PWM: what does it stand for?

AdDeRoo · Post by **AdDeRoo** » Sat Apr 20, 2013 11:42 am

Just out of curiosity: Would anybody know what the PWM in many factory programs stands for?

section4 · Post by **section4** » Sat Apr 20, 2013 11:48 am

Pulse Width Modulation

Sparker · Post by **Sparker** » Sat Apr 20, 2013 2:07 pm

PWM is applied to the oscillators to give a time varying change to the sound of the oscillators.

There is a table on Page 187 of the Parameter Guide that gives an indication of what types of oscillator modulation can be used.

Some TLA (Three Letter Acronyms) that it's useful to understand are:

PWM - Pulse Width Modulation
PCM - Pulse Code Modulation (the audio samples sometimes used for oscillators)
AMS - Alternate Modulation Source (again see Pg 187 of the Parameter guide)
MG - Modulation Generator - A 'general' Modulation source for the MS20 and Polysix engines
VPM - Variable Pulse Modulation - Korg's version of FM (Frequency Modulation [cross-oscillator-modulation]
VPL - what you don't want when playing live ...

MarPabl · Post by **MarPabl** » Sat Apr 20, 2013 8:42 pm

Read Synth Secrets, Part 10: Modulation

nambuco67 · Post by **nambuco67** » Sat Apr 20, 2013 9:27 pm

PWM, Pulse Width Modulation, is a method used to convert a digital waveform (sequence of digital values or samples) into a analog waveform (audio signal for speakers).

A fixed frequency square signal (on / off) is generated with pulse width (relation between on / off time) controlled by each value of digital waveform. Since square signal frequency is much higher than digital waveform, the high frequency can be filtered leaving only the digital waveform converted to analog.

I hope I could explain the principle.....

Regards,

Paulo

MarPabl · Post by **MarPabl** » Sat Apr 20, 2013 9:53 pm

nambuco67 wrote:PWM, Pulse Width Modulation, is a method used to convert a digital waveform (sequence of digital values or samples) into a analog waveform (audio signal for speakers).

A fixed frequency square signal (on / off) is generated with pulse width (relation between on / off time) controlled by each value of digital waveform. Since square signal frequency is much higher than digital waveform, the high frequency can be filtered leaving only the digital waveform converted to analog.

I hope I could explain the principle.....

Regards,

Paulo

This is true from an electronics perspective... but for musical usage, the concept is different: you have the oscillator of your synth configured to output a square wave, and then you modulate the width of the square wave, so it goes from:
- Fully square (50% duty cycle), for example

Code: Select all

    ---     ---
   |   |   |   |
---    ---     ---

- To rectangular (even 1% duty cycle)

Code: Select all

     -      -
     ||     ||
-----  -----  -----

Saxifraga · Post by **Saxifraga** » Sat Apr 20, 2013 10:37 pm

The really interesting thing is that the spectrum of any periodic function like sine, saw, square and triangel has a spectrum of discrete lines our ear hears as what makes up the characteristic sound. This can be visualized by fourier transformation. The more frequencies are needed to construct the signal the richer the sound. That´s also what you hear if you open and close the filter.
You can hear the filter cutting away those frequencies in the King Korg filter test video. http://www.youtube.com/watch?feature=pl ... Z3SX2Qtug0 So if you modulate the pulse width of a square wave you get a spectrum that changes with time and changes the character of the sound drastically. You need more and other frequencies to construct the signal if the square expands or tightens. Similar things go for gauss signals which no synth uses as far as i know, but from wich the term pulswidth originated.
A sine wave of course consist only of one frequency and can not be modulated with the filter. Only the amplitude which corresponds to loudness of that sine signal can be changed.

