Questions tagged [finite-impulse-response]
A finite impulse response (FIR) filter is a type of a signal processing filter whose impulse response is of finite duration.
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Is this FIR filter considered symmetric?
My goal is to generate linear phase FIR filters. Currently, I am generating an array of frequency response magnitudes for the positive frequencies and then I mirror those with the complex conjugate. ...
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Confused About Relationship Between Number of Points and Solving Wiener Filter Coefficients in Time Domain vs Fourier Domain
I'm interested in computing M, FIR filter taps / coefficients using a Wiener filter approach with known signal and noise timestreams. I know of two ways to compute the coefficients: one in the Fourier ...
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Compress filter coeficients?
if we have a FIR with 100 coefficients, is there a method to "compress" the length? so for example, have a similar response with 50 coefficients?
I am ok with the response degrading within ...
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FIR Filter Design in python
This is mine code for fir_filter_design in python. But I'm getting wrong output for low-pass, band-pass, and high-pass magnitude response.
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I'm struggling to understand the FIR Filter and its application
I'm new to GNU Radio, and I was watching this video However, I didn't quite understand the purpose of the FIR Filter. I know what decimation or downsampling is because I have the book "Digital ...
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Design a lowpass filter cutoff and sampling frequency is given
Design a FIR low pass filter with cutoff frequency of 1 𝑘𝐻𝑧 and sampling frequency of 4 𝑘𝐻𝑧
with 11 samples using Fourier series method. Realize the filter with suitable structure.
Determine the ...
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Algorithm to "serialize" impulse responses
New user to dsp.stackexchange here. I'd like to know what's a quick and effective way to take 2 impulse responses and create one that's equivalent to both of these "in series" for audio ...
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FIR filter design with nonlinear phase from measured amplitude and phase responses
I am having trouble when design FIR filter fitting to the complex data (i.e., amplitude and phase responses from measurements). I did try to use Matt. L's lslevin method here since this method is to ...
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On FFT, interpolating signal vs extending signal in time
When we interpolate, then FFT the output will have more bins.
When we extend the signal in time, Then FFT output will have more bins too but:
Interpolation increases max bin frequency but time ...
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Does block-based FIR filtering necessarily imply using the DFT?
Assume I want to filter a stream of data with a rather small FIR filter (in the order of 16 to 64 coefficients). In any case I have to split the input stream into chunks (blocks) of data.
Searching ...
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Impulse response aquired by ifft seems to require circshift
On the system below:
I have found W by inputing 20-1000 frequency and found amplitude and phase for each frequency. I've tested for minimum phase and none min S. But the answer is not logical for me. ...
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Total delay of cascade FIR filters
I have a doubt regarding the total delay in terms of taps for cascade filters.
Imagine that I have an SOS IIR Filter (2 delays) and an FIR filter of 10 taps (10 delays) and another of 20 taps (20 ...
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Impulse Response for an Input-Output Pair
Given an input-output pair of a LTI system
\begin{gather*}
x[ n] \ =\ 2\delta [ n+2] -\delta [ n+1] +\delta [ n-1]\\
y[ n] \ =\ 4 \delta [ n+2] +\ 4\delta [ n+1] -\delta [ n-1]
\end{gather*}
My ...
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Kaiser window but desired frequency response extends beyond pi?
Passband : 0 to 0.5π
Stopband : 0.7π to 2π
Is there a way to reduce this to a 0 to π scale?
If I had to guess, we should be able to simply divide the edge frequencies by 2 and use the time scaling ...
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Modeling an Acoustic Reflection from a Wall - a Paradox?
I am trying to simulate the reflection of a sound ray, that goes from a sound source, bounces off a wall, and is received by a microphone.
The wall has a an absorption coefficient, and a specular ...
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Scaling for windowed differentiator FIR filters
I recently took a dive into DSP and I was wondering what the appropriate scaling factor for a wide-band differentiator is.
Let's derive the impulse response of such a differentiator.
$$\begin{aligned}
...
