@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.

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.

This makes comparison much faster than comparing string characters.

True if you've already cached the symbols. If you have to make a call to gensym in order to do the comparison then it's slower than a string comparison in all cases. (Probably noticeably so since most libraries have fast paths for string comparisons.)

If many symbols are created (e.g. automatically in a patch), symbol lookup in the hash table will get slower - this can slow down the whole system as long as PureData is running.

That's not quite true. To be precise:

1. As the total number of symbols increases, the lookup time for an arbitrary symbol becomes less predictable. Importantly for realtime performance: your worst-case time for an arbitrary symbol lookup increases.
2. As the total number of symbols increases, the time to add a new symbol to the table increases.

For example: the time it takes to look up "set" in the symbol table stays consistent no matter how many symbols you generate in a running instance of Pd. This is because "set" is added to the symbol table at a predictable moment at Pd startup (since it's a method for a core Pd class).

Here's why my distinction matters: suppose you have a patch running. You want to find out if you're doing anything in your patch that is potentially realtime unsafe.

Now suppose I tell you to turn off dsp and run an abstraction that will spam the symbol table with a bunch of randomly generated symbols. Now turn DSP back on.

If your patch is truly realtime-safe then the added bloat shouldn't affect your performance at all. If you start getting dropouts you didn't have before it must mean you are generating new symbols during DSP which isn't realtime-safe.

I think I'm right on what I wrote above, but I haven't tested it. It might help with stuff like this to add a new introspection object to Purr Data for symbols.

Edit: clarification.

Hi guys, here is my new solo project. Interaction between synthesis sounds (karplus strong) and piezo feedback. The feedback sounds are actually produced via fft cross synthesis between the feedback itself and the synthesis sounds. The idea is to create an "organic" quality from synthesis sounds. I'll be touring with this project, contact me if you're interested!

https://archive.org/details/LeGrandIncendieDeLondres5

@whale-av hey, so I made this patch, it doesnt have abstractions, i havent practiced them yet, but would this be considered additive synthesis? 5 sinewaves, different frequencies, and start at the same volume but end differently...additive synthesis.zip :0,
for additive synthesis each sine must be at a different frequency and volume right? so I could be making glissandi in pitch and volume and it would still be considered additive synthesis?

Let's hope the bug is just in realtime or it is the os that is too lazy to report the time in time.

This patch seems to indicate this, but not definitely: realtime-test.pd

For me it looks like whenever the time is exactly zero, either realtime or the os is too slow to report the actual microseconds difference. Which would be good news for the delay and pipe objects.

THE GOAL:

I want my Raspberry Pi 2 to automatically start up the Jack server with realtime scheduling, and subsequently start Pure Data with realtime scheduling, load a patch &c. without any user intervention from a login shell.

As a performance artist working primarily with psychodrama (the technology is definitely NOT the important part here), fiddling around at a terminal right before or during a performance is kind of... psychically inconvenient. I need a box that I can plug in, give the audio output to the sound guy, and be ready to go.

PREREQUISITES:

I use Raspbian with a Linux kernel compiled with realtime goodness. I have hand-compiled Jack2 and Pure Data with realtime support in order to take advantage of this. Running a process with realtime priority requires the proper PAM directives set in /etc/security/limits.conf and related places, but that is beyond the scope of this little write-up.

Also somewhat relevant: I use a M-Audio MobilePre USB soundcard (sounds pretty awful by today's standards, but it's an extremely USEFUL box and sounds good enough for the work I do). For full-duplex sound, this requires the RasPi's USB to be set to single speed. In this configuration, I can get just under 2.9ms latency with good CPU overhead for Pure Data to run a few of my 64-voice wavetable and delay line granulators. Yeah!

THE PROBLEM:

Purely by happenstance, I had given the jackd command in my startup script the option “-s” which allows the server to ignore overruns and so on. So things seemed to be working as expected, but I noticed a lot more glitches than when I manually started up Jack and Pd from the terminal without the “-s” option. Upon removing it from my startup script, everything failed! WAH.

