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Is there any study on the magnitude of local oscillator leakage as a function of oscillator frequency? If RF input is disabled and there is local oscillator at the mixer, then due to leakage, this will give rise to dc. I want to understand how the dc magnitude vary as the local oscillator frequency is varied. Let's say... |
In general, How do we design a system that takes input x(t) and outputs x(t/2)? I know that if system outputs x(t/2) then the frequency domain of output signal has the form X(2f). But what should be the impulse response for such systems?
|
Let the figure of the Bode's Gain plot of a certain transfer function, estimate what could this transfer function be:
Here is what I tried to do: since at $\omega = 0 , |G(j \omega)|_{db} = 20\log(K) = 20$
I see that it goes up to $35$, I dont have an idea to translate it to something mathematical, because I think thi... |
I am trying to extract data from .iq file recorded by radio. The modulation is QPSK. I have implemented a GRC flow graph but the costas loop is not converging. Can anyone please help in understanding the issue
|
I’m guessing it’ll be more computationally complex considering the irregularly sampled image would have to be converted to a regularly sampled one and probably a different demosaicing algorithm would have to be used as well but is it something today’s image processors can’t handle ?? And what other issues could arise ?... |
Edit 2: I am trying to replicate results from this paper Compressed Sensing with Coherent and Redundant Dictionaries. On page 3 the "oversampled DFT" is mentioned as an example of an "overcomplete frame whose columns may be highly correlated".
Edit: Replaced magnitude plots with angle plots for all matrices, and includ... |
I am having an input signal x[n]=[1 2 3 4 5 6] and another signal y[n]=[2 3 4].
In order to check whether y[n] is present in the signal x[n],I did cross correlation and the result obtained was [ 0 0 0 4 11 20 29 38 47 28 12].
If the signal y[n] is present in x[n], then the correlation value will be equal to the ene... |
What are the methods available to calculate and compensate for IQ imbalance
using digital samples? What are the methods available: when one is allowed to send known signal and when data send is not known like a 5G system?
|
I am trying to implement the proposed method from this paper 1, to eventually have a more efficient PSD computation for feature extraction on a microcontroller.
The method is suitable since I am using a hamming window and have 50% overlap in the segments.
In short this method splits the segments in two parts, calculate... |
I searched on web and apparently can be due to truck drivers communicating at CB band which is around 20MHz. And apparently this communication may interference with the frequencies used by computer's audio card.
How is this electronically and physically possible that these MHz communications are able to interfere exiti... |
I'm reading Brian McFee's excellent book Digital Signals Theory.
He presents the DFT as a measure of similarity $S$ between a digital signal $x[n]$ with $N$ samples and a set of reference signals $y_m[n]$, where $m$ is one of the set of analysis frequencies $m = 0, 1, 2, ... N-1$. So we can think of the DFT as a set of... |
Let us suppose to have an image with size $28\times28$, and that we want to apply a 2D-FFT without any padding operation. Does exist any algorithm which allows to perform the 2D-DFT calculation with time complexity $O\big(N \log_2(N)\big)$?
|
Hello I have the following question in regards to what the units should be for the y axis for a continuous time Fourier transform and that for a discrete time Fourier transform.
In my understanding if my time domain signal is a a voltage over time signal when I take the continuous Fourier transform the x axis should be... |
The code below generates four sinusoids and considered as known signal. One needs to find the location of the signals under additive Gaussian noise.
The known signals is comprise of four sinusoids with random phase and amplitude and at random location in time samples.
As I understand Matched filter is the only option. ... |
I am new here and a beginner with SDR & FFT.
For antenna pattern measurements, a flying probe antenna is to be used to sample the electromagnetic near-field in front of an antenna. A lightweight spectrum analyser (SA) is to be used to provide a continuous stream of IQ signals. At defined times, this stream shall be sli... |
I have created exactly the array factor formula in matlab .
But when i add the beam steering delta member it does nothing to the main lobe.
Why it stays the same?
