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Re: change filter freq. w/o glitches

Hello Bernhard, 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 be expensive. The permanent modification of the IIR filter coefficients...

Re: An IIR filter gain questio

On 14 May 2006 18:51:14 -0700, "robert bristow-johnson" wrote: Hi, (snipped) > > > & 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 > day? Yep, that is correct. > man, wh...

Re: Happy Hmas!

Hi Eric, For those who have been touched by his noodly appendages, Rick's dismisal of other's beliefs must have been quite shocking and distressing. Steve Eric Jacobsen wrote: > David, there is a forum on venganza.org if you're interested. I've > scanned through it a few times but...

Re: AIC23, DC coupled?

> 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...

Another Goertzel question

Hi Guys, 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...

IIR Filter Difference Equation Implementation

Hi All, 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 lines: FeedForwardVariable = (b[0] * x[n]...

Re: Hilbert transform & analytic signals

Rick: [snip] "Rick Lyons" wrote in message news:3f135667.60818218@news.earthlink.net... > 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 : : > If...

Re: DSP technique to estimate the amplitude of the signal

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 http://www.it...

Fixed Point issues

Hello, I am trying to implement an FIR and an IIR filter in fixed point arithmetic. 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...

Re: Interpolation for Dummies

"Fred Marshall" wrote in message news:rSzTa.3405$Jk5.2403537@feed2.centurytel.net... > Ian, > > 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...

IIR filter - DC offset after swapping coefficients

Hello, 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...

Re: Hands on DSP books

You could try: Prentice Hall - C Algorithms For Realtime Dsp Christian_ wrote: > Hi, > 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. > > ...

Hilbert IIR filter implementation

Hello, 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 With regards praveen ...

Re: IIR quantization

ahgu@yahoo.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 > system? > > Where do you specifi...

Re: A fundamental question on 0-phase filter

On 19 Aug 2003 07:41:44 -0700, acoustictech_zhangtao@yahoo.com.sg (ZedToe) wrote: > Hi, > 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) ...

linear phase iir filters

sir, 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...

Re: Exam revision

Noway2 wrote: > 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...

"Time revresion technique"

sir, 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. ...

IIR filter design

Hi all, 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...

shrinking convolution remainders!

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 matlab. i have an impluse response defined as follows: h(t) = q(t) - q(t - p) where 0 ...

Re: wondering if this works

Peter H wrote: > I have been looking a bit on this scrambling method and tried to write the > source code for it. > > http://www.mathworks.com/access/helpdesk/help/toolbox/commblks/ref/scrambler.shtml > > As I understand the input is a sample value and the numbers from 1 to M-1 > ind...

Re: DC shift of the h(impulse response of the room) when using NLMS adaptive filtering

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...

Re: Nonlinear brick wall filters

"Matt Timmermans" wrote in message news:JFMdb.847$Tu2.135336@news20.bellglobal.com... > > "Fred Marshall" wrote in message > news:43Ldb.2188$v22.2108804@feed2.centurytel.net... > > Matt, > > > > Seems like all you have to do is cascade the ideal brick wall filter with > an...

Re: Arranging biquads

"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 filter The poles of a biquad are combined whith the zeros of the next biquads. Thank you for your answers, I'm currently looking at SPRA454 (Extend...

Exponential downdating and unequally spaced data

Hi all! 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 a...

Re: Hi-pass to pass thru with IIR

scott wrote: > > 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...

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 ...

Re: Can an analog filter really be linear-phase FIR-style?

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...

Re: Question about the z-transform for ARMA modelling

Martin wrote: > Hello! > > 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: > y(t)+a1*y(t-1)+a2*y(t-2)=x(t)+c1*x(t-1)+c2*x(t-2) > > After ...

Re: minor confusion

ma740988 wrote: > Jerry Avins wrote in message news: ... > > > Randy Yates wrote: > > > > > Jerry Avins wrote: > > > > > > > > > > Randy Yates wrote: > > > > > > > > > > > > > Jerry Avins wrote: > > > > > > > > > > > > > > > > ma740988 wrote: > > > > > > > > > > > > ...

Re: Cordic algorithm for atan in DSP processor

praveenkumar1979@rediffmail.com (praveen) wrote in message news: ... > Hello, > > 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). > ...

rounding errors : fixed point precision

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 ...

OT: Xmas wishes to you guys

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...

Re: newbie needs some sujjestions

"Jon Harris" wrote in message news: ... > "Fred Marshall" wrote in message > news:DJednSCfH68Iwn6iRVn-gQ@centurytel.net... > > > > "Rune Allnor" wrote in message > > news:f56893ae.0312191044.40e34b4a@posting.google.com... > > > konerusreeram@yahoo.com (sree) wrote in messa...

Re: Is anybody familiar with the Thiran allpass filter ?

On 21 Dec 2003 12:38:06 -0800, adibene@yahoo.com (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...

Re: OT: A winter morning in the arctic outback

robert bristow-johnson wrote in message news: ... > In article 3FF2F689.BD81CCCC@z-sys.com, Glenn Zelniker at glennz@z-sys.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 ...

Re: to administrators of the comp.dsp ,instancy!

"TheDoc" wrote in message news: ... > wrote in message > news:mt87vvcrl22ql7nbg8l26426s3mjeapjn3@4ax.com... > > 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...

Re: Gain of an IIR Filter

robert bristow-johnson wrote: (snip) > 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 ...

Re: What am I missing normalizing a FIR filter?

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...

Re: newbee fft question, decimate or mix??

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 ...