Cycling 74 » Topic: DSP: left, center, right.

DSP: left, center, right.

Roman Thilenius — Wed, 08 Sep 2010 03:14:14 +0000

i am still looking for different methods for stereo to left/mid/right
encoding, i.e. i want to distinguish between “mono” and “left only”
in a stereo source and distribute the “mono” content of a stereo
source onto a third channel.

did anyone implement something like this and wants to share his method?

take care: dont recommend the use of addition and substraction, that wont work. :)

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Re: DSP: left, center, right.

Roald Baudoux — Wed, 08 Sep 2010 19:23:07 +0000

Basically what you need is a measure of correlation. When correlation is poor, you would route L and R channels to their normal destinations. And when the correlation increase you would redirect more signal from both L and R to the third channel.

But how to estimate the correlation ? Maybe by comparing the energy (envelope follower) of the sum versus the energy of the difference ?

Just an idea, I don’t have the time to try it now. Some DSP gurus might have better ones…

Re: DSP: left, center, right.

AlexHarker — Wed, 08 Sep 2010 19:57:30 +0000

Stereo positioning is one cue that can be used to attempt blind source separation (using frequency domain techniques). You might google this. It’s not by any means perfect but it does work to a decent extent.

Re: DSP: left, center, right.

Roman Thilenius — Wed, 08 Sep 2010 20:10:40 +0000

yeah i was thinking like that, it is a maatter of phase/correlation.

but wouldnt i have to do it in several stages? at an rms of 50 milliseconds quite some
of the spectrum will remain unanalysed :)

this is funny: http://www.blindsourceseparation.com/

Re: DSP: left, center, right.

Roald Baudoux — Thu, 09 Sep 2010 09:10:33 +0000

You don’t have to perform the RMS over 50 ms, you can go down to 1 sample if you like…

However I think a few ms of integration time doesn’t matter as the ear’s integration time is around 50 ms. And with frequency domain techniques there is also a kind of integration time linked to the window size and window step of the FFT.

You could also try a filterbank with each band coupled an envelope follower (like in the analysis stage of a vocoder).

Re: DSP: left, center, right.

Roman Thilenius — Thu, 09 Sep 2010 16:18:25 +0000

yeah you are right, in best case it is just user controllable/dynamic.

i found a nice schemata, but i dont fully understand it, what does
“multiply and then integrate under the curve” mean?

maybe somebody could translate that from chinese to maxmsp for me.

http://www.bores.com/courses/intro/time/2_ave.htm

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Re: DSP: left, center, right.

Roman Thilenius — Thu, 09 Sep 2010 16:20:46 +0000

and of course i´d love to avoid fft, btw. i was dreaming to make it work with delay~

Re: DSP: left, center, right.

Floating Point — Fri, 10 Sep 2010 00:09:01 +0000

As far as I understand it, integrate is just the area under the curve, so in dsp I’d imagine you’d accumulate the values of the samples over the rms period, and divide by the sample rate (maybe also multiply by a constant like 2pi, depending on application?)

I’d also imagine the accumulator would need to be a running accumulation where you’d add the latest sample value and subtract the earliest from a running total. Maybe there’s an integrator external to make all this easier, but I don’t think it would be to hard to patch.
T

Re: DSP: left, center, right.

Roman Thilenius — Fri, 10 Sep 2010 01:51:42 +0000

yeah i´ve heard soemthing like that , too, that you can calculate the whole thing
as radians (at least as one of many options).

but i wasnt abel to patch it myself, a working example would be great.

i understand that not too mny people know how to do this, but i am a bit
surprised that noone ever asked for it?
a search at the forums for “correlation” returned nothing :D

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Re: DSP: left, center, right.

volker böhm — Fri, 10 Sep 2010 17:44:48 +0000

basic correlation of bipolar signals is quite easy: multiply the signals and sum the results.
for the running sum or integrator you could use [average~] in bipolar mode.

i *think* something like the sketch below should get you closer.
(the clipping is a trick from an old neumann document, if i remember well – otherwise you’d have to somehow cope with the changing input levels. you might also be able to calculate the rms of both signals, multiply them and devide the correlation signal???).
vb

max v2;
#N vpatcher 356 123 1106 761;
#P toggle 57 211 15 0;
#P window setfont “Sans Serif” 9.;
#P comment 305 282 100 196617 … and the sum;
#P newex 222 240 34 196617 *~;
#P comment 263 242 125 196617 calculate the product…;
#N vpatcher 20 74 402 374;
#P outlet 151 221 15 0;
#P window setfont “Sans Serif” 9.;
#P newex 151 129 44 196617 *~ 100;
#P newex 151 100 85 196617 clip~ -0.01 0.01;
#P newex 63 129 44 196617 *~ 100;
#P newex 63 100 85 196617 clip~ -0.01 0.01;
#P inlet 151 80 15 0;
#P inlet 63 80 15 0;
#P outlet 63 220 15 0;
#P connect 1 0 3 0;
#P connect 3 0 4 0;
#P connect 4 0 0 0;
#P connect 2 0 5 0;
#P connect 5 0 6 0;
#P connect 6 0 7 0;
#P pop;
#P newobj 222 199 34 196617 p clip;
#P newex 122 180 41 196617 *~ 0.5;
#P newex 80 180 41 196617 *~ 0.5;
#P newex 80 210 52 196617 dac~;
#P user scope~ 452 181 582 311 32 1 128 -1. 1. 0 0. 0 0. 102 255 51 135 135 135 0;
#P user multiSlider 276 359 43 187 -1. 1. 1 2681 15 1 0 2 0 30 0;
#M frgb 0 0 0;
#M brgb 255 255 255;
#M rgb2 127 127 127;
#M rgb3 0 0 0;
#M rgb4 37 52 91;
#M rgb5 74 105 182;
#M rgb6 112 158 18;
#M rgb7 149 211 110;
#M rgb8 187 9 201;
#M rgb9 224 62 37;
#M rgb10 7 114 128;
#P user number~ 222 335 286 350 9 3 3 2 0. 0. 0 0. 50 0. 0 0 0 221 221 221 222 222 222 0 0 0;
#P newex 222 279 79 196617 average~ 2000;
#P message 222 56 31 196617 open;
#N sfplay~ 2 120960 0 ;
#P newobj 222 86 58 196617 sfplay~ 2;
#P message 288 57 35 196617 loop 1;
#P toggle 261 56 15 0;
#P connect 2 0 9 0;
#P connect 15 0 8 0;
#P connect 9 0 8 0;
#P connect 2 1 10 0;
#P connect 10 0 8 1;
#P connect 3 0 2 0;
#P connect 1 0 2 0;
#P connect 0 0 2 0;
#P connect 2 0 11 0;
#P connect 11 0 13 0;
#P connect 13 0 4 0;
#P connect 4 0 5 0;
#P connect 2 1 11 1;
#P connect 11 1 13 1;
#P connect 5 1 6 0;
#P connect 2 0 7 0;
#P connect 2 1 7 1;
#P pop;