Here's a Pd-vanilla version of a linear-to-curved range mapping formula from SuperCollider. Published at https://github.com/jamshark70/hjh-abs . There's a message-rate and signal-rate version.

(Helpfile typo has been fixed but I won't bother to upload a new screenshot.)

This is, of course, close to cyclone's [scale]. It's close -- the symmetry is different (help file says "better" but actually that's a matter of taste, or situational need). For reciprocal values of [scale]'s "exponential" factor, the respective curves seem to be a reflection around the diagonal (x <--> y); for negated values of [lincurve]'s curve factor, one curve is a 180-degree rotation of the other (x --> -x and y --> -y). Compare -7 vs +7 lincurve against 0.2 vs 5.0 scale:

-7 curve, ~0.2 exp:

+7 curve, ~5.0 exp (lincurve and scale are very very close here!):

hjh

Wet-Dry AM.pd
Hey everyone,

I'm experimenting with trying to get a clean fade between the dry carrier sound and the wet modulated sound when using AM but am running into a couple problems.

1. When I make the modulating signal into a unipolar waveform, I can get a clean transition between the dry sound and modulated tremolo sound, but at faster modulation speeds the carrier can be heard with the other 2 frequency bands
2. When I use the bipolar waveform, the transition between the dry signal and the tremolo signal is sloppy and the rate begins to sound like it doubles

I'm wondering why the carrier sound is present when modulating at high speeds with the unipolar waveform but not with the bipolar waveform? And I'm also curious why the tremolo speed feels like it doubles in its rate when crossfading with the bipolar waveform?

Ideally, I would like to have the best of both worlds in the patch where it crossfades cleanly into a full tremolo, but also where the carrier signal doesn't exist when the modulation is fully wet. Not sure if videos work here, but I'll try and post one.[link text](Amplitude Modulation.mp4 link url)

there is no easy way to do a phase shift with sinesum

Correct-ish. You can shift any partial by 180 degrees bu multiplying it's amplitude by -1 I suppose

One could make a patch that could do a phase shift of a certain partials and then combine the two or three or more arrays

Yes. The solution is in this thread: https://forum.pdpatchrepo.info/topic/3506/phase-timing-help-basic

Alternatively, if you need phase control of each partial you could stack a bunch of [cos~] modules driven by the same phasor like this:

[phasor~]
|
[*~ n] (where n is partial#)
|
[+~ p] (where p is phase)
|
[cos~]
|
[*~ a] (where a is amplitude (duh))

@jameslo Analog components all have tolerances, +- 10% is not uncommon for pots, so any of the knobs between the two modules could be 20% off from each other given a worse case scenario, plus all those other components in the module could make it even further off, Generally circuits are designed to minimize the effects of component tolerances and high tolerance parts will be used in critical locations like frequency but amplitude and PW are not critical location, our ears will not notice small differences here. On top of that components age and their value changes, so two 30 year old modules can be very different despite once being identical. Things like phase and frequency are not constants in the analog world, things drift and oscillators which share a power source tend to sync when they get close. Most have to be very close in frequency to sync this way, the beat frequency will be well under 1hz but under certain situations you can hear it if you listen close. When trying to copy a patch from the analog world in the digital, knobs can only be used as a rough guide, need to use your ears.

When you distorted the wave you added harmonics and changed the strengths of the old harmonics, only harmonics of equal strength but opposite phase will cancel fully. You most likely did have cancellation just not full cancellation (depends on how you distorted it). Put both oscillators to the same frequency and opposite phase, listen to the distorted one and then add in the undistorted oscillator, you should hear a decrease in harmonic content from the cancellation. This will be most apparent with low to moderate levels of distortion, the more distortion, the more harmonics you are adding, the less the two waves have in common, and as a result less gets canceled.

Edit: didn't quite finish that last sentence.

@jameslo No time to check out your patch at the moment unfortunately..... maybe later..... looks interesting....

So this post might be total rubbish.

Absolute phase has some sort of meaning in analog audio because without equalisation a positive pressure at the microphone will produce a positive pressure at the speaker (unless it is wired out of phase) at exactly the same time... (not exactly true as electrons move at the speed of light...... but for us humans it holds true).
In the digital domain that cannot be true as there will be latency..... which means a phase shift..... which means the relationship no longer holds in time....... but it is just a delay that you can only hear if you are also listening to the original signal......... although the output is no longer technically in "absolute phase".

In the air it gets much more complicated..... conical speaker diaphragms...... reflections.... humidity.. pressure.....temperature...... wind...... comb filtering with stereo or if you really want to mess it up 7:1 surround..... and two ears placed where exactly......

