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Re: questions raised by reading and thinking with possibly missing background

Richard Owlett wrote: > Jerry Avins wrote: > > > Richard Owlett wrote: > > > > ... > > > > > I start with a basic presupposition that however humans recognize > > > speech is, in some sense, the "best" way. That leads me to believe > > > that any filtering should be constant g...

Re: multirate implementation problem

Design an FIR filter, with a sample rate of 4 Mhz and cutoff of 250 kHz. Every 8th new sample, calculated the FIR output ( but keep buffering the inputs every sample ). There, now you have a continuous flow at 500 kHz. Regards, Robert www.gldsp.com praveenkumar1979@rediffmail.com...

Re: Filter Options

Hi, I know its pretty obvious, but have you tried to increase the order of your FIR filter? Regards, Bobby "Àine Canby" wrote in message news:57ed59a.0306260936.3ba7244d@posting.google.com... > Hallo all, > > I'm processing a broadband signal. The problem I'm having is that when ...

Re: change filter freq. w/o glitches

"Bernhard Holzmayer" wrote in message news:5083003.DI8xdzyhYS@holzmayer.ifr.rt... > To be honest, I cannot oversee, if this "ripple switch" approach > would improve the behaviour. I guess that this depends on the data > again. > > Obviously it's important to describe the data stream ...

Re: FIR EQ questions (simple ones)

Tom St Denis wrote: ... > Hmm.. I thought there was a wobble in the passband where the level of > the filter would not be flat unless you used a lot of FIR taps. ... > > > > You can set it to whatever you like, as long as the length of the > > non-zero part of the IFFT of the filt...

Re: FFT questions

"Greg Aagard" wrote in message news:374a78f3.0307021546.733bc0f7@posting.google.com... > I haven't done fourier stuff for a while, and I had a couple of > questions. > > I'm trying to get the frequency spectrum of a time-sampled signal by > using the FFT function in Matlab. I get ri...

Re: uniform filter bank implementation.

I guess I am just a little confused with the constant notation switching, I am following Crochiere, since it is what I have available to me. The notation is rather abnoxious, and it isn't not clear what is happening step by step so that I can implement the algorithm, I did look at a lot of the ...

Re: A question about Frerking's DSP book

On Sun, 6 Jul 2003 21:17:11 -0400, "Clay S. Turner" wrote: > > "Rick Lyons" wrote in message > news:3f08187e.13298187@news.earthlink.net... > > Hello Rick, > It seems as though your picture doesn't look like the one in my book. I'm > looking at figure 7.8 (parts a and b) on page 33...

Re: Adaptive Filtering in Real Time

That is very true. One of the most confusing parts of implementing this filter for me has been that I have been considering it real time. In the back of my mind I have thought that it just didn't make sense that it could be real time. You're right though. I just need to keep the data stre...

Re: Impulse response starting with zeros?

"Steve Conahan" wrote in message news:beueuo$llj$1@ginger.mathworks.com... > "Ken" wrote in message > news:beudc9$iqd$1@dennis.cc.strath.ac.uk... > > > > Hello folks, > > > > Just a quickie: > > > > Have you ever come across or can you think of a reason why you would ha...

Re: Hilbert transform & analytic signals

On 15 Jul 2003 11:53:58 -0700, dirkman@erols.com (Dirk Bell) wrote: > Rick, > > Can you post your MATLAB filter design script? > > When you say 3-4 times more taps are you counting the half of the real > and half of the imaginary taps that should be 0 after the mix? > > Dirk > Hi D...

Re: Interpolation for Dummies

Ian McBride wrote: > I am trying to figure out interpolation (I have given up on decimation for > now). > > My scenario is that I have 8 ksamples per second audio (from a phone line), > and I want to interpolate it to 96 ksamples per second. I know my audio is > band limited within 0-4...

Re: ButterWOrth filter using matlab

Hello Suman, While I'm not a Matlab guy, it appears (If I read your program correctly) that you are convolving your data with an FIR filter whose tap values are the mangitude of a Butterworth filter. If this is the case, it will fail for several reasons. One: the Butterworth filter is an IIR d...

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: Blackfin two-word *fast* floating-point library

KG7HF wrote: > > Sounds like a fairly rigid and inflexible statement. So if I write my > application in binary, skipping the assembler and linker, am I a god or a No no. God always wrote in Lisp[1], you know? :-) So you can't be god if you write in binary. > fool? Use of the language ...

Re: Floating point to fixed point conversion.

HI Gowtham, Just Google for these PDFs "Fixed-Point Arithmetic: An Introduction" and "Practical Considerations in Fixed-Point FIR Filter Implementations" both by Randy Yates ( He is also a member of this group). They are very informative and will be useful for ur proj Regards, Bhanu ...

Re: LMS limitations?

"Zach R." wrote in message news:ZzkTa.61434$kI5.13224@nwrddc02.gnilink.net... > Hi, > The sample rate is 7040 Hz (the lowest that my ADC will go). As far as > the order goes, I would like it to be longer but the program takes too long > to run when I use a larger order (such as ...

Re: Sliding Goertzel References

Hi All, Yeh, the Sliding Goertzel (SG) is pretty cool, I have looked into it, and I have working code. The SG has interesting benefits over the Sliding DFT, and yet it also has some limitations. The Sliding DFT is best for sample by sample algorithms where R/I data is needed constantly (li...

FIR filtering using a lookup table

Hi, Currently I am using a polyphase FIR filter to perform an interpolation. To gain performance I have in mind to redesign the original filter with the principle of a lookup table. Does anyone know where I can find some helpful information of how to implement this principle? Regards, Ell...

beginer question

How may I determine cutoff frequency of FIR filter? Thanks Marcin ...