Thanks.
theta_0=pi*((20)/180);
amplitude_norm_V=[0.05,0.3,0.75,1,0.75,0.3,0.05];
delta=k*d*cos(theta_0);
x=sqrt(1/(sum(amplitude_norm_V.^2)));
real_voltage=a... |
What is the approach to fuse multiple IMUs in EKF framework?
Normally you take the IMU as the motion model and the sense step to consist of other sensors but when have multiple IMUs operating, what is the correct approach?
|
I am trying to decode QPSK data and these are the constellation and timing plots, what could be the reason for the non convergence of QPSK constellation
|
Most of the time, the baseband transmission chain is: modulation->oversampling->shaping filter (SRRC)->bandpass conversion on a carrier frequency $f_0$.
But can we change the previous order like that?
modulation (baseband)->bandpass conversion on a carrier frequency $f_0$.->oversampling->shaping filter (SRRC)
|
This is a standard thing in beamforming especially Mvdr the algorithms produce weights and if I wanted to transmit with the weights I need true time delay not phase shifters.
|
The utility electric power signal is periodic or not? Our nationalgrid frequency is 50 hz and in some other places in world, it is 60 hz
If I observe this signal on oscilloscope, will I get an exact repeating pattern ??
|
I am trying to implement the resampling algorithm as described in this paper: https://ccrma.stanford.edu/~jos/resample/resample.pdf
I managed to construct the $h$ (windowed sinc) table, and I have implemented the loop. I am having a hard time to compute the $\eta$ interpolation factor and the $l$ index index for the $h... |
I have a signal and i am trying to observe its approximations using complex exponential fourier series and trignometric fourier series but i am not getting exactly same result(graph of trignometirc fourier series approximation has more ripples compared to graph of complex exponential fourier series approximation)
Desp... |
I have 8 digital complex input channels digitized with 8 bits each and I want to combine them with certain complex weights. In the easiest case where all weights would be 1 the combination would be just a sum of the 8 channels. The resulting number would be an 10 bit number. Is it correct to say that the resolution has... |
I'm trying to understand the DCT, for this purpose I wanted to generate the image of all basis images of the DCT. And given the formula by matlab:
$$
B_{pq} = \alpha_p \alpha_q \sum_{m=0}^{M-1} \sum_{n=0}^{N-1} A_{mn} \cos \left( \frac{\pi(2m+1)p}{2M} \right) \cos \left( \frac{\pi(2n+1)q}{2N} \right)
$$
where $0 \leq p... |
I do not know much about signal processing and would appreciate if someone could confirm if my calculations are correct.
Here I have the Noise density of an accelerometer that has frequency 6400hz and will use full scale +-2g. I want to calculate the standard deviation of the sensor readings.
Here are my calculations:... |
There are several machine learning architectures specifically devoted to image segmentation (like UNet).
I am interested on identifying the location of an area but not interested in the precise boundaries. For example, identify the location of the head in the photo of a person. The input is a photo, the output is a do... |
I want to calculate the phase margin and the gain margin of my transfer function to have an idea about the system's stability. The transfer function is quite complicated since it contains an exponential term.
The corresponding Bode diagram is given as the following.
As you can see the phase is positive for the relevan... |
How to get an average of a sinusoidal signal? The frequency of a signal is 50 Hz and could have harmonics, i.e. high THD.
I tried to use exponential moving average with low cut-off frequency (about 1.6 Hz) but is not work. basically the amplitude of the sinusoidal is big and the filter return a small sinusoidal not the... |
I've been given a task to simulate a real-world scenario, in which a system receives a signal with a carrier frequency of 3.75 MHz. To filter the signal, either a very sharp and resource-intensive filter can be used, or a proposed series of frequency doubling and filtration of the signal that supposedly improves the SN... |
How can I find the impulse response for the following system in time domain? I actually would like to find my mistake in my attempt. Below is what I have tried according to the answer given for this question: Why is particular solution zero for an impulse excitation signal?
