Ok, the cat is out of the bag --> https://github.com/porres/pd-else/releases/tag/1.0-rc13 I'm officialy announcing the update and uploaded binaries to deken for mac (intel/arm), Win and Linux. It all looks ok but tell me if you see something funny please. There's also a raspberry pi binary but not working 100%yet and we'll still look into that. Hopefully someone could help me/us with it. I might make another upload just for the pi later on if/when we figure it out. Find release notes and changelog below.

RELEASE NOTES:

Please support me on Patreon https://www.patreon.com/porres I'll now try to add special content for subscribers. You can follow me on instagram as well if you like... I'm always posting Pd development stuff over there https://www.instagram.com/alexandre.torres.porres/

It's been a little bit over 7 months since the last update and I almost broke the record for taking too long to release an update (which had happened in my previous update). So yeah, there's just too much to talk about! I guess the delays in releasing updates is because it's been a little tricky and hard to sync the release cycles of ELSE with PlugData, which includes ELSE in its download.

Plugdata 0.9.2 should come out soon with ELSE RC13 and it's supposedly the last update before 1.0.0, so I've heard. And the plans was to get to that still in 2025! This means ELSE could be at its last "Release Candidate" phase as I'm aiming to sync the final release with PlugData. Until then, I'll still make breaking changes and I can't wait until I can't do that anymore as I really feel bad. On the other hand, it's kind of inevitable when I'm always adding new stuff and redesigning and reconfiguring objects to include more functionalities. And I always got a lot of new stuff! So I'm thinking that I will eventually try some mechanism like Pd's compatibility flag or something. I'll try to come up with something like that in the next update.

This update has 22 new objects for a total of 573 and 26 new examples in my tutorial for a total of 554 examples. Let's dive into the highlights (see full changelog below after the release notes).

• Multichannel Support: Last release had 92 MC aware objects, now it's 139! Over a 50% increase that include old and new objects (all the new ones have been coming with MC support). Virtually all oscillators and envelope generators now have MC support, plus some other random ones. Let me highlight the new [lace~]/[delace~] objects that are 'MC' tools that perform interleave/deinterleave in Multichannel connections. My bare minimum number of objects "to start with" would be at least a bit over half the number of signal objects. That was my target for 1.0! ELSE right now has 319 signal objects, so that'd be at least 160. I will definitely pass this milestone in the next update. I guess a good number of MC objects would be around 75% of the signal objects. I will aim for that as soon as I can. Some objects simply can't be MC at all, so 100% will never be the case, but maybe an ideal 90% eventually? We'll see... I am just proud and happy that ELSE is taking such a big jump on MC awareness in less than a couple years.

• Envelope generators ([adsr~]/[asr~]/[envgen~]/[function~]) now have more curve options. For [adsr~]/[asr~] the default is now a new log curve that you can set the curve parameter (and was 'stolen' from SuperCollider). A new [smooth~] family of objects perform the same kind of curved smoothening for alternating inputs - [envgen~] and [function~] also have that but also '1-pole' filtering, 'sine' and 'hann' curves. You can now trigger [adsr~] and [asr~] with impulses.

• The [play.file~] object now supports even more file formats besides MP3 and stuff. Hey, you can even stream the supported formats from weblinks! The [sfload] object (which loads files into arrays) also gained support for more formats and can download from weblinks as well! It also has a new threaded mode, so loading big files won't choke Pd. It now also outputs the file information, which is a way to tell you when loading finished in threaded mode. The [sample~], [player~], [gran.player~] and [pvoc.player~] objects are now also based on [sfload], so they support all these file formats!!! Now [sample~] and [tabplayer~] are integrated in a way that [tabplayer~] is always aware of the sample rate of the file loaded in [sample~] (so it reads in the "correct speed"). A new [sfinfo] object is able to extract looping regions and instrument metadata information from AIFF files (which is something I wanted for ages) - it should do more stuff in the future.

• [knob] has become the ultimate featured bloated creep GUI I always feared and avoided. MAX is envy! but I'm happy with this structure and I want to replicate in other GUIs in the future (yeah, I got plans to offer alternatives to all iemguis). I wanna highlight a new 'param' symbol I added that allows you to remotely set a particular method in an object, so you don't to connect to a "method $1" message and you can even do this wirelessly with a send symbol. [knob] now also acts like a number box, where you can type in the value, which may also be displayed in different ways or the value can be sent elsewhere via another send symbol so you can temper with it using [makefilename] or [else/format]. I've been using this for the MERDA modules and it's really cool.

• We finally have a [popmenu] GUI object! This was in my to do list forever and was crucial to improve the MERDA modules to set waveforms, instruments and whatnot.

• Let's about MERDA, the "Modular Euroracks Dancing Along" subset of abstractions in ELSE. It was first released in the last update and it's been driving lots of the development in ELSE as you can see. I now added a MIDI Learn feature for all knobs that feels great and quite handy! There are many fixes and improvements in general and some new modules. I wanna highlight the new [sfont.m~] module, which loads "sound font" banks and you can just click on a [popmenu] to choose the instrument you want. The default bank has numerous (hundreds) options and also comes with PlugData. The sequencer module [seq8.m~] was rather worthless but it's now a whole new cool thingie. It allows you to set pitches with symbols and even has quarter tone resolution. I added a right outlet to send impulses to trigger envelopes and stuff (there's still more stuff of course, see full changelog below).

