Questions tagged [filter-design]
Fiter design is the process of designing a filter.
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How to get the time domain of a pulse after an amplitude mask or a phase mask has been applied?
I am a chemist, and I am studying ultrafast spectroscopy. In these spectrscopies, usually a short (femtosecond) pulse of light is used to probe chemical reactions. Now, one of the topics in our ...
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Constrained Least Squares Filter Design
I would like to design a complex FIR filter, $h$, for a known signal that produces a desired output: $d$ = $s*h$ (where $s$ is my signal and $d$ is the desired filter output). Let $S$ be the ...
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Confusion understanding filter visualization tool output in MATLAB
My MATLAB code is
d = fdesign.bandpass(.3, .4, .6, .7, 80, .5, 60);
Hd = design(d, 'ellip');
fvtool(Hd)
when i run my script , i see two graphs on output ...
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Windowed Sinc vs Parks–McClellan filter design [duplicate]
In the context of FIR filters, what are the tradeoffs between using a low pass filter designed using a windowed sinc versus an optimally iterative algorithm such as Parks-McClellan? When would you use ...
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Single Kalman Filter: Resistance based on measurment of current and Voltage [closed]
Given measurement of voltage and current, which contain gaussian noise with known variance and a constant unknown offset $a_0$, I have attempted to estimate the resistance $R=\frac{V}{I}$. The ...
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How can I solve this [closed]
a) Consider the signal waveform g(t)= g1(t) + g2(t)
where
g1(t)= C1cos(w1t+θ1)
g2(t)= C2cos(w2t+θ2)
and w1≠w2
i. Determine the suitable signal size measure for g(t)
ii. Find the numerical value of (i) ...
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Designing of passband and stopband frequencies
good evening guys, let us consider following image
we can see from the picture designing of lowpass filter with given passband region and stopband region, only one thing what i am interesting is ...
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How to generate PSHC carrier used in vocoders
Vocoders are often used to acoustically simulate the information transmitted
by a cochlear implant. The different types of vocoders used in cochlear implants include noise-vocoding, sine-vocoding and ...
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Having a hard time understanding FIR filter transverse structures
I'm currently learning about FIR filtering from one of my modules and went through topics such as transverse implementation(or structures since I could not find transverse implementation on the net) ...
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Filter coefficients conversion from Matlab for FPGA
Filter in Maltab gives me a set of filter coefficients which I want to use in FPGA/VHDL design.
in Matlab I work with floating point, in FPGA I cant use it.
Will be it correct if I convert the ...
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Real parameters design and computation for a filter, modulator
I design in Gaussian filter for a modulator in MATLAB, which should be later implemented in FPGA.
I have an implemented design of the filter and the modulator as well and now I have to determine some ...
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What procedure to find the parameters of a given filter prototype to fit a desired frequency response?
To model the frequency response of a system, I am looking for a method to fit the response of cascaded bi-quad filters to the response of that system using an optimization algorithm. The $S$-domain ...
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A real-time (causal, non-symmetric) lowpass FIR filter with minimal passband ripple and phase
I am looking for a lowpass FIR filter that is causal (all samples used are current or past) and exhibits very low passband ripple and has passband phase very close to zero, which necessitates a filter ...
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Formula to compute FIR and IIR Filter coefficients
I am trying to understand filter design. I am trying to implement butterworth filter which would have high,low and bandpass using the difference equation representation for FIR and IIR. Given the low ...
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Particle Filter estimation: sinusoidal signal
I am designing a particle filter for the extraction of a sinusoidal wave with known frequency and sampling frequency from Gaussian noised biased measurements ($y=x+\text{bias}+\text{Noise}$)
My state ...
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Question in DSP signal flow graph
How to convert signal flow in Fig3 to Fig4? Is these two equivalent?
ZHANG Xiuli, ZHANG Ruihua, CHEN Weidong. Design of
digital parametric equalizer based on second-order function[C]//
IEEE ...
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Convert matlab line to scipy signal processing line
I am converting some legacy matlab code to Python and I'm struggling to find the right function in python from the legacy code.
The matlab filter is
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Quadrature Mirror Filter symmetry? How to understand this notation
The QMF symmetry of a two-channel filter bank is given by
$H_1(z) = H_0(-z)$
I don't understand why such a constraint is quadrature and mirror symmetry. Are $H_1$ and $H_0$ quadrature and mirror ...
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Best way convert transfer function to FIR filter without group delay if i have complex transfer function data
when i calculate acoustic transfer function, system identification using adaptive fir filtering is usually used and that have not tap/2 group delay maybe.
in simulation if i have complex number of ...
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How to implement cascaded biquad transfer functions in Python/SciPy
I am trying to implement in python/Scipy the biquad filter with the following S-domain transfer function:
$$
H(s)=\frac{w_0^{2N-1}(s+w_z)}{[s^2+(w_0/Q)s+w_0^2]^N}
$$
My problem here is that because of ...
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How do I get various peak values from a 2-D array or a matrix?
I'm working on 2D array analysis or an image analysis and get to know the different block of peak values marked "circle" in attached plot [e.g. 2*2 matrix] of the 2D array [as shown in Fig. ...
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How to convert real BPF to complex BPF
Suppose if I have a real BPF (band pass iir filter with co-efficients of numerator and denominator), can one get the complex BPF filter coefficients using these ?
Real BPF will be one sided and ...
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How design a filter to obtain a desired autocorrelation in output
I have a signal x(n) with a given autocorrelation function ACF, $R_x(\tau) \neq 0$ for $\tau=1, \cdots ,L$
My question is how can i design a filter (better with a few coefficients) to obtain a signal $...