HELP RE: FIR filter designing

Hallo, please listen my story . One day i went to a professor's room for asking a job..there i told him about myself .then he asked me to design a Finite Impulse Response Filter using VERILOG. !! I said OK for that moment !! He is not at all giving me any kind of guidance fo...

Re: A fundamental question on 0-phase filter

acoustictech_zhangtao@yahoo.com.sg (ZedToe) wrote in message news: ... > 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) = Hzp(w...

Re: Design of nonlinear phase FIR filters on magnitude only criteria

"Robert Rozman" wrote in message news:bi0c3v$ia8$1@planja.arnes.si... > Hello, > > I'm looking for some info and references on possibilities and methods for > designing non linear phase FIR filters based solely on magnitude error > criteria. Any help, info ? > > Thanks in advance...

Re: Minimum-Phase FIR Filters Tutorial in DSPguru.com

On Wed, 20 Aug 2003 23:28:40 -0700, Pepe Barbe wrote: > I am looking for a way to get the minimum phase version of a FIR filter. > I saw the tutorial in DSPguru.com: Sorry I didn't reply to your email. It is sitting in my "to be replied to" folder, but I have been a little busy lately... ...

Re: How to decide the filter length in ADNC using LMS Algo ?

Sandeep Chikkerur wrote: > > How to decide the FIR - filter length (no. of taps) required to design > the ADAPTIVE NOISE CANCELLER ? > > I am using the LMS Algorithm.. You need to know the model of the system. The parameters of interest are the flat delay and the impulse response...

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...

Decimation (CIC) filter in VHDL

Hi all, I am currently in the process of designing a decimation filter for a 16bit Sigma Delta ADC. I suspect it will be a CIC filter followed by a FIR filter. I am facing great difficulty understanding such filter, as my experience in usually with analog circuits, and not DSP. My ques...

Re: linear phase iir filters

In article f56893ae.0308271007.348925d5@posting.google.com, Rune Allnor at allnor@tele.ntnu.no wrote on 08/27/2003 14:07: > robert bristow-johnson wrote in message > news: ... > > > > While I haven't checked out the Clements & Pease paper, I don't understand > > > > what RBJ is talkin...

Re: Hilbert transform in MATLAB

R.Lyons@_BOGUS_ieee.org (Rick Lyons) writes: > On Fri, 10 Feb 2006 15:58:34 -0600, "Leah" > wrote: > > > I'm doing some real time work in MATLAB as well. I discovered that you > > cannot use the Hilbert transform for real time data because the Hilbert > > function is a non-causal filte...

Basic Filter Query

Hi, The order of a digital filter is the number of previous inputs (stored in the processor's memory) used to calculate the current output. My doubt is, how can one estimate the current output based on the previous inputs ? E.g in Adaptive Noise cancellation, if FIR filter is used, then t...

Re: Why (if) should be windows in spectral analysis nonnegative ?

"Jon Harris" wrote in message news:3f552fae$1_3@newsfeed.slurp.net... > I think you are confusing a "window" with "windowed filter coefficients". > The filter coefficients are created by multiplying the window by a sinx/x > (AKA sinc) function. The window itself is not a filter per se--...

Bandpass FIR Filter

I have a basic doubt regarding FIR Filter implementation. The input to my FIR filter is the data(16-bit out from ADC sampled at 20 KHz). i.e this data represents the amplitude of the sampled input signal. My doubt is "whether the output of the FIR filter denotes the amplitude or frequency...

Basic Filter Question

Hi, The order of a digital filter is the number of previous inputs (stored in the processor's memory) used to calculate the current output. My doubt is, how can one estimate the current output based on the previous inputs ? E.g in Adaptive Noise cancellation, if FIR filter is used, then t...

Minimum phase FIR filter algorithm

Hello, I am looking for algorithms to calculate a minimum phase FIR filter from a desired frequency responce or from zero phase impulse response. Thank you, Frederic ...

Re: Preamble detection in ofdm synchronization

On 8 Dec 2005 18:56:59 -0800, "Ant_Magma" wrote: > I'm currently studying the preamble synchronization of ofdm symbols in > power line comunication (in Simulink). > > I haven't finished reading yet but there's a few things i dont > understand. > > 1. In the auto-correlation of the prea...

Re: Low pass interpolation!!

"santosh nath" wrote in message news:6afd943a.0309180901.7ca83c48@posting.google.com... > Hi, > I am looking for low pass interpolation filter - C code or pseudo > code/algorithm descriptions etc. MatLab has one such function > "interp"- the algorithm is described in "programs for dig...

Halfband filter delay!

Hi! By definition the FIR filter delay is: (N-1)/2 The halfband filter has every other sample exactly zero. If the first and the last sample of the filter are exactly zero. What is the delay of the filter? (N-1-2)/2 ? Thats not what I am getting. Why? Thanks! Atmapuri ...

Multi-stage filter delay!

Hi! I am trying to compute the delay of a multi-stage half band FIR filter bank (upsampling by factor 2 in each stage). (N is number of taps). With one stage (half band filter) I get: (N1-1) /2 + 1. With two stages: (N1-1)/2 * 2 + 1 + (N2-1)/2 - 4 On stages higher then 2 filter length ...

Digital FM Demodulation - Filtering

Hi, I have no experience of 'software radio', however I am now trying to model a digital FM radio. I am using a simple arctan of the I and Q to generate the instantaneous phase and then differentiating by subtracting successive phases and dividing by the sample interval to give the FM output...

Re: most efficient way to downsample 32k to 8k?

The simple way is to low-pass filter, possibly using a half-band/quarter-band FIR filter. Then keep 1 out of every 4 samples. Or you can be smart about it and realize that since you are throwing away 3/4 of the data, there is no need to compute the filter for these samples. So just compute th...