"robert bristow-johnson" <rbj@audioimagination.com> escribi&#4294967295; en el mensaje 
news:f6f83b14-0aa5-4f1f-a85e-11080c7b1a40@d21g2000prf.googlegroups.com...
> On Dec 3, 11:44 pm, "Jaime Andres Aranguren Cardona" <j...@nospam-
> sanjaac.com> wrote:
> ...
>>
>> Robert, all... any comment on the approach by the Patent's authors?
>>
>
> i'm overwelmed, at the moment.  i got your stuff, but need more time.
>
> r b-j

Hello, Robert.

Thanks for showing up. Your input on the subject is very instructive, take 
your time.

Regards,

-- 

Jaime Andres Aranguren C.
SanJaaC Electronics
Soluciones en DSP
www.sanjaac.com 



-- 
Posted via a free Usenet account from http://www.teranews.com

On Dec 3, 11:44 pm, "Jaime Andres Aranguren Cardona" <j...@nospam-
sanjaac.com> wrote:
...
>
> Robert, all... any comment on the approach by the Patent's authors?
>

i'm overwelmed, at the moment.  i got your stuff, but need more time.

r b-j

<subscribemehere@yahoo.co.in> escribi&#4294967295; en el mensaje 
news:3bd2ce8b-90fe-4294-bb3d-541f42f20cc6@s19g2000prg.googlegroups.com...
> Hi All,
>
>>robert bristow-johnson: so, i would recommend designing the biquad 
>>boost/cut parametric EQ
>>using the simplest mathematical expressions and just try to get a good
>>definition for what you mean by Q or BW.
> Q = peak frequency of banpass/distance of 3dB points;
>
> What "Jaime Andres Aranguren" points is perfectly right.
> I have implemented the individual bandpass filters using "Robert
> Bristow-Johnson" method as in "http://www.musicdsp.org/files/Audio-EQ-
> Cookbook.txt".  And there is no confusion about it - "individual"
> bandpass filters are perfect.
>
> As Robert points out, if peking filters are used, problem is solved
> long back.  But as I went for "bandapass in parallel" instead of
> "peking in series" just to keep the gain outside the biquad loop.
>
> Input ------------------------------------------------------> + -------
>> Output
>            |                                                     |
>            |----------- BPF1 -----> Gain1 -----------> +
>            |----------- BPF2 -----> Gain2 -----------> +
>
> And here problem comes when gain happens to be negative - problem is
> when the individual filter output is added with the input.
> As Robert suggests, I had a thought about tweaking the Q, which again
> works well.  But wondering how do I find the "correct-Q" runtime
> (fixed-point implementation) - Quite challenging!
>
> Balaji.

Hello guys,

Well, the "tweaking" that has to be done to the Q in order to convert one 
boost filter to one cut filter with symmetric response is to change the Q 
for Q/k, and substract the filtered signal from the unfiltered one instead 
of adding, like on the boost case.

For a reference, see "Equalization Methods with True Response using Discrete 
Filters", by Ray Miller, AES Convention Paper 6088, presented at the AES 
116th convention in Berlin, Germany, May 8 - 11, 2004. It is mentioned in 
page 6, and proven in appendix II.

But again: it ONLY works for the SINGLE filter case. If you have several 
BPFs in parallel, such as the case for Interpolated Constant-Q Equalizer 
which emulates the easy to implement analog case, this approach will NOT 
work either.

It seemed like the Patent (the one originally introduced by Balaji (the OP) 
was going to solve the problem... But so far it solves nothing, there is 
some trick which does not match (at least my knowledge).

Robert, all... any comment on the approach by the Patent's authors?

Regards,

-- 

Jaime Andres Aranguren C.
SanJaaC Electronics
Soluciones en DSP
www.sanjaac.com 



-- 
Posted via a free Usenet account from http://www.teranews.com

Hi All,

>robert bristow-johnson: so, i would recommend designing the biquad boost/cut parametric EQ
>using the simplest mathematical expressions and just try to get a good
>definition for what you mean by Q or BW.
Q = peak frequency of banpass/distance of 3dB points;

What "Jaime Andres Aranguren" points is perfectly right.
I have implemented the individual bandpass filters using "Robert
Bristow-Johnson" method as in "http://www.musicdsp.org/files/Audio-EQ-
Cookbook.txt".  And there is no confusion about it - "individual"
bandpass filters are perfect.

As Robert points out, if peking filters are used, problem is solved
long back.  But as I went for "bandapass in parallel" instead of
"peking in series" just to keep the gain outside the biquad loop.