The given system/circuit can be described by... |
I am wondering if there is any implementation of Laplace Transform and Inverse Laplace Transform available for 2D data (i.e., images). For example, a batch of N input sequence of D can be reshaped into a 2D image with width of W and height of H and then 2D FFT is applied in both directions and then after processing it ... |
I've been asked to calculate the cross correlation in the 3-12 Hz band between two simultaneously recorded brain signals in two distinct brain areas. Data were acquired at 32 kHz then preprocessed by applying a 3-pole Butterworth filter (lowpass 1 kHz) then downsampling to 2 kHz. By applying the method in Adhikari et a... |
So, I have a problem where the input and output of a system are given :
\begin{align}
x(t) &= \sin(10t) \\
y(t) &=5\cos(10t+\frac{\pi}{6})
\end{align}
So, I need to determine whether this system is LTI.
First, when I saw this problem, I checked for zero input-zero output
condition of LTI. By substituting $t=0$, $x(t)=0... |
I want to check the time invariability of this recursively defined function
$$y(t) = y(t-4)+x(t-4)$$
We can check time invariability of functions expressed in terms of x(t), but I couldn't find anything for such recursively defined function.
In the book "Signals and System by A. Nagoorkani", we can test for time invari... |
I'm studying the perception of vowel formants (resonances of the vocal tract) and need to create stimuli where the signal below the first (lowest) formant is removed. I have some synthesised vowels which have a dynamic pitch curve because they sound much more natural this way, which is important for the study. But havi... |
I am implementing a real-time convolution reverb. I currently have a working implementation, doing frequency-domain multiplication after FFT, using uniformly partitioned block sizes. This answer on this platform was super helpful for me.
I was wondering if anyone could write a similar step by step guide for non-uniform... |
I’ve a general question about precoding.
The channel capacity after precoding is $\log_2(\det(I+F’H’HF))$ where $\det$ is the determinant, $I$ the identity matrix, $F$ the precoding matrix, $H$ the channel matrix and $’$ the complex conjugate transposition.
If $F$ is a square unitary matrix, then
\begin{align}\log_2(\d... |
I want to set the locations of two submatrices W22 and W21 taken from Hadamard matrix, with respect to W22 keeps being unitary and matrix T has non-zeros values in the first columns only (Regardless the number of columns that are non-zeros), T is sparse matrix. How to do that in Matlab?
For example this is the matlab c... |
Suppose we receive $R(t)=X(t)+W(t)$, where $X(t)$ is band-limited to $[-B/2, B/2]$ and $W(t)$ is white Gaussian noise with autocorrelation $R_W(\tau)=\frac{N_0}2\delta(\tau)$. If we filter $R(t)$ with an ideal LPF, i.e. $h(t)=B\cdot \mathrm{sinc}(Bt)$, and then sample it at Nyquist rate, i.e. $Y_n = \left. Y(t)\right... |
I am studying a 1D beamforming for wide-band signals, so that applying only phase weights to each individual antenna would steer the beam for the central frequency. For other frequencies the steering would not be the desired one. I understand the optimum way to apply the beamforming would be to apply delay weights inst... |
I have some audio files affected by an issue during recording, apparently chunks of data were duplicated, like this. Doing a search I realised that I already asked about this, but I have got a very open answer.
This could be posed as classification problem, where each sample is either included or excluded in the final ... |
Recently been running simulations where I run through simulating passing random data bits through LDPC Encoding, Modulation (4, 16 or 64) and then AWGN. The code itself is 90% developed by myself with the remaining 10% coming from a LDPC Encoder for 5G from Github (I mainly used this because the nrLDPCEncode function d... |
Suppose I have the FFT of an arbitrary real audio signal z samples long. Is there a way to discover the signal level (intended as the largest absolute value of its samples) from the FFT without computing the inverse FFT ?
I think I have read that a trivial method exists only if the spectrum is zero-phase, but in my exa... |
I’m observing drift and a stepped pattern in data from a piezoelectric pressure sensor measuring constant atmospheric pressure. the results are the following:
The data should ideally be a flat line with some noise. How can I correct these issues in real-time and/or in post-analysis?