• There are newly designed/renamed/recreated [resonbank~]/[resonbank2~] objects that are well suited for Modal Synthesis.

• What actually drives my development is my Live Electronics tutorial, which got a fair upgrade with a new chapter on Modal Synthesis amongst other things, such as new subtractive synthesis examples and a revision of envelope generators with examples on AHDSR and DAHDSR - by the way, there are new gaterelease~/gatedelay~ objects for handling envelopes (and other processes).

• I have to thank some people. Tim added 'zoom' to the [pic] object, as well as an image offset. Tim also implemented a new and better technique for bandlimited oscillators. Ben Wesh gave me a new [scope3d~] GUI object, pretty cool, that plots an oscilloscope in 3 dimensions, which is coded in LUA - and ELSE has been carrying a modified version of [pdlua] because it now depends on it for a couple of GUIs. Tim and Ben made many improvements to [pdlua] (as well as Albert Graef, of course).

• For more new objects, let me also tell you about the simple and cool [float2imp~], that is based on [vline~] and can convert floats to impulses with sample accuracy (don't know why I didn't think of that earlier). A new [tanh~] object has Multichannel support. A bit earlier I made an update to Cyclone that actually "borrows" and includes this one from ELSE instead of its original one (which does not have Multichannel support). PlugData users will load the one from ELSE. This is another tiny step that sort of integrates ELSE and Cyclone, specially for PlugData users.

happy patching.

CHANGELOG:

LIBRARY:

Breaking changes:

• [adsr~]/[asr~]: now a gate off before reaching the sustain point does not start the release right away (this allows you to trigger it with impulses). There's a new mode just for immediate release. There's a new exponential setting for curve factors, the old 'log' mode is renamed to 'lag' as it's the same as used in the [lag~] object. For [adsr~], a bang now is not "retrigger", but an impulse at control rate, there's a new 'retrigger' message for control rate retriggering (and now it only retriggers if the gate is on). For [asr~] a bang now also works like an impulse.
• [sample~]: no more 'load' message, args to 'open' message changed, size is now only in 'ms'.
• [format]: outputs are now always symbols, before you could get float outputs. Also, we just have a simplified symbol output, no more lists or anythings. Hopefully I'll be able to get the 'list' output back, but it involved some bugs that I couldn't fix so I just removed it. You cannot use bangs and lists in secondary inlets no more (this is cylone/max crappy paradigm we don't want here). Bang method was actually removed as well.
• [pack2]: no more support for anythings, also no more support for lists in secondary inlets and output has a list selector (I wanna make this more Pd like and not a silly clone from MAX's [pak], cause fuck MAX).
• [merge]/[unmerge]/[group]: no more '-trim' flag (again, respecting pd's usual list paradigm), in [merge] now there's no more 'hot' argument and a bang now represents an empty list and inlets initialized with empty lists
• [mono]: 1st argument is now 'glide' in ms.
• [sfont~] now uses 'mma' for bank selection (this alters how CC messages set the bank number).
• [player~]/[play.file~]: 'open' message does not play files right away anymore.
• [tabplayer~]/[player~]: play message without args now play at the default settings (whole file at regular speed).
• [envgen~]: removed the 'maxsustain' parameter, use the new [gaterelease~] or [gaterelease] objects instead. Removed the rightmost inlet just to set envelopes, now a list input only sets the envelope and doesn't trigger it. The 'set' message is then removed.
• [envgen~]/[function~]: simplified and got rid of '-exp' flag and message, also deleted 'expl' and 'expi' messages. A new 'curve' and cimpler message sets exponential factors for all or individual segments, and includes more curve formats.
• [knob]: 'esc' key now deactivates the object. The 'ticks' message is renamed to 'steps' and there is a new 'ticks' message that toggles showing ticks on and off. The 'start' message has been renamed to 'arcstart'. The 'outline' message has been renamed to 'square' for better clarity. Design changed a bit to make it like it is in PlugData (they won), so we now fill the whole background color when in 'square mode' and the knob circle has an 85% proportion in this case inside the full 100% square size (so it grows bigger when not in 'square' mode). Now, by default, the GUI is in a new 'loadbang' mode (I don't think this will influence old patches). I'm afraid some old patches might behave really weird since I added a lot of new stuff. I changed the 'load' message behaviour to not update the object (this can arguably be considered a bug fix).
• [wavetable~], [bl.wavetable~] and [wt2d~]: 'set' message now sets frequencies because of the MC support in [wt~] and [wt2d~], while there's a new 'table' method to set the table name.
• [gbman~]/[cusp~] list method is now for MC, old list method is now renamed back to an old 'coeffs' method.
• [f2s~]/[float2sig~] default value is now 10 ms.
• [op] now behaves like [*~] where the smaller list wraps til reaching the size of the longer one.
• [list.seq] does not loop anymore by default.
• [impseq~] list input removed, use the new [float2imp~] object to convert floats to impulses.
• [resonant~] now has 'q' as the default.
• [resonant2~] has been removed.
• [decay2~] has also been removed ([asr~] much better).
• [vcf2~] has been renamed to [resonator2~].
• [resonbank~]/[resonbank2~] have basically been deleted and replaced by new objects with the same name... [resonator~] is based on a new [resonator~] object which is similar to [resonant~] and [resonbank2~] is now based on [resonator2~] (old [vcf2~] instead of [resonant2~] that got deleted). These are well suited objects for Modal Synthesis.
• [oscbank~] now uses a 'partial' list and not a frequency list. The freq input now defaults to '1' and this makes [oscbank2~] completely obsolete.
• [oscbank2~] has been deleted since it became completely obsolete.
• [sfload] load message changed the behaviour a bit.