Those elements and more produce phase differences that our brains are highly sensitive to..... its an evolutionary life saver.
So you will hear phase changes but if the phase is fixed there is no reference for your brain to access.

If you put a jack socket in your neck and feed your signal directly to your brain then you will be certain that you hear no artefacts.

So you hear changes..... but once the audio wave becomes "fixed" again in time you will "just" hear that wave.
If the gate changes the shape of the wave then I think you should be able to hear that though....

But the brain has trouble with tones because we are slow to recognise pitch..... from the Pd/doc/Manual......... cognition.zip ....... change_perception.pd is a good one.......

Probably.
I think.
David

My friend who is a much more experienced synthesist than I was telling me how important it was to consider whether the oscillators were synchronized with the gate (and if so, at what phase angle) because it affects the time-domain shape of the attack. He also said that his FFT-based spectrum analyzer showed no change even as the attack shape changed, but that you could still hear the difference.

I was doubtful because my understanding has always been that the ear can’t hear absolute phase and so that’s why, in additive synthesis, even if you randomize the phase of the harmonics, things should still sound the same. It’s particularly striking with squares and ramps; get the phase of even one harmonic wrong, and it looks like something completely different on a scope.

So I wrote a test in Pd, and while I don’t hear any differences in attack, I do hear differences in tone as the phases of the harmonics change. If you’re interested, try my test for yourself. I’m approximating a ramp using 50 harmonics, and you can control the phase of each harmonic using the phaseTable. Trigger the AR envelope with and without sync and see if you can hear a difference. Next, open up the gate so you get a steady tone and play with the phaseTable. You’ll hear the timbre change subtly, especially when changing the lower harmonics. Now I’m not sure I could tell you what phase I’m hearing in a double-blind test, but I do hear a difference. So am I wrong that the ear can’t hear absolute phase? And if I’m not wrong, why do I hear a difference? It sounds different on different speakers/headphones/earbuds so I’m guessing it has to do with asymmetries between the positive and negative excursions of the diaphragm?

additiveSaw~.pd
harmonic~.pd

@whale-av Thank you very much for your reply! In my case I need to route signal outputs of each cloned object inside one module to another cloned objects inside another module. What I mean is that the first module receives signals from outside of it. I don’t need to route signals between instances of cloned objects inside the module, what i need is to direct multiple signals coming from different instances of cloned objects inside one module and send them outside and then send them to cloned instances inside another module. Hope I managed to explain well enough. Thanks a lot!

@djaleksei You can route separately to specific modules within [clone] by including the clone number in the [send~] and [receive~]
You do that using $ in the name.
Clone 1 will assign the value of 1 to $1...... clone 2 the value 2...... etc,
So if you use [receive~ $1-modulation] then each cloned module can receive separately.
If each osc module is to be paired with each modulation module thenit is a simple task...... putting [send~ $1-modulation] in the modulation cloned abstraction will have fmmod1 send to osc1, fmmod2 send to osc2 etc.
You can give [clone] more arguments..... giving you the possibility to use $2, $3 etc. for more complexity...... i.e. if you one day add another bank of cloned oscillators to your patch.......
https://forum.pdpatchrepo.info/topic/9774/pure-data-noob/4
David.

Hello all!

I’m new to this forum! However I’ve been working with Pd for a while. Currently I’m programming a polyphonic synth in Pd with FM capabilities and I’m facing a problem. I have created some sort of a module which includes one oscillator, a low pass filter and an ADSR envelope generator. The output of this module is an audio signal. It also includes some inputs and one of them is a signal input for frequency modulation (it receives a signal from the outside and uses it to modulate the frequency of the oscillator inside the module). I use clone command to create multiple instances of the same module to make it polyphonic. Then I tried to route a signal from another cloned module to the FM input of the first module and the obvious ocurred: when I play only one note the FM works as supposed to. But when I play multiple notes it sounds terrible and it is because the audio signal of all the notes from the second module goes to the FM input of the first module and then it is routed to all the cloned instances of it. What I need is to route only the signal corresponding to one note and route it to the same note of the first module. Is there any way to achieve this? I understand that a signal goes equally to all instances of a cloned object, but is it possible get separate signals from all instances of the second module and route them separately to the corresponding instances for the first module?

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

--DOWNLOAD-- NoxSiren for :

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

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

--DOWNLOAD-- ORCA for :

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

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

--DOWNLOAD-- loopMIDI for :

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

#-= Cyber Notes [v15] =-#

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

#-= Special Modules [v15] =-#

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

#-= Current Modules [v15] =-#

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

NoxSiren integrated modules menu system.