Stack Exchange Network

Stack Exchange network consists of 174 Q&A communities including Stack Overflow, the largest, most trusted online community for developers to learn, share their knowledge, and build their careers.

Visit Stack Exchange

FIR filters are filters with a finite duration impulse response.

1
vote
1answer
36 views

What is an “innovation filter”?

I'm a math postgrad student working through a paper on eigenvalue decompositions of matrices of FIR filters (used for stuff like total decorrelation, convolutive mixing, MIMO). Towards the beginning, ...
1
vote
1answer
24 views

Parallel Cascaded FIR Filters

I have a relatively simple question. I know if you connect multiple LTI systems in parallel, overall system's frequency response is the sum of each LTI system's frequency response. Therefore, what ...
1
vote
1answer
39 views

Design of a complex FIR filter

For apply least-squares linear-phase FIR filter design,with frequency domain specification is not symmetrical. The pass-band error function, $$E(\mathbf{h})_p=\int_{\omega_{p_1}}^{\omega_{p_2}}| \...
0
votes
0answers
19 views

How to select Input parameters of FIR band pass filter (narrow band e.g Alpha (8-12 Hz)). A basic question?

I have to filter eeg data (500 samples) offline using a FIR bandpass filter for alpha rhythms (8-12 Hz) providing the fixed order of filter (i-e., 128). Can some one explain what should be the ...
0
votes
0answers
21 views

Complex and nonlinear-phase equiripple FIR filter design

I want to apply cfirpm function for the design of a possibly complex FIR filter, The response is sampled at 100 linearly spaced frequency points. The passband spans [−0.4fs, 0.7fs] where fs is the ...
5
votes
4answers
190 views

Why is the first value in digital first-order IIR filter impulse response not the largest?

When I create a digital first-order IIR low-pass filter with scipy (code below), I get the following coefficients: B: [ 0.1367 0.1367] A: [ 1. -0.7265] The ...
1
vote
1answer
52 views

estimate the impulse response after equalization

I want to estimate the impulse response of the channel at the receiver. Assuming some arbitrary impulse response: h=[1 0.2 -0.4 0.0 0.6]. Once the equalizer is ...
0
votes
1answer
75 views

FIR filter design: evaluating the error function

Question 1 $$ H(e^{j\omega})=\sum_{n=0}^{N-1}h[n]e^{-jn\omega} =\mathbf{c}^H(\omega)\cdot \mathbf{h} \tag{1} $$ $$ =\mathbf{h}^H\cdot\mathbf{c}(\omega) \tag{2} $$ $$H(\mathbf{h})=\sum_{k=1}^Kh[k]e^{...
-1
votes
1answer
40 views

Is this a FIR algorithm?

I'm kind of out of my element here trying to understand this algorithm and I wanted to see if this is a FIR algorithm or not? If this isn't the right place to ask this, can someone point me to a place?...
0
votes
1answer
69 views

Design of FIR filters with arbitrary magnitude and phase responses

I found many information in this thesis "Algorithms for the Constrained Design of Digital Filters with Arbitrary Magnitude and Phase Responses",i want to understand it. This question is a ...
0
votes
1answer
35 views

Arbitrary-Response Filter Design

I want to design a FIR filter by using 'cfirpm' function , which can i use the group delay and grid frequency, my code is : ...
0
votes
1answer
36 views

FIR filter design and implementation meaning [closed]

What is the output of the convolution meaning which is used in the FIR filter? how can conclude the output of the FIR filter in the output according to the input $x[n]$?
2
votes
2answers
92 views

Mean delay for a M tap harmonically weighted filter

I am trying to derive the mean delay for $M$ filter taps harmonically weighted: $$y[n] = \frac{M\,x[n] + (M-1)x[n−1]... + 1 \cdot x[n−M+1]}{\tfrac12 M (M + 1)}$$. For a uniformly weighted filter, I ...
0
votes
1answer
125 views

16x interpolation using polyphase FIR - sum of the coefficients

I have an issue with polyphase implementation during interpolation. Lets assume I have 256 taps long FIR lowpass sinc filter. In order to interpolate by a factor of 2 I do use two FIR filters (...
0
votes
1answer
34 views

FIRFilter with low cut-off compared to sampling frequency

Sorry if my question is some FIRFiltering basics, but I couldn't find a specific answer after some search. I am trying to design a decimating low pass filter in C (actually in OpenCL with GPU) with ...
0
votes
0answers
39 views