Enhancements/fixes/other changes:

• [adsr~]: We have now a new mode for immediate release (see breaking changes above, I'm not repeating it). Fixed ADSR signal inputs (it was simply not really working, specially for linear). Fixed status output for MC signals. There's a new curve parameter that allows you to set the curvature.
• [asr~] I actually just made the new [adsr~] code into a new [asr~] code as a simplified version (as it was before)... so it's got the same impromevents/fixes.
• [play.file~]: added support for more file formats and even weblinks for online streaming!
• [sfload]: added an outlet to output information, added threaded mode, added support for more file formats and even weblinks for downloading.
• [sample~], [player~], [gran.player~] and [pvoc.player~] are now also based on [sfload], so they support more file formats!
• [sample~]: improved extension management with [file splitext].
• [sample~] and [tabplayer~] now are automatically integrated in a way that [tabplayer~] is always aware of the sample rate of the file loaded in [sample~], so it automatically adjusts the reading speed if it is different than the one Pd is running with.
• [numbox~]'s number display is not preceded by "~" anymore (that was just kinda stupid to have).
• [format]: fixed issues where empty symbols and symbols with escaped spaces didn't work. Added support '%a' and '%A' type. Added support for an escaped 'space' flag. Improved and added support for length modifiers. Improved syntax check which prevents a crash. Improved documentation.
• [knob]: added new 'param', 'var', 'savestate', 'read only', 'loadbang', "active", "reset" and 'ticks' methods. Added the possibility to type in number values and also modes on how to display these number values, plus new send symbols for 'activity', 'typing', 'tab' and 'enter'. New design more like plugdata. Changed some shortcuts to make it simpler. If you have the yet unreleased Pd 0.56-0 you can also use 'double clicking' in the same way that works in PlugData. Properties were also significantly improved (I'm finally starting to learn how to deal with this tcl/tk thingie). Yup, a lot of shit here...
• [autofade2~]/[autofade2.mc~]: fixed immediate jump up for 0 ramp up.
• [synth~]: fixed polyphony bug.
• [metronome~]: fixed bug with 'set' message.
• [midi2note]: fixed range (octaves 0-8).
• [pulsecount~]: fixed reset count to not output immediately, added bang to reset counter at control rate
• [click]: fixed regression bug where it stopped working.
• [else]: new 'dir' method to output ELSE's binary directory in a new rightmost outlet. The print information also includes the directory.
• [pic]: added zoom capability finally (thanks to tim schoen) and added offset message (also thanks to tim).
• [store]: added 'sort' functionality.
• [scales]: fixed octave number argument. Added functionality to allow octave number as part of the note symbol.
• [mono]: added 'glide' parameter, as in [mono~].
• [pluck~]: fixed list input.
• [rescale]/[rescale~]: added a "reverse log" mode.
• [limit]: added a new second ignore mode.
• [graph~]: added an external source input for plotting the graph and a 'clear' message.
• [canvas.setname]: added a new argument for "abstraction mode" and methods to set name, depth (and mode).
• [midi.learn]: added a new argument for "abstraction mode", fixed 'dirty' message sent to parent.
• [brickwall~]: fixed initialization.
• [list.seq]: added a loop mode and a 2nd outlet to send a bang when the sequence is done.
• [delete]: fixed index for positive numbers.
• [dust~]: added 'list', 'set' and '-mc' flag for managing the already existing Multichannel capabilities.
• Thanks to Tim we have many fixes and a whole new technique for band limited oscillators. Now [bl.saw~], [bl.saw2~], [bl.vsaw~], [bl.square~], [bl.tri~], [bl.imp~] and [bl.imp2~] have been redesigned to implement elliptic blep, which should provide better anti-aliasing.
• [parabolic~] now uses and internal wavetable for more efficiency.
• [resonant~]: added 'bw' resonance mode.
• [lowpass~]/[highpass~]: added 't60' resonance mode.
• [quantizer~]/[quantizer]: added a new mode, which combines floor (for negative) and ceil (for positive) values.
• [crusher~]: now uses the new [quantizer~] mode from above (arguably a breaking change).
• [envgen~]: fixed a bug (actually a misconception) where ramps started one sample earlier. Fixed 0-length lines. Added a possibility to set time in samples instead of ms. Maximum number of lines is now 1024. Added loop mode. Added many curve options (sin/hann/log curve/lag).
• [function~]: Added many curve options (sin/hann/log curve/lag).
• [The out~] family of abstractions now use [bitnormal~] so you won't blow your speakers beyond repair in edge cases.
• [trig.delay~]/[trig.delay2~]: fixed bug where impulse values different than '1' didn't work.
• Added MC support to: [trig.delay~], [trig.delay2~], [gatehold~], [vca.m~], [gain2~], [decay~], [asr~], [envgen~], [function~], [bl.osc~], [bl.saw~], [bl.saw2~], [bl.vsaw~], [bl.square~], [bl.tri~], [bl.imp~], [bl.imp2~], [imp2~], [tri~], [saw~], [saw2~], [vsaw~], [square~], [pulse~], [parabolic~], [gaussian~], [wavetable~], [wt2d~], [randpulse~], [randpulse2~], [stepnoise~], [rampnoise~] [pink~], [gbamn~], [cusp~], [gray~] and [white~].
• Also added MIDI input and soft sync to [imp2~], [tri~], [saw~], [saw2~], [vsaw~], [square~], [pulse~], [gaussian~] and [parabolic~].
• [wavetable~] and [wt2d~] gained args to set xfading.
• Updated pdlua to 0.12.23.
• M.E.R.D.A: Added MIDI-LEARN for all modules (this is only for the knobs). Replaced some number boxes that were attached to knobs by an internal number display mechanism (new feature from knob). Improved interface of [gendyn.m~]. Preset/symbol name fixes to [flanger.m~]. Now we have automatic MIDI mode detection for [plaits.m~] and [pluck.m~] when no signals are connected (still trying to get plaits right, huh? Yup! And bow MIDI input with monophony and trigger mode has been fixed in [plaits.m~]). Added MC support to [vca.m~]. Increased range of [drive.m~] down to 0.1. Changed some objects to include the new [popmenu] GUI. [vco.m~] now uses the new MC functionalities of oscillators and doesn't need to load abstractions into [clone], I hope it makes this more efficient and clean. The [seq8.m~] module was worthless and got a decent upgrade, it's practically a new module. Added new modules (see below). Note that MERDA is still at alpha development phase, much experimental. Expect changes as it evolves.
• 22 new objects: [float2imp~], [lace], [delace], [lace~], [delace~], [gatehold], [gatedelay],[gatedelay~], [gaterelease~], [gaterelease], [popmenu], [scope3d~], [tanh~], [resonator~], [sfinfo], [smooth], [smooth2], [smooth~], [smooth2~], [dbgain~], [level~] plus [crusher.m~], [sfont.m~] and [level.m~] MERDA Modules.

