Old 11-15-2011, 07:51 AM   #1
domino49
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Default BBE Sonic Maximizer

A JS modelisation of the "BBE Sonic Maximizer" directly by the electronic schematic.

Last version :
Code:
/* 
  BBE Sonic Maximizer modelisation

  (C) 20011, Dominique Guichaoua.

  Peek Detector code is Copyright (C) 2011 lubomir i. ivanov

  This program is free software; you can redistribute it and/or modify
  it under the terms of the GNU General Public License as published by
  the Free Software Foundation; either version 2 of the License, or
  (at your option) any later version.
  This program is distributed in the hope that it will be useful,
  but WITHOUT ANY WARRANTY; without even the implied warranty of
  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  GNU General Public License for more details.
  You should have received a copy of the GNU General Public License along
  with this program; if not, write to the Free Software Foundation, Inc.,
  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.

*/


desc:BBE Sonic Maximizer

slider1:0<0,10,.1>Low Contour (dB)
slider2:0<0,10,.1>Process (dB)
slider3:0<-30,0,.1>Output Level (dB)
slider4:0<0,10,.1>Peek Detector Level

@init
c_ampdB = 8.65617025;

//Electronic Components
R=21500;
C1=0.0000000033;
C2=0.000000033;

Te=1/srate;

//Input Gain
Gin=-2;
//Band Pass Gain
GBP=0.72;
//Low Pass initial Gain (dB)
GLP0=0.47;
//High Pass initial Gain(dB)
GHP0=0.47;

//High Pass Coefficient
a0HP = -Te/(2*R*C1);
b0HP = a0HP;
//Low  Pass Coefficient
a0LP = -Te/(2*R*C2);
b0LP = a0LP;

//Peek Detector Coefficients
//  Copyright (C) 2011 lubomir i. ivanov
a0cv = 0.0003;
b0cv = 1 - a0cv;
//end lubomir i. ivanov code

//Front Panel Controls
@slider
//Low Pass Gain
GLP = GLP0*exp((slider1)/c_ampdB);
//High Pass Gain
GHP = GHP0*exp((slider2+GHP0)/c_ampdB);
//Ouput Gain
amp=exp(slider3/c_ampdB);

//Peek Detector close loop gain
//  Copyright (C) 2011 lubomir i. ivanov
cv_k = slider4;
//end lubomir i. ivanov code

@sample
//Input Buffer
s0=spl0*Gin;
s1=spl1*Gin;

//Peak Detector
//  Copyright (C) 2011 lubomir i. ivanov
cv0 = 1 - abs(s0)*cv_k;
cv1 = 1 - abs(s1)*cv_k;

cv_rc0 = a0cv*cv0 + b0cv*cv_rc0;
cv_rc1 = a0cv*cv1 + b0cv*cv_rc1;
//end lubomir i. ivanov code

//Pseudo State-Variable Filter (Trapezes)
BP0 = a0HP*HP0 + b0HP*HP0tmp + BP0;
BP1 = a0HP*HP1 + b0HP*HP1tmp + BP1;
HP0tmp = HP0;
HP1tmp = HP1;

LP0 = a0LP*BP0 + b0LP*BP0tmp + LP0;
LP1 = a0LP*BP1 + b0LP*BP1tmp + LP1;
BP0tmp = BP0;
BP1tmp = BP1;

HP0 = s0 + 1.5*BP0 - LP0;
HP1 = s1 + 1.5*BP1 - LP1;

//Summing Output Buffer
spl0 = -(LP0 * GLP + BP0 * GBP + HP0 * GHP * cv_rc0)*amp;
spl1 = -(LP1 * GLP + BP1 * GBP + HP1 * GHP * cv_rc1)*amp;

