**1.661 filters questions.**

I'm trying MIM (Magnitude Invariance Method) and PIM (Phase Invariance Method) for to improve biquad LPF response at low sampling rates. I'm looking some help and examples of usage if available.
...

Suppose we have a grayscale image that contains vertical lines. Now suppose that not all vertical lines are the same, some of them have different thickness.
Question is, is there a way, in MATLAB or ...

I am now studying image processing in my spare time. My understanding of convolution is about 'response to a specific filter':
When we have a raw image, or raw signal; and a filter, aka kernel; we ...

can i increase or i change the sampling period of signal reference of PWM indirectly using kalman filter ? please provide me any reference or any other method to do that as soon as possible

It's a beginner question, but useful to users from python - signal.lfilter, I was using lfilter from
Find reverse one pole ...

Mornng;
I am working on condition monitoring f gears transmission.
I import a database of the current's signal using LMS test lab. and I want to apply TSA (time synchronous averaging ) on this signal.
...

Can anyone explain why exactly an "Overshooting" phenomena is observed when the fundamental harmonic is removed as seen on the figures? Is it technically right to call this "overshooting" at all ? If ...

Assume the following first order IIR Filter:
$$ y[n] = \alpha x[n] + (1 - \alpha) y[n - 1] $$
How can I choose the parameter $ \alpha $ s.t. the IIR approximates as good as possible the FIR which is ...

I have an array of trits... Some are zero, some are one, and most I don't know (yet).
In my current data set of the KNOWN bits, currently only 10% are "1". (and only 5% is known).
I want to figure ...

I have studied convolutions and filters a long time ago. Today, I am trying to review the basics using some notes of mine, but I am finding difficult to solve easy problems. Since I don't have ...

I am trying to design an experiment to determine the peak amplitude of an EEG signal in response to a stimulus. Till now, our team has been using MATLAB and since we wish to go open source, we are ...

for a Project I need to build an Audio Modem in GNU Radio.
I decided, that it would be the best to use DBPSK Modulation, because that is fast and relatively noise resistant.
But as it seems, the ...

I ran a finite-difference simulation and the behavior of an output signal, $s$, in time, $t$ (sampled with period $\Delta t$) behaves approximately as in the figure below. It is well-described by a ...

I have designed this bandstop filter to a desired specification:
Stopband attenuation = 40db
Passband ripple = 2db
etc ...
I understand that the stopband ripple can be given by the following:
...

I am trying to design a bandstop filter to the following specifications using a basic window hamming.
the code used :
The results:
Hamming gives the following N =3.1/0.06=51.6, therefore a length ...

$$h[n]=\begin{cases}a^n & \text{if } 0 \le n < N \\
0 & \text{otherwise}\end{cases}$$
And for which values of $a$ the filter is stable
I know that the transfer function will be
$$H(z)=\...

I have to create a third octave spectrum from a time signal on Octave GNU.
I found some code on the net to help me but I don't have all the parts of the algorithm.
I have a .csv file which contains ...

a figure for instance of size 500*500 has half above part with black and below half white should result in a white line where the
white meets the black (something like a single line at line 250 with ...

I've been trying to design a bandpass filter using scipy but I keep getting a LinAlg Singular Matrix error. I read that a singular matrix is one that is not invertable, but I'm not sure how that error ...

I have a blurred and noisy image $X$, I want to apply the Wiener filter on it and get a deblurred and denoised image $Y$ (i.e apply inverse of blurring filter while at the same time reducing some ...

I have been out of touch with DSP for a while. I was trying to refresh my basics and ended up with this doubt on IIR filter stability condition. I understood that for an iir filter to be stable, the ...

A filter like this:
$z^{4}\cdot\frac{2 + z^{-1} - 3z^{-3} + ...}{1 - 0.9z^{-1}}$ seems hard for me to implement. The fraction part is easy:
...

I have a signal with frequencies betwwen [0.5,500] Hz and i want to create a filter to best separate the signal in the following regions:
[0.5-3]Hz
[3-8]Hz
[8-13]Hz
[13-30]Hz
[30-500]Hz
My ...

For a given narrowband Gaussian filter with a specific central frequency and filter width, I need corners of a bandpass Butterworth filter whose amplitude response is close enough to the Gaussian ...

In [1], the author shows an efficient way of implementing the forward and backward filter using matrices. One can also implement this using filtfilt command in ...

