error "divide by zero" in "design of linear phase FIR filters using windows" plz help!!!!

Started by Anonymous in comp.dsp14 years ago 7 replies

I am doing homework on "design of linear phase FIR filters using windows" The unit sample response of the FIR filter...

I am doing homework on "design of linear phase FIR filters using windows" The unit sample response of the FIR filter is h(n)=sin((pi./6).*(n-12))./(pi.*(n-12)); The filter has 25 taps. I need to plot |H(omega)| in MATLAB where H is the Fourier Transform of h(n)=SUM(0-> 24)of (h(n)*exp(-j*omega*n)) but I got stuck here since at n=12, the h(n) is 0/0 and MATLAB gives an error: "Div


FIR Filter design tool

Started by eeh in comp.dsp14 years ago 4 replies

Hi, Is there any free or low cost software tools that could calculate FIR filter coefficients for bandpass filter? Thanks!

Hi, Is there any free or low cost software tools that could calculate FIR filter coefficients for bandpass filter? Thanks!


adaptive filter

Started by ldkha1979 in comp.dsp14 years ago 1 reply

hi all, i have to design an adaptive filter using Matlab. What i need is to find a symmetric FIR filter from 2 signals, desired and input. I...

hi all, i have to design an adaptive filter using Matlab. What i need is to find a symmetric FIR filter from 2 signals, desired and input. I don't know why my results cannot be converged. I'm using NLMS algorithm. Here are my code. Can anyone help me to find my wrong part, or can give me a suitable one? Thks so much function h = adp_filter(d,x,f0) % h = adp_filter(d,x,f0) % Inputs: % ...


Adaptive Feedback Path Tracing Mismatch Problem

Started by shengzheng1983 in comp.dsp14 years ago 8 replies

Hi, all I'm doing the acoustic feedback path tracing for hearing aids applications. My method belongs to non-continuous method. i have a random...

Hi, all I'm doing the acoustic feedback path tracing for hearing aids applications. My method belongs to non-continuous method. i have a random noise u(n)as the input of my adapitve filter and a prescribed 160-tap feedback path F(z). I use 160FIR adaptive FIR filter W(z) to model the original 160-tap feedback path. The echo is obtained by passing u(n) through the F(z), and outside there is a spee...


How to load a lookup table on to the flash of TMS320F2810 DSP?

Started by Byju C in comp.dsp14 years ago 2 replies

Hi everybody, Does anyone know how to load a lookup table (size-4K) onto the flash of TMS320F2810 DSP. I would like to store a file containing...

Hi everybody, Does anyone know how to load a lookup table (size-4K) onto the flash of TMS320F2810 DSP. I would like to store a file containing FIR filter coeeficients into the flash memory of the DSP. This need not be done at run time. Regards, byju


FIR filter coefficients - are they ever zero??

Started by elp02rap in comp.dsp14 years ago 5 replies

Hi, I am implementing a FIR filter. A typical application for the filter would be a wireless communications system. Assuming the filter...

Hi, I am implementing a FIR filter. A typical application for the filter would be a wireless communications system. Assuming the filter is programmable, is it possible for any of the filter coefficients to be zero? I ask this as the multiplier unit in my filter's tap can be simplified if the coefficients are never zero. Reference to a credible source will be much appreciated. Many th...


Any help about FIR filter algorithm

Started by mariya in comp.dsp14 years ago 12 replies

Hello Can anyone tell me how i can get an output array from the following piece of code. nm1=N-1; yn=0; for(k=0; k

Hello Can anyone tell me how i can get an output array from the following piece of code. nm1=N-1; yn=0; for(k=0; k


How to design a graphic equalizer

Started by dingke1980 in comp.dsp14 years ago 6 replies

I am doing a software project about a ten-band graphic equalizer. Is there any solutions about how to design it? Currently, I have the...

I am doing a software project about a ten-band graphic equalizer. Is there any solutions about how to design it? Currently, I have the following ideas: 1.Use FFT, apply different gains to each frequency bin. But for low frequency band like 31.25Hz, 62.5Hz, 125Hz, there are few frequency bins even using large length FFT. The frequency precision is roughly influencing. 2.Use FIR filter bank. D...


How to design a graphic equalizer

Started by dingke1980 in comp.dsp14 years ago 1 reply

I am doing a software project about a ten-band graphic equalizer. Is there any solutions about how to design it? Currently, I have the...

I am doing a software project about a ten-band graphic equalizer. Is there any solutions about how to design it? Currently, I have the following ideas: 1.Use FFT, apply different gains to each frequency bin. But for low frequency band like 31.25Hz, 62.5Hz, 125Hz, there are few frequency bins even using large length FFT. The frequency precision is roughly influencing. 2.Use FIR filter bank....


Baseband OFDM simulation - noise?

Started by Snowball in comp.dsp14 years ago 2 replies

Hi, I am simulating a baseband OFDM system in MATLAB Simulink. It looks similar to the adsl_sim demo, though I am not doing any CRCing,...

Hi, I am simulating a baseband OFDM system in MATLAB Simulink. It looks similar to the adsl_sim demo, though I am not doing any CRCing, scrambling or interleaving. I use a 256-carrier DMT modulator, and transmit my data over a channel which is basically a lowpass FIR filter with real coefficients. I am not sure how to model the noise though. Is it correct to add AWGN noise on each freque...