Last edited by domino49; 03-21-2013 at 09:04 AM.
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Old 11-15-2011, 08:04 AM   #2
matey
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great Domino, it sounds very good on my workplace laptop...
thanks
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Old 11-15-2011, 08:24 AM   #3
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Does the same thing as 3-Band-EQ, no? Except for fixing the x-over frequncies.
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Old 11-15-2011, 08:57 AM   #4
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Yes it's based on the 3Band-splitter and the 3Band-Joiner because the électronic works similarly.
http://www.radio-flier.com/PDF%20Files/BBE%20482i.pdf
In the schematic you can look a analog 3 bands splitter and 3 inputs summator.
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Old 11-16-2011, 03:38 AM   #5
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Thanks! Working really well on some muddy guitar picking I'm working on. Cleans it up nicely and brings out the detail. I think I'll find this extremely useful.
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Old 11-16-2011, 09:12 AM   #6
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Thank you for your valuable contribution, domino49 !
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Old 11-16-2011, 10:07 AM   #7
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I haven't tried this yet, but it seems like a cool plug-in. It certainly is way ahead of its time...

Quote:
Originally Posted by domino49 View Post
// (C) 20011, Dominique Guichaoua.
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Old 11-17-2011, 06:43 PM   #8
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Quote:
Originally Posted by domino49 View Post
A JS modelisation of the "BBE Sonic Maximizer" directly by the elctronic scematic.

// (C) 20011, Dominique Guichaoua.

//BBE Sonic Maximizer modelisation

desc:BBE Sonic Maximizer
slider1:0<0,10,.1>Low Contour (dB)
slider2:0<0,10,.1>Process (dB)
slider3:0<-30,0,.1>Output Level (dB)

@init
cDenorm=10^-30;
c_ampdB = 8.65617025;

freqHP = 2192.2;
xHP = exp(-2.0*$pi*freqHP/srate);
a0HP = 1.0-xHP;
b1HP = -xHP;

freqLP = 153.92;
xLP = exp(-2.0*$pi*freqLP/srate);
a0LP = 1.0-xLP;
b1LP = -xLP;

@slider
band1 = exp((slider1+3)/c_ampdB);
band2 = exp(0/c_ampdB);
band3 = exp((slider2+3)/c_ampdB);
amp=exp(slider3/c_ampdB);

@sample
s0 = spl0;
s1 = spl1;

sp0 = (tmplLP = a0LP*s0 - b1LP*tmplLP + cDenorm);
sp1 = (tmprLP = a0LP*s1 - b1LP*tmprLP + cDenorm);

sp4 = s0 - (tmplHP = a0HP*s0 - b1HP*tmplHP + cDenorm);
sp5 = s1 - (tmprHP = a0HP*s1 - b1HP*tmprHP + cDenorm);

sp2 = s0 - sp0 - sp4;
sp3 = s1 - sp1 - sp5;

spl0 = (sp0 * band1 + sp2 * band2 + sp4 * band3)*amp;
spl1 = (sp1 * band1 + sp3 * band2 + sp5 * band3)*amp;
Can someone tell me how to compile this or if someone did compile this can they post it as a DLL thanks
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Old 11-17-2011, 09:12 PM   #9
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There's no need to compile it...

just create a text file in the "effects" folder in reaper folder and then paste the code there and that's all

I just tried it and it sound great! I make some A/B test with the original BBE Sonic Maximizer plugin which I have and they are very similar in sound. Seems like it's going to be very used in my studio
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Old 11-18-2011, 08:18 AM   #10
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I have tried this out briefly and it seems pretty cool. Easy to change the overall sound of the track. But can someone explain this sliders a little better or describe what this is actually doing in more detail.
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Old 11-18-2011, 09:03 AM   #11
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Awesome, this one's a keeper. They should upgrade the JS list, I mean who really depends on SlightlyLouderator or whatever that one was.
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Old 11-18-2011, 11:10 AM   #12
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Default code

Hi all
I am a little mature and do not know which part of this should be left out of "The Code" on the text file.
(I know what a text file is I think?)
Would this include the Author and BBE Sonic Maximizer modelization or just the script below these two?
..................................

A JS modelisation of the "BBE Sonic Maximizer" directly by the elctronic scematic.

// (C) 20011, Dominique Guichaoua.