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 ...

Consider the LTI system given by: $H(z) = 1 - \frac{1}{2}z^{-1}+\frac{3}{4}z^{-2}$
Let $x[n] = (\frac{1}{2})^nu[n]$ be the input to the system. We want to find the output for $n = 0,1,...,N_a$, using ...

In explication ''the geometric interpretation of least squares'' Typically, the number of frequency constraints is much greater than the number of design variables (filter coefficients). In these ...

With scipy I can use signal.butter and signal.lfilter on a time signal.
Performing a Fourier ...

Is anybody familiar with Gustafson's algorithm for minimizing transients in forward backward filtering [1]? I'm trying to implement it and my first guess was to check Matlab's filtfilt.m, since they ...

For AR HP filter
zeros, to the right of the imaginary axis
poles outside the unit-circle
zeros on the real axis
poles, to the left of the imaginary axis
Apparently, the right answer ...

I understand to an extent various filter like low pass filter, high pass filer, Wiener filter Kalman filter etc. I also understand some of this filter will decorrelate/uncorrelate the signal. The ...

I am reading a paper inwhich there is a plot of log-Gabor quadrature pair which are even and odd.
I'm wondering how these two are plotted in time-domain, since log-gabor is real valued in frequency-...

I have a general question in image processing. I have a noisy image. I would like to classify the noisy image into some regions. Two famous approaches I can use are:
MRF/Gibbs MRF: Model the spatial ...

I am new to signal processing. I am trying to simulate something similar to IIR/FIR filter with $k$ delays to imitate acoustic echo reflection. The difference equations for FIR and IIR respectively ...

I'm currently working with a dataset of $5000$ pulses of $N=15000$ samples each. I managed to implement the RLS algorithms with a FIR M-Tap filter such that $M\leq 15000$ ($150$ seems to achieve the ...

I believe that there is no connection between the sampling frequency used for converting an analogue filter to digital filter and the one used to sample a signal that the filter will be used on. But I ...

I sample a continuous signal $s(T)$ over time. This leads to $s[t]$, which depends on several factors of which some are (pseudo-) periodic. I am interested in the effect of one such periodic factors ...

I have signal. I do Fast Fourier transform on it and now I have this output.
I want to delete the other except two that have high altitude. Can you help me ?
My f sample is 2500.
My complete time ...

I work with depth time series data from electronically tagged fish. When the fish spend time on the bottom we get a prominent tidal signal (approx 1.96 cycles per day). This interferes with our ...

I have a 50-60Hz signal being sampled at a frequency of 25 kHz & I'm trying to filter all frequencies beyond 70 Hz with a reasonable steep transition band (approx 4 Hz) & good attenuation (~40+...

Good Morning!
I am new in signal processing and I am trying to do a work in noise control of an electronic steering lock device (ESL). My aim is to calculate the loudness (Zwicker Method- ISO 532 B) ...

I'm trying to understand how to show that with real coefficients, the phase response of a filter is 0. Here is the impulse response
$h[n] = b_1d[n+1] + b_0d[n] + b_1d[n-1]$
How should I approach ...

I consider a signal of length $N = 2^n$ for some $n$. I want to derive two signal from it, one containing only the odd frequencies and one only the even frequencies. Each of these signals have length $...

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:
...

I've got a basic bandpass filter design that works great at higher frequencies, but breaks down at around 150 Hz. Here's the MATLAB code:
...

This is the simple code to find transfer function between sigout and sigin signals and then are the filter coefficients ...

I have been told in the university that the natural frequencies (also called $\textit{eigenfrequencies}$) are the poles of the transfer function, however, Matlab compute them as the modulus of the ...

I want to determine optimal initial states for a FIR/IIR filter to get rid of oscillations and shifts at the initial interval.
I've found that these initial states depend on filter type in the ...

I am currently studying two Butterworth and Chebyshev low-pass filters of order $n =3$ and $n=2$ respectively, whcih are in fact two prototypes to make a bandpass filter. The transfer function that I ...

- filter-design
- matlab
- image-processing
- lowpass-filter
- discrete-signals
- fft
- signal-analysis
- noise
- audio
- infinite-impulse-response
- fir
- convolution
- bandpass
- sampling
- frequency-spectrum
- fourier-transform
- python
- frequency-response
- digital-filters
- kalman-filters
- transfer-function
- adaptive-filters
- frequency
- filtering
- phase

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