**Fixed point implementation of 4'th order IIR filters**Hi
Does anyone have some guidelines on how to implement a 4'th order low-pass
Butterworth IIR filter in fixed point. My cut-off frequency is relatively
close to the DC frequency so high precision is needed for the coefficients.
What about realization structure and so on!
I have implemented the bit-flipping algorithm in
http://www.cmsa.wmin.ac.uk/~artur/pdf/Paper16.pdf for quantization of
coefficients
and it indeed works, but does some other techniques allow for further
reductions of number of bits used to represent the coefficients.
Thomas
I've found the normalized lattice-ladder...

**couple of questions on practical IIR filter implementation on fixed point systems**Hello.
I am a professional programmer, but I am doing DSP just as a hobby.
Three years ago I designed a morse / PSK31 decoder on a 16bit no-
multiplier controller. After rouhgly 3 years of absence in the DSP
field I am back to design a HAM shortwave software defined radio using
a beautiful new part from Texas Instruments - TLV320AIC3254 codec with
miniDSP core. The part could do all the signal processing I need with
just a 5x5mm board space including earphone driver. That's an
engineer's dream come true.
The goal is to downconvert a SSB voice channel down to audio with a
quadrature m...

**IIR fixed-point implementation**HI everyone,
I was wondering if anyone can recommend me good books or papers on
fixed-point IIR implementation strategies on FPGAs or CPUs. Especially
regarding fc versus Fs and coefficient quantization. I have issues
with an order-1 DC-removal filter with Fs = 100 MHz and Fcut = 250
kHz. Basically, I need lots of precision in the intermediate
calculations which impact the number of bits for the multiplier and
thus the maximum frequency of my FPGA design.
Best regards
On 11/14/2011 1:15 PM, Benjamin Couillard wrote:
> HI everyone,
>
> I was wondering if anyone can recommend me good...

**Fixed Point IIR implementation**Hi pals,
I would like to implement a IIR Biquad filter using the fixed point
arithmetics...
Hence to reduced the intermediate states I plan to use the following trick:
s(k) = x(k) -a1*s(k-1) -a2*s(k-2)
y(n) = b0*s(k) + b1*s(k-1) + b2*s(k-2)
The I can compute each y, saving only 2 states (s(k-1) a�nd s(k-2))...
BUT ... in order to scale my coefficients or input, I need to know what
are the boundaries of the s(k) serie... in order to find a proper Fixed
Point format for my coefficients..
So my questions are :
* for which conditions, s(k) is bounded
* If the s(k) is bounded, what are the b...

**IIR Filter fixed point DSP**Hi
I am trying to design a IIR filter using a 16-bit fixed point DSP. I nee
to calculate the coeff in the form of second order sections. I would lik
to do this with MATLAB. Does anyone have some MATLAB code that will d
this. I found a routine on Texas Instrument site but as well as the coef
it creates some scaling factors for the inputs. I dont want this as I hav
seen other programs such as Filter Express that just give the Coeff.
Thanks
J
This message was sent using the Comp.DSP web interface o
www.DSPRelated.com
OK. You have several kinds of IIR filter, for instance elliptic (the
be...

**IIR filter on a fixed set of data points**I have a fixed 3600 data points(samples) and I want to apply a notch 2nd
order Butterworth IIR Notch filter to remove the 1/4Fs frequency, the
results start to converge(or filter start to work) after about 250 or so
samples. The issue is the it is a continuous stream of data and I need to
remove the harmonics on the initial 250 samples. What can I do to filter
the whole data set?
thank you,
Andrew
ahgu <honestgold@n_o_s_p_a_m.gmail.com> wrote:
> I have a fixed 3600 data points(samples) and I want to apply a notch 2nd
> order Butterworth IIR Notch filter to re...

**IIR biquad implementation in fixed point / integer**Hello,
I tried implementing a LPF biquad (that works perfectly in floating
point) with: a = [1.00000, -1.90887, 0.91129], b = [6.1150e-04,
1.2230e-03, 6.1150e-04] in fixed point / integer, with 16 bit
coefficients, 10 bit input signal and a 32 bit accumulator. I suspect
the coefficient resolution is not enough.
I tried:
- the direct form 1 and direct form 2 transposed structures, but the
output is zero if the input signal is not very high -- because of the
small b coefficients, the signal is attenuated very much in the FIR
portion of the filter and is lost;
int biquad_ab(d, a0...

**Maximizing dynamic range of fixed point IIR filter**
Let's say we need to implement H(z) = P(z)/Q(z) in the fixed point. The
typical implementation would be a cascade of biquads. So we factor P(z)
and Q(z) and distribute poles and zeroes between the stages.
The dynamic range of a filter is limited by overflow at the top, and by
quantization artifacts at the bottom. We can try all variants of
assignment of poles and zeroes to different stages to maximize the
dynamic range from rms quantization noise to full scale sine wave at the
"worst" frequency. So far so good.
However, this doesn't tell if some stage of f...