//BBE Sonic Maximizer modelisation

desc:BBE Sonic Maximizer
slider1:0<0,10,.1>Low Contour (dB)
slider2:0<0,10,.1>Process (dB)
slider3:0<-30,0,.1>Output Level (dB)

@init
cDenorm=10^-30;
c_ampdB = 8.65617025;

freqHP = 2192.2;
xHP = exp(-2.0*$pi*freqHP/srate);
a0HP = 1.0-xHP;
b1HP = -xHP;

freqLP = 153.92;
xLP = exp(-2.0*$pi*freqLP/srate);
a0LP = 1.0-xLP;
b1LP = -xLP;

@slider
band1 = exp((slider1+3)/c_ampdB);
band2 = exp(0/c_ampdB);
band3 = exp((slider2+3)/c_ampdB);
amp=exp(slider3/c_ampdB);

@sample
s0 = spl0;
s1 = spl1;

sp0 = (tmplLP = a0LP*s0 - b1LP*tmplLP + cDenorm);
sp1 = (tmprLP = a0LP*s1 - b1LP*tmprLP + cDenorm);

sp4 = s0 - (tmplHP = a0HP*s0 - b1HP*tmplHP + cDenorm);
sp5 = s1 - (tmprHP = a0HP*s1 - b1HP*tmprHP + cDenorm);

sp2 = s0 - sp0 - sp4;
sp3 = s1 - sp1 - sp5;

spl0 = (sp0 * band1 + sp2 * band2 + sp4 * band3)*amp;
spl1 = (sp1 * band1 + sp3 * band2 + sp5 * band3)*amp;

Many thanks in anticipation

Grinder
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Old 11-18-2011, 12:56 PM   #13
benj.calderon
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Hello grinder

the code starts with the "//"
those are mark for comments which means that they are part of the code but only for information to the user and the computer does not use that...

just start copy from the first "//" to the last ";"

-benj
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Old 11-18-2011, 01:09 PM   #14
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Great Benj, thanks for that, have a great Christmas.

Grinder
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Old 11-18-2011, 03:23 PM   #15
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Hi
A little bank of some presets (nothing fancy, just handy)
http://www.box.com/s/lxd63pimrq9zrxnpcr1i
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Old 11-18-2011, 05:08 PM   #16
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Quote:
Originally Posted by Link0126 View Post
But can someone explain this sliders a little better or describe what this is actually doing in more detail.
Yes please.
I also wondered about it.
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Old 11-19-2011, 03:55 AM   #17
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Talk about synchronicity! I was just looking at this plugin and hardware unit from a Sweetwater magazine earlier today. When I get some time tomorrow ill play around with your JS version.

I know this is probably a long and complicated question...but...How do you take the schematics from the circuitry and convert that to a JS plugin? I know it can be done but its way above my head.


Thanks for your effort.
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Old 11-19-2011, 05:13 AM   #18
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Sounds great. I'll be trying this on some sources in coming mixes.

Thanks very much for sharing.
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Old 11-19-2011, 05:17 AM   #19
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Quote:
Originally Posted by nofish View Post
Yes please.
I also wondered about it.
http://www.google.fr/search?q=bbe+so...ient=firefox-a
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Old 11-19-2011, 09:31 AM   #20
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i've seen plots where there was an allpass and also phase cancelation when boosting, which is perhaps for a different model and suggests different filters. its definitely not "just an eq" effect though...having said that the peak-detector in that schematic, did look a little bit odd.

here is a quick go at the dynamic part + "modeling" some other bits...adding CV will "duck" the "process" amount.

Code:
/*

  sonic enhancer thing

  Copyright (C) 2011 lubomir i. ivanov

  This program is free software; you can redistribute it and/or modify
  it under the terms of the GNU General Public License as published by
  the Free Software Foundation; either version 2 of the License, or
  (at your option) any later version.
  This program is distributed in the hope that it will be useful,
  but WITHOUT ANY WARRANTY; without even the implied warranty of
  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  GNU General Public License for more details.
  You should have received a copy of the GNU General Public License along
  with this program; if not, write to the Free Software Foundation, Inc.,
  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.