Edit Date: 24/03/2024
I've updated ... |
in 2.104 he put a condition that x(t) must be an exp to use this result ,but in 2.108 he did't even know what the value of x but use the result to get y
i know that he decompose x(t) to group of exp but still the amplitude for each exp isn't one
isn't it supposed to be
H(F)x(t)=lim summation{y(t)x(f)delta(f) ???
|
In spread spectrum systems like DSSS or CDMA, we can take a pseudo-random sequence composed of 1 and -1. But can we use a complex number sequence between -1 and 1? Why is this type of sequence never used?
Here are the Matlab codes:
pn=2*(rand(1,128)>0.5)-1; %pn code
comp=(2*(rand(1,128)>0.5)-1)+1i*(2*(rand(1,128)>0.5)-... |
I am encoding and decoding a randomly generated bitstream of data using a [5 7] convolutional code. For the decoding part I am using a Viterbi Decoder and trying both HDD and SDD. However, my results for HDD seem incorrect as the BER exceeds the theoretical upper bound that I have calculated. I used a bitstream of 100,... |
Does anyone have any good books/videos or courses to learn about radio waves,frequencys,phase spread spectrum and modulation?
I need a something that goes through the basics until advanced.
Thank you in advance
|
I have created a code which created the plot below.
The heart of the code is the AF expression.(array factor)
I want to do the inverse process , to take the plot do the inverse of it and somehow to find using optimization the coeffects(genetic algorithm) which will recreate the plot.
my plot is from -90 to 90 degrees... |
I am aiming to create a digital windowing FIR filter on MATLAB and have been given specs such as a:
sampling frequency of 360Hz
maximum filter order of 5
cut-off frequency of 98Hz/ 1.710rad/sample (normalised)
The cut-off frequency was approximated by doing a DFT on the noisy signal that needs filtering, i have put a... |
I again need your help. You were already nicely able to help with predicting the signal I'm dealing with. Thanks a lot. (Forecasting of a periodic movement based on sensor data)
Now I need to remove to effects from the sampling. Unfortunately I don't really have back ground in signal processing
The signal is sampled mo... |
I used Intel 5300 NIC and the CSI tool to get CFR. I also want to apply for a IFFT to get CIR. Do I need to rearrange CFR before I apply ifft?
The OFDM subcarrier indexes (for 20MHz) are:
subcarrier_index_L = [-28, -26, -24, -22, -20, -18, -16, -14, -12, -10, -8, -6, -4, -2, -1]
subcarrier_index_H = [1, 3, 5, 7, 9, 11,... |
The heat equation
$$
\frac{\partial u}{\partial t} = \alpha \nabla^2 u + f(x,t) \tag 1
$$
Here, $u$ represents the temperature field, $\alpha$ is the thermal diffusivity of the material, $\nabla^2$ is the Laplacian operator, and $f(x,t)$ represents the sink/source term. I guess that it would depends on what kind of sin... |
I need to apply a low pass filter up to a period of 10 hours to my data. My sensor has a sampling rate of 1 sample every 5 minutes. I need to know how much data I at least need to have to be allowed to use a lowpass filter.
Fourier is fairly new to me, but I would say that I need at least 10 hours of data (so 120 datap... |
I recently started learning about wavelet transforms, and there's something that is confusing me. My understanding was that most wavelet decompositions are simply a decomposition of a given signal in an orthonormal basis. This seems to be true for something like Haar wavelets. If I generate a signal of length $N = 2^k$... |
I understand tha the wavelet transform is about computing the coefficients to assign to scaled and translated versions of the chosen mother wavelet. The coefficients measure the correlation between the signal and shifted/scaled wavelet. That said, I know that the CWT is used to compute the scalogram while the DWT is us... |
Let $s(t)$ be a complex signal. I would like to know how to find a signal from its instantaneous frequency. I programmed the following function which calculates the instantaneous frequency.
function [f]=instfreq(s)
z=unwrap(angle(s));
f=diff(z)/(2*pi);
end
Example on a chirp:
clear all;close all;clc;
B=20e3;
fs=100... |
How can one estimate the ML bound for Massive Mimo detection with 128 transmitting antenna and 128 receiving antennas using 4 QAM modulation?
