Questions tagged [digital-filters]
The digital-filters tag has no usage guidance.
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Design of digital filters with negative group delay
I am currently working on digital filters that can predict my input signal(assume that input signal is bandlimited). In other words, I want my filter to have a flat magnitude response in bandwidth of ...
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Understanding how to implement high-pass filter with cascaded biquad filters
I'm trying to understand how to implement a high-pass filter using, quote:
a cascasded biquad direct form II IIR filter
with a cut-off of 8kHz.
DSP is a complete new field for me, and the ...
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Mixing multiplied audio signals without clipping
I'm building a vocoder on an FPGA, and now trying to mix $N$ signals that are each the product of some carrier signal in one of $N$ bands and an envelope of a modulator in that same band (essentially ...
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The output length of upfirdn in MATLAB doesn't match that of the theory
I have some doubt about the upfirdn built-in function in MATLAB. First I generate the root raised cosine transmit filter as
tRRC = rcosdesign(0.25, 20,4,'sqrt');
...
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Truncating the output of a digital filter. Which part to discard?
Suppose I have a signal $\mathbf{x}\in \mathbb{C}^{N}$ and a digital filter with impulse response $\mathbf{h}\in\mathbb{C}^L$, where $L<N$. If we pass the signal through the filter, the output will ...
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Trying to understand scipy.signal.kaiserord
I am trying to understand scipy.signal.kaiserord example. The example says
We will use the Kaiser window method to design a lowpass FIR filter
for a signal that is sampled at 1000 Hz.
We want at ...
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How to design low pass filter for this case
From control systems perspective, I have two subsystems that are connected in a cascade structure. An outer loop represents a joystick that provides position that runs at 1kHz and an inner loop ...
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Why will the output be imprinted with delay effect or echo after convolving two direct dry signals' FFT values (convolution filtering)
I am recently working on the reproduction of the filtering effect of target play back devices, like phone, or speakers, etc using convolution techniques in MATLAB.
I firstly created a function called &...
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Matlab - dfilt filter produces different results for SOS and normalized SOS
I think the title says it all. Anyone could provide a solid answer to why it happens? Here is a sample code to replicate the issue.
...
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Can the standard deviation of the Gaussian window in a Gabor filter be made infinitesimally small?
My understanding is that the standard deviation of the Gaussian window in a Gabor filter dictates the temporal resolution.
Wouldn't it always be better then to make the window smaller, thus achieving ...
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What is the difference between STFT and Gabor filter?
My understanding is that a Gabor filter gives us information about when in the input signal does the frequency of interest (equal to the frequency of the sinusoid in the Gabor Filter) occur, thus ...
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How to implement a filter for a sigma delta modulator?
I am working on a project involving an accelerometer that outputs a pdm signal, which needs to be decoded to an acceleration. The device itself outputs a clock and data line, we just have to sample ...
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What does a Gabor filter "filter" out?
Almost all of the searches online have returned results related to image-processing, but I am looking for a more rudimentary example with some physical intuition (eg; a high-pass filter only lets the ...
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57
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Lattice Structure of system $H(z)=1+0.7z^{-1}+1.2z^{-2}-z^{-3}$
When solving
$$A_{m-1}(k) = \dfrac{A_m(k)-A_m(m)*A_m(m-k)}{1-A_m(m)^2}\,,$$
the $A_m(m)$ here is $A_3(3)=-1$, which results in infinity if put in above equation.
How to solve it?
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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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How to preserve constant segments in the signal during digital filtering
I have a sampled signal, which has constant segments(I pad the signal if some of the BLE packets were lost). I want to apply a digital filter to the signal, say Butterworth and I don't want it to ...
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How to recover the LTI system step response by the known output and input signals?
Having the input signal as a step-like pulse and the output as its distorted version after passing through the system:
is it possible to somehow recover the step response of the system?
In the Figure ...
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73
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Cfar Algorithm Implementation
I am trying to implementation Cfar algorithm to our radar signal processing algorithm. I obtained some output but I want to be sure that is right. Cfar algorithm's output below like, I'm interpreting ...
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Decoding an analogue signal for a given sampling and signal frequency that aren't in sync
I am trying to decode an analogue signal, that is intended to be converted to a digital one and decoded as USART communication.
In the image shown the blue trace is the analogue signal, and the ...
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Z-Transform of a Precoder with XOR
I am reading a paper called "Design and Comparison of Three 20-Gb/s Backplane
Transceivers for Duobinary, PAM4, and NRZ Data" and got stuck on seemingly easy thing, how the Z-domain ...
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67
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How to trade poles of All pole filter with zeroes?
I have filter of order 2000 IIR all-pole. I want to implement it via cascade IIR filter. Is there any way to reduces poles order and increase zeroes as its cost. To balance zero and poles. Since all ...
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Converting transfer function to cascaded IIR filter [closed]
I have two function, But I don't know which one is correct to convert my Transfer Function(tf) to secound order sections(sos).
I'm using GNU octave, and it have intrinsic ...
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How to find system output by its step response?
Inspired by this post, I tried to reproduce the procedure described in the answer in Python considering rectangular pulse:
...
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102
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How to create a filter that is a single period of a time-warped raised cosine?
I am attempting to implement a filter described in the paper Functional Count-Comparison Model for Binaural Decoding by Pulkki et al 2009:
Here is Figure 9:
However, I am quite certain that the ...
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What are the tradeoffs between shifting a lowpass FIR filter vs using a bandpass FIR filter?