Objects count: total of 573 (319 signal objects [139 of which are MC aware] and 254 control objects)!

• 323 coded objects (210 signal objects / 113 control objects)
• 227 abstractions objects (87 signal objects / 140 control objects)
• 23 MERDA modular abstractions (22 audio / 1 control)

TUTORIAL:

• New examples and revisions to add the new objects, features and breaking changes in ELSE.
• Added the MERDA modules into the examples for reference.
• Revised section on envelopes.
• New subtractive synthesis examples.
• New chapter on Modal Synthesis.
• Total number of examples is now 554! (26 new ones)

@oid said:

hopefully no stupid errors like last night. No time to test now.

And first object after the inlet is a [*~ 0] that I forgot to delete, really doing a great job at being consistent with this one. And it turns out the attack does nothing and the release affects overall response instead of just release despite this being copy and pasted directly from the patch where I have been developing it and where the attack and release work properly. Would love an explanation as to why, guessing one of those bugs that goes away after restarting pd but which is the bug, the working one or the not working one?

@whale-av Analog world always clips and will do it hard but may do it soft for awhile before it gets to hard. it is limited by the power supply and circuitry. In an analog compressor the gain computer will never see a signal greater than it was designed for, the circuitry before it will just clip it back down to size if you feed it a ridiculously large signal. With this compressor in its current state you can feed it a -200 - 200 signal and it will happily do its job and a very different job than that analog compressor in the same situation which would clip the signal back down to its limits and then let you compress that heavily distorted signal to a level manageable by what ever gear the compressor is running into. We can use this compressor to compress ridiculously large signals to manageable levels but you will probably get aliasing from the heavy gain reduction; upsampled clipping on the input alleviates that, if executed well can help manage troublesome peaks and reduce lookahead/delay of the signal as well.

Not sure if I will make a ui for this, I am trying to keep it generic enough that people can use it to make the compressor they want with it instead of use my idea of a compressor which would lack things like reassuring visual confirmations or knees and would be made fun of by all the other compressors. It is mostly an exercise to get better with the audio side of pd, but I have considered doing a ui so maybe there is a gain reduction meter in the future.

Hi, it's been a while, here we go:

RELEASE NOTES:

Hi, it's been almost 8 months without an update and I never took this long!!! So there's a lot of new stuff to cover, because it's not like I've been just sleeping around

The reason for the delay is that I'm trying to pair up with the release cycles of PlugData and we're having trouble syncing up. PlugData 0.9.0 came out recently after a delay of 6 months and we couldn't really sync and pair up then... we had no luck in syncing for a new update now, so now I'm just releasing it up cause enough is enough, and hopefully in the next plugdata release we can sync and offer the same version.

As usual, the development pace is always quite busy and I'm just arbitrarily wrapping things up in the middle of adding more and more things that will just have to wait.

First, I had promised support for double precision. I made changes so we can build for it, but it's not really working yet when I uploaded to deken and tested it. So, next time?

And now for the biggest announcement: - I'm finally and officially releasing a new pack as a submodule, which is a set of abstractions inspired by EuroRack Modules, so I'm thinking of VCV like things but into the Pd paradigm. Some similar stuff has been made for Pd over the years, most notably and famously "Automatonism", but I'm really proud of what I'm offering. I'm not trying to pretend Pd is a modular rack and I'm taking advantage of being in Pd. I'm naming this submodule "Modular EuroRacks Dancing Along" (💩 M.E.R.D.A 💩) and I've been working on it for a year and a half now (amongst many other things I do). PlugData has been offering this for a while now, by the way. Not really fully in sync though.

