Questions tagged [digital-filters]

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1answer
38 views

Take FFT of audio represented in PDM

I have a PDM data input from the MEMS microphone on the Pynq-Z1 development board. My eventual goal is to take the FFT of the audio samples and display the frequency bins on some LED. I am struggling ...
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4answers
68 views

What is the correct gain of an RRC Filter?

Breaking my brain all morning with this reading previous questions and googling ... I have made an RRC filter from the equation on wikipedia. It works fine and I compared it to commpy library in ...
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5answers
2k views

Why do digital filters work?

So I just started reading on FIR and IIR filters and am amazed by how "simple" the theory seems, so far. But what confuses me is, why does filtering work by creating a weighted sum of previous ...
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1answer
147 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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1answer
78 views

Sub-Band Extraction Techniques for Complex (I/Q) Signal

There is an application which processes Complex I/Q signal of 100 MHz bandwidth and it is required to extract independent sub-bands from this bandwidth at different center frequencies and bandwidths. ...
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1answer
44 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
53 views

Setting up RRC filter and useful equations

I have looked at previous answers but cant seem to get everything I need in one place to help me and my textbook doesnt explain this in practical terms. Signal Parameters: $F_s = 44100$ $R_s = 300$ $...
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0answers
55 views

Wave Digital Filter Diode Equation?

I'm trying to understand how diode circuits are implemented in wave digital filters, particularly for clippers. The research papers and other sources I've looked at use the equation $$I(V) = 2I_s \...
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1answer
222 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
39 views

Help me with the impulse response please [closed]

Using the separability property where appropriate, write the minimum total number of arithmetic operations (number of additions and multiplications summed together) per sample needed to filter a 2D ...
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1answer
114 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
87 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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0answers
46 views

How to break a second-order filter into two first-order filter

Let's assume the transfer function of a continuous-domain filter consists of two poles and one zero: $H(s) = \frac{k_c (s-\omega_{z_1})}{(s-\omega_{p_1})(s-\omega_{p_2})}$. Let's consider we do the bi-...
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1answer
248 views

How do I initialize the state of a digital filter in Direct Form II?

Suppose I have a digital filter implemented in Direct Form II. How do I initialize the state of the filter as if the input $x[n]$ had a fixed value $x_0$ for all $n<0$?
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6answers
1k views

Why can't a causal digital filter have an infinitely sharp transition between the passband and the stopband?

In DSP book by Proakis and as well as in this pdf, it is mentioned that practical causal digital filters cannot have an infinitely sharp transition from Pass-band to Stop-band. Why is it so? Can you ...
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0answers
58 views

The optimal estimator using FIR Wiener filter

The received signal $y(n)$ is defined as $$y(n) = h(n)*x(n) +v(n)$$ where $h(n)$ is unknown impulse response and $v(n)$ is a white Gaussian noise. The value of $x(n)$ for $n=1, ..., 32$ are given. The ...
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1answer
430 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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0answers
37 views

How to design IIR digital filters

Practical Infinite-Impulse-Response (IIR) filters are usually based upon analogue equivalents (Butterworth, Chebyshev, etc.), using a transformation known as the bilinear transformation which maps the ...
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2answers
36 views

Scipy butter filter - remove DC offset w/ high sample rate

I am having a hard time figuring out how to employ a high pass filter to remove the DC offset of my data signal with the "scipy butter" function because my sample rate is quite high. The ...
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1answer
139 views

Not able to reach minimum phase using Hilbert transform

My problem is pretty simple, I've designed a magnitude response and I would like to find the corresponding minimum phase filter. I'm using the code below and unless there is a bug my eyes don't want ...
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1answer
35 views

Transposed Direct Form II

In the book digital Signal Processing by Proakis 4th Edition, at page 593 a table is drawn about some second order modules for discrete time systems. In that table there Transposed Form II drawn But ...
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1answer
34 views

Echo Removal from PCM 16 bit arrays (time frames)?

I am keen to know the basics to remove echo knowing FarEnd Signal and NearEnd Signal. I have read research papers in which its commonly said: y(n) = (hT)x where h is the room impulse response and T ...
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1answer
49 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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0answers
27 views

filter single radio station with sdr [closed]

I want to write a program to filter a single radio station using SDR. Not for any application just a stepping stone to learn. I am going to use C so it is easier for me to see the steps and things ...
2
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1answer
56 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
58 views

Summation of two IIR filters

Actually my question is how to get the weighted average of two or three IIR filters given their transfer functions, which is nearly equal to this question. How to calculate the total transfer function ...
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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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1answer
181 views

Adaptive Band pass filter for extracting respiratory waveform from accelerometer data