*/

desc: Sonic Enhancer

slider1:0<-12, 12, 0.1>Input
slider2:0<0, 10, 0.1>Low Contour
slider3:0<0, 10, 0.1>Process
slider4:2<0, 10, 0.1>CV
slider5:0<-12, 12, 0.1>Output
slider6:1<0, 1, {Off,On})>Noise Floor

@init
r_6 = 1/6;

nf_k = pow(2, -96 * r_6);

rc0_k0 = 0.0003;
rc0_k1 = 1 - rc0_k0;

fc = 700;
fs = srate;
pi2 = 2*$pi;
q = 0.23;

omega = pi2 * fc / fs;
sn = sin(omega);
cs = cos(omega);
alpha = sn / (2.0 * Q);

b0_ap = 1.0 / (1.0 + alpha);
b1_ap = (-2.0 * cs) * b0_ap;
b2_ap = (1.0 - alpha) * b0_ap;
a0_ap = b2_ap;
a1_ap = b1_ap;
a2_ap = 1;

b0_lp = 1.0 / (1.0  + alpha);
b1_lp = (-2.0 * cs) * b0_lp;
b2_lp = (1.0 - alpha) * b0_lp;
a1_lp = (1.0 - cs) * b0_lp;
a0_lp = a1_lp * 0.5;
a2_lp = a0_lp;

b0_hp = 1.0 / (1.0 + alpha);
b1_hp = (-2.0 * cs) * b0_hp;
b2_hp = (1.0 - alpha) * b0_hp;
a1_hp = -(1.0 + cs) * b0_hp;
a0_hp = -a1_hp * 0.5;
a2_hp = a0_hp;

@slider
g_in = pow(2, slider1 * r_6);
g_lp = slider2*0.5;
g_hp = slider3*0.5;
cv_k = slider4;
g_out = pow(2, slider5 * r_6);
noise_floor = slider6;

@sample
in0 = spl0*g_in;
in1 = spl1*g_in;

nf0 = 0.0;
nf1 = 0.0;
noise_floor ? (
  nf0 = (rand(2.0) - 1.0)*nf_k;
  nf1 = (rand(2.0) - 1.0)*nf_k;
);

out_ap0 = a0_ap*in0 + a1_ap*x1_ap0 + a2_ap*x2_ap0 - b1_ap*y1_ap0 - b2_ap*y2_ap0;
x2_ap0 = x1_ap0;
x1_ap0 = in0;
y2_ap0 = y1_ap0;
y1_ap0 = out_ap0;

out_ap1 = a0_ap*in1 + a1_ap*x1_ap1 + a2_ap*x2_ap1 - b1_ap*y1_ap1 - b2_ap*y2_ap1;
x2_ap1 = x1_ap1;
x1_ap1 = in1;
y2_ap1 = y1_ap1;
y1_ap1 = out_ap1;

out_lp0 = a0_lp*in0 + a1_lp*x1_lp0 + a2_lp*x2_lp0 - b1_lp*y1_lp0 - b2_lp*y2_lp0;
x2_lp0 = x1_lp0;
x1_lp0 = in0;
y2_lp0 = y1_lp0;
y1_lp0 = out_lp0;

out_lp1 = a0_lp*in1 + a1_lp*x1_lp1 + a2_lp*x2_lp1 - b1_lp*y1_lp1 - b2_lp*y2_lp1;
x2_lp1 = x1_lp1;
x1_lp1 = in1;
y2_lp1 = y1_lp1;
y1_lp1 = out_lp1;

out_hp0 = a0_hp*in0 + a1_hp*x1_hp0 + a2_hp*x2_hp0 - b1_hp*y1_hp0 - b2_hp*y2_hp0;
x2_hp0 = x1_hp0;
x1_hp0 = in0;
y2_hp0 = y1_hp0;
y1_hp0 = out_hp0;

out_hp1 = a0_hp*in1 + a1_hp*x1_hp1 + a2_hp*x2_hp1 - b1_hp*y1_hp1 - b2_hp*y2_hp1;
x2_hp1 = x1_hp1;
x1_hp1 = in1;
y2_hp1 = y1_hp1;
y1_hp1 = out_hp1;

cv0 = 1 - abs(in0)*cv_k;
cv1 = 1 - abs(in1)*cv_k;