MERDA modules are polyphonic, thanks to multichannel connections introduced in Pd 0.54! There are 20 modules so far and some are quite high level. I'm offering a PLAITS module based on the Mutable Instruments version. I have a 6-Op Phase Modulation module. A "Gendyn" module which is pretty cool. I'm also including an "extra" module that is not really quite a modular thing at all but fits well called "brane", which was a vanilla patch I first wrote like 15 years ago and is a cool granular live sampler and harmonizer. You'll also find the basics, like oscillators, filters, ADSR envelope and stuff I'm still working on. Lastly, a cool thing is that it has a nice presets system that still needs more work but is doing the job so far.

There are ideas and plans to add hundreds more MERDA modules, let's see when and if I can. People can collaborate and help me and create modules that follow the template by the way

Thanks to Tim Schoen, [play.file~] is now a compiled object instead of an abstraction and it supports MP3, FLAC, WAV, AIF, AAC, OGG & OPUS audio file extensions. A new [sfload] object can import these files into arrays (but still needs lots of more work). There are many other player objects in ELSE that can load and play samples but these don't yet support these new formats (hang in there for the next version update).

Tim also worked on new [pdlink] and [pdlink~] objects, which send control and signal data to/from Pd instances, versions and even forks of Pure Data (it's like [send]/[receive] and [send~]/[receive~], all you need is a symbol, no complicated network or OSC configuration!). And yes, it works via UDP between different computers on the same network. And hell yeah, [pdlink~] has multichannel connections support! By the way, you can also communicate to a [pd~] subprocess. This will be part of ELSE and PlugData of course, and will allow easy communication between PlugData and Pd-Vanilla for instance.

The great pd-lib-build system has been replaced for a 'cmake' build process called 'pd.build' by Pierre Guillot. This was supposed to simplify things. Also, the [sfont~] object was a nightmare to build and with several dependencies that was simply hell to manage, now we have a new and much simpler system and NO DEPENDENCIES AT ALL!!! Some very rare file formats with obscure and seldom sound file extensions may not work though... (and I don't care, most and the 'sane' ones will work). The object now also dumps all preset information with a new message and backwards compatibility broke a bit

I'm now back to offering a modified version of [pdlua] as part of ELSE, which has recently seen major upgrades by Tim to support graphics and signals! This is currently needed in ELSE to provide a new version of [circle] that needed to be rewritten in lua so it'd look the same in PlugData. Ideally I'd hope I could only offer compiled GUI objects, but... things are not ideal

The lua loader works by just loading the ELSE library, no need for anything "else". I'm not providing the actual [pdlua] and [pdluax] objects as they are not necessary, and this is basically the only modification. Since PlugData provides support for externals in lua, if you load ELSE you can make use of stuff made for PlugData with lua without the need to install [pdlua] in Pd-Vanilla.

For next, we're working on a [lua] object that will allow inline scripting and will also work for audio signals (again, wait for the next version)! Also for the next version, I'm saving Ben Wesch's nice 3d oscilloscope made in lua (it'll be called [scope3d~]). There's a lot going on with the lua development, which is very exciting.

As for more actual new objects I'm including, we have [vcf2~] and [damp.osc~]. The first is a complex one pole resonant filter that provides a damping oscillation for a ringing time you can set, the next is an oscillator based on it. There's also the new [velvet~] object, a cool and multichannel velvet noise generator that you can also adjust to morph into white noise.

I wasn't able to add multichannel capabilities to many existing objects in ELSE in this one, just a couple of them ([cosine~] and [pimp~]). Total number of objects that are multichannel aware now are: 92! This is almost a third of the number of audio objects in ELSE. I think that a bit over half might be a reasonably desired target. More multichannel support for existing objects to come in the next releases.

Total number of objects in the ELSE library is now 551!

As for the Live Electronics tutorial, as usual, there are new examples for new objects, and I made a good revision of the advanced filter section, where I added many examples to better explain how [slop~] works, with equivalent [fexpr~] implementations.

Total number of examples in the Live Electronics Tutorial is now 528!

There are more details of course, and breaking changes as usual, but these are the highlights! For a full changelog, check https://github.com/porres/pd-else/releases/tag/v.1.0-rc12 (or below at this post).

As mentioned, unfortunately, ELSE RC12 is not yet fully merged, paired up and 100% synced in PlugData. PlugData is now at version 0.9.1, reaching the 1.0 version soon. Since ELSE is currently so tightly synced to the development of PlugData, the idea is to finally offer a final 1.0 version of ELSE when PlugData 1.0 is out. Hence, it's getting closer than ever Hopefully we will have a 100% synced ELSE/PlugData release when 0.9.2 is out (with a RC 13 maybe?).

