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Module that I am using is KLD-7 radar Transceiver from RFbeam
I am new to FMCW radar systems and I recently started working on distance and velocity calculations from such a system. Although the module provides the built-in functionality to calculate distance and velocity, I want to replicate those measurements using t... |
I've tested out the Chamberlin digital state variable filter (DSVF) as a sine generator as per this well-known approach. At $f_0=686\text{Hz}$ tone and $f_s=48000\text{Hz}$ it sounds fine and the FFT shows exactly two peaks at -220Hz and 220Hz; at $f_0=687\text{Hz}$ it buzzes.
The Chamberlin filter looks as below. Se... |
I don't understand why the frequency spectrum of a trapezoidal signal looks like this. I was expecting either a $\mathrm{sinc}$ or a $\mathrm{sinc}^2$, but it seems to be a combination of both.
|
I have obtained the following z-domain transfer function:
$\frac{Y(z)}{U(z)}=\frac{3.3641×10^{-7}×z^{6}+1.5584×10^{-5}×z^{5}+6.7263×10^{-5}×z^{4}+
5.5016×10^{-5}×z^{3}+8.525×10^{-6}×z^{2}+1.2303×10^{-7}×z+9.7492×10^{-20}}{z^{7}-5.5998×z^{6}
+13.5548×z^{5}-18.3685×z^{4}+15.0393×z^3-7.4352×z^2+2.0541×z-0.2245}$
This is a... |
I have an audio recorded, recorded by a 2 mics array. I want to a Python code that get the audio as input and estimate the RT60 reverberation time. I do have a function that estimates RT60 from a impulse response (IR). See below.
In theory I could try to extract the IR from the recorded audio, but since it involves dec... |
I have an anisotropic image that is anisotropic both in terms of number of voxels and also in terms of voxel resolution.
1. Number of voxels: 256x512x96 (wxhxd).
2. Voxel resolution: 0.5x0.5x2.0 micrometer (wxhxd).
Now, to perform uniform Gaussian blurring, I need to choose different sigma values in each direction. ... |
I'm want to generate the IQ components of a FSK signal using numpy and then try to demodulate the resulting IQ samples by looking at the derivative of its phase in time.
This is the script I have:
import numpy as np
import matplotlib.pyplot as plt
from scipy import signal
fs = int(1.8e6) # Hz
rb = int(1e3) # bps
mess... |
I found an image hash library for python and this blog
In the block Getting Funky With pHash, the author writes:
Further reduce the DCT. This is the magic step. Set the 64 hash bits
to 0 or 1 depending on whether each of the 64 DCT values is above or
below the average value. The result doesn't tell us the actual low
f... |
A theoretical question in echo cancellation performance evaluation: $\mathrm{ERLE}$ = $\mathrm{E}$cho $\mathrm{R}$eturn $\mathrm{L}$oss $\mathrm{E}$nhancement, is defined as
$$
\mathrm{ERLE} = 10 \log_{10}\left(\frac{\mbox{power before EC}}{\mbox{power after EC}}\right)
$$
Modeling the echo canceller as $x = d + e$ whe... |
My question is:
Which type of diversity technique is used by the rake receivers?
In Bahai, et. al. "Multi-Carrier Digital Communications Theory and Applications of OFDM,"
Second Edition, 2004, on page 27, it is said that RAKE, OFDM, and Equalization, are all using frequency diversity, but here and there on the internet... |
In Boaz Porat's book about signal proccessing, at part 8 he mentions the example:
$$ H\left(z\right)=1-z^{-1}\Rightarrow H^{f}\left(\theta\right)=1-e^{-j\theta}=\left(e^{\frac{j\theta}{2}}-e^{-\frac{j\theta}{2}}\right)e^{-\frac{j\theta}{2}}= \\
2j\sin\left(\frac{\theta}{2}\right)e^{-\frac{j\theta}{2}}=2\sin\left(\frac{... |
I am working on a radar signal processing project using MATLAB. I am required to simulate range doppler maps for Linear Frequency-Modulation (LFM) targets (non-fluctuating) in noise at various SNRs (40dB to 0 dB) at the range doppler output. I am using the AWGN function to add noise to my received echoes.