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Understanding FIR windows
We learnt about the various windowing techniques recently and I can't seem to wrap my head around why one would use anything other than a rectangular window.
I created a signal with 10 evenly spaced ...
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How to normalize FIR filter gain from the amplitude response in frequency domain?
I've got a set of measured data (amplitude and phase responses) of a filter. And I want to normalize its gain to be unit (equal to 1) for further processing. I know that we can convert to time-domain ...
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How to Dereverberate Speech taken in an Auditorium with Reverberation Time of 3.8 to 4 seconds
I'd like to cancel echoes from a Talk recorded in a large extremely reverberant auditorium. It's unintelligible as recorded, and I'm hoping to make it intelligible by echo cancellation.
Audio was ...
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Why the difference of a vectorized VS non-vectorized FIR convolution differ?
I have an FIR filter that is a low pass at 11025Hz for 44100Hz, 461 taps. The reference implementation, i.e. naive convolution, works as expected. Now I've written a vectorized implementation to take ...
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IIR/FIR equations for custom frequency response
Given a frequency response plot where gain is 0dB up to Fc and after Fc gain increases linearly by factor K (example chart below):
I am seeking how to arrive at its IIR/FIR equations so it can be ...
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How to speed up convolution for symmetrical property of a half-band FIR filter?
I've discovered that such convolution can be further sped because of symmetric coefficients but I'm not able to get it right.
The filter is a low pass filter at 11025Hz for 44100Hz, with 461 taps.
The ...
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Low-pass Filter with constant phase response
I would like to design (e.g. in GNU-radio) a low-pass filter where up to the cut-off frequency the phase-response is constant (I do not mean a "linear-phase response"). The rationale is to ...
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Design a Real FIR with arbitrary Phase Response
Intro
My question is related to this one: Correcting phase response of a signal.
I need to design an FIR low-pass filter with real coefficients for data received from a sensor. An additional ...
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Half-band FIR filter is not filtering as expected
I recently discovered that one may be able to further optimize an FIR filter processing time by skipping calculations of coefficients whose values are zero, providing that the filter is about halving ...
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Rayleigh flat fading channel and pulse shaping
I'm trying to implement the Rayleigh flat fading channel in MATLAB using SRRC (Square-root raised cosine) pulse shape. The modulation scheme is BPSK. Here is my code:
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Finding impulse response/polynomial zero on the unit circle
The polynomial here is impulse coefficients of minimum phase FIR filter, or it's impulse response.
A code somewhere tries to finding roots of polynomial on the circle. It have roots of polynomial <...
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Resampling a signal with variable sampling rate
I’ve created an upsample and a downsample functions (x2) using polyphase FIR filter.
I have a signal which comes in chunks where sampling rate of each chunk can vary between 1x and 2x (e.g. 48000 Hz ...
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Trade offs in window based design of FIR filter
Please find attached snap of proakis dsp book. It shows rectangular window has minimum value of stop band attenunation and transition width and as we go on towards Hamming and Blackman window,stop ...
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Random Peak at the end Impulse Response
I am measuring a room's acoustic impulse response by playing a log sine-sweep through a speaker from 20hz to 24 khz, and then recording it using a microphone.
The sweep is 10 seconds long, followed by ...
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Significantly different results between matlab's fft and FFTW for FIR-filter implementation
I am trying to implement the fast fir-filter in a C++ application.
I have implemented the same code in matlab as a way to double-check my implementation and have ran into a problem.
When running ...
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Controllability matrix is full rank identity matrix
What does it mean, when my system has controllability matrix that is full rank and identity matrix? Having full rank means that the system is controllable, but does the Identity feature of my ...
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Parallel processing of FIR filter multiple samples per clock cycle
I have an FPGA that is limited in clock frequency, but receives data from a higher sample rate ADC by receiving N samples per clock cycle (all N samples are from a single ADC channel). So the FPGA has ...