I have seen for a 8 by 8 system it is possible to use K-best detection but for a 128 by 128 system I am not really convinced that K-best detection is the way to go.
|
Let system G(s) be:
$$ G(s)=\sum_{i=0}^{10}\frac{(-1)^i}{(2i+1)^2}\frac{\omega_i}{s^2+2\zeta_i \omega_i s+\omega_i ^2}$$
$$\omega_i=\frac{(2i+1)\pi}{T}\, T=1\, \zeta_i=0.2$$
Its impulse response is an approximation (truncated Fourier series) of a slowly decaying triangular wave. Using the balanced truncation method, fi... |
I have a single "pulse" signal (looks like a gaussian but it is not exactly), followed by several bounces, with a constant period, that probably come from the electronics used (I don't have access to this).
Specifically, it is a light pulse (I don't have access to measurement of it, I assumed Gaussian for simplicity), ... |
I am trying to calculate the Blackman-Tuckey (BT) PSD in Python to check my understanding (getting started with signal processing). I have tried making the calculation myself and compare it with Scipy's periodogram and pyspectral CORRELOGRAMPSD. I expected some differences between my result and the periodogram (it is a... |
I am currently taking an graduate-level Advanced Signal Processing class and I have a midterm soon. However, the midterm is not only open-book but it is also open-internet and untimed. Now I have no previous exams of the professor but I do have one exam of a lower level signal processing course where the professor basi... |
I have a temperature sensor data, I want to denoise it. The first thing that came to my mind was to take the moving average, it was very smooth but it is still not stationary. If I take the log difference, it becomes stationary according to ADF and KPSS, but is this the right thing to do? Is it the right thing to do di... |
I am currently writing an essay on Wavelet transforms, and as part of such, I am trying to show that the Morlet wavelet satisfies the standard criteria:
$$
\int^\infty_{-\infty} \psi(t) dt = 0
$$
$$
\int^\infty_{-\infty} |\psi(t)|^2 dt = 1
$$
Using the following definition from this article (as cited on the Wikipedia p... |
What kind of filter would produce this kind of frequency response?
I'm pretty sure there's a moving average involved, maybe two, but that's as far as I got.
|
I've been working from this reference document, essentially trying to recreate Figure 7 with the understanding that the two functions place_signal and signal_shift were able to undo one another.
import numpy as np
import matplotlib.pyplot as plt
def place_signal(size,location,signal,sinc_length=100):
bin = int(n... |
I am studying the seismo-ionospheric anomalies associated with the Japan earthquake using total electron content derived from a GPS signal. I am trying to detect acoustic-gravity waves in the signal caused by the earthquake if possible.
I am trying to perform spectral analysis for this task using matplotlib's built-in ... |
I am trying to implement convolution using fourier transform property but i am not getting exactly same shape of output that i get by using conv command
First portion of graph obtained using conv command is same as that obtained using fourier transform property but second portion of graph obtained using conv command ... |
I want to investigate the attenuation in signal due to various environmental factors such as rain. Is it sensible to try to investigate some function that relates the time delay due to those environmental factors on the RF communication signal. In short to relate the impact of environment to the time delay experience
|
Because it causes confusion within my team I wanted to create a simple overview on how to convert from one PSD Unit to another. I know it is really simple but I've obsessed over it for a few hours. I feel like I'm blind to simple mistakes at this point. Would you agree with what I compiled in that overview?
Conversion ... |
In order to compute the power spectral density of a signal, we can use np.fft.ftt in python in the following formula:
def direct_fft(f,f_s):
'''
- f: signal
- f_s: sampling frequency
'''
return np.fft.fft(f) / f_s
def PSD_fct(f, f_s):
Fourier = direct_fft(f,f_s)
T_obs = np.real(len(f))/f... |
While I'm aware of the fact that causality implies that the present output is only dependent on present and past inputs, something that is bugging me is what if the input is dependent on future outputs. Would the system be causal or anti-causal in that case? I look at this with the assumption that since input depends o... |
This is not a duplicate of Filter secondary bounces of a pulse signal because in that question, a convolution model is finally not appropriate, whereas here, the question is about convolution.