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Comparison between convolution filter and spectral division filter?
Consider a signal $x=s+n$ where s is the desired signal and n is the noise we want to remove. If we know that noise is just low frequency we can remove noise by designing a low pass filter say $w$ and ...
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How to implement the direct form of FIR filter if in the transfer function there is no poles and power of z's are positive?
I confuse about the implementation of the FIR filter with no poles. If the transfer function is given in the form
$$ H(z) = 1+az^{-1}+bz^{-2} $$ direct form implementation is just delay operator ...
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overlap save method zero padding fft gibb's phenomenon
i'm doing overlap save method in frequency domain. i make hankel function to digital filter using frequency sampling method(to make arbitrary magnitude phase filter). and i do ifft and zero padding ...
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How to determine the transfer function of the second order FIR filter?
I need to find the transfer function of the second order FIR filter. I searched a lot in the standard textbooks, but I could not get the mathematical expressions and derivations.
I want that the zeros ...
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Designing an FIR filter with Gain = 1 on TI DSP processor
I'm trying to design an FIR filter with gain = 1 on DSP processor that does not support floating representation. I'm using MATLAB to design the filter and then transport the taps into my code. The ...
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I can't hear my sound after filtering
I am studying on a project which my supervisor gave me this semester. It has multiple steps. Basically, I have my own audio. I added AWGN to this sound using awgn() ...
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SOS-matrices' order does not correspond to given parameter when designing bandpass with scipy.signal.butter
I do not understand why this scipy operation:
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Range of difference equation coefficients in practical FIR/IIR filters
In the design of practical FIR and IIR filters using difference equations, what is the range of coefficients that is employed for practical filters. How many orders might be used? What might the range ...
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Arbitrary shape frequency response filter [duplicate]
i want to make arbitrary shape digital filter from equation. that have own magnitude and phase response.
the equation is bessel function. the frequency response is below zero phase and real value
and ...
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frequency sampling method) why the coefficient is not symmetric?
Here is matlab code
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...
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question about time delay of practical filter design with sampling frequency
most system have 48khz and if i have 40tap low pass fir filter that cutoff freuqency is 300hz, filter delay is 1/48000*20 = 0.0004s
but the frequency resolution is very sparse and i think if the ...
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Minimal group delay filter to remove harmonics but preserve fundamental
I have a signal that contains energy at a fundamental frequency (60 Hz) plus significant energy at harmonics (2nd, 3rd etc up to 15th). The signal is sampled by an ADC at 1920 Hz. I want to construct ...
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High Dynamic Range FIR Filters
Related to this question on using the Windowing method for FIR filter design versus the optimized algorithms such as least squares (firls in MATLAB, Octave and ...
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Estimating change in RMS of noise given FIR filter coefficients
Filtering noise leads to a change in the overall RMS of the noise as demonstrated by the following code-block. which uses scipy's firwin2 to calculate the taps. The ...
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Implementing HPF using frequency sampling method
I'm on it for a few hours trying to tweak it in all sort of ways but the output comes out scrambled and I can't understand why.
I am trying to implement an HPF with a stopband frequency of 500Hz and ...
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Create filter without regard to phase response (amplitude only)
Why I only care about amplitude response
I'm working on equalizing a DAC where I only care about the flatness of an AWGN spectrum.
Since AWGN has random phase for any frequency bin, I do not care ...
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FIR filter | how can I change the axis to unnormalized?
I don't understand why my output graphs and not showing the real frequencies, tried to change it in a number of ways with no luck. so far I'm stuck.
This is what I have done so far while implementing ...
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How to derive a 2d jinc highpass filter from a lowpass filter?
The following code define a jinc lowpass filter, how to derive a corresponding highpass filter? Is it possible?
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Find the output signal in a Low Pass Filter for a given cutoff frequency range
I have a scenario where 𝑥(𝑡)=2⋅sin150𝜋t +sin250𝜋t and g(t)=𝑥(𝑡)sin250𝜋 . The signal g(t) is passed as input through an ideal lowpass filter with cutoff frequency(fc)= 300𝜋 and passband gain= 3....
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Signal filtering with physical limitations and estimated STD
I have a prediction mechanism that predicts some signal value. Generally the prediction look like this:
The are some ranges of systematic error and I cannot do anything with this. But in many points ...
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Filter for ISI Channel
I know that OFDM has many advantages in removing ISI. But I think there is one more way of removing ISI(equalizing or inverting technique).
Syppose there is a channel $H(\omega)$ and we have our ...
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implement bandpass filter on complex valued signal? (C++)
I want to filter out complex valued samples - Bandpass filter.
I have 100 complex samples and I wish to filter out a range of frequencies from it, as you can see in the result of the FFT, there are 3 ...
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Frequency Sampling Filters (FSFs) NOT the Frequency Sampling Method
A class of digital filters called Frequency Sampling Filters (FSF) has been mentioned in a side note by Richard Lyons in another related question comparing IIR digital filter design using mapping ...
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Mapping of Classic Filters for Digital Filter Design
Of the four classic analog filter types: Butterworth, Chebyshev, Elliptic and Bessel- are any of these relegated to obsolescence for purposes of digital filter design in comparison to optimized ...
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How to design a digital Butterworth bandpass filter?
I am looking into designing a Bandpass Butterworth filter in python, but, I was not sure I am designing my filter correctly. What I have are the following:
High cutoff frequency = 200Hz
Low cutoff ...
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how would gradient descent work in optimizing a practical frequency response of a sinc function?
The gradient descent algorithm should work by minimizing the ripples in the stop and pass band, then increase the slope of the transition band. So basically the parameters of interest will be passband,...
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