cv_rc0 = rc0_k0*cv0 + rc0_k1*cv_rc0;
cv_rc1 = rc0_k0*cv1 + rc0_k1*cv_rc1;

out_sv0 = out_ap0 + out_hp0*cv_rc0*g_hp + out_lp0*g_lp;
out_sv1 = out_ap1 + out_hp1*cv_rc1*g_hp + out_lp1*g_lp;

out_rc0 = rc0_k0*out_sv0 + rc0_k1*out_rc0;
out_rc1 = rc0_k0*out_sv1 + rc0_k1*out_rc1;
out_dc0 = out_sv0 - out_rc0;
out_dc1 = out_sv1 - out_rc1;

spl0 = out_dc0*g_out + nf0;
spl1 = out_dc1*g_out + nf1;

@gfx 0 10
gfx_r = 0.8;
gfx_g = 0.5;
gfx_b = 0;

gfx_a = min(1, 0.55 + cv_rc0*0.25 + g_hp*0.1);
gfx_x = 5;
gfx_y = 5;
gfx_rectto(gfx_w*0.5 - 3, gfx_h - 5);

gfx_a = min(1, 0.55 + cv_rc1*0.25 + g_hp*0.1);
gfx_x = gfx_w*0.5 + 3;
gfx_y = 5;
gfx_rectto(gfx_w - 5, gfx_h - 5);
--
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Old 11-19-2011, 02:02 PM   #21
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Quote:
Originally Posted by Cisco Kid View Post
Talk about synchronicity! I was just looking at this plugin and hardware unit from a Sweetwater magazine earlier today. When I get some time tomorrow ill play around with your JS version.

I know this is probably a long and complicated question...but...How do you take the schematics from the circuitry and convert that to a JS plugin? I know it can be done but its way above my head.


Thanks for your effort.
Hi
It is very difficult for me to explain in English. Electronics, signal processing, scientifique computing and digital modelisation it's my job. Reaper with JS script is a ideal tool for signal processing and digital modelisation. To model an electronic circuit you have a choice between modeling electronic components (as in simulation software) and then model the function performed by a set of components. I chose the second solution.
The schematic : http://www.radio-flier.com/PDF%20Files/BBE%20482i.pdf
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Old 11-19-2011, 04:26 PM   #22
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Hi, Liteon
Thank you !

This eye candy blinking window looks good, and the ducking can help us achieve interesting effects.

http://img249.**************/img249/3...icenhancer.gif

Would there have any way to somewhat compensate, level-wise, the volume boosting when we engage the effect ?
(I mean, if we had roughly +2dB of "Low Contour" and +3dB of "process", the output will be 5dB louder than the input, the volume increase making us thing louder = better, so we have to decrease manually the output to get a proper before/after comparison. If it was possible to automatically decrease the input volume at the same time and by he same amount of "low" and "process" boosting, it would allow a better A/B comparison... Does it make sense ?)

Oh, and what does the "noise floor" feature do ?
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Old 11-19-2011, 04:34 PM   #23
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Like, how friggin' cool is this thread - like Christmas, without the heavy financial burden.

