@whale-av

It is often forgotten that [metro] is the Pd metronome and it will take beat length messages though not quite as "musically" as quant for TempoClock.

The limitation there is that it necessarily enforces a certain time granularity. (Though, modular sequencers have the same limitation.)

In SC, you can schedule for any floating-point time, on or off a grid. A [metro]-driven clock in Pd could calculate an offset from the preceding grid line; if the granularity is tight enough, it would be reasonably accurate even through moderate tempo changes.

So bars and beats can be sent separately with two tied [metro] clocks or one and a counter.

Or take the modular "clock divider" approach (wherein, after a night's sleep, I see that I messed up the connections out of the [select 1] but I won't bother to fix the screenshot ).

hjh

@raynovich technically, yours is a polynomial function. It would be an exponential function if the input was used as the exponent of some constant, like {mtof].

I've only used Hann windows in 2 circumstances: as an FFT windowing function, and for granular synthesis. In the FFT case, the Hann window allows you to overlap each window by a factor of 4 so there are no discontinuities in the inverse transform. In granular synthesis, I've noticed that the Hann window injects fewer harmonics of the looping frequency so that the tone of the wavetable comes through more. I think there's a mathematical explanation of how that works in that same Miller Puckette book.

I'm sure there are other uses I don't know about, maybe someone more knowledgeable can chime in.

@g.arj I am sure it is not exactly what you wish but there should be a few clues in there.
I have added recording to the patch...... Microsound-granular-record.pd
I don't have any headphones at the moment so I am not sure what it does when you start to record again.
I get some feedback....... but it is probably a lot worse.
In fact it might be easier to do with two recordings each with a [delwrite~] [delread~] pair looping and an additional [delread~] randomly accessing the delay line..
David.

@g.arj Something like this although it has no audio input...... but I am sure that could be easily added.
Write over the table when you want a new sample. I am a bit brain dead today and not shure straight away how to resize the table for a recording but that can be done too.
Microsound-granular.pd
Pretty sure it's Vanilla.
Take a look at Katja's SliceJockey too........ http://www.katjaas.nl/slicejockey/slicejockey.html
Was Pd Extended........ but I think someone said on this forum that it can work in Vanilla.
Google is your friend.
David.

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.

my "audiolab" abstraction library is now available on deken. You'll need Pd-0.50 or later to run this.
Please report any bugs on github: https://github.com/solipd/AudioLab

here is a picture to draw you in (:

Edit:
list of objects:

Soundfle processing
pp.sfplayer~ ... variable-speed soundfile player
pp.grainer~ ... granular sampler
pp.fft-stretch~ ... pvoc time stretching & pitch shifting

Spatialization
pp.pan~ ... constant power stereo panning
pp.midside~ ... mid-side panning
pp. spat8~ ... 8-channel distance based amplitude panning
pp.doppler~ ... doppler effect, damping & amplitude modulation
pp.dopplerxy~ ... xy doppler effect

Effects
pp.freqshift~ ... ssb frequency shifter
pp.pitchshift~ ... pitch shifter
pp.eqfilter~ ... eq-filter (lowpass, highpass, resonant, bandpass, notch, peaking, lowshelf, highshelf or allpass)
pp.vcfilter~ ... signal controlled filter (lowpass, highpass, resonant)
pp.clop~ ... experimental comb-lop-filter
pp.ladder~ ... moogish filter
pp.dynamics~ ... compressor / expander
pp.env~ ... simple envelope follower
pp.graindelay~ ... granular delay
pp.rev~ ... fdn-reverberator based on rev3~
pp.twisted-delays~ ... multipurpose twisted delay-thing
pp.shepphaser~ ... shepard tone-like phaser effect
pp.echo~ ... "analog" delay

Spectral processing
pp.fft-block~ ... audio block delay
pp.fft-split~ ... spectral splitter
pp.fft-gate~ ... spectral gate
pp.fft-pitchshift~ ... pvoc based pitchshifter
pp.fft-timbre~ ... spectral bin-reordering
pp.fft-partconv~ ... partitioned low latency convolution
pp.fft-freeze~ ... spectral freezer

Misc.
pp.in~ .... mic. input
pp.out~ ... stereo output & soundfile recorder
pp.out-8~ ... 8 channel output & soundfile recorder
pp.sdel~ ... samplewise delay
pp.lfnoise~ ... low frequency noise generator
pp.spectrum~ ... spectrum analyser
pp.xycurve

@yannseznec Thanks for sharing your patch.
I have made a granular sampler with vline~ which you might find interesting. The point of using vline~ instead of phasor~ is that this way the grain stream can be set as synchronous or asynchronous, or grains can be even triggered manually. It has 8 voices and 500 maximum grains per voice, but this is completely arbitrary. Also the way I'm using switch doesn't produce any clicks that I can hear, and it's used to turn on and off individual grains as well as whole unused voices. The grain window as well as other bits and pieces were taken from a granular delay called grannie basher that was posted in the forum years ago.
graindeurv1.0.zip
Main patch is graindeur-help.pd
I hope I have included all the abstractions needed, please let me know if something's missing. It requires the bsaylor external for the envelope.

I have also included your piano sample, hope you don't mind!

Hey folks.

Here's a polyphonic granular playback (and sampling) patch, I've been working on it for a while, I've used some different versions of it for a few different projects like the EN-1 video I posted last week. Full patches here: https://github.com/yannseznec/Granular-Polyphonic

It's currently 8 voice polyphonic, and uses 8 grains. These could both be raised significantly on any standard computer, I've kept them low for running on low-spec boards. It currently uses MIDI note input to play, with the note values controlling pitch. All other settings are controlled via UI sliders.

It's getting to a pretty decent point now, where it's relatively flexible...my long-term goal has been to make it somewhat drag-and-droppable.

It still needs work, of course! A few things I'd like to get some feedback on, if anyone has the energy to download all of the patches and have a play:

• At certain pitch values (particularly low ones) there is a fair amount of high frequency noise. I guess this is aliasing? I've tried to use the lop~ object to filter this out but it hasn't proven to be a satisfying solution. Any tips? I can just about understand why it's happening, with all of the fast phasors going on and the repitching of a sound file and so on, but it's still somewhat frustrating.
• Does the patch generally make sense? Any glaring mistakes?
• Any ideas for how you would use a patch like this, which could lead to different features, UI elements, or approaches to a hardware build?

In any case, enjoy! Hope you like it.

My new standalone granular sampling synth, made entirely in PD and running on a Bela in a custom box. The patches need some cleanup, so I'll be updating them over the coming weeks but feel free to have a look (and tell me how to improve them): https://github.com/yannseznec/Granular-Sampler

@metaphysician Yes, [phasor~] is often used. There should be some useful bits in this...... Microsound-granular.pd
David.