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Started by bakoguz in Matlab DSP22 years ago

I have a problem of for loops. I saw the messages about them but it did not help me. I want to filter my matrix(actually an...

I have a problem of for loops. I saw the messages about them but it did not help me. I want to filter my matrix(actually an image) by a filter and I want to convolve(2D) my filter with my image. If I use for loops, I need two of them, but the execution is too slow. Can


How to inverse a FIR Filter using Matlab

Started by Lê Anh in Matlab DSP12 years ago

Hi Everybody Im new here und i have a question from my Bachelorthesis In a Matlab system ,I have the estimated transfer function from the...

Hi Everybody Im new here und i have a question from my Bachelorthesis In a Matlab system ,I have the estimated transfer function from the channel, simply ch_est (in frequency domain), and that is a 64x1 Matrix You can say that the channel is a FIR Filter with finite impulse response. I have heard that, the inversion of a FIR Filter is always a IIR Filter, with infinite impulse response...


Designing a Notch Filter

Started by maha devi in Matlab DSP16 years ago 2 replies

Hi friends, I need to design a notch filter using matlab.. can anyone tell me if any inbuilt command is available for designing a...

Hi friends, I need to design a notch filter using matlab.. can anyone tell me if any inbuilt command is available for designing a notch filter in matlab.how can i specify the frequency which should be attenated and the bandwidth of notch???? also, i need to design a FIR FILTER using window method. Can i use FDATOOL for designing it...If so..where should i mention the Number of taps and ...


16-QAM with root raised cossine filter

Started by "diogo.rodrigues" in Matlab DSP16 years ago 1 reply

Can someone help me with a simulating 16-qam using square root raised cossine filter? This is the problem: even without noise, the filter are...

Can someone help me with a simulating 16-qam using square root raised cossine filter? This is the problem: even without noise, the filter are generanting errors. What am i doing wrong? Thanks, Diogo This is the code: M = 16; Simbolo = randint(750,1,M); % Simbol generator SimboloModulado = qammod(Simbolo,M); % Modulation % Tx filter SimboloEnviado = 2*rcosflt(SimboloModulado,...


filter function

Started by cherrymata in Matlab DSP20 years ago

Hi All, is there any filter function in matlab that can filter the signal disturbed by awgn (additive white gaussian noise) ? ...

Hi All, is there any filter function in matlab that can filter the signal disturbed by awgn (additive white gaussian noise) ? can anyone help me... thx very much, arida


Freqz function- plssssss help

Started by discreteworld in Matlab DSP18 years ago 1 reply

I am using notch filter, below are the commands for calculating filter coeff: f0 = 0.4 % freq to be notched b = [1...

I am using notch filter, below are the commands for calculating filter coeff: f0 = 0.4 % freq to be notched b = [1 -2*cos(2*pi*f0) 1]; % compute the filter numerator coeff "b" r =0.995; a = [1 -2*r*cos(2*pi*f0) r^2]; % compute the filter denominator coeff "a" [h,f1]


analog filter order for cheby1

Started by mao mu in Matlab DSP20 years ago 1 reply

Hi My target filter is a bandpass analog filter. With N=7, cheby1 returns a 14th B(s), A(s). But if the target is a LPF or HPF,...

Hi My target filter is a bandpass analog filter. With N=7, cheby1 returns a 14th B(s), A(s). But if the target is a LPF or HPF, the results are 7th order B(s),A(s). Could somebody explain why bandpass filter design using cheby1 doubles the order on B(s) and A(s)? Thanks


Filter and boundary conditions

Started by Sergio Dominguez in Matlab DSP20 years ago

hi all, how can I filter a trame trace so that the initial and final value remains the same? I need to filter only chunks of...

hi all, how can I filter a trame trace so that the initial and final value remains the same? I need to filter only chunks of time series (a long one) depending on one condition, and then put the filtered part where the original was. therefore I need the initial value and


Filter Design(Complex Bandpass Filter)

Started by adnan kiani in Matlab DSP16 years ago

Hello; I need some help regarding complex bandpass filter design. The task is to design a complex BPF with passband at 8-12MHz and stopbands at...

Hello; I need some help regarding complex bandpass filter design. The task is to design a complex BPF with passband at 8-12MHz and stopbands at -16-7MHz and 13-16MHz. The idea is to design with through complex modulation of proper lowpass prototype filter. I was wondering how to include the negative frequencies if i design it using firpm/remez function. The LPF has passband at -2 to 2MHz and stop...


filter

Started by brandon_etheridge in Matlab DSP21 years ago 1 reply

I am trying to filter a column of values to obtain a smooth curve. What is the best method in Matlab to filter these data points. I ...

I am trying to filter a column of values to obtain a smooth curve. What is the best method in Matlab to filter these data points. I seem to run into a problem because I am not filtering a function but rather a column of values. Thanks in advance, Brandon


adaptive lms notch filter is IIR or FIR?