Is there a difference between shifting a lowpass FIR filter to an arbitrary frequency offset versus creating an equivalent bandpass filter, such that the passband and cutoff frequencies are identical? ...
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66
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multi h CPM implementation
I have understood the math behind CPM. but recently I came across multi h CPM. I haven't found any good document to explain about multi h CPM. Can anybody explain the math behind it. and how it ...
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143
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How to detect decrease of a digital signal?
I have below given digital signal (sampled with period $T = 1\,\mathrm{ms}$)
for which I need to evaluate its decrese. Namely I am interested in regions where the signal decreases faster than $-1000\,\...
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263
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Is this a leaky integrator?
We have implemented a way of approximating a data set with the following lines of code based on the dx/dt of wikipedia of leaky integrator.
...
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How to initialize states of second order direct form 2 transposed filter
Background
This question is an extension to this post. I need to calculate states ($S_1$ and $S_2$) of a second order direct form 2 transposed (DF2T) filter that result in minimum 'transient' when ...
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How to initialize a second order direct form 1 filter for no transient upon filter reset
I need to calculate initial states of a second order direct form 1 (DF1) filter that result in zero transient upon reset (initialization). I'm starting with setting filter input (x) equal to filter ...
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Tap and Impulse Response of MATLAB Filter
This is a simplified version of the Savitzky-Golay MATLAB functions (sgolayfilt, sgolay).
...
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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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93
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Denoising Signal With Butterworth-Filter
im trying to denoise a signal to which i added AWGN. Here is what ive done so far:
...
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Filtering with scipy sosfiltfilt generates flat edge artifact
I need to filter some array of data before differentiating it, I do this so I don't drown the signal in noise.
It's also important that no delay is applied to the signal, that's why I decided to use a ...
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Correct baseline for slow sinusoidal noise signal
I acquired some noise signal (5GHz samplerate, 520 samples) data from a device and recognized that it is not gaussian distributed. There seems to be some slow sinusoidal signal underlying. I tried to ...
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Poor Attenuation With High Pass Filter
I am having issues with my implementation of a FIR high pass filter. The low pass filter has an acceptable level of attenuation in the stopband, but the high pass filter has poor attenuation in the ...
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Why do inverted impulse responses sound awful when applied to an audio system?
In this answer, the responder claims that acquiring a measured impulse response and straightforwardly inverting it (e.g. by directly in the spectrum, after zero-padding, or by determining the inverse ...
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Calculating the filter coefficients for 2nd order bandpass and bandstop Butterworth filter
I want to calculate the filter coefficients for both bandpass and bandstop filter for 2nd order Butterworth filter, like
...
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221
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Fast-settling CIC/FIR filter design
I am investigating the properties of the cascaded integrator comb (CIC) filter (for a $\Sigma\Delta$-modulator). I have two questions:
What's the correct method to design a fast settling decimation ...
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"There's no ideal lowpass filter" - really?
Sinc is $\propto 1/t$. If $x(t)$ is bounded, then there exists $t: |x(t)/t| < \epsilon_M$, where $\epsilon_M$ is machine epsilon. If $x(t)$ is also time-limited, it also means there's $\tau$ such ...
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Real coefficient linear phase FIR filter design based on least square
I want to build up linear phase filter by least square approach.
I have formulated a least square problem similar to Matt's blog. To approximate the desired complex response $F \times 1 $ vector $d$ ...
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How to remove PWM cycles present in a signal?
I have applied PWM across a DC motor. The voltage measured across the motor is shown above.
I want to count the ripples present in the data. The ripples exist at a different frequency than the PWM ...
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transformation methods for digital filters
i have the following question:
in digital filter design what's the difference between the methods of transformations :
bilinear vs impulse invariance vs Euler vs step invariance.
thank you!
Here is a ...
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Band pass filter = Band stop filter if you switch the cutoff frequencies?
The normalized frequency response of a N points bandpass digital filter is:
$\frac{sin(\omega_{c2}(n-M))}{\pi(n-M)}-\frac{sin(\omega_{c1}(n-M))}{\pi(n-M)}$ where $\omega_{c1}$ is the normalized lower ...
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Difference between FIR and IIR filters
I understand that FIR filters have a finite impulse response and IIR filters have an infinite impulse response. Here's the issue:
A signal is finite in time if and only if it is infinite in frequency.
...
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FIR filter for flipping the phase in negative phase
I have built up a FIR filter based on least square approach.
I deploy the IDFT matrix $$F_{inv}$$ in above document page 7 and the desired response for compensated is H_F, the objective is
$$
min_{...
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Fixed point implementation of IIR filter
I am trying to implement an IIR filter followed by downsampler with the following specifications:
fs=18000,order=8,3db freq=[500 1000],downsampling factor=6.
...
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Using Zero-Phase Anti-Causal Filters in Real-Time Embedded Systems
Wanted to know the feasibility and usefulness of implementing Zero-Phase Anti-Causal filters such as those mentioned at this link in modern embedded signal processing applications given the ...
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Implementing a high pass filter in MATLAB
I am trying to implement a high pass digital filter in MATLAB.
High pass filter:
...
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Behaviour of difference equation does not match that of its z-domain transfer function
I have obtained the following z-domain transfer function:
$\frac{Y(z)}{U(z)}=\frac{3.3641×10^{-7}×z^{6}+1.5584×10^{-5}×z^{5}+6.7263×10^{-5}×z^{4}+
5.5016×10^{-5}×z^{3}+8.525×10^{-6}×z^{2}+1.2303×10^{-...
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