So I started piping STDERR and STDOUT to text files so I could read what either Jack or Pd were complaining about. As it turns out, Jack was unable to start with realtime priority due to a permissions problem. (I assume this is one of the things the “-s” options allows jackd to ignore, and thus start up with non-realtime priority. Problem is that Pure Data can’t connect to a non-realtime Jack server when its “-rt” option specified.)

Now, I had already been through the whole rigamarole of setting proper memory and priority limits for the “audio” group, to which the user “pi” belongs. So I thought, okay, I have to execute these commands as “pi”, and while simulating a login shell because the security limits in question are only set during login.

So I did this:

su -l pi -c "/usr/local/bin/jackd -R -dalsa -dhw:1,0 -p128 -n3 -r44100 -S >> /home/pi/jackd.log 2>&1 &"

This says “login as user ‘pi’ and then run the jackd command with these options, piping the outputs to this log file and run it in the background”. Well, I still got all the same errors about not being able to set realtime priority. WHYYYYYYYYY?

THE SOLUTION:

I hunted and hunted and hunted on a Very Popular Search Engine til I decided to try searching “security limits not loaded with su -l” and found this.

(Makes me think of that Talking Heads lyric, “Isn’t it weird / Looks too obscure to me”.)

So by uncommenting the line # session required pam_limits.so in /etc/pam.d/su everything started working as expected.

CONCLUSION:

I now know a LOT MORE about PAM and how important it is to keep in mind when and in what order scripts and other little subsystems are executed; but also that sometimes the problem is EXTREMELY OBSCURE and is to be found in some seemingly far-flung config file.

I hope this helps anybody out there working with Pure Data and the RasPi. The second generation board really packs quite a punch and can run several hundred audio grains (run by vline~ and enveloped by vline~ and cos~) simultaneously without a problem. And I'm pretty sure this is just using ONE of the 4 cores!

I'm by no means an expert Linux sysadmin, so if you have any other suggestions or corrections, please let me know! I wouldn't have been able to get this far without all the generous and helpful writeups everybody else has contributed, both within the RasPi and Pure Data communities. If you have any questions about anything I glossed over here, I'll do my best to answer them.

@AlbertoZ said:

@lorenzosu said:

@mjpalumbo said:

Hi ! I would really like to try out this patch, but i don't use gitorius (and i don't have an SSH Key). any chance you could upload a .zip file to this thread? Many Thanks!

No need

In gitorious you can dirctly download a tar.gz of the source tree:

http://gitorious.org/granita/granita/archive-tarball/master

Lorenzo

Hi Lorenzo, very good patch, thanks for sharing it.
Maybe it is not fully clear how to download it from the main granita gitorious page:
first one have to press the "Source Tree" button and in the new page that opens there is the "download master as tar.gz".
Otherwise there is your donwload link...

Yes... I guess it's a bit convoluted, but once you get used to the intricacies of GIT it's rather custom
Actually it is also linked from e.g. http://puredata.info/downloads/granita-minimalist-granular-synthesis

Concerning the pitch control, how is it handled? The units you provided are kind of "arbitrary" (0-5000, with 4000 being the
"no pitch modification"). Wouldn't it be more informative to have something linked to the frequencies
(double the frequency, half the frequency etc.) or maybe this is not possible because of grain implementation?

Absolutely... the 'pitch' control is probably the roughest part of granita, although it can be fun (especially with the live looper). Redoing the whole GUI with more comprehensible controls and tidier patching is in the todo ever since

By the way, It's true, this is "an incredibly original aesthetic problem "

Credit where due: it's a sentence from Tommaso Landolfi's 'Dialogo dei massimi sistemi' http://it.wikipedia.org/wiki/Dialogo_dei_massimi_sistemi

Lorenzo

Hi there!
Lots of fine and talented ladies and gentlemen here, i feel i need to finally stop lurking and get my concept together.

I'm banging my head around a concept of realtime granular modulation to the incoming signal.
Every granular synth i have encountered is built upon samples - either recorded or read from a disk.
I would like to make grains out of heavily timestretched first couple of ms of the incoming signal and then continue with the rest of the signal, probably fed from a continuously refilled buffer.