My simulation... |
What is the best way to define a detection statistic for a time series event trigger generator, which computes the wavelet transform of some time series data? I want such quantity to be similar to a blind SNR, but based on wavelet coefficients.
|
I am trying to implement the discrete Fourier transform.Here is my code in MATLAB:
sym k
n = 1:4;
disp(n);
y = n-2>=0;
z = n-4>=0;
x = y-z;
X(k) = sum(x.*exp(-1i*2*pi*k.*n/4));
l = 1:4;
subplot(1,2,1);
stem(l,abs(X(l)));
subplot(1,2,2);
stem(l,angle(X(l)));
I get these results for magnitude and phase:
However when ... |
Let's say I have a noisy raw signal with 1GS/s sampling rate (That means the time resolution is 1 ns). Now I apply a FIR lowpass filter (window filter) of 25 MHz on it. The length of Window filter will be 40 ns in time domain. Since it is doing convolution in time domain, does it mean that I get one "non-correlated" po... |
In my previous question, I tried to implement the DFT. Now I want to compare it to Matlab's FFT
My input signal is:
x(n) = n-2>=0
Here are the results:
As you see, my DFT doesn't match Matlab's results.
This online calculator agrees with my snippet of code. What is going on?
|
i am currently struggling to find the Transfer Function of the following Block Diagram, since i have never done it for a non recursive System, could somebody please walk me through the process of it?
My first step was setting up the Difference Equation:
$$g(n) = a*s(n)+s(n)*(a+1)*z^{-1}+a*s(n)*z^{-2}$$
Now I went on t... |
I want to analyze the motion X(t) of a very flexible structure, which is excited/forced by unknown perturbations, in order to characterize this perturbations.
Hence, I plan to record the motion at one point of the structure thanks to a sensor, and then to analyze the results with a FFT to find the greatest contribution... |
Let say I perform an experiment and I record data at a given frequency sampling $F_s$ and then I perform FFT analysis on these results.
As I do not know the characteristics of the perturbation that I record before conducting the experiment, especially the maximum frequency that exists, I am not able to choose $F_s$ acc... |
For the following system:
We have:
$$r(t) = s(t) + Ks(t-\tau)$$
where $|K|< 1$, with the following impulse response $h(t)$ and frequency-response $H(f)$:
$$h(t) = \delta_0(t) + K\delta_0(t-\tau)$$
$$H(f) = 1 + Ke^{-j2\pi f\tau}$$
The following question is asked:
For a band-limited input signal $s(t)$ with cutoff freq... |
I am trying to implement a high pass digital filter in MATLAB.
High pass filter:
function [hd,n] = idl_highPass(wc,N)
if mod(N,2)==0
a = N/2;
else
a = (N-1)/2;
end
m = 0:1:N-1;
n = m-a+eps;
hd = -sin(wc.*n)./(pi.*n);
end
I use the freqz() function to find the DTFT(or frequency response)of the filter but I... |
I am trying to design a 2nd Order CIFB DSM ADC on Simulink based off the values I have extracted from Richard Schreier's Delta Sigma Modulation Toolbox. However, when I ran the model according to the values I used, it seems like there is not a sufficient amount of noise shaping in the system. Since this is second-order... |
I am working on a project in which I need to calculate the range of a target using an FMCW radar. Dependent on the target's range, I expect various beat frequencies to occur in the data from my ADC. I would like to calculate the frequencies so that I can predict the distance of my target. If I use the FFT directly on m... |
I have an incoming stream of time-series data which looks like in the below figure :-
I want to detect the end of each cycle in real-time (marked with red arrows) using C language. What would be the best mathematical way to do it?
Edit:- I have some liberty for delay here. I know I can't do it in a sample by sample ba... |
I read on Wikipedia MP3 that the MP3 standard uses a modified discrete cosine transform.
Why does it use a modified transform and not the original DCT like the JPEG standard? Are there reasons specific to sound signals compared to images?
Reading the MDCT article:
This overlapping, in addition to the energy-compaction... |
Is there a general formula I could use to calculate the ideal window for the window argument in the function scipy.signal.coherence, maybe taking into account sampling rate and number of time points?
For example, if I acquired data at a sampling rate of $1.5\texttt{Hz}$ and I have $100$ time points, what window size wo... |
I asked a question in SO regarding what kind data I should pass to the wasserstein_distance function from the scipy module.
The documentation says that the input data are "Values observed in the (empirical) distribution".