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Solving the Cocktail Party Problem when the channel response of each input is known
I have a measurable $y[n]$ that is the sum of two input channels $x_1[n]$ and $x_2[n]$ according to
$$ y[n] = x_1[n] + h_0 x_2[n] + h_1 x_2[n-1] $$
where the scalar weights $h_0$ and $h_1$ are known. ...
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Derivation of the 4 types of real-valued linear-phase FIR filters
Is there any resource or literature where I can find derivations/proofs for all four linear phase filter types starting from an arbitrary real valued impulse response? I looked at some of the ...
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Jointly determining the weighting matrix and target vector in weighted least square
I have the same weighted least square form weighted least square composed of diagonal weighting matrix $ W $, IDFT matrix $F({\omega})$, desired response vector $D({\omega})$ and FIR Filter ...
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Optimize FIR filter kernel in Matlab
I am currently using the firls function in Matlab to create my FIR filter kernel. This works well but I typically have to play with the filter order to get the gain to 1 and the shape to be somewhat ...
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Frequency Translation after FIR Filter
I am trying to translate a signal to baseband by multiplying by the complex exponential. The issue is when I do the calculation in MATLAB the signal seems to disappear. I originally thought the signal ...
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LCCDE filter classification
I am trying to figure out if
$$y[n]-2.56y[n-1]+2.22y[n-2]-0.65y[n-3]=x[n]+x[n-3]$$
is either IIR or FIR, causal or non-causal, stable or unstable (BIBO).
I got the transfer function from the ...
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Integral Calculation in Matlab for Filter Optimization
I am trying to calculate the area under my FIR filter kernel using trapz but I'm getting weird results. The reason I want to know the area is eventually I would ...
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How does rate reduction work in a decimator with even valued gain?
Cascaded integrator-comb (CIC) filters are used in decimators, and decimators with a gain that is a power of 2 are popular (at least at my workplace) because the gain can be normalized with a right ...
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FIR filters: is it possible to manipulate phase without change in magnitude response
Here's response of FIR design:
which shows SPL 0dB and here's a wave file exported from ( DRC ) FIR design software: FIR48kHz.wav.
Audio software (internally uses FFTW routines) where this filter is ...
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Troubleshooting distortion in room impulse response measurements:
I am collecting data for a project, where I am measuring the room impulse response by playing a sine sweep from a speaker (QSC K8 1KW Active Loudspeaker) and measuring it from the other side of the ...
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Stringent filter requirements
I am trying to resample a signal using Matlab's resample function. The documentation explains that conceptually, it is upsampling by p, then applying an interpolation filter, then downsampling by q, ...
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Implementing the NMCFLMS Algorithm for a 2 channel WGN input signal
I am just trying to implement the Normalised Multi-channel Frequency Domain Least Mean Square (NMCFLMS) algorithm for a simple 5 tap filter to estimate the filter coefficients of a 2 channel system ...
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Can Synchrosqueezing be use to Derive IIR Filter From Impulse Response
I am looking for alternate solutions to determine the IIR filter coefficients from the impulse response, or more specifically determine the closest IIR filter for a given FIR filter. Much of this is ...
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What is the position of all zeros of a minimum phase, Type 1 Linear Phase FIR Filter?
Let me write down all the facts that I know of.
In context of the z plane:
Minimum phase system: All zeros and poles of such a system lie inside the unit circle.
Linear phase FIR filter: For every ...
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Linear phase impulse response causes non linear phase frequency response in GNU Octave/ Matlab
Assume linear phase FIR symteric impulse response like [1 2 3 4 5 5 4 3 2 1],shoud resault in a frequency response with linear phase negative sloped phase. but it ...
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Subtract Lowpass from Wire to Get Allpass
One technique to make a high pass filter is to subtract a lowpass filter from a wire (meaning subtract the output of the filter from the input as I have demonstrated in this post).
What lowpass can ...
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Super Resolution Techniques for Tighter Transition Frequencies in FIR filters
In this related post:
https://dsp.stackexchange.com/a/86776/21048
I have demonstrated the similarity of FIR filters and their related frequency response to antenna arrays and their beam patterns (...
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