Impulse response

Started by Gert Baars in comp.dsp14 years ago 13 replies

Hello, I'm trying understand designing a FIR filter from scratch because I want to experiment with home-made windows. With H(W) = 1 for -W0...

Hello, I'm trying understand designing a FIR filter from scratch because I want to experiment with home-made windows. With H(W) = 1 for -W0 < W < W0 = 0 else After IFT(F[W]) the result f[t] is a sinc function. This function is symmetrical to t=0 Turning this function into h(n) without a window is the translation t = Ts(n-(L-1)/2) ( so h(n) = f[Ts(n-(L-1)/2] ) c


need help - converting from a Frequency spectrum to a FIR filter

Started by Anonymous in comp.dsp14 years ago 5 replies

Hello, Im doing a room equalization project, at the moment by sweeping the room with sine waves, and getting the frequeuncy response of the...

Hello, Im doing a room equalization project, at the moment by sweeping the room with sine waves, and getting the frequeuncy response of the room. I now have to convert from this 'room information' to a FIR filter (and then an inverse FIR filter) that will be applied to sound sources to compensate for the response of the room. Any ideas anyone? How to get from FFT (magnitude and phase i...


windowed FIR

Started by Gert Baars in comp.dsp14 years ago 3 replies

Is lets say a Hamming window the same for either LPF,BPF or HPF FIR filter desings? (My guess is it is.)

Is lets say a Hamming window the same for either LPF,BPF or HPF FIR filter desings? (My guess is it is.)


Equiripple FIR filter design

Started by jaan in comp.dsp14 years ago 6 replies

I have to build a lowpass filter: cuttoff freq = 0.1; attenuation = 90 dB; transition = 0.01; passband ripple = 0.000275 dB; Using a...

I have to build a lowpass filter: cuttoff freq = 0.1; attenuation = 90 dB; transition = 0.01; passband ripple = 0.000275 dB; Using a single eruiripple FIR, the attenuation is too small or transition is too long or the number of taps is enormous. When using set of filters, the passband ripple will be the problem. Is it practically possible to build at all? This message was sen...


coefficients compared

Started by Gert Baars in comp.dsp14 years ago 12 replies

Hello, For a low-pass filter with hammingwindow I can calculate the coefficients manually. Lets say for n = (L-1)/2 the middle tap (#taps...

Hello, For a low-pass filter with hammingwindow I can calculate the coefficients manually. Lets say for n = (L-1)/2 the middle tap (#taps is odd) coefficient is 2*Fc/Fs. (The Hammingwindow here is 1). Compared to the FIR filter program ScopeFIR I get exactly the same values for manually calculated middle-tap coefficient if Fc > = 1 KHz < Fs/2 while Fs = 12 KHz. The problem with Fs


Choosing a FIR filter windows type

Started by Mook Johnson in comp.dsp14 years ago 3 replies

Wen inplemening a FIR filter, what considerations are given when choosing between, rectangular, Bertlet, Hanning, Hamming, Blackman, kaiser, etc...

Wen inplemening a FIR filter, what considerations are given when choosing between, rectangular, Bertlet, Hanning, Hamming, Blackman, kaiser, etc windows for the coefficients? Basically, aside from the attenuation differences at the first hump in the stop band, and transition band width, what other characteristics (trade offs) are to considered? thanks


Odd symmetric FIR filter

Started by Umutesi Faith in comp.dsp14 years ago 13 replies

Hello I am trying to implement an odd symmetric FIR filter (using C code) but i am struggling with the algorithm . Here is the piece of code...

Hello I am trying to implement an odd symmetric FIR filter (using C code) but i am struggling with the algorithm . Here is the piece of code that I am trying to understand. Can anyone give me some steps on how the array index are working here. I?am still reading my C book on arrays but still I cannot get the whole idea.Please if someone out there can give me some hints I will be grateful to le...


FIR filter change filter banks (tms320c50)

Started by Przemek in comp.dsp14 years ago 4 replies

Hi How to change code TMS320C50 (asm or c) from filter FIR make filter banks 8-chanels? Best regards

Hi How to change code TMS320C50 (asm or c) from filter FIR make filter banks 8-chanels? Best regards


band-pass FIR filter low stopband attenuation problem

Started by Michel Rouzic in comp.dsp14 years ago 10 replies

I have designed a band-pass filter that uses two windowed-sinc functions that has one which is spectral inverted, and the two are convolved...

I have designed a band-pass filter that uses two windowed-sinc functions that has one which is spectral inverted, and the two are convolved together. The problem with that, is the stopband attenuation on the low side is very poor. It's all well above -50db, as on the high side, it's mostly all under -60db. Both my windowed-sinc functions look fine when looking at their frequency respons...


About FIR filters

Started by I. R. Khan in comp.dsp14 years ago 9 replies

Hi, I cannot recall but I think I saw somewhere some specific term for the ratio (or difference) of the magnitudes of largest and smallest...

Hi, I cannot recall but I think I saw somewhere some specific term for the ratio (or difference) of the magnitudes of largest and smallest coefficients of an FIR filter. Could some one please help me? Is the smaller value of this ratio (or difference) a desirable feature for a filter? Why? Thank you. Ishtiaq.