Please support me on Patreon https://www.patreon.com/porres
You can follow me on instagram as well if you like... I'm always posting Pd development stuff over there https://www.instagram.com/alexandre.torres.porres/

cheers

ps. Binaries for mac/linux/windows are available via deken. I needed help for raspberry pi

CHANGELOG:

LIBRARY:

Breaking changes:

• [oscope~] renamed to [scope~]
• [plaits~] changed inlet order of modulation inputs and some method/flags name. If a MIDI pitch of 0 or less input is given, it becomes a '0hz'.
• [gbman~] changed signal output range, it is now filtered to remove DC and rescaled to a sane -1 to 1 audio range.
• [dust~] and [dust2~] go now up to the sample rate and become white noise (removed restriction that forced actual impulses, that is, no conscutive non zero values)
• [cmul~] object removed (this was only used in the old conv~ abstraction to try and reduce a bit the terrible CPU load)
• [findfile] object removed (use vanilla's [file which] now that it has been updated in Pd 0.55-0)
• [voices] swapped retrig modes 0 and 1, 'voices' renamed to 'n', now it always changes voice number by default as in [poly] (this was already happening unintentionally as a bug when one voice was already taken). The 'split' mode was removed (just use [route], will you?)
• [voices~] was also affected by changes in [voices] of course, such as 'voices' message being renamed to 'n'.
• [sr~]/[nyquist] changed output loading time to 'init' bang
• [sample~] object was significantly redesigned and lots of stuff changed, new messages and flags, added support for 64-bit audio files (Pd 0.55 in double precision and ELSE compiled for 64 bits is required for this). Info outlet now also outputs values for lenght in ms and bit depth.
• [sfont~] uses now a simpler build system and this might not load very very rare and unusual sound formats.

Enhancements/fixes/other changes:

• builds for double precision is now supposedly supported, by the way, the build system was changed from pd-lib-builder to pd.build by Pierre Guillot.
• [play.file~] is now a compiled object instead of an abstraction thanks to Tim Schoen, and it supports MP3, FLAC, WAV, AIF, AAC, OGG & OPUS file extensions.
• Support for double precision compilation was improved and should be working for all objects (not yet providing binaries and fully tested yet by the way).
• The ELSE binary now loads a modified version of [pdlua], but no [pdlua] and [pdluax] objects are provided.
• added signal to 2nd inlet of [rm~].
• fixed 'glide' message for [mono~].
• fixed [voices] consistency check bug in rightmost outlet and other minor bugs, added flags for 'n', 'steal' and offset.
• [gain~] and [gain2~] changed learn/forget shortcut
• [knob] fixed sending messages to 'empty' when it shouldn't, ignore nan/inf, prevent a tcl/tk error if lower and upper values are the same; added "learn/forget" messages and shortcut for a midi learn mechanism.
• [mpe.in] now outputs port number and you can select which port to listen to.
• Other MIDI in objects now deal with port number encoded to channel as native Pd objects. Objects affected are [midi.learn], [midi.in], [note.in], [ctl.in], [bend.in], [pgm.in], [touch.in] and [ptouch.in].
• [pi]/[e] now takes a value name argument.
• [sr~]/[nyquist~] take clicks now and a value name argument.
• fixed phase modulation issues with [impulse~] and [pimp~].
• [cosine~] fixed sync input.
• added multichannel features to [cosine~] and [pimp~].
• [plaits~] added a new 'transp' message and a functionality to allow MIDI input to supersede signal connections (needed for the 'merda' version [see below]), fixed MIDI velocity.
• [pluck~] added a new functionality to allow MIDI input to supersede signal connections (needed for the 'merda' version [see below]).
• 26 new objects, [velvet~], [vcf2~], [damp.osc~], [sfload], [pdlink] and [pdlink~], plus abstractions from a newly included submodule called "Modular Euro Racks Dancing Along" (M.E.R.D.A)! Warning, this is all just very very experimental still, the object are: [adsr.m~], [brane.m~], [chorus.m~], [delay.m~], [drive.m~], [flanger.m~], [gendyn.m~], [lfo.m~], [phaser.m~], [plaits.m~], [plate.rev.m~], [pluck.m~], [pm6.m~], [presets.m], [rm.m~], [seq8.m~], [sig.m~], [vca.m~], [vcf.m~] and [vco.m~] (6 of these are multichannel aware).

Objects count: total of 551 (307 signal objects [92 of which are MC aware] and 244 control objects)!

• 311 coded objects (203 signal objects / 108 control objects
• 240 abstractions (104 signal objects / 136 control objects)

TUTORIAL:

• New examples and revisions to add the new objects, features and breaking changes in ELSE.
• Added a couple of examples for network communication via FUDI and [pdlink]/[pdlink~]
• Section 36-Filters(Advanced) revised, added more examples and details on how [slop~] works.
• Total number of examples is now 528!

I was just pondering a signal-rate flip-flop toggle.

The "easy way" is a pulse counter, modulo 2 (left side). This works, but will eventually overflow.

Then I thought... if you have a "true" signal a, and a "false" signal b, and a condition signal x (0 or 1), then if(x, a, b) == (a-b) * x + b (algebraic reduction of a*x + (b * (1-x)) = ax + b - bx).

If the "true" signal flips the last output -- a = 1 - y1[-1] -- and the false signal just outputs the previous value -- b = y1[-1] -- then:

((1 - $y1[-1]) - $y1[-1]) * $x1[0] + $y1[-1]
(1 - (2 * $y1[-1])) * $x1[0] + $y1[-1]

... which does satisfy the cases:

• if $x1[0] == 0, the first term disappears and it outputs the previous sample, ok.
• if $x1[0] == 1:
  • if $y1[-1] == 0 (as it is initially), then: (1 - (2*0)) * 1 + 0 == 1, ok.
  • if $y1[-1] == 1, then: (1 - (2*1)) * 1 + 1 == -1 + 1 == 0, ok.

