/VGZ - Sega Master System/Mark III, Sega Genesis/Mega Drive,BBC Micro

The externals use the game-music-emu library, which can be found here: https://bitbucket.org/mpyne/game-music-emu/wiki/Home

[gme~] has 2 outlets for left and right audio channels, while [gmes~] is a multi-channel variant that has 16 outlets for 8 voices, times 2 for stereo.
[gmes~] only works for certain emulator types that have implemented a special class called Classic_Emu. These types include AY, GBS, HES, KSS, NSF, SAP, and sometimes VGM. You can try loading other formats into [gmes~] but most likely all you'll get is a very sped-up version of the song and the voices will not be separated into their individual channels. Under Linux, [gmes~] doesn't appear to work even for those file types.
Luckily, there's a workaround which involves creating multiple instances of [gme~] and dedicating each one to a specific voice/channel. I've included an example of how that works in the zip file.

Methods

• [ info ( - Post game and song info, and track number in the case of multi-track formats
  • this currently does not include .rsn files, though I plan to make that possible in the future. Since .rsn is essentially a .rar file, you'll need to first extract the .spc's and open them individually.
• [ path ( - Post the file's full path
• [ read $1 ( - Reads the path of a file
  • To get gme~ to play music, start by reading in a file, then send either a bang or a number indicating a specific track.
• [ goto $1 ( - Seeks to a time in the track in milliseconds
  • Still very buggy. It works well for some formats and not so well for others. My guess is it has something to do with emulators that implement Classic_Emu.
• [ tempo $1 ( - Sets the tempo
  • 0.5 is half speed, while 2 is double. It caps at 4, though I might eventually remove or increase the cap if it's safe to do so.
• [ track $1 ( - Sets the track without playing it
  • sending a float to gme~ will cause that track number to start playing if a file has been loaded.
• [ mute $1 ... ( - Mutes the channels specified.
  • can be either one value or a list of values.
• [ solo $1 ... ( - Mutes all channels except the ones specified.
  • it toggles between solo and unmute-all if it's the same channel(s) being solo'd.
• [ mask ($1) ( - Sets the muting mask directly, or posts its current state if no argument is given.
  • muting actually works by reading an integer and interpreting each bit as an on/off switch for a channel.
  • -1 mutes all channels, 0 unmutes all channels, 5 would mute the 1st and 3nd channels since 5 in binary is 0101.
• [ stop ( - Stops playback.
  • start playback again by sending a bang or "play" message, or a float value
• [ play | bang ( - Starts playback or pauses/unpauses when playback has already started, or restarts playback if it has been stopped.
  • "play" is just an alias for bang in the event that it needs to be interpreted as its own unique message.

Creation args

Both externals optionally take a list of numbers, indicating the channels that should be played, while the rest get muted. If there are no creation arguments, all channels will play normally.

Note: included in the zip are libgme.* files. These files, depending on which OS you're running, might need to accompany the externals. In the case of Windows, libgme.dll almost definitely needs to accompany gme(s)~.dll
Also, gme can read m3u's, but not directly. When you read a file like .nsf, gme will look for a file that has the exact same name but with the extension m3u, then use that file to determine track names and in which order the tracks are to be played.

gme.zip

Hi everyone,

I'm having an issue with audio output in PD : when setting audio to output via portaudio, all the audio that gets out of PD has audio glitch in it, some kind of random-spaced crackles or clicks. It is completly useless as such. Increasing the delay in PD's audio settings doesn't change a thing.
However, using Jack as an audio outlet supresses all this problem, and the audio is clean that way.

Would anyone know the reason for this? I would be glad to know it as there may be a link with another issue about audio on my hardware : system audio of mac os X has the same kind of glitches that I got with PD on portaudio, only it's only occasional glitches so it's bearable. Solving this issue would be a big relief to me.

My hardware is a hackintosh (pc with osX installed on it), and mbox2 as audio interface (but I have tested with an m-audio transit and I have the same crackling issue). My soft is osX Snow Leopard 10.6.5.

Any help appreciated!