I am recording data from an accelerometer attached to the chest (1000Hz). I need to extract the respiratory waveform. I tried an adaptive bandpass filter based on a dominant frequency in my signal ...
4
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2answers
113 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
128 views

Frequency conversion in IIR lowpass-to-lowpass transformation

We use $\ z^{-1} \xrightarrow{} \frac{z^{-1}-\alpha}{1-{\alpha}z^{-1}}$ and $ \alpha = \frac{\sin(\omega_c^{'}-\omega_c)/2}{\sin(\omega_c^{'}+\omega_c)/2}$ for a lowpass-to-lowpass frequency ...
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1answer
314 views

Weighting function for output error IIR filter design

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 ...
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0answers
42 views

FIR filter design using Kaiser window - passband ripple and stopband attenuation

I have been given the following parameters : lower cutoff frequency = $\pi/3$ lower cutoff freq = $2\pi/3$ beta = 5.6533 M = 51 Now I have to get magnitude and phase response of the filter using ...
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4answers
47 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
112 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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1answer
297 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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1answer
89 views

Digital Signal Processing Filters Problem

Hello I need some help with this question. $$๐‘ฆ[๐‘›] = ๐‘Ž_1๐‘ฅ[๐‘› โˆ’ ๐‘˜_1] + ๐‘Ž_2๐‘ฅ[๐‘›โˆ’ ๐‘˜_2] + ๐‘_1 ๐‘ฆ[๐‘›โˆ’ ๐‘˜_3]$$ Set $a_1, a_2$, and $b_1$ to any positive or negative real numbers of your choice. Set $...
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1answer
43 views

How to remove frequency offset with RRC as pulse shaping filter

I am working on a project with QPSK modulation format and a RRC filtering at the transmitter (TX) side. The channel introduces a constant dropper shift ($f_d$) to the received signal. To make the ...
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2answers
196 views

How Come the Low Pass Filter in Sobel Operator Isn't Normalized?

I am relatively new to the field of computer vision and I have just learnt about the sobel operator. The sobel operator in the x direction is a convolution of the finite difference kernel $[1,0,-1]$ ...
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1answer
21 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 ...
0
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1answer
48 views

Design of discrete time second order all pass filter

I want to design a second order digital all pass filter with the transfer function given by $$H(z)=\frac{a_2 +a_1 z^{-1}+z^{-2}}{1+a_1 z^{-1}+a_2 z^{-2}}$$ The input to this filter has frequencies ...
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0answers
21 views

interpreting the post filter features in a sensor data

I am new to signal processing and I am working on Filtring signals(using basic analog filters, FIR filters, and IRR filters), for Human activity recognition, using sensor data, I Extract the features ...
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2answers
112 views

High-pass filter with quantized (fixed-point) coefficients

I have a high pass IIR filter with numerator b = [0.997688673703782,-1.995377347407564,0.997688673703782]; and denominator ...
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1answer
38 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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0answers
14 views

How to make 2 finger rake receiver to counter intersymbol inferference?

I have to transmit Walsh sequence (spreading codes) modulated with 8 psk with a channel impaired by intersymbol interferences, fast fading . Suppose i have a stream of data for instance 1 0 1 1, 0 0 0 ...
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1answer
54 views

High pass filter not attenuating signal at start of input

I have implemented digital high pass filter with below specs: Sampling frequency = 5333 Hz Cutoff frequency = 10 Hz Order = 3 We are facing problem like below: As ...
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1answer
57 views

Matlab filter implementation

I have a filter response in the $z$-domain. $$H_0(z) = 2 + 6z^{โˆ’1} + z^{โˆ’2} + 5z^{โˆ’3} + z^{โˆ’5}$$ How do I implement this filter on MATLAB? I can do this if I have the coefficients but I do not know ...
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1answer
69 views

How to determine the type of a digital filter given its expression?

I'm trying to understand a variant of the filter presented in http://www.stockspotter.com/files/PredictiveIndicators.pdf in the "Code Listing 1.". Its expression is given as follow: $$y_n = ...
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2answers
237 views

A path to get the foundation of SP and DSP for a pure math

I have a Ph.D. in pure math (interested in Harmonic analysis and operator theory). I am looking forward some proper references to lead me get the foundation of discrete/signal processing more and ...
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0answers
55 views

Richardson-Lucy iterative deconvolution is producing erroneous deconvoluted images

I am working on producing deconvoluted image from degraded image where the process of degradation is known to be linear and space invariant. I have used Richardson-Lucy iterative deconvolution method ...
0
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1answer
56 views

Cascading first order digital filters in C++

In a related question a probable solution was given to build a first-order digital filter and then cascade three of them in order to turn white noise into pink. I have applied the C++ as follows but ...

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