My data arrays range between -4 and 8:
x = np.array([0.12,-1.29,-3.23,-3.21,-0.13, 1.52, 4.45, 6.4... |
On the Wikipedia article about Discrete cosine transform it is said:
For strongly correlated Markov processes, the DCT can approach the compaction efficiency of the Karhunen-Loève transform (which is optimal in the decorrelation sense)
My question is: how can we show that the DCT approaches the efficiency of the Karh... |
I know that if you have a linear time invariant system defined by
$$ \dot{X} = AX+BU $$ $$Y = CX$$ by "Laplacing" the previous equations, you get the following transfer function in the Laplace domain :
$$Y(s) = C(A-sI)^{-1}BU(s) = H(s)U(s)$$ Let us focus on : $$H(s) = C(A-sI)^{-1}B$$
Indeed, many libraries allow to pa... |
I have a black box-like deterministic model $h(t)$, which provided with a real-valued input $x(t)$ returns a real-valued response $y(t)$. The model should produce little to no noise. My goal is to determine an input signal $x^*(t)$ that produces a specific output $y^*(t)$.
I'm trying to use deconvolution in the frequen... |
I have two pandas DataFrames, dfb and dfv, where dfb has a higher sampling rate than dfv. I want to downsample dfb to align it with dfv. However, I am aware that I need to apply a low-pass filter to avoid aliasing. Can you suggest any improvements to the following function and what is the best way to apply a low-pass f... |
I am working on a project which uses radar imaging. The radar imaging takes place at a relatively small distance from the radar (0.25m to 3m). The data is collected using an FMCW radar. The FFT of the collected data is taken to calculate the target range. I want to reduce the sidelobes of the calculated range asymmetri... |
I have non-uniformly sampled data and I'm trying to look at its Power Spectral Density (PSD).
For that, I looked at the Periodogram PSD estimate of an interpolated (with uniform sampling period) version of my data and compared it to the Lomb-Scargle PSD estimate of my original data.
I'm surprised to see that the resu... |
I would expect the FFT of a periodic pulse signal to look like a sinc function, like shown here. However, if I try to replicate that, what I get is a single peak at the signal frequency, and then another peak at 3 times the signal frequency. Shouldn't I see a sinc function, as a pulse contains a wide range of frequenci... |
I have a transfer function in z-plane with two poles and two zeros. I plotted the function with matlab
k = 0.15;
z = [0.8 -1]';
p = [(0.51+1i*0.68) (0.51-1i*0.68)]';
[b,a] = zp2tf(z,p,k);
What type of filter is that? The closer I can think of is bandpass. But it has a really non -smooth peak so bandwidth would be zer... |
I have been using FIR filters for float variables, can I know how can I pass complex float variables ( complex float buf[1024] ) to the filter equations? I mean pointers to the variables of type complex float needs to be passed. I am reading complex float samples from the RTL SDR dongle and here is what I use for float... |
If i am not mistaken, the two following hold :
a. In OFDM Conceptualization, the constellation mapping at Tx (or de-mapping at Rx) is done in Frequency domain
b. However the basic theory (Including the math) of constellation mapping, example PSK/QAM, is always in Time Domain. I have not found any explanation in the fre... |
I am trying to convolve the two discrete sequences
$$\left(\frac34\right)^nu(n-2)$$
and $$2^nu(-n-5)$$
where $u(n)$ is the unit step function.
Here is how I approached the problem ,using the convolution sum
$$\sum_{k=-\infty}^\infty x(k)h(n-k)$$
in this case ,
$$\sum_{k=-\infty}^\infty\left(\frac34\right)^ku(k-2)2^{n-k... |
I'm trying to compute the Discrete Cosine Transform via FFT in C, using a response to this question, but the recommended solution gives me wrong results.
Function DCT in matlab:
x = [0:1:7]
x = 0 1 2 3 4 5 6 7
dct(x)
ans = 9.8995, -6.4423, 0, -0.6735, 0, -0.2009, 0, -0.0507
Type 2 DCT usin... |
When calculating Foreshortened Area as in the image below, why are we using cosine(theta) to calculate the scaled length in the x-axis? I understand intuitively why no correction is needed in the y-axis, but can't understand exactly why cosine for the x-axis
|
I'm sampling from the functions $x(t)=\cos(2\pi f_0t)$ and $y(t)=\cos(2\pi f_0t^{1.005})$, for one second, where $f_0=300$ and my sampling rate is
$f_s=1000$. ($N=1000$).