As I'm only a humble piano player the idea of making two tables that somehow manage to synchronize the out-spitting of the little grain-building chunks of audio makes me wonder if there isn't any widespread solution to my problem?

my signal would need to be timestretched at least 8 times its size, so I think there should be plenty of computational time to spare.
For added value I plan to implement beat synchronization so on every next beat the grain catches up on the incoming signal (scraps or saves the buffer and fills it anew)

So putting a long story short: how to feed a realtime signal into a variable length grain window, saving the residual realtime signal somewhere else?
should the timestretching be constant or is it possible to modulate this also?

I hope i made myself clear and someone will offer a helping hand so I can start coding. Of course if I succeed I will share the results

Nikola Kolodziejczyk
http://music.thinkbay.net

I did.

For command jackd I get this terminal output:

matjaz@matjaz-NV59C:~$ jackd
jackdmp 1.9.6
Copyright 2001-2005 Paul Davis and others.
Copyright 2004-2010 Grame.
jackdmp comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
Cannot create thread 1 Operation not permitted
Cannot create thread 1 Operation not permitted

usage: jackdmp [ --no-realtime OR -r ]
[ --realtime OR -R [ --realtime-priority OR -P priority ] ]
(the two previous arguments are mutually exclusive. The default is --realtime)
[ --name OR -n server-name ]
[ --timeout OR -t client-timeout-in-msecs ]
[ --loopback OR -L loopback-port-number ]
[ --port-max OR -p maximum-number-of-ports]
[ --midi OR -X midi-driver ]
[ --verbose OR -v ]
[ --clocksource OR -c [ c(ycle) | h(pet) | s(ystem) ]
[ --replace-registry ]
[ --silent OR -s ]
[ --sync OR -S ]
[ --temporary OR -T ]
[ --version OR -V ]
-d backend [ ... backend args ... ]
Available backends may include: alsa, dummy, freebob, firewire or net

jackdmp -d backend --help
to display options for each backend

and for qjackctl:
The same control GUI opened where I clicked start.

15:38:11.262 Patchbay deactivated.
15:38:11.263 Statistics reset.
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started
15:38:11.286 ALSA connection graph change.
15:38:11.482 ALSA connection change.
15:38:19.241 Startup script...
15:38:19.242 artsshell -q terminate
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started
sh: artsshell: not found
15:38:19.646 Startup script terminated with exit status=32512.
15:38:19.646 JACK is starting...
15:38:19.646 /usr/bin/jackd -dalsa -d/dev/dsp -r44100 -p256 -n2 -Xseq
15:38:19.649 JACK was started with PID=2544.
Cannot create thread 1 Operation not permitted
Cannot create thread 1 Operation not permitted
jackdmp 1.9.6
Copyright 2001-2005 Paul Davis and others.
Copyright 2004-2010 Grame.
jackdmp comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
JACK server starting in realtime mode with priority 10
Cannot lock down memory area (Cannot allocate memory)
ALSA lib control.c:882:(snd_ctl_open_noupdate) Invalid CTL /dev/dsp
control open "/dev/dsp" (No such file or directory)
ALSA lib control.c:882:(snd_ctl_open_noupdate) Invalid CTL /dev/dsp
control open "/dev/dsp" (No such file or directory)
audio_reservation_init
Acquire audio card Audio-1
creating alsa driver ... /dev/dsp|/dev/dsp|256|2|44100|0|0|nomon|swmeter|-|32bit
ALSA lib control.c:882:(snd_ctl_open_noupdate) Invalid CTL /dev/dsp
control open "/dev/dsp" (No such file or directory)
Cannot initialize driver
JackServer::Open() failed with -1
Failed to start server
15:38:19.743 JACK was stopped with exit status=255.
15:38:19.745 Post-shutdown script...
15:38:19.746 killall jackd
jackd: no process found
15:38:20.153 Post-shutdown script terminated with exit status=256.
15:38:21.703 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info.
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started
15:38:31.152 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info.
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started
15:38:42.143 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info.
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started
15:39:02.300 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info.
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started

I'm lost.