Questions tagged [digital-filters]

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1answer
551 views

How can I get the continuous-time transfer function coefficients (or poles and zeros) from the corresponding discrete-time TF and vice versa?

Let's say I have a continuous time transfer function which I know its numerator coefficients $(B^c = [b^c_m, ..., b^c_1, b^c_0])$ and denominator coefficients $(A^c = [a^c_m, ..., a^c_1, a^c_0])$. ...
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1answer
42 views

Filter model of passive attenuation of headphones

It is possible to model passive attenuation of headphones as a Matlab/Python filter ? For example I would like to find digital filter which will be simulating the way of the passive attentuation of ...
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2answers
64 views

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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2answers
26k views

Transfer function of second order notch filter

Can anyone mention the transfer function of second order notch filter to remove the line frequency of 50 Hz, in terms of frequency and sampling rate. Just like for Low pass Butterworth filter as, $$ ...
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1answer
44 views

What is the duration of the transient response in FIR filters

I have read that for FIR filters, the duration of the transient response is equal to the order of the filter. But, for linear phase FIR filters constructed with the windowing method, the transient is ...
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2answers
279 views

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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1answer
64 views

GNU Radio Low Pass Filter doesn't work as expected with different transition widths

I'm new to GNU Radio and signal processing. I don't know if the following is a simple question, but I have been scratching my head about it. I created a simple flow graph with a random source (0-3) ...
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1answer
107 views

How can I find the transform function, amplitude and frequency response of a digital filter in python?

I have applied a laplace filter mask to an image and now I want to find the amplitude and freqency response of a laplacian filter: [[1,1,1], [1,-8,1], [1,1,1] ]. I know I need to first find the ...
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1answer
26 views

Modelling and testing fixed point IIR filter with freeware tools

I understand from this other question that prior to implementing a fixed point IIR filter on a particular microcontroller target it's best to create a model using a DSP tool and compare the output to ...
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2answers
45 views

Issue Understanding Time Delay of a Digital Filter

I am new to the world of digital filters and am educating myself with the book Introduction to Digital Filters by J.O Smith III,. The author derives the frequency response of a very simplistic filter: ...
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2answers
117 views

Implementing a Butterworth Filter Manually in C/C++ via Second Order Sections

Short background: I want to implement a lowpass butterworth filter in C/C++. The end goal is to use this in a low-latency Python program, for which of course ...
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1answer
71 views

Phase lag changes cross spectrum

I have a same noise passing through two different ARMA filter. The autospectrum of first filter is given by S11=G1*G1*S_in where S_in is input spectrum created by set of random numbers. Similarly the ...
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2answers
3k views

highpass filter design in Matlab

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 ...
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0answers
17 views

Pressure signal

I have got an raw file which has pressure and time and temperature format and after coding I have got the plots but my main task is to analyse the data and understanding the influence of Bessel and ...
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1answer
46 views

What is the warm-up period of a Butterworth filter?

I am not sure if I use the correct term which may be the reason why I was not able to find the answer, but what I mean by the warm-up period is the number of samples required for the filter to produce ...
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1answer
19 views

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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1answer
200 views

Help in designing filter for DUC (Digital Up-Converter) and DDC (Digital Down-Converter) using the built-in DUC and DDC blocks in MATLAB Simulink

Can anyone please help me in simulating a simple BPSK transceiver system that correctly receives a signal through a DUC and DDC? I have attached the screenshot of my Simulink system along with ...
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0answers
73 views

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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1answer
48 views

Digtial FIR Impulse response & transfer function

I am currently working the figure through below. as it is an FIR Filter i have worked out using convolution that the output is 4,2,4,6,0,0. i am trying to obtain the 'z' domain transfer function of ...
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1answer
414 views

Generating FIR filter in Web Audio API

I'm creating an HTML that applies a low-pass Windowed Sinc filter with a window size of 8192 to white, pink, or Brownian noise. There are nine radio buttons, eight of which set a cutoff frequency and ...
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5answers
3k views

What's the Q factor of a digital filter's pole?

For the analog S-plane, the Q factor depends on the angle of the complex pole $p$ from the horizontal axis, $\theta = \arg p$: damping factor $\zeta = \frac{1}{2Q}$, and $\zeta = \cos(\theta)$, and $\...
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1answer
211 views

Filter: IIR Filter

How IIR Filter coefficients formula derived? for LPF, HPF, BPF, BSF there are coefficients formula for a0, a1, a2, b0, b1. How it came? I would like to know the derivation for all those coeffs w.r.t ...
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1answer
202 views

How to predict the cramped frequency of a digital filter based on an analogue frequency?

I have a second order analogue high pass transfer function (unity gain at infinity). It's magnitude response hits the -20 decibel line at a frequency of 5706 Hz (the corner frequency is 18000 and ...
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1answer
50 views

Digital lowpass filter causing aliasing

I'm writing a biquad filter program in C++, using the RBJ Cookbook as the source for the formulas. https://www.w3.org/TR/audio-eq-cookbook/ The lowpass filter that I've made seems to half work. The ...
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4answers
108 views

Why do FIR linear interpolators have linear phase?

I wonder why linear interpolation is a linear-phase FIR filter. In fact, I am reading Dr. Rice's great book "Digital Communications; A Discrete-Time Approach," and he mentioned on page 466 ...
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1answer
34 views

Load cell noise filteration

I'm currently working in a project as apart of a student team. Setup A platform is attached (placed over and screwed) two 1 Kg load cells in order to measure the weight of the system. A vibration ...
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1answer
778 views

Why is interpolation a time varying system

I was reading about interpolation (Interpolation and Decimation of Digital Signals - A tutorial Review, Ronald E. Crochiere) and found that Interpolation filter is a time varying system. Can someone ...
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1answer
54 views

Composing digital filters

I'm interested in composing filters for realtime audio processing on an microcontroller (MCU). Ideal frequency response is unity as a default, with deviations up and down at specific freq-domain ...
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1answer
129 views

Is the sampling period arbitrary for a bilinear transform, and why?