A plot of the coherence between the signal samples, $x$ and $y$ is shown below, which indicates a maximum coherence of 0.74
bin $m=299$, which is wha... |
I'm working on a simple 1D FFT and iFFT for two days now but no matter what I do, the code fails. I'm using FFTW library and libsndfile on linux and my goal is to use FFT for filtering audio.
The code reads a wave file (input.wav) performs a global FT , modify frequencies, inverse FT and writes to file (output.wav). I ... |
Suppose I have a channel matrix $\mathbf{H}$ and I estimate it as $\mathbf{\hat{H}}$. How can I measure the accuracy of the estimate? Is using the $\mathtt{MSE}=\mathbb{E}\left\{\|\mathbf{H}-\mathbf{\hat{H}}\|^2\right\}$ a good measure metric?
|
I found about one of the BM3D implementation in Python through this post and tried to use the same, however, I don't see any difference after using the algorithm.
Here is a small snippet of my code,
import numpy as np
import matplotlib.pyplot as plt
import cv2
from numpy import pad
from math import log10, sqrt
from ski... |
I'm trying to understand how video compression works by looking at the HEVC encoder, and I was wondering how a compressed video looks like; I mean, in the encoder, the last step is entropy coding (CABAC), therefore, assuming we code a single frame:
Is that frame actually watchable or do we need to decode it first?
If ... |
I was doing some calculations in my LTI systems course and I stumbled in an interesting question I wasn't really sure how to answer so I'd appreciate any direction or solution you can give me:
I'm trying to determine whether there's an invertible system $g[n]$ that will satisfy the equation:
$$e^{j\omega_0 n} \star g[n... |
The question is pretty self explanatory. I want to know if it is possible to recover a signal originally injected into noisy data (coloured gaussian noise) after whitening the data with an AR minimum phase filter. I want to recover the phase.
|
I have been performing digital image upscaling of pixel art in one of my hobby projects, and the two simplest upscaling algorithms are point sampling or bilinear upscaling.
Point sampling exhibits tricky aliasing/pixel column/row duplication effects if the upscaled image size is not an integer multiple of the source im... |
I am trying to design a bandpass filter in MATLAB with following specs:
$$\texttt{f_s = 20000Hz}$$
$$\texttt{3dB freq = [392 464]Hz}$$
$$\texttt{filter order = 8}$$
When I tried to implement the filter using inbuilt function :
h = fdesign.bandpass('N,F3dB1,F3dB2', N, Fc1, Fc2, Fs);
Hd = design(h, 'butter');
I g... |
Every paper/article I've read on the subject of TDOA localization is concerned with some combination of 2+ stationary receivers (or, sometimes, 1 stationary + 1 moving). There are also some papers that employ combinations of 1 moving receiver and several stationary reflectors with modifications to the algorithm.
What I... |
everywhere says that phase is los but WHY?
it is that when we measure x rays we cannot measure phase?
is it that phase is lost because of such high scattering?
what is the REASON for the phase lose in x-ray crystallography and NOT in cryoEM
Thanks!
|
I have implemented the Continuous Wavelet Transform using the pycwt library(https://github.com/regeirk/pycwt/blob/master/pycwt/wavelet.py) and its inverse using Morlet wavelets, however, upon calculating the inverse, the signal generated is accurate but with a constant discrepancy.
The code to produce icwt on pycwt lib... |
Two signals have arrived with different phases. The sum signal is given by
$$x(t) = A\cos(2\pi f_ct) + B\cos(2\pi f_ct + \theta)$$
What is the complex envelope of $x(t)$? Need advice on how to get started.
What is the analytic signal $x_+(t)$? For this, can I not apply the $x(t) + \mathscr{H}\big\{x(t)\big\}$ to both $... |
I am new in this community. I hope the question I want to ask suits the people here.
Here, we are dealing with two-ring Positron Emission Tomography (PET) scanner (figure 1), also called Compton-PET, because the inner detector ring is optimized for Compton scattering of gamma rays. In PET, we inject radiopharmaceutical... |
I'm currently working on a project where I have to measure the latency of a given system.
Below you can see a exemplary image which shows the input pulse (green) and the response (blue).
In order to compute the latency I have to detect when the blue curve starts to rise.
I'm wondering what approach would be the most s... |
Wanted to know the feasibility and usefulness of implementing Zero-Phase Anti-Causal filters such as those mentioned at this link in modern embedded signal processing applications given the advancement of devices on which these applications are now running, resource wise it should not be difficult to implement.