**Implementing 2nd Order IIR High Pass Filters**Hi
I tried implementing a 2nd order High Pass filter in Fixed Point with
cutoff of 20 kHz with a Sampling Frequency of Fs = 4MHz to remove D
offsets. Despite quantizing the coefficients to 31 bits, when I supply
DC signal, the values dont settle to zero. I used the butter function i
Matlab to generate the coefficients and subsequently quantize it
butter(2,1,20e3/4e6)
It would be helpful if somebody could share their experience and tell m
the right way to implement these filters. Incidentally, if there is
small signal riding on the DC offset, then the signal settles to a leve
whose mean ...

**Help to interpret impulse response of an fixed point IIR filter**I have done an impulse response of an IIR filter. The impulse response of
the filter does not return to zero but stops at a level close to zero (I
think it�s because limitations in fraction bits). The filter is implemented
in fixed point arithmetic�s.
And when I make an FFT of the impulse response de 0 Hz gain is not 0 dB as I
would like it to be.
But when I run a step response I get almost perfect 0 Hz gain of 0 dB.
So the problem is that the FFT of the impulse response says the gain is not
0 dB at 0 Hz but the step response says the gain is 0 dB at 0 Hz?
What should I do?
Not ...

**Matlba butterworth IIR filter implementation**Hi all,
I want to implement an butterworth low pass filter to filter the DC compone=
nt of a signal inside a PLL - phase locked loop. I am processing sample-by-=
sample. Therefore I have to filter the signal sample by sample (Not the who=
le signal or block by block). I have implemented a filter to filter the who=
le signal. But Im struggling to modify it make it work sample by sample bas=
is.
I highly appreciate if someone can advice me regarding this issue. I have i=
ncluded the code Im using to filter.
[b,a] =3D butter(10, 4*fc/Fs); % 10th Order butterworth filter coefficie...

**floating point to fixed point conversion-in implementation of navigation system algorithm**Hi,
I have a floating point c code to implement a navigation system
algorithm.
In order to implement it in the 64x+ fixed point DSP processor, i need to
have the code in fixed point. I learnt that there are few processor
specific IQmath libraries aiding this purpose. Is it practical to write
the code using these libraries or should i manually write codes for each
operation by scaling the inputs for each line of the code, in which case i
have to create look up table for the trigonometric functions etc which is a
harder process to begin with..
Thanks,
Divya
On 2/13/12 11:38 AM, divya_di...

**Simulating higher order IIR filters using cascaded low order sections**I'm trying to make a tool that implements various types of IIR filters up t=
o a certain maximum filter order (8 in my case). I've been learning about t=
he bilinear transform and how to use it to turn an analog transfer function=
into a something that can be implemented as a digital filter.
I already have working implementations for 1-5th order butterworth low pass=
filters, but I've noticed some issues with filter stability for filter ord=
ers above 3 when the cutoff frequency is a small with respect to the sample=
rate. Going to double precision helps a little bit bu...

**IIR filter Implementation**Hello Guys!
I have a brief signal, about 100 values, having unknown frequenc
components including a single frequency of interest to me. I have designe
an IIR filter using MATLAB that will filter out the single frequency.
However, since the signal is brief, I face the problem that the filter ha
a non-zero transient response while handling most of my signal. This cause
weird results.
Will it help if I pass a long sinusoid of the passband frequency befor
passing my signal? Is this a good way of preventing the transient respons
from affecting the filtering?
Thanks a million!
--- Par.
Thi...

**IIR filter order**I'm finally about ready to release an IIR filter design program called -
you guessed it - ScopeIIR (TM).
It currently supports the basic design types of Butterworth, Chebyshev,
and Elliptic. I'm trying to finalize the default and maximum IIR filter
order that the program will support. To help me do that, please tell me
about your experiences with IIR filter order, assuming the above types
and implementation via biquads, where applicable:
- Minimum order: (I assume 1)
- The typical order you use for most problems
- The maximum order you've ever used in a real sys...

**fixed point implementation**Hi,
i want to implement the equation
p3 = p3*0.5/cos(3*PI/8);
into 16bit processor, where p3 is a 16 bit value;
any pointers as to how to go about it
-vijay
n_vijay_anand wrote:
> Hi,
>
> i want to implement the equation
>
> p3 = p3*0.5/cos(3*PI/8);
>
> into 16bit processor, where p3 is a 16 bit value;
>
> any pointers as to how to go about it
It looks like '0.5/cos(3*PI/8)' is a constant, so simplify it.
cos(3*PI/8) = 0.99978861
0.5/0.99978861 = 0.50010571
Depending on the algorithm you might get away with simply dividing by 2:
p3 >...

