Some time ago I encountered a problem similar to yours.
The simple solution is to run the old and the new filters in parallel
and mix their outputs slowly changing the proportion. However that might
The permanent modification of the IIR filter coefficients...
On 14 May 2006 18:51:14 -0700, "robert bristow-johnson"
> > & I'm doomed forever to take
> > four metformin tablets every stinkin' day.
> had to look that one up. is it in lieu of sticking yourself 4 times a
Yep, that is correct.
> man, wh...
For those who have been touched by his noodly appendages, Rick's
dismisal of other's beliefs must have been quite shocking and distressing.
Eric Jacobsen wrote:
> David, there is a forum on venganza.org if you're interested. I've
> scanned through it a few times but...
> I can use an all-pass filter to fix the phase distortion for low
> frequency signals, correct?
I would rather build a amplitude equalizer which also linearizes the
phase. Say you have 10Hz -3dB cutoff and want to shift that to 2Hz, so
you must apply 15dB gain (10/2) at 2Hz. One implements th...
our DSP pal Jon Harris and I have exchanged a few
E-mails regarding the Goertzel algorithm. If you
recall, the Geortzel algorithm is implemented with
an IIR filter structure with a 2cos(2*pi*k/N) feedback
coefficient and an -exp(-j2*pi*k/N) feedforward
coefficient. The va...
I have a beginner-esqe question: all of my DSP literature shows the
IIR Difference Equation as:
y(n) = b(0)x(n) + b(1)x(n-1) + b(2)x(n-2) + ... + a(1)y(n-1) +
a(2)y(n-2) + ...
But then the C code implementations I see are something along the
FeedForwardVariable = (b * x[n]...
"Rick Lyons" wrote in message
> Hi Guys,
> Another idea for generating an analytic signal, that's
> bounced around this newsgroup for quite a while, is
> designing a lowpass filter and translating its center
On Fri, 18 Jul 2003 04:48:53 -0700, praveen wrote:
> But i don't know the exact freqency of f0 i.e if f0 is 200kHz then it
> will be anywhere between 195 to 205 kHz.
There are many techniques for determining the parameters of an unknown
sinusoid. For an overview, check out
I am trying to implement an FIR and an IIR filter in fixed point
The actual filter is of no importance(so I have implemented the
simplest filter possible),as long as it is in fixed point,and moreover
as I can give as input the desirable precision.
I wrote the following piece...
"Fred Marshall" wrote in message
> The answer is "it depends". If you can meet all the specs with one filter
> then why not? - subject to breaking the filter up into stages, etc.
> This implies that you want to upsam...
I have an application that has a standard IIR high pass filter with a
long time constant. During certain conditions, a shorter time
constant is required, so new coefficients .. with a shorter time
contant ... are swapped in. After a designated period, the original
coefficients are r...
You could try:
Prentice Hall - C Algorithms For Realtime Dsp
> are there any good books out there that actually explain how to implement
> DSP in C. In particular I am interested how to caluculated FIR/IIR filter
> coeffiencents and window them in C.
I wanted to know how to implement Hilbert transform using IIR filter.
Any reference or article or suggestion will be great.
I wanted to implement it on a DSP processor.
Hardware structure, filter coefficient?????
waiting for reply
email@example.com (Andrew Xiang) writes:
> In the matlab filter design filter box, when I click the Quantization
> button after design a IIR, the freq response seems very different. I
> cannot find the place where you can configure how many bits for the
> Where do you specifi...
On 19 Aug 2003 07:41:44 -0700, firstname.lastname@example.org
> Thanks for your concern in advance.
> I was told that a zero-phased filter Hzp(z) can be used to 'off-line'
> filter a time sequence x(n). Since its response Hzp(w) is real, so its
> output Y(w) ...
iam doing my postgraduation in communication in south india at Anna
University. Iam doing my project in Linear phase IIR filters. I have
the base materials for Linear phase iir filters. I wish to implement
the Linear phase IIR filter system in Matlab. so yet i haven't get an
idea how to im...
> Jerry Avins wrote:
> > Noway2 wrote:
> > ...
> > > It sounds like you are being asked to describe the pros and cons of
> > > both FIR filters implemented using the DFT and IIR filters.
> > ...
> > There are few FIRs that can be implement...
iam trying to implement Linear phase IIR filter. To acheieve Linear
phase in IIR filter we have the main technique of Time reversion of
the section. If anyone is doing in the same please reply.
thanks in advance.
As we all know, designing IIR filters is easy enough
for low filter order but is far more difficult when
higher order filters are required to meet design
contraints like low passband ripple, narrow transition
bands and high levels of attenuation in the stopband.
I'm currently w...
this post is similar to previous posts but an effort has been made to
clarify and emphasize points; i'm new to dsp. A lot of the "notation" is
i have an impluse response defined as follows:
h(t) = q(t) - q(t - p)
where 0 ...