I.e., s... |
I working on a project called " eulerian video magnification " and i want to apply FFT on video and get the frequency response from the video . I have gotten 4D array ( frames,width,high,RGBchannel). Which steps should i do ? I am doing the implementation on Labview but it will he helpfully to know just the algorithm t... |
How can I obtain the fluctuations of a timeseries at a specific scale using the ssqueezepy library for inverse continuous wavelet transform (ICWT)?
I have a minimum reproducible example that uses the squezepy library to perform the continuous wavelet transform (CWT) and estimate the ICWT. However, what I would like to ... |
I want to isolate the harmonic at 440Hz and I need to design a filter with zero gain at 440Hz.
I tried this with matlab:
bandpass(ym, [2*pi*1*440/Fm-0.005, 2*pi*1*440/Fm+0.005], 2*pi);
However when I plotted the DFT i noticed larges values in y - axis (which means gain I guess). I cannot find in the documentation of b... |
In technical terms, feedback occurs when the gain in the signal loop
reaches "unity" (0dB gain).
How to Eliminate Feedback
Peak-to-peak amplitude (abbreviated p–p) is the change between peak
(highest amplitude value) and trough (lowest amplitude value, which
can be negative).
Amplitude - Wikipedia
Does that mean th... |
I have 20kHz sampled dc current data of 3 second (i.e signal length of 60000). I want to use undecimated wavelet transform to extract statistical features from it. But i don't know how to choose best performing mother wavelet and decomposition level. Please anyone help me.
|
I need to design a short, symmetric low-pass FIR filter with a monotonic frequency response (no ripple). In other words, the differential of its amplitude response will be zero or less for all frequencies. How can I compute N coefficients such that I minimise stopband energy?
|
I am trying to implement an IIR filter followed by downsampler with the following specifications:
fs=18000,order=8,3db freq=[500 1000],downsampling factor=6.
First I have designed the filter using inbuilt function ,then downsampled by a factor of 6 and gave a sine wave of frequency 700Hz as input to the filter to p... |
I am struggling to calculate a mixer output noise, but I am stuck in the understanding how to calculate it formally, could someone please help with reasoning or give a reference? I assume that we use the device as up convertor. The goal is to understand the effect of LO and IF to the output, as far as I know that it sh... |
After designing a "least square optimal" FIR filter I wanted to find the group/phase delay, defined as: $P(\omega) \triangleq - \frac{\Theta(\omega)}{\omega}$ (Depending on the literature group and phase are used inconsistently).
In scipy I've done this as follows:
import numpy as np
from scipy import signal
import mat... |
I am currently building a Kalman filter to add positions derived from high-sample rate velocity data (from an IMU) to lower-sample rate GNSS data. Normally, I understand you would use acceleration data, however this is the only data I have available. I am looking for advice on whether I have built my Kalman filter corr... |
I'm very new to signal processing and attempting to implement a spectrogram generator to learn more about DSP in general. So far, I have built up an understanding of how I might do that.
Get input signal
Window the data (e.g. split up the samples into 1024 sample long chunks)
Apply a windowing function
Zero-pad the da... |
Suppose we have an linear time-invariant (LTI) system which acts on discrete signals. Suppose someone tells us the transfer function is: $$H(z) = \frac{1}{z-2},$$ but doesn't specify the ROC. Now the ROC must be an annulus, so our options are: $$0 < |z| < 2$$ or $$2 < |z| < \infty.$$ Now, my thinking (which is probably... |
I have built up a FIR filter based on least square approach.
I deploy the IDFT matrix $$F_{inv}$$ in above document page 7 and the desired response for compensated is H_F, the objective is
$$
min_{h_F} || F_{inv} * h_{F} - H_{F} ||
$$
h_F is a complex number obtained by above approach.
I plot the frequency respons... |
I understand that FIR filters have a finite impulse response and IIR filters have an infinite impulse response. Here's the issue:
A signal is finite in time if and only if it is infinite in frequency.
A signal is finite in frequency if and only if it is infinite in time.
So let's talk about the ideal LPF as an example.... |
I want to estimate the impulse response of an LTI system using frequency-domain deconvolution.
