Hello,

Some time ago I did an expander/compressor/limiter abstraction. It proved to serve well, therefore I'd like to share it now. It is done with an experimental technique using an analytic signal, hence the term quadrature compander, in short [qompander~] for the abstraction. It is just a mono, single band compander. Fast response is it's major characteristic.

The abstraction uses two objects specific to Pd extended: [expr] and [cyclone/pow~]. A help patch is included, with test signals and some demo presets which should give an impression of the compander's possibilities.

Illustrated documentation is at http://www.katjaas.nl/compander/compander.html

qompander~.pd and qompander~-help.pd are in the attached zip file.

Katja

update: as Maelstorm indicated, [expr] is in vanilla Pd. However, I found yet another Pd extended object in my patch, [iemlib/peakenv~]. I will soon upload a vanilla version of the compander.

http://www.pdpatchrepo.info/hurleur/qompander.zip

i did some qompandering and found some weirdness: see the attached patch

also the clip after pd pyth~ could be replaced by a max~ 1e-5 or just been thrown out altogether (like in my last post)
and the help patch should be named qompander-help~ not qompander~-help

thats all for today.. bye

http://www.pdpatchrepo.info/hurleur/slur-is-qompandering.pd

@slur said:

and the help patch should be named qompander-help~ not qompander~-help

Why? There's nothing wrong with that.

Thanks for testing Slur. Your patch shows that qompander~ can not handle excessive DC steps properly. Interesting matter. I have checked that compander~ the not-quadrature version can not handle this as well. It is not the DC level of say 200 dB itself (amplitude 100000 times unity) which is the problem, but the steps, the discontinuities. Although this is not regular use of a compander, the problem points to subtle but irritating artifacts which I could not reproduce in test conditions but which I heard in real life sound processing. I also checked limiter~ against your test conditions. Limiter~ can handle this. Now I am beginning to believe that limiter~'s convolution routine is crucial. After all, such extreme DC steps like in your test patch would be smoothened with convolution.

So, possibly we need limiter~'s convolution combined with qompander~'s curve and amplitude method but compensated for the phase distortion, to get fast and good quality noise reduction / compression / limiting. Can we do a FIR filter with Pd objects?

[clip~ 1e-05 100] can be replaced by [max~ 1e-05] that's right. The minimum value is there to prevent instability with very small numbers division which is more than a theoretical problem. The help patch name is qompander~-help otherwise Pd can not find that patch from the right-click menu.

Katja

@katjav said:

Can we do a FIR filter with Pd objects?

You can do them with [rzero~] and [czero~]. You should be able to do them with [fexpr~] as well, though my guess is it would be more cpu intensive.

well over here on windos the help patch couldn't be found until i renamed it
interesting what you found out... i was more talking about the clip 100 problem and the bump, at 4db (completely insignificant), which appears at high compander point settings

All tech aspects aside, I've been using qompander~ for about six months and now it finds its way into every patch I make.

It is a beautiful thing: simple and powerful. Thanks Katja.

I was wondering: why not apply the compander curve to the original signal instead of the complexified one? Wouldn't we get rid of the harmonic distortion?

@kyro said:

I was wondering: why not apply the compander curve to the original signal instead of the complexified one? Wouldn't we get rid of the harmonic distortion?

The filters used to create the analytic signal cause phase shifts (= frequency-dependent time shifts). From http://www.katjaas.nl/compander/compander.html:

"Since the measurement delay with the analytic signal is frequency-dependant, there is no practical way to sync the amplitude curve with the original signal. To cut things short, I decided to use the analytic signal itself as the audio, by summing real and imaginary phases and reduce the resultant amplitude with factor 1/sqrt(2). Then, the measured amplitude will by definition always exactly match the processed signal."

Oh. Turns out I was absent-minded.