So, in theory, that should work in fexpr~. But I only get zeros.

23-0224-toggle-ff.pd

I can imagine only a couple of possibilities. One is that I misunderstood what is $y1[-1]. The other is that there might be a bug.

(Yes, I know fexpr~ is slow... but it should be able to do this, no...?)

Just before posting, I tried replacing $x1[0] with $x1[-1] and... it starts toggling! Uh... why? In theory I should be taking the current trigger sample vs the previous output sample, which is what I wrote, but it only works to take the previous trigger sample...? (Point being, if I did this in a class, I couldn't explain why $x1[0] is bad while $x1[-1] is good. These are all deterministic operations, so it should be explainable.)

EDIT: Checking the tables' list views, I found that the fexpr~ with $x1[-1] is one sample late... confirming that $x1[0] is theoretically correct. So I'm puzzled why it fails.

EDIT EDIT: After quitting and relaunching Pd, $x1[0] works soo... never mind... weird.

hjh

Is the following possible:

Connecting dialectic mic to pd like that one or that one (does it need to pass through arduino? how can I first convert the analog signal to digital signal?

After receiving the signal in pd - is it possible to analyze the signal based on frequency and/or intensity ? so if the signal is at bandwidth x (between range of frequencies) it will trigger audio file x, if the signal is in bandwidth y it will trigger audio file y and so on

Is the above possible?

Thanks for any advising

@lo94 The old way to use audio outputs to control CV gear is to use a high frequency sine (highest your interface can put out) as a carrier for your CV, amplitude modulate the sine with what you really want and send it to [adc~]. Then you run it through a full wave rectifier and maybe some filtering and you have your CV. If you have some soldering skills you can make a two channel CV output that you can plug into your computer headphone jack for maybe $5, if not you can always go with simple passive rectifier and filter which is as easy as it get but you loose some of the signals amplitude in the process, if you have a VCA on that modular you use that too make that loss up or if your modular has a fullwave/halfwave rectifier and filter that can go down to DC than you can use those to do the lifting for you. It can be tricky to get well calibrated CVs this way, probably not too bad with a decent audio interface but if you don't need perfect key tracking than it is not a problem anyways. You can do very complex CV stuff this way on even simple analog synths with just a 1v/oct and expression input for CV, create all the modulation in pd or the like, sum and scale them then send them out, you can even piggy back gate and trigger signals on the audio signal by exploiting DC offset or just gating the CV itself if you are OK with the CV ceasing when the release cycle of the envelopes starts.

Let me knwow if you need more info, I kept it simple since I have no idea of your skill level.

Edit: if you want more info it would also be helpful to know your modulars CV/gate standards and what sort of CVs you want to send, plain modulators like LFO are much simpler than well calibrated key CV.

Edit 2: just remembered you said midi note information, not modulation. so needs to be calibrated. A bit more work but not much, very helpful to have access too a meter for calibration but can be done by ear if you do not.

Edit 3: we can probably use pd for calibration instead of a meter. Have not done this since the 90s, but it has me thinking and kind of makes me wish I still had some analog gear so I could play with this, will be much simpler in these days of modern interfaces, although the tendency for modern interfaces to have an analog volume control does complicate things some.

@MarcoDonnarumma Just had a cursory look at you video.... very good.
So you will understand more musical tech terms.
With your low sample rate the steps in the waveform are massive. The result is approximately what you would ask of a "fuzz" box.

Your ears only hear the rapid changes where the waveform rises or falls vertically. Your brain can only interpret the audio that way as it gets no intermediate information.

A rapid rise or fall like those you see in the scope are..... because the signal rate is now (after [sig~] ) 44100 samples per second..... actually a very high frequency signal....... one half (the left or right half) of a 22KHz "note".
Usually called "aliasing"...... they are there because there was no information before or after to give the real analogue slope of the wave before it was sampled.

For CD audio the rate is also 44100Hz. The Nyquist is 22050Hz and a low pass filter in the DAC removes everything above 20000Hz so as to remove such artefacts.

Putting a [lop] will smooth out those steps and approximate the original waveform as it was originally sampled. The downside is that you will no longer hear the audio because it is likely outside the range of most peoples hearing (in this case only!.... with audio below 40Hz..... if it was a 200Hz note you would hear it).

If you put a [lop~] (between [sig~] and [dac~] with a fader connected to its right inlet with a range of say 0.3 to 40 you can play with the fader to get an audible signal that is close to what you want to hear.
A sort of "depth of fuzz".
You might need a slightly different range on the fader..... 0.1 to 25 or something.
Looking at your video....... seeing (yes, ok, hearing) that you need audible sound.... not 192KHz audiophile sound.... and bearing in mind the original 40Hz analogue signal is inaudible to a lot of people, I don't think the clock problem is going to have a significant effect.

But if you are mistaken about the 40Hz maximum and there is actually more information that you need....... think heartbeat (low) + blood rushing (high) then upping your sample rate from the arduino will be necessary to get the higher frequencies.
500Hz sample rate will only give you up to 250Hz of audio information, just like 44100Hz only renders 22050Hz for our delectation.
David.

