Questions tagged [digital-filters]

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172 views

How can I oversample the received signal by double of sampling time and have two signals

I have a question, how can I make the sampling in Matlab for the received the signal by half of time duration of symbol (Ts), and have two copies of signal (that's the process of fractional sampling ...
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228 views

Why does noise prevent a (kalman) filter from diverging?

I'm using a filter (not exactly kalman) of the following form to estimate angles by fusing gyroscope with accelerometer and gyro with magnetometer: $(1)\quad \hat{\theta}_k = \hat{\theta}^-_{g,k} + \...
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2answers
2k 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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28 views

Help on understanding Gardner's original paper for QPSK symbol timing recovery?

I am fairly new to DSP and RF receivers in general, and I was recommended to read Gardner's original paper for symbol timing error recovery for QPSK signals. I have a question regarding an equation ...
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0answers
132 views

Early-Late Symbol Synchronizer on Non-Triangular Pulses

I've been studying the basics of digital communications using the free book at desktopsdr.com. One of the examples is a 4-QAM demodulator which where symbol timing recovery is done with some Early-...
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140 views

Choosing Right Type of Filter for time series data

I am working on a temperature time-series data which is very noisy. I am trying to measure true low and ...
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122 views

GNU Radio - Bandwidth and Band Reject Filter

I am currently trying to generate a psk modulated carrier with gnu radio. Attach is an image of my flow graph. As can be seen I am generating a bit stream using the Random Source module and ...
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30 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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0answers
223 views

Determining window Size for Savitsky-Golay filter (Savgol_filter)

First, I am aware that an ideal optimal window task is really signal/task-specific, i.e. nature of signal/sampling rate/characteristics of interest etc. What I am interested in: "Is there an optimal ...
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21 views

Converting the equalizer coefficients from VSA at 2 sample/Symbol to 1.5 sample/Symbol

I was looking for some feedback on conversion of equalizer/filter coefficients at different sampler rate. Problem Statement: We have built a transmitter system whose output is at 1.5 sample/symbol (...
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1answer
67 views

bibliography about implementation of IIR filters

I need to implement a 6th order IIR filter. The filter I need to implement is the A-weighting. I've simply used the bilinear function of Matlab with an fs of 20Khz. I need to implement this over the ...
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0answers
75 views

Why do additional harmonics arise after the filter bank?

I'm trying to implement direct implementation of MDCT filter bank. So, some additional artefacts have appeared after all of the routines. The spectrum of the initial signal looks like: ...
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45 views

persist state of digitalFilter

Suppose I created a digitalFilter using designfilt (in matlab). I am interested in using it to filter windows (I dont have the whole data set offline). When I call filter() with this filter and a ...
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43 views

Frequency warping while creating filter coefficients

If I have an audio signal at 16kHz with frequencies band-limited to 8kHz. I want to low-pass filter this signal such that fc=4000Hz. Meaning when I reconstruct the signal I do not want any content ...
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1answer
168 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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239 views

signal interpolation (upsampling) by factor 2

The task is to interpolate the signal by increasing the sampling rate by factor 2. Matlab has a dsp.FIRHalfbandInterpolator function. Let's try using it. ...
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102 views

Band Edge Filters

I am interested in using Band Edge Filters to do carrier frequency recovery but this is kind of secondary to my question which is: How do you build an FIR filter which isn't symmetric about the ...
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1answer
295 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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90 views

Digital or analogue filtering?

I'm trying to self learn the art of signal processing whilst moving through my third year pure maths degree. Sorry if my terminology is incorrect however I hope I am understandable! I am looking at ...
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245 views

How use FIR filter for simulating an integral implemented using the trapezoidal integration?

Suppose I have this equation $$ \phi = \frac{60}{T^5} \int_0^T \left( T^2 - 6T \tau + 6\tau^2\right) y(t-\tau) d\tau - \frac{30 \alpha}{T^5} \int(T-\tau)^2\tau^2 u(t-\tau) d\tau $$ and I want to use ...
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50 views

Is there any audio DSP without biquad constraint?

