Dear all, A very basic question actually. Just found an issue with IIR low pass filter in decimation low pass chain after a 3rd...

Dear all, A very basic question actually. Just found an issue with IIR low pass filter in decimation low pass chain after a 3rd order SINC filter in a Sigma Delta A/D system. The one bit input code from Sigma Delta modulator is 1010101010... forever which is supposed to

Hi all, I'm trying to implement a low pass filter on a microcontroller, getting coefficients in MATLAB. I have the transfer function in...

Hi all, I'm trying to implement a low pass filter on a microcontroller, getting coefficients in MATLAB. I have the transfer function in s-domain, so I am using bilinear() to get the coefficients. This gives num and den (a and b coefficients) for an IIR implementation. I would like to know if it possible to get the coefficients for an FIR implementation of a filter, when you have the transf...

Hello, I would like to investigate the effects of filters on a signal. I have tried different nyquist filters like Sinc and RRC. I would...

Hello, I would like to investigate the effects of filters on a signal. I have tried different nyquist filters like Sinc and RRC. I would like to work with non-nyquist filter such as guassian. However, I am not sure of the choice of 'sigma' for the filter. The signal is a complex gaussian signal and has a certain variance, say v1. I normalize this variance, so that the variance of real...

I have a wav format audio file. I need to operate with fft and I have to filter it between 30hz to 100hz. I can read it, but I dont...

I have a wav format audio file. I need to operate with fft and I have to filter it between 30hz to 100hz. I can read it, but I dont know how to filter it. I don't know much about filters. Please help me

Hello everybody, I would like to use the Kalman filter (and the unscented kf, the particle kf, etc) in prediction for time series, but I...

Hello everybody, I would like to use the Kalman filter (and the unscented kf, the particle kf, etc) in prediction for time series, but I haven't any experiences with the KF. I wonder if someone would be so kind to help me in using the KF,and what toolbox could be better. Thanks in advance. Alberto Rossetti

Hi,   I want to implement an FIR High pass filter by using FIR1 function . The parameters of the filter are as follows filter...

Hi,   I want to implement an FIR High pass filter by using FIR1 function . The parameters of the filter are as follows filter order normalised stopband frequency normalised passeband frequency I have some difficulty to enter these parameters in the suited place in the function.  thanks for any suggestion ------------------------------------------------------- Sonia Rezk PhD Student, ...

hi everyone, im a student doing an assignment about development of an adaptive notch filter for the removal of 50hz noise from an ecg...

hi everyone, im a student doing an assignment about development of an adaptive notch filter for the removal of 50hz noise from an ecg signal..after i read about filters i chose low pass filter with woindowing method(kaiser)..now i tried this command.. > > fsamp = 60; fcuts = [5 10]; mags = [1 0]; devs = [0.05 0.01]; [n,Wn,beta,ftype] = kaiserord(fcuts,mags,devs,fsamp); hh = fir what i s

Does any out there know how to accuratley filter frequencies in 0.2 Hz bins....exactly? Which filter, order number, etc. would...

Does any out there know how to accuratley filter frequencies in 0.2 Hz bins....exactly? Which filter, order number, etc. would be most useful. Thanks Magnut

Hello all, I want to use a kalman filter and i don't know exactly how to do that. I'd appreciate if someone could give me an idea or a...

Hello all, I want to use a kalman filter and i don't know exactly how to do that. I'd appreciate if someone could give me an idea or a simple example of kalman filetring for an one-dimensional signal (state space). Thak you! Best regards!

I am working on a school project, and I need to create a Linear phase, causal, digital Bandpass FIR filter using windowing methods. with the...

I am working on a school project, and I need to create a Linear phase, causal, digital Bandpass FIR filter using windowing methods. with the following specifications. Sampling fs= 24khz Passband freq= p1= 2*pi*4000, and p2= 2*pi*8000 rad/sec stopband freq= st1=2*pi*3000 st2=2*pi*9000 rad/sec Minimum stopband attenuation = 40db having the lowest possible filter length. I fou...

Hello, I'm currently analyzing wind speed records, and it turns out that type of filter used influences the power spectrum. Does any one...

Hello, I'm currently analyzing wind speed records, and it turns out that type of filter used influences the power spectrum. Does any one know of a good first reference/book for Digital Signal Processing? I'm also looking for the filter functions for block averages as I also need to understand this for my research. Regards, Juan Antonio

Hi, I have always been confused about this two functions. Can someone please clarify why they behave this way? Let me give an example To me...

Hi, I have always been confused about this two functions. Can someone please clarify why they behave this way? Let me give an example To me it looks like that if we compute the impulse response of a filter, it should be the same thing as taking the FFT of that filter. But how come there is a difference? For example, let's say if I take the simple FFT of the following hamming window: L...

Greetings everyone in the group. This is my first message. I am not a DSP engineer but a programmer with a very limited knowledge of filters. I...

Greetings everyone in the group. This is my first message. I am not a DSP engineer but a programmer with a very limited knowledge of filters. I am writing in the hope that someone can help me with his/her expertise in the following problem: I have a butterworth filter implemented in software.(C++) So far it works exactly the same as the functions: butter() and filter() in Matlab. Therefore wh...

Hi, I am presently working on Filter design, i need to introduce the finite wordlength effects. does anyone worked on this ...

Hi, I am presently working on Filter design, i need to introduce the finite wordlength effects. does anyone worked on this please suggest me some tips. how to implement it. thanx in advance, sree.

Hi everyone, I am trying to implement a gabor filter to a 1D signal. Are there any built in functions or toolboxes available. ...

Hi everyone, I am trying to implement a gabor filter to a 1D signal. Are there any built in functions or toolboxes available. thank you in advance,

Hello,               I tested digital filter by giving the sweep signal.  Is there any other method to...

Hello,               I tested digital filter by giving the sweep signal.  Is there any other method to basically check, filter is working or not ??     

Hello everybody. Can any one let me know about good Digital Filter Design Packages available and its cost? How do they compare...

Hello everybody. Can any one let me know about good Digital Filter Design Packages available and its cost? How do they compare with others? Thanks arun

Dear List: I have a doubt in filter implementation! I wish to include the effect of quantization after...

Dear List: I have a doubt in filter implementation! I wish to include the effect of quantization after multiplication with the filter coefficients. Eg. I need, y[n] = Q(x[n] b0) + Q(x[n -1] b1) -Q(y[n -1] a1) ; where Q(.)

hi, I am designing an IIR filter for my application ... the problem is i am confused with the transfer function... some books...

hi, I am designing an IIR filter for my application ... the problem is i am confused with the transfer function... some books say it is: y[n]=a0*x[n]+a1*x[n-1]......+b1*y[n-1]+b2*y[n-2].... some other books says, y[n]=a0*x[n]+a1*x[n-1]......-b1*y[n-1]-b2*y[n

Hello, I'm trying to create an FIR filter to cancel some noise in an audio sample by looking back at the previous samples (some m samples back)...

Hello, I'm trying to create an FIR filter to cancel some noise in an audio sample by looking back at the previous samples (some m samples back) to create the filter weight for the current sample and then incrementing (similar to a queue where a new sample enters). I am using the auto and cross-correlation between the noise and the noisy signal to find the filter coefficients and not having ...