Given the signal y below, is there a way to recover the original signal x (before convolution), without knowing the impulse re... |
What I am trying to do:
I am trying to extract features based on the difference between a reference signal (white noise in my case, let's call it ref) and different processed versions of it (let's call one proc_k).
Although the difference signals are not the equal, the MFCCs extracted from these differences are equal... |
I need to show that the impulse response of every stable system has finite energy.
I have trouble solving this problem.
Since impulse input signal is unbounded, I can't apply BIBO stability of system well.
I'd like to know how can I approach to this problem.
|
So using this resource:
https://flylib.com/books/en/2.729.1/the_zoom_fft.html
I think I understand that the zoom FFT allows you to calculate the FFT for a region of interest with a lower order FFT by shifting the region of interest to near zero Hz and filtering out frequencies outside of the shifted region of interest,... |
im working on a program in matlab for calculation of sampled signal spectrum, and I have a sinusoidal signal with frequency f=100 Hz, sampling frequency fs=250 Hz, every sample is repeated 4 times (fr=1000 Hz), my spectrum got this series of frequencies:
100, 150, 350, 400, 600, 650, 850, 900 [Hz]
so:
f, fr/4-f, fr/4+f... |
The following code creates a cosecant squared radiation pattern.
There are phases and an amplitude_norm_V vector which are the coefficients of an Array Factor (AF) expression as the response of an array of isotropic antenna elements.
I am looking to create a Genetic Algorithm (GA) that given known AF data will recreate... |
Context:
$\bar{\Theta}$ is the room regression filter coefficients (RRC);
$$X_{t} = \bar{\Theta}^{H}\bar{X}_{t-1} + s_{t}$$ means in words: the filter that defines how the room causes reverberation to the signal (how the past of the speech signal x affects the current speech signal).
$s$ is drawn from $\mathcal{N}(0,\l... |
I've implemented second order biquad filters using the famous RBJ Cookbook. I also want to implement first order lowpass and highpass filters. I was hoping to use the same code, so I need to define these first order filters in Direct Form I.
I found examples of first order filters in Direct Form II, but nothing for Dir... |
Im new to DSP and i wanna know how i can add overlap-add into this to remove artifacts.
#include <SoapySDR/Device.h>
#include <SoapySDR/Formats.h>
#include <stdio.h> //printf
#include <stdlib.h> //free
#include <fftw3.h>
#include <complex.h>
#include <sndfile.h>
#include <string.h>
#define BUFFER_SIZE 65536
#define SA... |
I have simple sine wave as input $x(t) = \sin(6\pi t)$ and then take a Fourier transform of the signal. Afterwards, I multiply that to a transfer function. I see a discontinuity 3GHz. Is this just an artifact?
$
Y(\omega) = G(\omega)\cdot X(\omega)
$
where $G(\omega) = \frac{-j\omega}{\omega_m^2(1+j\eta) - \omega^2}$
|
I have found this piece of code in Infenion DSP Library (asm code):
Fir_Blk_16()
...
;(ACC)=(ACC)+h(i)*x(n-i)<<1 as comment
I don't understand why the result is shifted (<<1)?
Is this for rounding purposes?
Thank you in advance,
Anton
|
I have a waveform where I sliced the flat top so I can do some analysis on the amplitudes of each frequency. I have a dataset of N measurements each with M samples of current (A) over time (microseconds). I expect that there is a frequency on the 100 Hz sideband of the expected frequency. I curve fitted the expected fr... |
I want to estimate the power spectrum of a given signal using Bartlett's method.
I understand the general purpose of the method. However, I do not get my head around the algorithmic description of the method given in the Wikipedia article.
