**638 filter-design 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 ...

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

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 ...

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 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 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 ...

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 ...

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 ...

Following this paper , I am trying to make a least-squares algorithm in MATLAB, but for type I (I know about firls()).
...

I'm looking construct a stable pole-only filter where the feedback coefficients start with a block of zeroes, i.e.
\begin{align}
a_0 &= 1\\
a_i &= 0, \textrm{ for}\ 1 \le i \lt k\\
a_i &\...

I hope to design 1st order highpass filter from transfer function. In the example of 1st order lowpass filter, I first get the coefficients of numerator and denominator in the variable 'b' and 'a'. In ...

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 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 ...

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 want to test some code generated by this site. I selected Bessel LP 1st sample rate $600\textrm{ Hz}$ corner $8\textrm{ Hz}$ long $10\textrm{ bit}$. If I adjust the code for octave to be:
...

I am trying to figure out how to use the matlab gausswin function which constructs a Gaussian window for $N$ samples with a given standard deviation $\sigma$. The function is defined by
...

Trying to compute Biquad Coefficients for use in a codec chip TLV320AIC3100.
Bass Shelf(boost and attenuation) work as expected. Treble Shelf attenuation works as expected but Treble Shelf boost is ...

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 noticed, when I try to fit filter coefficients to a given complex transfer function with the output error method, implemented for example in the MATLAB function ...

Global interpolation or sinc interpolation is an ideal filter since its frequency response is a rect function.
The impulse response of this filter is the sinc function (same as the coefficients of the ...

I am trying to implement a FIR filter on FPGA and trying to have a solid understanding of the FIR filter tap delay and sampling frequency.
Does the “one tap” delay equal to “1/Fs (sampling frequency)”...

I'm working with signals with variable sampling rate (the time space between samples is not constant). I know the delay between samples but I don't wont to interpolate the signal.
Is there a way to ...

buzzer frequency range
recording of audio signals(environmental sounds with speech) with buzzer soundI have audio signals(environmental sounds with speech) with buzzer sound, I have to attenuate only ...

I come from Computer Science so please pardon for my possibly wrong terminology.
I need to design a filter which has coefficients
$$h_0, h_1, \ldots, h_n, \ldots \quad\text{such that}\quad h_0 > ...

I have code like below that applies a bandpass filter onto a signal. I am quite a noob at DSP and I want to understand what is going on behind the scenes before I proceed.
To do this, I want to know ...

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....

I am new to digital filters and trying to understand parameters behind them.
Scenario
I am trying to get DC value from signals sampled at 100kHz sampling rate. such high frequency is used due to ...

I'm trying to make a simple bandpass filter using Pole-Zero Placement method, which has unity gain at frequency w=0.37pi and 3 dB cutoff frequency wc=0.42pi (has a gain of 1/sqrt(2) at that frequency)....

The wavelet transform has a problem as it gives poor time resolution for low frequencies and poor frequency resolution for high frequencies according to uncertainty conditions.
This appears well ...

I am new to Signal Processing. From my understanding -- FIR/IIR just refer to the placement of poles and zeros in the z-domain helping us achieve convolution, if FIR and ??? in IIR.
Chebyshev and ...

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 ...

The axes are wrong in the spectrogram. I have an audio wav file, and I know the sampling rate. this is read into the audioData variable. The audio is 1 channel. selected window length I believe (can't ...

One of the main disadvantages of realizing digital filters using impulse invariance is aliasing. According to the Nyquist sampling criterion, in order for the frequency response of the digital filter ...

I have the strain signal of a lateral beam of a car measured at sampling rate 1,200Hz from data acquiring system. Even after using temperature compensation in strain gage, we are getting drift. So I ...

This question is from the book Discrete Time Signal Processing by Oppenheim and Schafer (3rd ed). I was going through the problems in the chapter concerned with filter design, and stumbled upon this ...

Have made difference equation of signal flow graph of FIR filter is it correct or not?

Please refer to the paper Splitting the Unit Delay - Tools for fractional delay filter design by Laakso, Valimaki et.al.
I am not able to visualize how the fractional delay is obtained by resampling ...

I am working on ECG signals, to eventually extract features in order to detect an arrhythmia and classify it.
I am using Discrete Wavelet Transform with biorthogonal wavelet bior6.8
During my research,...

first I need to mention I'm new to signal processing. here is the situation:
I have an acceleration time-series derived from an accelerometer
I wanted to imply a filtering method like high pass ...

I started to learn MATLAB and reading paper.i have faced difficulty where to start, how to start in the field of biomedical signal filtering (fNIR)?

Also draw its normalized frequency response.
What is the ROC?
This has to be done in z-plane so there must be two poles at $+i$ and $-i$ since they cannot be included in region of convergence. Is my ...

I read some example of design LPF which I didn't understand something.
The stop-band in that example is $\frac { 22 }{ 25 } $ from the over-all frequency, and I want to filter some white noise.
...

Let's say I have a signal, $x(t)$, defined as such,
$
x(t) =
\begin{cases}
0 &\text{if} \,\,\, t < -\alpha/2 \\
\frac 1\alpha t+\frac12 &\text{if} \,\,\, -\alpha/2 \leq t \leq \alpha/2 \...

I am looking at a tracking problem. It can be modelled similarly to the Extended Kalman Filter:
$$
\begin{array}{rcl}
\mathbf{x}_k &=& \mathbf{f}(\mathbf{x}_{k-1}, \mathbf{u}_k) + \mathbf{w}...

I'm learning DSP slowly and trying to wrap my head around some terminology:
Question 1: Suppose I have the following filter difference equation:
$$y[n] = 2 x[n] + 4 x[n-2] + 6 x[n-3] + 8 x[n-4]$$
...

For an FIR filter, with symmetrical tap values $h[N-1-n]=h[n]$,
why is the group delay $\frac{N-1}{2} T$ (where $N$ is the number of taps of the FIR filter and $T$ is the sampling time)?
Why is ...

Suppose we have a mixture audio signal, consisting of very short transients and some that are a bit longer (in ms, or s) along with other material, like harmonics, etc. But focusing only on transients,...

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

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