>
> y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2] -
> (a1/a0)*y[n-1] - (a2/a0)*y[n-2]
>
>
> e.g. for samplerate=44100Hz, Q=1.0, frequency=100Hz:
>
> b0/a0 = 5.03944E-05
> b1/a0 = 1.00789E-04
> b2/a0 = 5.03944E-05
> a1/a0 = -1.98565E+00
> a2/a0 = 9.85854E-01
>
> I am working with 24-bit samples and need to keep everything within
> 32-bits. Therefore I can afford to multiply the above coefficients by
> 256 for fixed point calculations. I need to have coefficients within the
> range 1/256 to 256. In the above example the first three coefficients
> will all be 0 (even after multiplying by 256) so I will not get any
> sound out.
>

[b0 b1 b2 a1 a2]/a0*256 = [0.0129 0.0258 0.0129 -508.3264
252.3786]
the first 3 terms are not 0. In fixed point, once you chose your fixpt
format,
in this case, 24bit, should be able to expressed in Q10.14.
You said, to have the coefficients in 1/256~256, should multiply 128,
[b0 b1 b2 a1 a2]/a0*128 = [0.0065 0.0129 0.0065 -254.1632
126.1893]
Why set range on the coefficients, you may scale the singal and leave
some space for accurate coefficients. Besides, you may try some
multirate filter in this design.
Lin. S.

Reply by Vic●August 19, 20032003-08-19

> b0/a0 = 5.03944E-05
> b1/a0 = 1.00789E-04

Hi,

> b2/a0 = 5.03944E-05

will be 1 i.e. scaler=round(1/5.0E-05) to 2^n. If overflow will be by
adding then scaler :2, but values b2/a0 rest 1 if you afraid 0. You
don't see difference after division on scaler(>>n). Error of round up
isn't reason for absent of sound.
Cheers

Reply by Mike Rosing●August 17, 20032003-08-17

Malcolm Haylock wrote:

> Hi Everyone,
>
> I'm a newcomer to DSP and am trying to write a Fixed Point
> implementation of the bilinear transform lowpass filter as outlined in
> the Audio EQ Cookbook
> (http://www.harmony-central.com/Computer/Programming/Audio-EQ-Cookbook.txt):
>
>
> y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2] -
> (a1/a0)*y[n-1] - (a2/a0)*y[n-2]
>
> where for a lowpass filter:
>
> b0 = (1 - cos)/2
> b1 = 1 - cos
> b2 = (1 - cos)/2
> a0 = 1 + alpha
> a1 = -2*cos
> a2 = 1 - alpha
>
> where :
> cos = cos(2*pi*frequency/sampleRate)
> alpha = sin/(2*Q)
>
> The trouble is that I can't find a way to get the coefficients within
> the range of my fixed point calculations. For 44.1kHz sample rate things
> are fine with cutoff frequencies down to about 400Hz but below that
> things get tricky.
>
> e.g. for samplerate=44100Hz, Q=1.0, frequency=100Hz:
>
> b0/a0 = 5.03944E-05
> b1/a0 = 1.00789E-04
> b2/a0 = 5.03944E-05
> a1/a0 = -1.98565E+00
> a2/a0 = 9.85854E-01
>
> I am working with 24-bit samples and need to keep everything within
> 32-bits. Therefore I can afford to multiply the above coefficients by
> 256 for fixed point calculations. I need to have coefficients within the
> range 1/256 to 256. In the above example the first three coefficients
> will all be 0 (even after multiplying by 256) so I will not get any
> sound out.
>
> Am I being realistic and is there a way around this using the above
> design or should I be using a different design altogether?

If you have 32 bits then you have plenty of room for scaling. One
thing to do is to scale the b coef's as 1, 2, 1, do the sum for x terms,
then multiply by 1/a0 to add in the y terms.
The other thing to note is that you're assuming the y terms are going to
be pretty small, so work with a0*y[n] = b0x[n] + b1x[n-1] + b2x[n-2] +
a1/a0*(a0*y[n-1]) + a2/a0*(a0*y[n-2]) This puts all your coefficients in
the same range as well as your results. Your final output has to be
scaled back by a0.
To make life easy, change a0 in my formula to 2^n, and then you can scale
by a simple shift. Adjust the coefficients accordingly too!
Patience, persistence, truth,
Dr. mike
--
Mike Rosing
www.beastrider.com BeastRider, LLC
SHARC debug tools

Reply by Malcolm Haylock●August 16, 20032003-08-16

Hi Everyone,
I'm a newcomer to DSP and am trying to write a Fixed Point
implementation of the bilinear transform lowpass filter as outlined in
the Audio EQ Cookbook
(http://www.harmony-central.com/Computer/Programming/Audio-EQ-Cookbook.txt):
y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2] -
(a1/a0)*y[n-1] - (a2/a0)*y[n-2]
where for a lowpass filter:
b0 = (1 - cos)/2
b1 = 1 - cos
b2 = (1 - cos)/2
a0 = 1 + alpha
a1 = -2*cos
a2 = 1 - alpha
where :
cos = cos(2*pi*frequency/sampleRate)
alpha = sin/(2*Q)
The trouble is that I can't find a way to get the coefficients within
the range of my fixed point calculations. For 44.1kHz sample rate things
are fine with cutoff frequencies down to about 400Hz but below that
things get tricky.
e.g. for samplerate=44100Hz, Q=1.0, frequency=100Hz:
b0/a0 = 5.03944E-05
b1/a0 = 1.00789E-04
b2/a0 = 5.03944E-05
a1/a0 = -1.98565E+00
a2/a0 = 9.85854E-01
I am working with 24-bit samples and need to keep everything within
32-bits. Therefore I can afford to multiply the above coefficients by
256 for fixed point calculations. I need to have coefficients within the
range 1/256 to 256. In the above example the first three coefficients
will all be 0 (even after multiplying by 256) so I will not get any
sound out.
Am I being realistic and is there a way around this using the above
design or should I be using a different design altogether?
Thanks very much,
Malcolm Haylock
smaugNOSPAM@kagi.com (remove NOSPAM)