Questions tagged [digital-filters]

The tag has no usage guidance.

54 questions with no upvoted or accepted answers
Filter by
Sorted by
Tagged with
2
votes
0answers
167 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 ...
2
votes
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 ...
1
vote
0answers
125 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-...
1
vote
0answers
99 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 ...
1
vote
0answers
91 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 ...
1
vote
0answers
29 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 ...
1
vote
0answers
164 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 ...
1
vote
0answers
20 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 (...
1
vote
1answer
59 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 ...
1
vote
0answers
73 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: ...
1
vote
0answers
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 ...
1
vote
0answers
42 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 ...
1
vote
1answer
147 views

How do I initialize the state of a digital filter in Direct From 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 $n<0$?
1
vote
0answers
231 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. ...
1
vote
0answers
84 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 ...
1
vote
1answer
245 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 $(A^c = [a^c_m, ..., a^c_1, a^c_0])$. ...
1
vote
0answers
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 ...
1
vote
0answers
211 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} + \...
1
vote
0answers
238 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 ...
1
vote
0answers
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., $...
1
vote
0answers
344 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 ...
1
vote
0answers
149 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'...
0
votes
0answers
19 views

Do we need a transmit and a receive filter in a OFDM transceiver system?

can anyone specify in brief what are the filters that can be used as a transmit and receive filter in OFDM. Any material to start with will be very much appreciated. Thank you!
0
votes
0answers
24 views

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

I am still struggling to understand the filter updating process, the computation of the demodulated result and the computation of the error. Attached you can find the simulation I've done so far, the ...
0
votes
1answer
43 views

What does filtering do to a signal in time domain? Does the signal gets distorted?

When considering a transceiver system, if we do a certain filtering operation on the signal, will the BER of the system be affected? My point is, since filtering in frequency domain filters a portion ...
0
votes
0answers
57 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?
0
votes
0answers
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 ...
0
votes
1answer
23 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 ...
0
votes
0answers
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 ...
0
votes
0answers
14 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) ...
0
votes
0answers
7 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. ...
0
votes
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 ...
0
votes
0answers
29 views

Design of frequency sampled filter in matlab

I have written following matlab code for frequency sampled filter: ...
0
votes
1answer
58 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 ? ...
0
votes
1answer
146 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 ...
0
votes
1answer
164 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 ...
0
votes
1answer
125 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 ...
0
votes
0answers
54 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 ...
0
votes
1answer
77 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 ...
0
votes
0answers
34 views

Correctly interpret Butterworth filter frequencies

I am using the butter function from the signal package in R to filter my data. The data is Fourier-transformed (complex numbers) - and I want to filter out specific frequencies (I have a 256Hz ...
0
votes
0answers
53 views

How to find the filter coefficient of matched filter to detect the pulse of a signal?

I would to understand how to find the coefficient of a matched filter. Let's consider we have a signal and the signal is buried in the following noise. we want to find the coefficient of a matched ...
0
votes
0answers
39 views

IFFT, negative frequencies and noncausal impulse

When using the IFFT to compute an approximation to the IDTFT, frequency range is 0 to 2$\pi$ versus $-\pi$ to $\pi$. This produces a different impulse sequence(causal I think). How do you calculate ...
0
votes
0answers
35 views

What is a simple smoothing filter for signals sampled at non-uniform times?

An answer: A simple smoothing filter for signals that has a DC gain of one, is a simple RC low pass filter. This filter has a exponentially rising step response, $$O(t) = 1 - e^{\frac{-t}{\tau}},$$ ...
0
votes
1answer
144 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 ...
0
votes
0answers
149 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.
0
votes
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 ...
0
votes
0answers
53 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 ...
0
votes
0answers
21 views

“Necessary condition for lpf to lpf transformation is that a magnitude response must keep its original values at $\omega=0$ and $\omega=\pi$”?

"Necessary condition for lpf to lpf transformation is that a magnitude response must keep its original values at $\omega=0$ and $\omega=\pi$"? Why is it so? But does that not essentially read that ...
0
votes
0answers
72 views

Digital filter: selection

I am going through digital filter design. But I really get confused with so many Digital filters : IIR FIR Moving Average Linear phase filter Allpass filter Comb filter Analog filter: ...
0
votes
1answer
280 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 ...