Greetings all,

I have just posted a collection of student patches for an interaction design course I was teaching at Emily Carr University of Art and Design. I hope that the patches will be useful to people playing around with Pure Data in a learning environment, installation artwork and other uses.

The link is: http://bit.ly/8OtDAq
or: http://www.sfu.ca/~leonardp/VideoGameAudio/main.htm#patches

The patches include multi-area motion detection, colour tracking, live audio looping, live video looping, collision detection, real-time video effects, real-time audio effects, 3D object manipulation and more...

Cheers,
Leonard

http://www.VideoGameAudio.com

Pure Data Interaction Design Patches

These are projects from the Emily Carr University of Art and Design DIVA 202 Interaction Design course for Spring 2010 term. All projects use Pure Data Extended and run on Mac OS X. They could likely be modified with small changes to run on other platforms as well. The focus was on education so the patches are sometimes "works in progress" technically but should be quite useful for others learning about PD and interaction design.

NOTE: This page may move, please link from: http://www.VideoGameAudio.com for correct location.

Instructor: Leonard J. Paul
Students: Ben, Christine, Collin, Euginia, Gabriel K, Gabriel P, Gokce, Huan, Jing, Katy, Nasrin, Quinton, Tony and Sandy

GabrielK-AsteroidTracker - An entire game based on motion tracking. This is a simple arcade-style game in which the user must navigate the spaceship through a field of oncoming asteroids. The user controls the spaceship by moving a specifically coloured object in front of the camera.
Features: Motion tracking, collision detection, texture mapping, real-time music synthesis, game logic

GabrielP-DogHead - Maps your face from the webcam onto different dog's bodies in real-time with an interactive audio loop jammer. Fun!
Features: Colour tracking, audio loop jammer, real-time webcam texture mapping

Euginia-DanceMix - Live audio loop playback of four separate channels. Loop selection is random for first two channels and sequenced for last two channels. Slow volume muting of channels allows for crossfading. Tempo-based video crossfading.
Features: Four channel live loop jammer (extended from Hardoff's ma4u patch), beat-based video cross-cutting

Huan-CarDance - Rotates 3D object based on the audio output level so that it looks like it's dancing to the music.
Features: 3D object display, 3d line synthesis, live audio looper

Ben-VideoGameWiiMix - Randomly remixes classic video game footage and music together. Uses the wiimote to trigger new video by DarwiinRemote and OSC messages.
Features: Wiimote control, OSC, tempo-based video crossmixing, music loop remixing and effects

Christine-eMotionAudio - Mixes together video with recorded sounds and music depending on the amount of motion in the webcam. Intensity level of music increases and speed of video playback increases with more motion.
Features: Adaptive music branching, motion blur, blob size motion detection, video mixing

Collin-LouderCars - Videos of cars respond to audio input level.
Features: Video switching, audio input level detection.

Gokce-AVmixer - Live remixing of video and audio loops.
Features: video remixing, live audio looper

Jing-LadyGaga-ing - Remixes video from Lady Gaga's videos with video effects and music effects.
Features: Video warping, video stuttering, live audio looper, audio effects

KatyC_Bunnies - Triggers video and audio using multi-area motion detection. There are three areas on each side to control the video and audio loop selections. Video and audio loops are loaded from directories.
Features: Multi-area motion detection, audio loop directory loader, video loop directory loader

Nasrin-AnimationMixer - Hand animation videos are superimposed over the webcam image and chosen by multi-area motion sensing. Audio loop playback is randomly chosen with each new video.
Features: Multi-area motion sensing, audio loop directory loader

Quintons-AmericaRedux - Videos are remixed in response to live audio loop playback. Some audio effects are mirrored with corresponding video effects.
Features: Real-time video effects, live audio looper

Tony-MusicGame - A music game where the player needs to find how to piece together the music segments triggered by multi-area motion detection on a webcam.
Features: Multi-area motion detection, audio loop directory loader

Sandy-Exerciser - An exercise game where you move to the motions of the video above the webcam video. Stutter effects on video and live audio looper.
Features: Video stutter effect, real-time webcam video effects