**Fixed Point Filter**Hi,
i have a problem with a filter that i have constructed using the
fdatool. The Filter is a IIR Chebychev Bandstop with 8 sections. I
have exported the Filter to my Simulink model where i generate Code
from it with the RTW. I have a fixed point DSP(tms320f2812). Since
the filter is generated with floating point values the generated code
is very slow-> the floating point calculations are emulated by
code. For a single Multiplication it takes some hundred CPU cycles.
If I had a fixed point filter the calculations could be done much
faster(in best case 6 cycles per multiplication).
I have ...

**Fixed-Point Filtering**Hi guys,
I have a question:
What is the best, shortest way to filter a fixed-point signal and get the spectra of input and output signals?
I have the filter designed using fdatool and then used sfi to transform a sinusoidal signal into fixed-point. But I cannot import a fixed-point object to sptool and then filter it and get the spectra and etc.
...

**about IIR Filter implementation...**hi,
This is my first filter implementation and is made in Scilab, also the
design of the 3ord Chebyshev Type I IIR Filter (with the iir function). I
actually have 3 filters, so i can use them as an equalizer for an audio
application. I´m using Direct Form I Realization. Since I have to
implement this on a 24bit Fixed-Point DSP I had to scale my coefficients
by a factor of 256 (same as shifting 8 bits). The problem that I've been
having is that the output signal generated by the scilab script gets some
attenuation but only with my coefficients scaled, without the scale factor
I get 0 ...

**LMS fixed point implementation**Hi everyone!
I need a fixed point implementation for the LMS algorithm;I have to
implement it on a TI dsp board (c6711 or a c5000); can anyone suggest
me where I can find this code?
thank you guys!
f
Use this:
It is a single step for fixed point LMS.
c_ptr points to coefficient, r_ptr points to history.
muErr = e * mu;
mu = 2048(trial), using fixed mu;
accA, accB are at least 32 bit accumulators.
/* Macro: Performs single step of filter and adapt */
#define ADAPT_AND_FILTER(accA, accB, muErr, c_ptr, r_ptr) \
accB = 16384; \
accB += ((i3...

**Implement IIR Filter on FPGA**hi people,
I'm designing filter system called IIR filter on the FPGA kit, but it
doesn't work when I implement on FPGA. When i iput the signals, the
output results seem to not get any thing. I do not know whether my
source code is wrong or another reason. The FPGA kit operate normally
with other sources which i loaded in the past.
Can anyone give me some advices to test what parts in my project do
not work or give me some idea to test anything. I am in the mess. I
hope everyone can show me.
I am looking forward hearing from people soon,
Gordon Freeman wrote:
> hi people,
> ...

**IIR filter implementation in C**Hi all,
I have designed an algorithm in MATLAB, and am currently trying t
translate it into C code so that I can run it in real time. At the momen
I am stuck on the IIR filter. I am trying to implement the function
[B,A] = cheby1(4,1,[80/48000,16000/48000],'bandpass')
I have found two promising sources online that provide source code for a
IIR filter in C:
1) http://www-users.cs.york.ac.uk/~fisher/mkfilter/trad.html
2) http://www.csois.usu.edu/people/yqchen/filter.c/F.C
Both sources seem to work reasonably well for LPF cases (source tw
provides a sample data set that shows how c...

**Implementing cascade IIR filters**Dear comp.dsp,
I am just beginning my foray into practical DSP after learning the
bookish stuff in class. To try out various things, I manipulating wave
files in C, and in parallel, trying to make XMMS (http://www.xmms.org)
effect plugins to get the effect in real time.
My query is about IIR filtering. What is the best way to implement an
IIR filter, assuming that I get data in blocks of, say 1024 samples? To
filter, I need to maintain state information, and I have several
filters in cascade. Right now, what I am doing is:
1. Use pen and paper to get the difference equation to implement th...

**fixed point IIR coeffs**Hi i am using built-in Matlab functions to design IIR fixed point filter. For example:
[b a] = cheby1(2,6,0.25)
hd = dfilt.df1(b,a)
set(hd,'arithmetic','fixed')
Is there a simple method to check the coefficient values for fixed arithmetic? Precisely the ones which will appear in HDL code ( generatehdl(hd) ). Because i want to make a parametrized filter with a coefficient lookup table for different cutoff frequencies.
...

**IIR Notch Filter Implementation**Hi,
I want to implement a digital IIR notch filter on FPGA.
I used FDA tool HDL coder, but it seems not to work properly.
However, can you give me a vhdl code or example of it?
Thanks...
On Thu, 16 Aug 2007 09:05:55 -0500, m.nabaee wrote:
> Hi,
>
> I want to implement a digital IIR notch filter on FPGA.
> I used FDA tool HDL coder, but it seems not to work properly.
> However, can you give me a vhdl code or example of it?
>
> Thanks...
You have pointed up what I hate most about "wizards" and other software
that purports to do my thinking for me -- by the...