Suppose that the input signal $x[n]$ of length $N$, and the unknown system impulse repsonse $h[n]$ of length $L$, I have the output signal $y[n]$ of length $M=N+L-1$.
$$
y[n] = x[n] * h[n]
$$
Now convert them to the frequency... |
I have a VNA response (S11 parameters) from 3.5 to 40 GHz. Then I apply a hermitic transformation (complex conjugate) to it since we only have the spectrum for the positive frequencies. Then I use the IFFT to know the time domain representation. Then I would like to apply a Bandpass Filter to clean the time domain sig... |
I am trying to run 'wsprd' on a micro-controller for reduced power consumption and fun.
'wsprd' is open-source at https://github.com/pavel-demin/wsprd.
This program currently processes ~15 seconds of audio data in one go leading to a peak memory usage of around 21 MB.
We would like to bring this memory usage under 8 MB... |
I am trying to understand the system level impact on the receiver sensitivity of a UWB radio in the presence of a systematic elevation of the noise floor above thermal.
I know from the data sheet for both BPRF and HPRF the receiver sensitivity is in the range of -98 dBm/500MHz which puts the PSD around -124 dBm/MHz. I ... |
I am trying to follow some notes on "Analytical Computation of the Convolution Sum (Graphical Method)" but am getting stuck on what I am doing, or if what I am doing is correct.
There are 5 given steps:
Plot both the sequence $x[n]$ and $h[n]$ as functions of $k$ and flip one of them (either one) about $k = 0$ to obta... |
Assume $\tilde{h}(f)$ is a filter estimated by an algorithm for (room) impulse response estimation working in the frequency domain.
Is there a way to assert if such filter is causal?
Specifically, without resolving it in the time domain and convolving it with a known signal or estimating its zeros and poles.
I read som... |
I've been advised to ask here, I hope that it fits.
Original question:
I want to create a more straightforward and lighter version of this fantastic repo (which itself is based on Fingerprint Image Enhancement:
Algorithm and Performance Evaluation).
The tradeoff here will be between the running time and the quality of ... |
I am new to DSP. Please bear with my question. In QPSK why are we multiplying InPhase (Odd Bit) with Cosine and Quadrature (Even Bit) with Sine wave in this QPSK formulae
$$ s \left( t \right) = I \left( t \right) \cos\left(2 \pi f c t \right) − Q \left( t \right) \sin \left( 2 \pi f c t \right)$$
|
I want to know how I can find the type (algorithm) of a binary sequence at hand.
For instance, I'd like to know whether the following (binary) sequence is m-sequence, Barker, Gold, etc.
AC DD A4 E2 F2 8C 20 FC (hex) = 1010110011011101101001001110001011110010100011000010000011111100 (binary)
|
I recently had my first proper class of DSP and upon completing the review of Signals and Systems my professor started working on the concepts of upsampling and downsampling. Of course as is their style, they first introduce the topic and explain it in its entirety and then tell us what the topic is.
Now from what I un... |
Suppose I have a function that is not square integrable such as the zeroth Bessel function of the first kind. How do I generate a random sequence, with mean zero of course, such that its autocorrelation satisfies this function?
|
When I selected highpass filter effect in Audacity (an open-source audio editor), it lets me choose the cutoff frequency and the roll-off but it doesn't specify which filter type is used (IIR Butterworth, IIR Chebyshev or FIR types etc.).
Do you know which highpass filter is implemented in Audacity's highpass filter?
M... |
Is it possible to calculate one of two interfering waves, if the interference (the sum) $v$ and the other wave (summand) $v_1$ is known? The waves have same frequency but may differ in phase and amplitude.
I have found the formulas for the interference (the sum) but my algebra/analytical skills are not good enouth to r... |
The normalized frequency response of a N points bandpass digital filter is:
$\frac{sin(\omega_{c2}(n-M))}{\pi(n-M)}-\frac{sin(\omega_{c1}(n-M))}{\pi(n-M)}$ where $\omega_{c1}$ is the normalized lower cutoff angular frequency , where $\omega_{c2}$ is the normalized upper cutoff angular frequency and $M = floor(N/2)$.
Ho... |
In the time discrete convolution the order of convolution of 2 signals doesnt matter :
$x_{1}(n)\ast x_{2}(n) =x_{2}(n)\ast x_{1}(n) $ When we use the tabular method does it matter which signal we put in the x axis(which signal's points we write 1 by 1 in the x axis) and which we put in the y axis(which signal's points... |
The solution to an inhomogeneous differential equation can be split up into homogeneous solution and a particular solution (forced response).