The grid frequencies In the design of a digital filter

I want to find many difference 'h' with same group delay and length filter using this code : https://gist.github.com/mattdsp/17f508d2b368db61f47689e4622d8841/c1518f22a5aa451951601b56c5d9f72743868bfe ...
0
votes
3answers
38 views

Phase response of a FIR Hilbert transformer

The graph below shows a linear phase response of FIR Hilbert transformer with a constant diff of -0.06135923. I am not sure if I understand the concept of the phase response correctly, but as the ...
0
votes
1answer
64 views

FIR Impulse response & transfer function

I am working though a dsp past paper and I have come across the questions "find the impulse response in the time domain" & "find the transfer function in the time domain" I know its really simple ...
1
vote
1answer
56 views

Is it possible to get the envelope of a signal via FIR filter?

I was reading an article about ultrasonic signals, and I read in it that it is possible to get the envelope of a signal by using a band pass FIR filter. My question: Is it possible to get the envelope ...
0
votes
0answers
37 views

IIR (non linear phase) filters

As you know, a nonlinear phase filter (in passband), distorts the frequency contents of passband region of signal. In general, Why do we use such filters in the case they cause distortion in the ...
1
vote
1answer
49 views

How do I plot the square of the amplitude response?

I have calculated the transfer function of an FIR filter $$ y[n] = x[n] + α · x[n − R] $$ This is what I have $$ H(z) = 1 + αz^{-R} $$ Now I should plot the square of the amplitude response. So I ...
0
votes
0answers
25 views

How to remove 750Hz sinusoidal background from audio using FIR filter? [duplicate]

Kindly help me to design a FIR filter for eliminating the 750 Hz sinusoidal beep from audio. I shall be thankful to you
0
votes
1answer
63 views

What is meant by Two pass FIR Filter?? (A basic question)

I have a confusion about what does a two pass FIR (bandpass) filter with order 40 means?? Passband frequencies are [8 13]. Type 2 FIR is same as two pass?? I have check some previous literature ...
2
votes
1answer
124 views

Discrepancies between FFT-based Hilbert Transform and FIR filter results

I compared the results of Hilbert Transform as provided in scipy.signal.hilbert() (which is uses I/FFT as you can see in the source code) with its approximation implemented as FIR filter using coeffs ...
1
vote
3answers
80 views

No-lag causal bandpass filter kernel formula

Can anyone point me towards a formula for a good (for any values of "good") no-lag causal bandpass filter? I am doing sound processing and I need to identify starting points of certain sounds (target ...
0
votes
1answer
50 views

Applying multiple IIR and FIR filters at once

I would like to appy multiple filters to a signal at once. Let's assume I got two frequency bands I want to extract. I compute a Butterworth Filter for 300-500Hz and one for 700-800Hz. I would like to ...
0
votes
0answers
79 views

Differences between Matlab resample() and CMSIS decimate_f32()

I am pretty new to DSP and I need some help to pinpoint my mistake. I am using Matlab to simulate an embedded environment. I have three signals all centered on 50Hz (6.4KHz sampling rate) which have ...
0
votes
1answer
53 views

Compensating for group delay in FIR with symmetrical coefficients

Suppose I have a sinusoidal signal of 50Hz sampled at 6.4kHz and I wished to filter it at: Band pass: 160Hz, Bands stop: 800Hz, Apass:0.1db, Astop: 106db For a FIR filter order of 40 I would get 41 ...
0
votes
1answer
62 views

Z-domain transfer function to difference equation

So I have a transfer function $ H(Z) = \frac{Y(z)}{X(z)} = \frac{1 + z^{-1}}{2(1-z^{-1})}$. I need to write the difference equation of this transfer function so I can implement the filter in terms of ...
0
votes
0answers
69 views

Estimating frequency when there's little isolation between signal and noise frequencies

Say I have a signal which is guaranteed to have a frequency between 110-120 Hz but is corrupted by interference signals that're very close to this frequency range. For example, let the interference ...
1
vote
0answers
39 views

Implementation of halfband pass filter

I need to interpolate a complex valued bandlimited periodic function using local interpolation. I can have the signal sampled at any frequency I want over at equispace intervals. I am aware that for ...
0
votes
1answer
139 views

Interpolated FIR filter group delay

I'm trying to design a digital low pass filter with a narrow transition band. My sampling rate is 25 kHz, the cut off frequency is 60 Hz & the transition band width is 4 Hz. I'm looking for about ...
1
vote
0answers
77 views