P.S. I don't much like maths..... not true..... it is fascinating but too hard for my feeble brain.
But without bothering to work it out, in theory, to preserve the signal the [lop] should be just below half the nyquist... so for 500Hz sampling a [lop~ 240] ?? should do that.
So I was way off base with the [lop~] values I posted above.
The signal would be very reduced (much lower I think) with the values I gave above.
However, [lop~] is not a very "sharp" filter... low dB/octave...... so maybe I was not in fact wrong...... and anyway you will need some distortion so as to properly hear your 40Hz note.
I hope you like to have someone struggling alongside you while you work.
I have to research what that means when the signal has already been oversampled by [sig~]..... but I think it doesn't matter.
Someone clever will tell us first no doubt.

@jameslo 's solution above should give a better outcome though than with a rather uncertain value for [lop~].
It will depend on what you need from your patch.

@Coalman said:

if I make a simple patch like this:
[osc~ 220]
| [bang]
|
[print~]

I get an array of numbers for each bang in the console.
Can someone tell me what these numbers represent?

are they amplitudes of the signal at various time points in chronological order?

Yes. Kind of. It's the value of each sample within the current DSP block (buffer) at the time of the bang. To test the chronology you could use a [phasor~] instead of [osc~] and have the same bang trigger a [0( message connected to phasor's phase input.

if I use [tabwrite~] to store them in an array in my patch instead of printing them to the console are the indices then in chronological order?

Yes, from the time of the bang and throughout the table. The difference here is that printing the buffer will give you as many samples as the block size (64 by default), where as you can set your table/array to be any length you like (you can also change the DSP block size on the fly with [block~] or [switch~] but only to powers of 2)

how does one get the frequency content of the signal back from this array of amplitude/time points?
I guess this is a question that is related to how FFTs work maybe...

FFT is indeed how you analyse frequency content. Samples stored in an array are a representation of your signal in the time domain. An FFT block is a representation of your signal in the (complex) frequency domain.

I am giving away the fact that I am not a signal processing wizard here, but I would be very interested to know how these different numeric representations of these signals show their (the same?) information..
thanks!
J

Your intuition is spot on, so don't beat yourself up.

Here is some good reading adressing everything you've touched on: http://www.pd-tutorial.com/english/ch03s08.html

One of the very early lessons that I teach in my interactive multimedia class is range mapping, where I derive the formulas and then leave them with ready-to-use patch structures for them.

1. If it's based on incoming data, first normalize (0 to 1, or -1 to +1, range). If you're generating a control signal, generate a normal range (e.g. [phasor~] is already 0 to 1).

2. For both linear and exponential mapping, there's a low value lo and a high value hi. (Or, if the normalized range is bipolar -1 to +1, a center value instead of lo.)

3. The "width" of the range is: linear hi - lo, exponential hi / lo.

4. Apply the width to the control signal by: linear, multiplying ((hi - lo) * signal); exponential, raising to the power of the control signal = (hi / lo) ** signal.

5. Then (linear) add the lo or center; (exponential) multiply by the lo or center.

One way to remember this is that the exponential formula "promotes" operators to the "next level up": + --> *, - --> /, * --> power-of (and / --> log, but that would only be needed for normalizing arbitrary exponential data from an external source). So if you know the linear formula and the operator-promotion rule, then you have everything.

• linear: (width * signal) + lo
• exponential: (width ** signal) * lo

(Then the "super-exponential" that bocanegra was hypothesizing would exponentiate twice: ((width ** signal) ** signal) * lo = (width ** (signal * signal)) * lo.)

0723-triple-mapping-demo.pd

[mtof~] is a great shortcut, of course, but -- I drill this pretty hard with my students because if you understand this, then you can map any values onto any range, not only MIDI note numbers. IMO this is basic vocabulary -- you'll get much further with, say, western music theory if you know what is a major triad, and you can go much further with electronic music programming if you learn how to map numeric ranges.

hjh

thank you, will study this, could you please add the missing

clone oddHarmonic~ 25
... couldn't create

@jameslo said:

what does it mean to low pass filter a signal at Nyquist, since the filter itself is subject to the Nyquist limit? Or maybe you have other reasons to do so?

Yes, that was wrong, I made many mistakes in that patch.

I'm still not convinced that comparing a signal with its upsampled version is meaningful. How do we know that the upper harmonics in the upsampled version are present and aliasing at the normal sample rate?

I see your point and I am not sure.
If this is the case, that whole concept would plop.

for measurement I wonder whether we should be instead passing the normal signal to an upsampled subpatch and then measuring how much energy is above 22050. Then again, that signal would look like a stepped signal in the upsampled environment, and that seems like it would have a lot of additional high frequency energy

No, I don't think so, as niquist reflects before upsampling. Even when using some upsampling filter, as in [inlet~]'s help, you can't un-alias an aliasing signal by upsampling !?

maybe you should try fft-split~

Yes, I was thinking about some steep FFT-filter,
https://forum.pdpatchrepo.info/topic/12245/perfect-filter-square-shape-filter
but don't know how to tweak it for different sample rates and with fairly precise cuttoff frequency, yet.

Bessel was not a good idea, high order Butterwoth is stronger, and everything needs to get calibrated, because the filter-slopes are messing with the result.