Input ------------------------------------------------------> + -------
> Output
            |                                                     |
            |----------- BPF1 -----> Gain1 -----------> +
            |----------- BPF2 -----> Gain2 -----------> +

And here problem comes when gain happens to be negative - problem is
when the individual filter output is added with the input.
As Robert suggests, I had a thought about tweaking the Q, which again
works well.  But wondering how do I find the "correct-Q" runtime
(fixed-point implementation) - Quite challenging!

Balaji.

"robert bristow-johnson" <rbj@audioimagination.com> escribi&#4294967295; en el mensaje 
news:c516eca8-5c69-4695-8cb6-327bfb650850@w34g2000hsg.googlegroups.com...
On Dec 3, 8:24 am, "Jaime Andres Aranguren Cardona" <j...@nospam-
sanjaac.com> wrote:
> <subscribemeh...@yahoo.co.in> escribi&#4294967295; en el 
> mensajenews:cfb41a00-f558-4317-8a66-1e9ac1bebbe9@a35g2000prf.googlegroups.com...
>
>
>
> > Hi,
>
> > I am trying to implement Symmetric boost and cut digital Equlizer
> > using bandpass filters.  I found this link which has some help:
> >http://72.14.235.104/search?q=cache:UIvZnXeDalgJ:www.freepatentsonlin...
>
> > In fact I did implement in Matlab.  But for -ve gains, output is
> > incorrect.  (gain is applied after the bandpass filter.).  Actually I
> > am still not clear how to calculate "k1" as given in Fig.14 of that
> > document.
>
> > I am using bandpass-butterworth-IIR-biquads; where b1=0 (in
> > b0*x0+b1*x1+b2*x2-a1*y1-a2*y2)
>
> > Has anybody implemented this before? Or any other general symmetric
> > cut-boost bandpass equalizer design? Help please!
>
> > Thanks,
> > Balaji
>
>

i tried the USPTO site:

http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1&Sect2=HITOFF&d=PALL&p=1&u=%2Fnetahtml%2FPTO%2Fsrchnum.htm&r=1&f=G&l=50&s1=5524022.PN.&OS=PN/5524022&RS=PN/5524022

and i still couldn't view the images (i'm using linux, and Firefox
says there is a missing plugin, but i can't figure out what it is and
how to get it).

---

Hi Robert, Vladimir, Original Poster, all:

For those of you that can not access the patent, you can access it on 
www.freepatentsonline.com You can suscribe for FREE and will be able to get 
the patent in .pdf format, which you can read everywhere. Here is the link 
http://www.freepatentsonline.com/5524022.html

Robert: I think that it is clear the equivalence on the various design 
methods for biquad IIR filters, regardless of their function. And you 
summarized many years of knowledge, BUT that is not the point on this 
thread; that problem is "solved". However, it works out fine ONLY for single 
filters, and not for a bank of parallel filters, like one needs to implement 
in several equalizer configurations, specially Interpolated Constant-Q 
Equalizers.

The point here is: that patent seems to solve the problem of the 
Interpolated Constant-Q Equalizers, namely the problem of the feedback 
delay, which is needed to implement the cut filters. Remember: we are 
talking here of parallel BP filters which behave as a whole, and not single 
2nd order IIR biquads.

More specifically, the point is that the solution given by the authors seems 
to have some special trick that seems to be wrong, and the implementation of 
the original poster (and mine too!) does not seem to work, although the 
patent says that it works AND they even provide frequency response graphs to 
prove that. It could be due to an error on the patent itself, that is 
exactly the point that the original poster (and I) want to disucuss 
publicily with gurus here.

Hopefully several of the audio gurus jump in to discuss that document.

-- 

Jaime Andres Aranguren C.
SanJaaC Electronics
Soluciones en DSP
www.sanjaac.com







-- 
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On Dec 3, 2:43 pm, Vladimir Vassilevsky <antispam_bo...@hotmail.com>
wrote:
> robert bristow-johnson wrote:
> >>I was thinking of a way to represent those degrees of freedom with a set
> >>of the obvious parameters, so every possible response type can be
> >>represented. Such as:
>
> >>1. Gain at zero
> >>2. Gain at Nyquist
> >>3. A frequency
> >>4. Gain at this frequency
> >>5. Q of this frequency
>
> >>What do you think?
>
> > it's exactly what i was thinking 14 years ago about this.
>
> ..And then what?

as long as we all know what we are talking about when we say "Q", then
it's just an issue of 5 equations and 5 unknowns.  not much more
designing to do.