Another way to split up the solution to an inhomogeneous differential equation is in a zero-input response and a zero-state response.
The zero-input response is the system's resp... |
In the PySDR online course, to the question "Why can’t we directly transmit the ethernet signal [directly in the antenna]", one answer is the following :
Square waves take an excessive amount of spectrum for the bits per
second – recall from the Frequency Domain chapter that sharp changes in
time domain use a large am... |
My understanding of the Fourier transform is that the FT of a cosine wave should be non-zero in the real part and all zero in the imaginary part. This follows from the orthogonality of sine and cosine:
$$
FT(\omega_0) = \int_{-\infty}^{\infty} \cos(2\pi \omega_0 t) e^{-i2\pi \omega_0 t} dt = \\
\int_{-\infty}^{\infty}... |
Let's say we have two signals with fast ascending in the beginning, then one slowly decays while another remains at a constant level. An illustration plot is attached. How to compare the similarity between these two with an optimal metric? Cross-correlation seems insensitive to the delay and shape difference. Would dyn... |
I have a triangular mesh (generated from a Lidar point cloud).
On this mesh I do point measurements using two instruments. One records M amplitudes of frequencies in the visible range and the other records N amplitudes of frequencies in the infrared range.
This picture shows two such measurements: blue visible range a... |
In the Audio EQ CookBook there is $$\omega_0 = 2\pi\cdot\frac{f_0}{f_s}$$so frequency warping will be $$\omega_r = \frac{2}{T}\tan(\frac{\omega_0}{2})$$
Then $$s \longleftarrow \frac{2}{T}\frac{1-z^{-1}}{1+z^{-1}} = \frac{\omega_r}{\tan\frac{\omega_0}{2}}\frac{1-z^{-1}}{1+z^{-1}}$$
But in the in the CookBook, we're gi... |
i have the following question:
in digital filter design what's the difference between the methods of transformations :
bilinear vs impulse invariance vs Euler vs step invariance.
thank you!
Here is a picture of what I got.
|
If a VMD analysis were fairly successful, in the sense that the spectrum is unimodal and relatively sparse, but still with a bit of bandwidth is it possible to extract a meaningful local amplitude and phase from it?
For instance, would it make sense to use the Hilbert transform of the output and use the usual magnitude... |
My problem is the following:
I am sampling data from a sine signal
Samples come in a online fashion (more and more data incoming through time).
The device does not have a lot of memory so it is impossible to store a lot of points -> the algorithm needs to be a online fashion.
The sine is noisy (assuming gaussian noise... |
I have applied PWM across a DC motor. The voltage measured across the motor is shown above.
I want to count the ripples present in the data. The ripples exist at a different frequency than the PWM frequency. I would like to figure out a way to count the ripples directly from this data or to remove/filter out the PWM c... |
When considering a direct conversion SDR receiving or emitting architecture, we can "represent" the received or transmitted signal as the sum of two components, the I and Q component.
Each component, represented by a complex number in the "IQ plane", or by a magnitude and a phase, is then mixed with a sin and cos origi... |
I want to build up linear phase filter by least square approach.
I have formulated a least square problem similar to Matt's blog. To approximate the desired complex response $F \times 1 $ vector $d$ by the length $N$ FIR filter's , which coefficients are formed as $N \times 1 $ vector $h$. The basis $F \times N$ matrix... |
I have to compare the response of an 8th order IIR filter and its equivalent 4 stage cascaded biquad structure. A sinusoidal signal is given as input to both the systems and the output responses are plotted and the difference between the outputs of two cases are plotted.Why there is an error of the range of 15 between ... |
I want to generate a stepped frequency modulated waveform (SFMW), similar to a linear frequency modulated chirp.
While approaching this, I wanted to consider the issue that there should be some continuity at the points where the frequency jumps, such that there are no jumps in the signal waveform like here:
In my case... |
We know overlap-add of a en-framed signal can be done easily by following code
function y = overlapAdd(frame, hop)
frameLen = size(frame, 1);
nframes = size(frame, 2);
xlen = frameLen + (nframes-1)*hop;
y = zeros(xlen, 1);
for l = 1 : nframes
y(1+(l-1)*hop : frameLen+(l-1)*hop) = y(1+(l-1)*hop : frameLen+(l-1)*hop)... |
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