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I'm designing a parametric equalizer using a spartan 3e starter board and a TI PCM codec I
wired to it, running at 48kHz.  I decided to use the biquad IIR filter core from opencores.org
which accepts 5 16bit normalized coefficients.  I'm using
http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt as a guide for calculating coefficients. 
(this is not platform specific and applicable for dsp chips too)

If I want to have a LP filter at, say 200Hz, I get coefficients that saturate to about 1 LSB. 
This seems like way too much aliasing or quantization or whatever you call it.  One solution
I'm considering is to downsample to increase my Fo/Fs and get more descriptive coefficients.  I
assume I would use averages or something when downsampling to avoid aliasing.  I would still
run my higher filters (2kHz, etc) at full sample rate.  Is this a reasonable solution or will
it suffer quality problems downsampling low bands?  Is there another type of digital filter
more suitable for say subwoofer crossover type of applications?

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D Gens-

> I'm designing a parametric equalizer using a spartan 3e starter board and
> a TI PCM codec I wired to it, running at
> 48kHz.  I decided to use the biquad IIR filter core from opencores.org
> which accepts 5 16bit normalized coefficients.
> I'm using http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt as a guide
> for calculating coefficients.  (this is not
> platform specific and applicable for dsp chips too)
>
> If I want to have a LP filter at, say 200Hz, I get coefficients that saturate
> to about 1 LSB.  This seems like way too
> much aliasing or quantization or whatever you call it.  One solution I'm
> considering is to downsample to increase my
> Fo/Fs and get more descriptive coefficients.  I assume I would use averages
> or something when downsampling to avoid
> aliasing.  I would still run my higher filters (2kHz, etc) at full sample
> rate.  Is this a reasonable solution or will
> it suffer quality problems downsampling low bands?  Is there another type of
> digital filter more suitable for say subwoofer crossover type of applications?

FPGA is a poor solution for high quality audio/acoustic applications, with one reason being
what you're discovering
the hard way:  difficulty in implementing longer precision.  For effective IIR filters
frequencies lower than Fs/Fc
ratios of about 250, many more bits than 16 are needed.  For a recent 16-bit TI DSP project we
used 64-bit precision
for the core biquad calculation sections.  We implemented this in asm language, used
instruction cache, dual MAC, etc
and the result was fairly fast.

An FPGA solution would be slower for each IIR filter, due to sequential sections, but would
overall be faster
depending on how many parametric filters were implemented in parallel.  But, the main points in
the real world are a)
how long to get it working, and b) how can the code be maintained and supported over the
long-term?  DSP = days, FPGA
= weeks.  C + small asm lang, easy to document between engineers.  Complex Verilog to run
multiprecision calculations,
not easy.

-Jeff

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Two suggestions--

For a biquad implementation, you can consider doing the sections which 
challenge 16 bit precision in double precision. Most DSP chips offer 
this  feature. It is commonly used when frequency gets small relative to 
the sampling rate, or when the rate of gain change with respect to 
frequency is large.

2nd suggestion
There is a second order filter algorithm used by audio engineers which 
realizes parametrics in which sensitivity to coefficients and other 
concerns are eased and low frequency performance is superior. Another 
advantage of this realization is that the user parameters (Fo, Q, and 
level of boost or cut) are mapped to the DSP EQ with virtual 
independence of each other so look up tables are smaller. The technique 
uses a second order allpass filter embedded in a matrix of forward paths 
where the allpass outputs are added and subtracted. Look for these 
references or key words--
	textbook by Mitra (most current one)
	Gray and Markel All Pass Lattice
	journal article by  Regalia and Mitra
	AES preprint paper by Dana Massie
	papers by Dutilleux (AES Journal 1989) and Maher

Good luck!

Christopher Moore

d...@engmail.uwaterloo.ca wrote:
> I'm designing a parametric equalizer using a spartan 3e starter board 
> and a TI PCM codec I wired to it, running at 48kHz. I decided to use the 
> biquad IIR filter core from opencores.org which accepts 5 16bit 
> normalized coefficients. I'm using 
> http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt 
> <http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt> as a guide for 
> calculating coefficients. (this is not platform specific and applicable 
> for dsp chips too)
> 
> If I want to have a LP filter at, say 200Hz, I get coefficients that 
> saturate to about 1 LSB. This seems like way too much aliasing or 
> quantization or whatever you call it. One solution I'm considering is to 
> downsample to increase my Fo/Fs and get more descriptive coefficients. I 
> assume I would use averages or something when downsampling to avoid 
> aliasing. I would still run my higher filters (2kHz, etc) at full sample 
> rate. Is this a reasonable solution or will it suffer quality problems 
> downsampling low bands? Is there another type of digital filter more 
> suitable for say subwoofer crossover type of applications?
> ------------------------------------------------------------------------
> 
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.503 / Virus Database: 269.15.12/1096 - Release Date: 10/27/2007 11:02 AM

-- 
-----------------------------------------------------------------------
Seven Woods Audio, Inc.                Christopher Moore
Concepts, Products, Circuits for Audio    Analog/Digital
m...@SevenWoodsAudio.com
http://www.SevenWoodsAudio.com

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Options:

1. Increase your number of bits
2. Decrease your sampling rate
3. Downsample

Apply 1 in any case. Apply 2 prefereably than 3, only in case you can not apply 2 go for 3.

Cheers.

Jaime Andrés Aranguren Cardona
j...@ieee.org
j...@computer.org

----- Original Message ----
From: "d...@engmail.uwaterloo.ca" <d...@engmail.uwaterloo.ca>
To: a...@yahoogroups.com
Sent: Friday, October 26, 2007 9:02:55 AM
Subject: [audiodsp] Biquad filtering with Fs/Fo > 200

I'm designing a parametric equalizer using a spartan 3e starter board and a TI PCM codec I
wired to it, running at 48kHz. I decided to use the biquad IIR filter core from opencores.org
which accepts 5 16bit normalized coefficients. I'm using http://www.musicdsp .org/files/
Audio-EQ- Cookbook. txt as a guide for calculating coefficients. (this is not platform specific
and applicable for dsp chips too)

If I want to have a LP filter at, say 200Hz, I get coefficients that saturate to about 1 LSB.
This seems like way too much aliasing or quantization or whatever you call it. One solution I'm
considering is to downsample to increase my Fo/Fs and get more descriptive coefficients. I
assume I would use averages or something when downsampling to avoid aliasing. I would still run
my higher filters (2kHz, etc) at full sample rate. Is this a reasonable solution or will it
suffer quality problems downsampling low bands? Is there another type of digital filter more
suitable for say subwoofer crossover type of applications?

__________________________________________________
 

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