> My next step was the joint optimization, search for global optima. Slow
> like hell, although it usually converges to a pretty optimal solution.


what's getting optimized?  there are no additional knobs left to twist
or twiddle.  you have these constraints:

1. Gain at zero
2. Gain at Nyquist
3. A frequency
4. Gain at this frequency
5. Q of this frequency

and you satisfy those constraints by adjusting a1, a2, b0, b1, and
b2.  what's left?

r b-j

robert bristow-johnson wrote:


>>I was thinking of a way to represent those degrees of freedom with a set
>>of the obvious parameters, so every possible response type can be
>>represented. Such as:
>>
>>1. Gain at zero
>>2. Gain at Nyquist
>>3. A frequency
>>4. Gain at this frequency
>>5. Q of this frequency
>>
>>What do you think?
> 
> 
> it's exactly what i was thinking 14 years ago about this.

..And then what?

My next step was the joint optimization, search for global optima. Slow 
like hell, although it usually converges to a pretty optimal solution.


Vladimir Vassilevsky
DSP and Mixed Signal Design Consultant
http://www.abvolt.com

ooops, forgot to reply to this.

On Dec 3, 2:12 pm, Vladimir Vassilevsky <antispam_bo...@hotmail.com>
wrote:

> I was thinking of a way to represent those degrees of freedom with a set
> of the obvious parameters, so every possible response type can be
> represented. Such as:
>
> 1. Gain at zero
> 2. Gain at Nyquist
> 3. A frequency
> 4. Gain at this frequency
> 5. Q of this frequency
>
> What do you think?

it's exactly what i was thinking 14 years ago about this.

r b-j

On Dec 3, 2:12 pm, Vladimir Vassilevsky <antispam_bo...@hotmail.com>
wrote:
> > but the problem with this is that is that the boost and cut are not
> > symmetrical (and i cannot find a drawing on the web that shows this).
> > from its *appearance*, the BW of the cut filters (of the same Q, same
> > w0, and opposite dB) is squished in comparison to the BW of the boost
> > filter.  if you want your boost/cut behavior to be symmetrical, you
> > must fudge the meaning of "Q", either in the boost, or in the cut, or
> > a little in both.
>
> Yes, of course. The classic definition of Q works only if the gain is
> higher then 1. If the gain is smaller then 1, then the design Q should
> be multiplied by gain.
>
> >     1.  you have only 5 degrees of freedom regarding the design
> >         of the coefficients to implement a 2nd-order paramtric
> >         boost/cut filter or *any* 2nd-order IIR filter.
>
> Actually you have 4 degrees of freedom per each section plus the common
> gain which is one more degree of freedom.

no, these are not cascaded BPFs where you can roll the common gain
into one big K at the beginning or end.  each of these sections have a
DC and Nyquist gain speced (usually to 0 dB) and the boost or cut is
relative to that spec.

each section has 5 specs to nail down, and you are fortuitously given
5 degrees of freedom (a1, a2, b0, b1, b2) to do it.

oh, i guess you're right, that you can factor out b0 from all of the
numerator coefs and roll that into one big K.  i tried to mention that
in the cookbook.

nonetheless, it doesn't change the main thesis i was making in the
"Equivalence of Various Methods..." paper.  i probably should have
sought to publish that, or something like it in the JAES, but i
thought, at the time, the potatoes were too small.  so it got a
convention preprint.

r b-j

> but the problem with this is that is that the boost and cut are not
> symmetrical (and i cannot find a drawing on the web that shows this).
> from its *appearance*, the BW of the cut filters (of the same Q, same
> w0, and opposite dB) is squished in comparison to the BW of the boost
> filter.  if you want your boost/cut behavior to be symmetrical, you
> must fudge the meaning of "Q", either in the boost, or in the cut, or
> a little in both.

Yes, of course. The classic definition of Q works only if the gain is 
higher then 1. If the gain is smaller then 1, then the design Q should 
be multiplied by gain.


>     1.  you have only 5 degrees of freedom regarding the design
>         of the coefficients to implement a 2nd-order paramtric
>         boost/cut filter or *any* 2nd-order IIR filter.

Actually you have 4 degrees of freedom per each section plus the common 
gain which is one more degree of freedom.

I was thinking of a way to represent those degrees of freedom with a set 
of the obvious parameters, so every possible response type can be 
represented. Such as:

1. Gain at zero
2. Gain at Nyquist
3. A frequency
4. Gain at this frequency
5. Q of this frequency

What do you think?



Vladimir Vassilevsky
DSP and Mixed Signal Design Consultant
http://www.abvolt.com