I have been given the specifications for a digital highpass filter (stopband, passband, stopband attenuation and maximum passband ripple). I am expected to design a prototype lowpass filter in the ...
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3answers
260 views

Can I set a constraint on the first tap of an FIR filter such that its inverse is stable?

Let's say I have the following FIR filter $h[n]$, so the output $y[n]$ for an input $x[n]$ is $$ y[n] = \sum_{k=0}^{m-1}x[n-k]h[k] $$ The inverse of this filter is given by the IIR difference ...
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1answer
79 views

How to identify low pass, bandpass, band reject, high pass filter etc from given $|H(e^{j\omega})|$?

1) How is this a low pass filter? $0.99 \leq |H(e^{j\omega})|\leq1.01 {\rm\ for\ } 0\leq|\omega|\leq0.19\pi$ $|H(e^{j\omega}\;)|\leq0.01{\rm\ for\ } 0.21\pi\leq|\omega|\leq\pi$ 2) What kind of filter ...
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1answer
342 views

Understanding and Correcting phase shift in General Linear Phase FIR filters

I have been learning about General Linear Phase Filters and the four standard FIR implementations. I was wondering in the case $\beta = \frac{\pi}{2}$ or $\beta = \frac{3\pi}{2}$ if there was a way ...
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2answers
98 views

Pulse-shaping FIR filter

I understand that pulse-shaping is performed to limit the bandwidth of the transmitted signal, but I would like to know how pulse-shaping can be implemented using an FIR filter.
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1answer
50 views

Expression of output for discrete time causal filters

I have a question about the expression of the output of a discrete time filter described by its impulse response $h(k)$. Looking at the defintion of a discrete filter with input $u(k)$ and output $y(k)...
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4answers
118 views

Filtering Angular Measurements

I am trying to apply a filter to reduce noise in angular measurements. Since the data comes with noise, the filter needs to be able to handle measurements that are not smoothly progressing; for ...
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1answer
32 views

Phase in FIR filters [closed]

Does anyone have any idea what is Clear Phase technology ? How is it implemented ? And which audio channels whether left ,right center , sub woofer should have this technology implemented and why? ...
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1answer
185 views

CIC Filter implementation for Normalized Correlation

This is a follow-on to the original post ... I am trying to implement real-time correlation of two signals. I noticed the CIC Filter discussion here: How to find correlation/cross-correlation of two ...
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2answers
169 views

DSP Book with More Applications

The process of my DSP class used the Oppenheimer DSP book as text. Although I can easily understand the math in the book, I found it lack examples and applications, hence I still cannot build a ...
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1answer
68 views

How to design a filter that can filter out noise accurately, after setting the parameters of the filter using standard signal?

I meet a problem with designing a filter. I have two different instruments that could measure the same AC signal (usually ~200hz, always <1kHz), A and B. A can carry out signal measurement during ...
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1answer
413 views

What is the slope of a transfer function?

Lets say I have a Transfer function (H) plot as below. I need to find the SLOPE of the transfer function in dB/MHz. That is... i need a plot (db/MHz) vs frequency. How can i go about it ? ...
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0answers
27 views

Subband Tuning in Matlab

Suppose you have an i/q stream with a sample rate of 20 MHz. You want to bandpass filter a 1 MHz bandwidth (ie passband from 5 to 6 MHz). You then want to heterodyne that 1 MHz chunk. How would you do ...
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1answer
59 views

What is the transfer function of the filtfilt?

I designed a Parks McClellan filter to design a lowpass filter. First, I created the order of the filter with firpmord(). Then I designed the filter with firpm() and created the impulse response. I ...
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1answer
135 views

How do I know what is causing the noise in my signal

I have a noisy audio file, available here (It is safe to download if you wish to) https://ufile.io/pstrt Now, I inspect it in matlab: And then i plot it on normalised frequency: Now I need to ...
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1answer
185 views

How to implement a series of second-order, digital state-variable filters in MATLAB?

How to implement a series of second-order, digital state-variable filters, using the high-pass output from each section to effect the low-pass to high-pass transformation in MATLAB? Hello, I am trying ...
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1answer
52 views

Why an does an integrate-and-dump stage in a CIC filter provide the same functionality as the integrator and comb in series?

I have a hard time understanding why the combination of integrator and comb (integrate-and-dump) is the same as a separate integrator and comb in series like in this picture: Please explain why these ...
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2answers
180 views

Can digital filters actually separate signals?

I've come to realize what appears to be a hard limit in convolution theorem: to avoid time-domain aliasing, we must pad the signal/filter, but the padding distorts the spectrum. Consider a ...
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1answer
42 views

Fourier transform and Zero Order Hold

I was asked to explain why a zero order hold (ZOH) filter is not a good choice for signal reconstruction. To answer this question, I was thinking about studying the Fourier transform of the ...
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2answers
41 views

Amplitude and phase compensation of analog filter

My analog filter perform much needed antialiasing and input protection. Unfortunately, the filter adds undesired changes in the gain/phase across frequency. For example, here is an example of ...
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1answer
156 views

Moving average filter implementation in the floating point [closed]

I have following C++ implementation of the moving average digital filter. ...
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1answer
49 views

Prove the following property of All-Pass Filters

If A(z) is an all-pass filter given by $$A(z)=\frac{z^{-1}-d_1}{1-d_1 z^{-1}}$$ where $d_1$ is a real coefficient Then prove that $$ |A(z)|\left\{ \begin{aligned} &<1 &|z|>1 \\ &=1 &...

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