Best you read some book about "Introduction to signal theory" (search on iTunesU). A synth makes much more sense if you know about the physics and mathematics of time signals and fourier transformation between time and frequency domain. It´s not that hard.

cobi · Post by **cobi** » Mon Apr 22, 2013 9:46 pm

MarPabl wrote:Read Synth Secrets, Part 10: Modulation

... very helpful link. Thanks.
It has been a long time since I used this stuff with my good old MS20 and my little Moog Prodigy.
The articles whet the appetite for tweaking some sounds ...

cobi

AdDeRoo · Post by **AdDeRoo** » Tue Apr 23, 2013 7:22 pm

Thanks all. Very helpfull

Sparker · Post by **Sparker** » Tue Apr 23, 2013 9:51 pm

I'll second that link to Synth Secrets

Plenty of good reading there for a week or three ...

taki · Post by **taki** » Wed Apr 24, 2013 12:27 am

Saxifraga - very good explanation in a paragraph about how time domain signals can be translated into the frequency domain to better explain what PWM signals may look like in that domain, where they are easier understood. Being an Electrical Engineer myself, you explained it perfectly in a short paragraph.

For those that it did not make sense, it might be useful to do some simple google searches and read about this topic for an hour, especially if you are doing any sound editing/creation using keyboards.

Taki

Saxifraga · Post by **Saxifraga** » Wed Apr 24, 2013 6:18 am

@Taki - Thank you!

1jordyzzz · Post by **1jordyzzz** » Wed Apr 24, 2013 5:43 pm

Saxifraga wrote:The really interesting thing is that the spectrum of any periodic function like sine, saw, square and triangel has a spectrum of discrete lines our ear hears as what makes up the characteristic sound. This can be visualized by fourier transformation. The more frequencies are needed to construct the signal the richer the sound. That´s also what you hear if you open and close the filter.
You can hear the filter cutting away those frequencies in the King Korg filter test video. http://www.youtube.com/watch?feature=pl ... Z3SX2Qtug0 So if you modulate the pulse width of a square wave you get a spectrum that changes with time and changes the character of the sound drastically. You need more and other frequencies to construct the signal if the square expands or tightens. Similar things go for gauss signals which no synth uses as far as i know, but from wich the term pulswidth originated.
A sine wave of course consist only of one frequency and can not be modulated with the filter. Only the amplitude which corresponds to loudness of that sine signal can be changed.

Best you read some book about "Introduction to signal theory" (search on iTunesU). A synth makes much more sense if you know about the physics and mathematics of time signals and fourier transformation between time and frequency domain. It´s not that hard.

Hey saxifraga, i'm interested in learning fourier transform since i'm attracted to music and being a computer science student myself, i think it'll be helpful for my future papers... but can you provide the link that is very simple on explaining fourier transform?? looking at wikipedia and suddenly there's a star ringing around my head.. my maths are suck, i want to change it.. thanks!

Saxifraga · Post by **Saxifraga** » Wed Apr 24, 2013 10:48 pm

1jordyzzz wrote: Hey saxifraga, i'm interested in learning fourier transform since i'm attracted to music and being a computer science student myself, i think it'll be helpful for my future papers... but can you provide the link that is very simple on explaining fourier transform?? looking at wikipedia and suddenly there's a star ringing around my head.. my maths are suck, i want to change it.. thanks!

At the moment I know only a good book in german language (Lüke, Signalübertragung, Springer-Lehrbuch).
(As a primer I would read this one. Sadly, I lost my copy at the University:
Waves (Berkeley Physics Course, Vol. 3) [Hardcover]
Frank S. Crawford Jr. (Author) )

Do you have Mathematica Home Edition? Any other symbolic computation package with FFT (fast fourier transformation) will also do.
Mathematica is very useful because it can play the periodic waveforms directly via audio.

A different approach is to use Logic or another DAW with an AU/VSTi plugin that shows you the spectrum of the sounds on your tracks (like VoxengoSPAN_241_MacAU_setup.dmg (MAC)). You can experiment then how your waves look like in the frequency domain and experiment with the filters.

Ojustaboo · Post by **Ojustaboo** » Wed Apr 24, 2013 11:02 pm

Sparker wrote:I'll second that link to Synth Secrets

Plenty of good reading there for a week or three ...

I'll third it.

They are supposed to be releasing the entire series as a book at some point. Will definitely get it if they finally get round to it.

I haven't read the series in a while but last time, a few of the pic links were to the wrong diagrams, I sent them a list of the ones I noticed.

http://www.soundonsound.com/forum/showf ... t=1#991455