Peter H wrote:
> I have been looking a bit on this scrambling method and tried to write the
> source code for it.
> As I understand the input is a sample value and the numbers from 1 to M-1
DC is high cut off, however, there is some low freq stuff. The HPF is
a second order IIR filter with slow roll off but about -80dB at DC.
Vladimir Vassilevsky wrote in message news: ...
> Andrew Xiang wrote:
> > Anyone encountered this problem when using NLMS? The h shif...
"Matt Timmermans" wrote in message
> "Fred Marshall" wrote in message
> > Matt,
> > Seems like all you have to do is cascade the ideal brick wall filter
"Curl" a écrit
| Rule No. 1 already respected ;o)
oops.. Too fast
The exact aswer is : I'm using a cascade form of 2nd order Form 1 IIR
The poles of a biquad are combined whith the zeros of the next
Thank you for your answers, I'm currently looking at SPRA454 (Extend...
I have an equation to update a correlation matrix: Rxx(n) = beta*Rxx(n-1) +
(1-beta)*x(n).herm(x(n)) where beta is the forgetting factor.
I am struggling to wrap my head around what needs to happen to beta in the
case of unequally spaced data, i.e. what happens if data comes in at
> Hi all
> I have an IIR high pass filter at say 50Hz that uses say the following
> coef's (not exact but to illustrate)
> a0 = 0.995
> a1 = -1.990
> a2 = 0.995
> b1 = -1.990
> b2 =0.990
> if i wanted to change the coefs on the fly to make the filter pass...
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 ...
Richard Dobson wrote:
|| Hello all,
|| I hope this is not too off-topic for this list - I am assuming that
|| dsp experts are more likely to be conversant with analog design than
|| vice versa!
This is something I really doubt!
|| I have this irresistible assertion, that analog fil...
> I have some questions about the z-transform and what to use it for in
> for example ARMA-filters. I know it is used to find poles and zeros,
> but what else?
> Consider an ARMA filter:
> After ...
email@example.com (praveen) wrote in message news: ...
> what is the accuracy of estimation of atan using cordic algorithm?.
> If i use look up table it will be huge since my step size of LUT is 5
> microradian (half octant ie pi/(8*5microradians) is huge).
I know that bits precision required for
an IIR filter is proportional to the number of IRR
stages. I learnt this aftermaking a 9 stage IIR using doubles
years ago, only the 2nd and third order ones worked correctly.
Anything more went to the rails (generally), due to overflows.
I also used ...
Hi, before I forget:
Here's wishin' you guys a Merry Christmas and
a Happy New Year!
Of course, these good wishes are also directed at any
of you who do not celebrate Christmas such as: Muslims,
Hindi, God-less Atheists, Buddhists, motorcycle mechanics,
Sikhs, Jews, IIR filter design...
"Jon Harris" wrote in message news: ...
> "Fred Marshall" wrote in message
> > "Rune Allnor" wrote in message
> > news:firstname.lastname@example.org...
> > > email@example.com (sree) wrote in messa...
On 21 Dec 2003 12:38:06 -0800, firstname.lastname@example.org (Alberto) wrote:
> I have the need to adjust the differential delay between two audio
> streams by a value less than one sampling period. So I thought to
> delay one stream by a fixed amount N, let's say 128 samples, and the
> other by a variabl...
robert bristow-johnson wrote in message news: ...
> In article 3FF2F689.BD81CCCC@z-sys.com, Glenn Zelniker at email@example.com
> wrote on 12/31/2003 11:17:
> i can't "one-up" Rune or Fred (but Burlington VT is kinda a cool town and
> before a recent warming spell i wuz ...
"TheDoc" wrote in message news: ...
> wrote in message
> > MathWorks watching like hawks, hey? Ha.
> Yep, gotta get their ideas from somewhere!!
Not from comp.dsp, that's for sure. Check out their FILTFILT rout...
robert bristow-johnson wrote:
> i tend to agree with Andor and JOS3. filters do two things to a
> sinusoid; they (may) change the amplitude and they (may) change the
> phase. "frequency response" means the total effect that the LTI
> system has on a complex sinusoid of a given ...
Sorry about the confusion.
The raised cosine filter is
H(w) = .5 * [1 - cos((w - w0)/wf)] for w0 - pi*wf < w < w0 + pi*wf,
H(w) = 0 otherwise.
where 'w0' is the center frequency
'wf' is the "width" of the filter and
'w' is, well, the frequency.
This function should rise cont...
On Thu, 08 Jan 2004 23:27:47 -0500, albert wrote:
> I am attempting to use a PC soundcard and Spectrum Lab software to
> detect the presence of a very weak input signal at 600 Hz.
> I know the frequency of the input signal and I know it does not drift
> (drift is less than 1 ppm), so I ...