The part I am stumbling over, is in the 2nd step:
For each segment, compute the... |
I have done a power spectrum density estimate using Bartlett's method. However, I was wondering if it is also possible to get a sense of the variation of the averaged spectra.
Unfortunately, I have not found a method for this problem, so any reference is most welcome!
I have plotted the mean and std of the DFT amplitud... |
I have couple of analog synthesizers and I want to turn their oscillators into code. I've been reading about how to do it for a while and making a lookup table seems just right for me. I've been modeling some linear/nonlinear effect units, but I'm new to modeling synthesizers, so at this point, there is a great possibi... |
I am very new to Kalman filter. I am doing a project using one sensor the track the sensor's position. I developed 2 methods to solve the position, one in better accuracy and one is less accurate.
I would like to know that can I use Kalman filter to fuse the 2 position-solving algorithms to achieve a better performance... |
I have done a power spectrum density estimate using Bartlett's method. However, I was wondering if it is also possible to get a sense of the variation of the averaged spectra.
I have plotted the mean and std of the DFT amplitudes (no normalisation), which at least does not look completely off.
However, when I plot the... |
I just started a while ago with DSP and slowly staring to understand the concept of I/Q and what it means. I successfully could decode some bfsk, bpsk and wanted to move to msk which as not easy to understand. Lost of blog posts explain very well how to deal with as OSQSK or CPFSK. In those posts we always work with I/... |
My question is: after upsampling a signal, does the output signal contain the original signal values? The diagrams I saw so far (on wikipedia and different forums) have always shown the original values but when I convolute with the fir filter coefficients, the output signal doesn't contain the original values or not al... |
I am struggling to understand the meaning of the covariance of PSD (Power Spectral Density) between two frequency points ($f_1$ and $f_2$).
The covariance of a function $s(t)$ in time domain $t$ is a Fourier dual of the power spectral density and that is understandable. However, in some papers, the covariance of the po... |
I am trying to bandpass filter an EEG signal, nothing fancy but it's coming out pretty distorted.
EEG data is taken from forehead. Sampling rate is 250 Hz. Cutoff is 2.5 Hz & 120 Hz.
Tried in both matlab & python, getting same results.
Matlab code:
data = load("rawdata.mat");
data = data.data;
figure
bandpass(data,[2.5... |
I have a sample of a signal lets say 8 values of a 16 value sequence. I am trying to obtain the full DFT from the partial.
I am thinking that I am going to work back, I read the matlab documentation and have come acrosss an inverse function. Am I on the right track? I was not able to see a pattern in the results..
yt =... |
Sorry if this is a trivial question, but I am not doing signal processing everyday. I will try to express what I think I have understood as best as possible.
Suppose I apply a (linear) sweep signal to a system (like a loudspeaker in a room) and measure the response. To the input and output of that measurement I apply a... |
So I am trying to understand the scattering transform on graphs (GST). There are many formulations but I am stuck on equation 8 in this paper. Here they give a condition on a wavelets that looks like Equation 10 on page 7 of Mallat's foundational paper,and claim that this means that the wavelets were derived from a mul... |
When reading about SIFT I read halving resolution is the same as increasing $\sigma$ of the gaussian in terms of feature detection but reducing resolution has the advantage of reducing processing.
What I am curious about is suppose we keep the image the same size and only modify $\sigma$ and apply DOG as usual. When a ... |
Say I have a signal x(n) and I want to resample from rate 30Hz to 20Hz. In Matlab, I can make a filter with:
fs1 = 30;
fs2 = 20;
f = [0 6 7 fs2/2]/(fs2/2);
a = [1 1 0 0];
n = 31;
b = firpm(n, f, a);
This way my cutoff frequency is 7Hz. Now I want to use this as my interpolation filter. I can run
x2 = resample... |
What can be recommended as challenging tests for cross-fading/mixing of music?
I am looking for examples of music-titles that exhibit severe volume dips when cross-faded or mixed.
Ideally answers should provide pairs of (free) music-titles and the length of the cross-fading interval and also other information that is n... |
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