Equalizer design for spectral gain flatness

I'm trying to design an equalizer for spectral flatness over a 100MHz signal for an LTE channel emulator. A channel emulator simply multiples channel coefficients with an external signal from a signal ...
3
votes
2answers
386 views

A basic question about the use of moving average vs low-pass filters in DSP

I'm newbie novice in digital signal processing. If one has a signal with some noise superimposed in time series, for which type of information/analysis one would use moving average or LP filter in ...
0
votes
1answer
62 views

Arbitrary Phase FIR design using FFT

I want to design a digital FIR filter to cancel the phase of an analog Butterworth filter (i.e. the phase is smooth) i.e. the filter has unity gain with just an inverted phase component. The ...
-1
votes
1answer
120 views

Get the frequency response curve from FIR filter coefficients + sampling rate

What's the fastest way (if possible in the browser thanks to an online tool, or if not possible easily, with Python), to get the frequency response curve (x-axis: Hz, y-axis: dB), when giving just: ...
1
vote
1answer
32 views

Help generating coefficients for basic FIR LPF

I'm trying to generate coefficients for a FIR low pass filter to be used in an FPGA. I'm using python (scipy.signal) to attempt to do this, but am having trouble getting coefficients in a usable form....
0
votes
1answer
25 views

What does “ripple limit cutoff” mean?

So I'm reading the datasheet for MAX98357A. It's an audio amplifier that receives digital signal using I²S bus and has integrated digital low-pass filters of the IIR and FIR types. In addition to ...
1
vote
0answers
48 views

Interpolated FIR (IFIR) for decimation

I'm interested in implementing an optimized 5x decimation stage. However it seems the resources needed are significantly more than 4x or 8x composed of cascaded 2x stages. For general purpose FIR ...
0
votes
0answers
136 views

Understanding Scipy's Savitzky- Golay filter implementation

Scipy's cookbook documentation gives the following code ...
0
votes
0answers
34 views

Which FIR filter window to choose?

I need to design a high pass FIR filter based on parameters given. However I'm stuck at which window i should select. How do I calculate the parameter M? The guide I referred myself to was behind ...
0
votes
0answers
87 views

FIR Filter Signal Flow Graph

Have made difference equation of signal flow graph of FIR filter is it correct or not?
-1
votes
2answers
154 views

Fir filter Hamming window 161 order bandpass filter implementation

I'm running an FIR filter hamming window order 161 bandpass with pass frequency range 20Hz-200Hz, but output doesn't sound like the filter is working. It's just distorted output. Following is the ...
0
votes
1answer
48 views

Some doubts in the matlab code of computing Wigner distribution

I found that a discrete form of the Wigner distribution was $$WD(n,k) = \sum_{m = -N/2}^{N/2}f(n+m)f^*(n-m)e^{-j\frac{2\pi}{N+1}2mk}$$ where $f(n)$ is the signal and time limited within $|n|\le N/2$. ...
0
votes
1answer
67 views

Downsampling in MATLAB - why scaling filter coefficients?

Trying to undertsand how MATLAB does resampling, looking at toolbox/signal/resample.m. This is the part where (downsampling) filter coefficients are calculated: <...
0
votes
1answer
54 views

Eigenvalues of FIR Convolution Matrix

Are the eigenvalues of the FIR convolution matrix the zeros of the corresponding FIR filter? Suppose I have an FIR filter $H(z) = h_{0} + h_{1}z^{-1} + h_{2}z^{-2}$. I want to implement it using a ...
2
votes
2answers
57 views

How to classify filters?

In class, we have defined four types of filters: low-pass, high-pass, band-pass and band-stop. I have understood that in order to classify a filter into one of the fourth, one can use the signals: DC:...
1
vote
0answers
54 views

When Is a Power of 2 FFT Slower than Smaller Sized Non Power of 2 FFT in MATLAB

Knowing that computing an FFT is faster if the amount of samples is a power of 2 I have always tried to pad the inputs to Matlab's FFT with zeros until the next power of 2 is achieved. Matlab's ...
0
votes
1answer
78 views

Amplitude problem by puting 5 FIR in a row

I'm trying to use 5 FIR in a row to improve the S/N ratio on my signal, but I have some difficulties doing it. In fact I have 2 signals, let's call them sig_I and sig_Q. I want to apply my 5 filter ...
0
votes
1answer
67 views

Understanding and Correcting phase shift in General Linear Phase FIR filters

I have been learning about General Linear Phase Filters and the four standard FIR implementations. I was wondering in the case $\beta = \frac{\pi}{2}$ or $\beta = \frac{3\pi}{2}$ if there was a way ...