I found a codec with audio DSP, ADAU1772. It supports 32 biquads, that is $32\cdot5=160$ parameters. I need to calculate filter coefficients on-line. The result is in transfer function form, i.e., $...
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346 views

Time-domain filtering to match a target time-varying frequency response

I would like to implement a filter in the time domain based on a desired frequency response that changes over time. The initial goal is a bandpass filter with a time-varying center frequency, where ...
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152 views

Basic “filtering” in digital communications

I am completely stuck on solving some problems for my digital communications class. Junior level undergrad class, so fairly basic. I have been searching around all day for related articles etc but can'...
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25 views

27 points median filter

I trying to build 27 points long median filter on FPGA using Verilog. The filter is 1-D (used to clean noise from a spectrum - I want to clean some random peaks). I already built an 9 points median ...
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35 views

IIR filter relaxation time computation

Let's define the relaxation time of a filter as the delay needed for the impulse response of a filter to reach and remain under an amplitude threshold. Is there a closed-form solution to the ...
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1answer
25 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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9 views

chromatic and polarization mode dispersion compensation in electrical dmain

I am currently working on adaptive compensation of chromatic and polarization dispersions in the electric domain using LMS (least squares method) and CMA (constant module algorithm) algorithms. I am ...
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0answers
26 views

How to use background noise information to de-noise a (non-stationary) signal

I performed some measurements, which unfortunately include some noise. The measured signals are non-stationary, because the phenomena I am capturing is a transient. Before the transient, I recorded a ...
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10 views

Estimating velocities from IMU Acceleration

I hope someone can help me to figure out the best way to solve my problem I'm trying to drive a Cuboid on V-rep simulator with velocities What I'm receiving is accelerations from an IMU (MPU9250 acc+...
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2answers
40 views

Q format doubts

Is my understanding correct w.r.t Q format. signed Q15.1 format means, 15 bits for fractional and 1 for integer bits SO max range is - 2^15 / 2^15 to + (2 ^ 15 - 1) / 2^15 signed Q7.0 means, 7 bits ...
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22 views

Remove known sound from signal

I'm currently getting FFT values from the music playing from a user's PC. However, my application also generates noise which interferes with the FFT data as my app's sound is also coming out of the ...
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1answer
30 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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66 views

First order IIR high pass filter

The cutoff frequency is 0.05 Hz. The DC offset of the signal is 5mv. How to determine when the signal will stabilise to the baseline?
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43 views

Can someone explain or elaborate on the channel estimation and noise variance estimation algorithms used in HF receiver?

For different type of communication, there is different type of algorithms to acquire CSI. For instance 4G (LTE) network have SRS or DMRS to estimate the channel based on pilots. I am designing a ...
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1answer
41 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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29 views

Hydrophone signal calibration by frequencies

First of all excuse me if anything I post is completely or partially wrong (or poorly explained), I am new in signal processing and still learning a lot from this community, I will do my best to ...
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0answers
25 views

How to compute the DFE based MMSE equalizer FF and FB coefficients in a non adaptative way?

I seek your assistance, I am reading the Proakis 5th edition and I came across a paragraph explaining the tuning of DFE filters, i.e, how to compute the FF(FeedForward) and FB (Feedback Filter) ...
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20 views

HF ionospheric Channel Model Example in matlab : Watterson model

In the example file in matlab HFIonosphericChannelModelExample.m, the watterson model is simulated. In this model the channel is represented as one tap (for the sake of simplification) delay channel. ...
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1answer
17 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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29 views

Design of frequency sampled filter in matlab

I have written following matlab code for frequency sampled filter: ...
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1answer
93 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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1answer
157 views

DC Spur in DAC output

I have a design in which a signal from ADC is filtered and then played by DAC. In FPGA the real signal is converted to quadrature using DDS with 70 MHz frequency. After filtering, the baseband signal ...
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1answer
202 views

How to perform spectral inversion in the frequency domain to convert a low-pass filter into a high-pass filter?

To convert a linear-phase FIR low-pass filter into a high-pass filter with the same cut-off frequency, we can invert the sign of the low-pass filter's impulse response $h(n)$ and then add one to the ...
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1answer
128 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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61 views

What is the threshold to prefer polyphase channelizer over FIR filtering?

A wideband signal need to be processed to extract several narrowband channels from it. There are two straightforward approaches to do so: For each narrowband channel shift frequency and filter it ...
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1answer
104 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 ...
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1answer
185 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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230 views

Please explain Multiplication property in Z transform?

I am having problem visualizing contour any example will be great help. As far as as where I need it I was trying to find Z transform of a contracted signal by first multiplication by impulse train.
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1answer
86 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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54 views

Can we sample the signal with any period as long as the Nyquist rate is respected

I have a question regarding the sampling rate. Suppose that we transmitted a signal $m(t)$ then at receiver side, we received the continuous signal $x(t)$, which is time varying and sampled with a ...