Started by ashw...@yahoo.com in Matlab DSP16 years ago 2 replies

hi im doing a project regarding development of an adptive digital notch filter for the removal of 50hz noise from an ecg signal.i have the...

hi im doing a project regarding development of an adptive digital notch filter for the removal of 50hz noise from an ecg signal.i have the code.now my problem is when do the full report for methodology.first I explained about adaptive filter and followed by adaptive notch filter..then I explained about LMS algorithm seems the sign regressor algorithm comes under LMS..the thing is im confused with ...


filter bank : design n application

Started by Sanjay S in Matlab DSP19 years ago

I'm currently doing my major project on Non-uniform filter bank theory and application. I need your help and suggestions based the...

I'm currently doing my major project on Non-uniform filter bank theory and application. I need your help and suggestions based the design of a non-uniform cosine-modulated filter bank (a 27 or 32 channel filter bank)and find it's application in the existing MPEG-1, Layer-III model ~ "MP3" a


Stable Numerical MATLAB Routine for FIR MISO Wiener Filter

Started by Emanuele in Matlab DSP11 years ago

ear all, I have a basic knowledge of signal processing, but unfortunately I'm not a specialist of this field. I have to build an adaptive ...

ear all, I have a basic knowledge of signal processing, but unfortunately I'm not a specialist of this field. I have to build an adaptive MISO (multiple-input single-output) FIR Wiener filter, which receives three (or more) signals as inputs and produces an output according to the Wiener filter theory (for example referring to the book of Haykin, Adaptive Filter Theory). I need to use the W...


Volterra filter design

Started by mmni...@yahoo.com in Matlab DSP17 years ago

How can I design a 2nd order Volterra filter with any "input + noise". Output should be noise free. If anyone can give me an example? Also how...

How can I design a 2nd order Volterra filter with any "input + noise". Output should be noise free. If anyone can give me an example? Also how can I represent above filter in state variable form. Any help, pointers, are greatly appreciated. Thanks


how to resample a z transformation???

Started by stefanosorrentino in Matlab DSP21 years ago 5 replies

Hello friends. I have an apparently unsolvable problem: I have a given IIR filter (they give me the poles and the zeros). The...

Hello friends. I have an apparently unsolvable problem: I have a given IIR filter (they give me the poles and the zeros). The filter was originally sampled at a given (very high) frequency. I also have a signal, which is sampled at a much lower rate. When I filter my signal, I


Changing sampling frequency

Started by kn srivastava in Matlab DSP14 years ago 3 replies

I have a digital filter that is designed for 1kHz sampling frequency. Everything else remaining the same, I want that filter to work for 4...

I have a digital filter that is designed for 1kHz sampling frequency. Everything else remaining the same, I want that filter to work for 4 kHz sampling frequency. What do I need to do? I am not a filter guy and my background is power system. That is why I am asking you here this basic question. Kailash


RE: filter function in matlab for awgn

Started by Glen Ragan in Matlab DSP20 years ago

Yes, the filter() function will do filtering. Since you seem to be doing signal detection, you would also be interested in...

Yes, the filter() function will do filtering. Since you seem to be doing signal detection, you would also be interested in convolution, conv(), with the time-reversal of the pattern you wish to detect (matched-filter detection). Glen Ragan -----Original Mes


Remez Filter Design in MATLAB

Started by Anonymous in Matlab DSP22 years ago 1 reply

Hi There, Can any one tell me how to design a low pass FIR filter using remez exchange algorithm in MATLAB. An...

Hi There, Can any one tell me how to design a low pass FIR filter using remez exchange algorithm in MATLAB. An example will be highly appreciated. I tried with some parameters but it says filter could not converge. Thanks!!! --Sur


comb filter design using matlab

Started by nagi_nov3 in Matlab DSP21 years ago

can any one suggest me how to get the required result,Iam not getting response plz any one help me......... Software...

can any one suggest me how to get the required result,Iam not getting response plz any one help me......... Software Implementation in Matlab or Octave In Matlab or Octave, this type of filter can be implemented using the filter function. For example, the followin


query about optimizing all pass filter parameters

Started by kimm...@yahoo.co.in in Matlab DSP13 years ago

I have a function H(f) = 10 ?(?*L/10)exp ?j[ D*?0^2*L*?^2/ (2*c)] . And ?ap(phase of all pass filter in frequency domain) is a function of T, r,...

I have a function H(f) = 10 ?(?*L/10)exp ?j[ D*?0^2*L*?^2/ (2*c)] . And ?ap(phase of all pass filter in frequency domain) is a function of T, r, ?0 and f. Expression of ?ap is known. I have to optimize all pass filter parameters T, r, ?0 such that minimum mean square condition is achieved MMSE( T, r, ?0 )= ? { arg( 1/ H(f) ) ? ?ap( T, r, ?0, f) }^2 [Note: Summation