Questions tagged [digital-filters]

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55 views

Filtering a square signal with a median filtering to preserve the edges

If needed, you can find my first post for this problem here. I am trying to clean the following signal: As proposed in the comment, I tried to use this post proposing 2 methods: median filtering and ...
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0answers
26 views

Effect of time duration in bandpass filtering

I am new in DSP, but I am trying to process EEG data to decompose into the bands, theta, alpha, and others. The raw data is processed: Remove the DC offset by subtracting the average value from the ...
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181 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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250 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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1answer
64 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
90 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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0answers
50 views

The size of an FIR filter for PDM-PCM conversion

I have been trying to understand how the conversion from PDM to PCM happens so that I could interface with a microphone. I have been doing some research online and I found that the most common way to ...
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38 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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138 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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203 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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192 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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1answer
44 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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393 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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22 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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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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55 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
214 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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143 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
362 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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267 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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352 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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157 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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19 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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13 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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53 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 ...
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1answer
36 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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21 views

Filter using frequency and magnitude

I'm used to filtering using FIR filters with a cutoff and dB attenuations etc. The issue I have is that I have pulses in my signal which contain similar frequencies as the noise. I'd like to totally ...
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27 views

Interpolationg phase and magnitudes, Transformation function

I am trying to filter signal x(n), n = 150. I made a filter with few frequency points on the x axis , [-11., -9., -3., -2., -1., 1., 2., 3., 9., 11.]) ...
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22 views

cascade realisation from direct realisation

I am trying to convert filter realisation from Direct form 1 to cascade using function provided from Digital signal processing by Proakis. I designed lowpass IIR filter and extracted numerator and ...
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24 views

Fixed Point, Time Domain and Freq Domain Filter Difference

I am trying to compare my IIR filter in time domain and freq domain, but i am getting different results which i dont understand why ?. So please help me out in understanding the issue. My code is : <...
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28 views

Remove noise from distance calculated from accelerometer data

I got distance calculated from accelerometer raw datas using double integral accelerometer.I attached the image below.But it shows noise with it.Please help me to remove this noise.
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30 views

Increase signal to noise ratio

I work with the K-LC6 microwave module. Previously, when the microwave module was assembled on discrete elements, the signal-to-noise ratio was 5. Now, with the transition to specialized microcircuits,...
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22 views

Threshold Estimation from a Linear FM Pulse Position Modulated signal

I have a signal chain to detect multi channel LFM (Linear Fm - Hann window being applied as shaping filter for the pulses) pulses as shown in picture below. (Numbers in the picture is in Hz., for ...
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30 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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42 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
41 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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12 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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30 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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11 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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25 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
40 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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78 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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44 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
66 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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33 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) ...