Here's some prefab to save you 30 seconds:
Attached Files
File Type: txt BBE_Model_Domino_JS.txt (930 Bytes, 388 views)
File Type: txt BBE_Model_LiteOn_JS.txt (3.4 KB, 376 views)
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Last edited by Kundalinguist; 11-19-2011 at 04:40 PM.
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Old 11-20-2011, 01:11 AM   #24
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Quote:
Originally Posted by sinkmusic View Post
Ok so supposedly it splits the lo, mids, and highs and offsets them by milliseconds so that they all reach your ear at the same time, thereby correcting phase differences amongst the frequencies. That's pretty bad ass if actually true.
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Old 11-20-2011, 10:44 AM   #25
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I've been using the Refined Audiometrics CLAS which is very similar to the BBE but it's much more subtle. I find this JS version much better for single tracks because it can do more extreme settings. It's amazing what we can get for free these days. I love js sonic maximizer for analog synths. Wow.. Thanks a lot guys!!!
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Old 11-21-2011, 04:58 AM   #26
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Hi,
New version with more comments and new modelisation
Attached Files
File Type: txt SonicMaximizer.txt (1.3 KB, 401 views)
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Old 11-21-2011, 10:07 AM   #27
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Thanks, Domino.
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Old 11-21-2011, 04:44 PM   #28
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Getting some weird pumping effects from the second version that aren't present in the first.
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Old 11-21-2011, 06:16 PM   #29
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Quote:
Originally Posted by Link0126 View Post
Getting some weird pumping effects from the second version that aren't present in the first.
Have you tried calling tech support?
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Old 11-21-2011, 06:22 PM   #30
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Quote:
Originally Posted by Link0126 View Post
Getting some weird pumping effects from the second version that aren't present in the first.
i noticed the same thing when i tried the second version as well.
i'm happy with the first version so i'm going back to that one for now.
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Old 11-22-2011, 01:54 AM   #31
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Quote:
Originally Posted by slops View Post
i noticed the same thing when i tried the second version as well.
i'm happy with the first version so i'm going back to that one for now.
Fin the line :
Code:
//Peek Detector close loop gain
cv_k = 8;
And adjust variable "cv_k" with less value (2 for example)...
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Old 11-22-2011, 02:10 AM   #32
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New version with a "Peek detector" front panel adjustment.
Code:
// (C) 2011, Dominique Guichaoua.
//BBE Sonic Maximizer modelisation

desc:BBE Sonic Maximizer
slider1:0<0,10,.1>Low Contour (dB)
slider2:0<0,10,.1>Process (dB)
slider3:0<-30,0,.1>Output Level (dB)
slider4:2<0,10,.1>Peek Detector Level

@init
c_ampdB = 8.65617025;

//Electronic Components
R=21.5e3;
C1=0.0000000033;
C2=0.000000033;

Te=1/srate;

//Input Gain
Gin=1;
//Band Pass Gain
GBP=1;
//Low Pass initial Gain (dB)
GLP0=3;
//High Pass initial Gain(dB)
GHP0=3;

//High Pass Coefficient
a0HP = -Te/(2*R*C1);
b0HP = a0HP;
//Low  Pass Coefficient
a0LP = -Te/(2*R*C2);
b0LP = a0LP;

//Peek Detector Coefficients
a0cv = 0.0003;
b0cv = 1 - a0cv;

//Front Panel Controls
@slider
//Low Pass Gain
GLP = exp((slider1+GLP0)/c_ampdB);
//High Pass Gain
GHP = exp((slider2+GHP0)/c_ampdB);
//Ouput Gain
amp=exp(slider3/c_ampdB);
//Peek Detector close loop gain
cv_k = slider4;

@sample
//Input Buffer
s0=spl0*Gin;
s1=spl1*Gin;

//Peak Detector
cv0 = 1 - abs(s0)*cv_k;
cv1 = 1 - abs(s1)*cv_k;

cv_rc0 = a0cv*cv0 + b0cv*cv_rc0;
cv_rc1 = a0cv*cv1 + b0cv*cv_rc1;

//Pseudo State-Variable Filter (Trapezes)
BP0 = a0HP*HP0 + b0HP*HP0tmp + BP0;
BP1 = a0HP*HP1 + b0HP*HP1tmp + BP1;
HP0tmp = HP0;
HP1tmp = HP1;

LP0 = a0LP*BP0 + b0LP*BP0tmp + LP0;
LP1 = a0LP*BP1 + b0LP*BP1tmp + LP1;
BP0tmp = BP0;
BP1tmp = BP1;

HP0 = s0 + 1.5*BP0 - LP0;
HP1 = s1 + 1.5*BP1 - LP1;

//Summing Output Buffer
spl0 = (LP0 * GLP + BP0 * GBP + HP0 * GHP * cv_rc0)*amp;
spl1 = (LP1 * GLP + BP1 * GBP + HP1 * GHP * cv_rc1)*amp;

Last edited by domino49; 12-02-2011 at 01:23 AM.
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Old 11-22-2011, 05:09 AM   #33
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bugfix:

too many zeros in the (c) year.
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Old 11-22-2011, 05:54 AM   #34
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Quote:
Originally Posted by reapercurious View Post
bugfix:

too many zeros in the (c) year.
Corrected Thanks
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Old 11-23-2011, 12:06 AM   #35
Evan
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Is this an effect that translates well to all playback mediums (mono/stereo speakers, headphones)?
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Old 11-23-2011, 09:10 AM   #36
IXix
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Quote:
Originally Posted by Evan
Is this an effect that translates well to all playback mediums (mono/stereo speakers, headphones)?
Yes.

Could somebody please explain how the new version differs from the original? What does changing the peak level slider actually do?
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Old 11-25-2011, 06:12 AM   #37
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Quote:
Originally Posted by IXix View Post
Could somebody please explain how the new version differs from the original? What does changing the peak level slider actually do?
its a bit hard to tell what CLAS and the BBE plugins are doing dynamically, but they are doing something similar. my version above, is pretty much the same frequency response wise to these plugins, apart from adding different amounts of gain!

the "CV" part though is something in the lines of "what may sound good". it grabs the current peak value and smooths it out with a one pole LP filter then multiplies this value to the gain of the "process" (or high band) to create a more "gentle" ducking effect. it is also controllable.

looks like domino has copy pasted my CV code into his version, which is fine, by me, of course. but his filters are still different (or wrong if you will). both the schematic and the plugins which i examined use a biquad state variable.

also this part:
Quote:
Originally Posted by the manual
the BBE sound processing system adds progressively longer delay times to lower frequencies
is missing in his version, due to the used filters.

another thing i've noticed is that he is adding "process" and "low contour" amounts by default, which i don't think is present in the bbe schematic, bbe plugin, CLAS etc.


disclaimer:
no intent to hijack the thread. just posting more info on the subject...

--
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Old 11-25-2011, 09:29 AM   #38
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Thanks for the explanation liteon.
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Old 11-25-2011, 05:59 PM   #39
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But in defense of both plugins, I think what matters is how they sound. I think both are excellent. And if in trying to do one thing someone invents something entirely different, then we have simply done as many famous inventors and explorers of the past, including Columbus, who never did make it to India that trip.
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Old 11-28-2011, 03:37 PM   #40
domino49
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Quote:
Originally Posted by liteon
the "CV" part though is something in the lines of "what may sound good". it grabs the current peak value and smooths it out with a one pole LP filter then multiplies this value to the gain of the "process" (or high band) to create a more "gentle" ducking effect. it is also controllable.
Thanks Liteon for your explication.

Quote:
Originally Posted by liteon
looks like domino has copy pasted my CV code into his version, which is fine, by me, of course.
Exactly, I copy and paste your CV code because I don't find any infos in the schematic or in the datasheet of the component NJM2153 regarding the operation of peak detector and CV feedback.

Quote:
Originally Posted by liteon
but his filters are still different (or wrong if you will). both the schematic and the plugins which i examined use a biquad state variable.
Look the filter on the schematic of the BBE Sonic Maximizer 482i :


The filter is a pseudo state variable filter composed of two integrators and a subtractor like in my filter. There are two cutoff frequencies at 224Hz and 2443Hz and not only one at 700Hz !!!

Quote:
Originally Posted by liteon
also this part:

Originally Posted by the manual
the BBE sound processing system adds progressively longer delay times to lower frequencies

is missing in his version, due to the used filters.
Look the complete schematic there are no delay line in the lower frequencies process.
http://www.radio-flier.com/PDF%20Files/BBE%20482i.pdf
The delay is only caused by phase shift of two integrators.

Quote:
Originally Posted by liteon
another thing i've noticed is that he is adding "process" and "low contour" amounts by default, which i don't think is present in the bbe schematic, bbe plugin, CLAS etc.
In the schematic the input buffer gain is setting at +3dB and the Band-pass gain at -3dB. I set the input buffer gain and the Band-pass gain at 0dB and the low-pass and high-pass gain at +3dB. it's the same...

Quote:
Originally Posted by liteon
disclaimer:
no intent to hijack the thread. just posting more info on the subject...

--
quite agree
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