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I am trying to solve the following exercise, where $y(t)$ is the sum of two signals $x_1(t)$ and $x_2(t)$ with each of them being the product of the convolution of $e_i(t)$ with $h_i(t)$.
So far I have found that the PSD of $y(t)$ is equal to
$|H_1(\omega)|^2\sigma_1^2 + \rho\sigma_1\sigma_2[H_1(\omega)^*H_2(\omega) +... |
Fft question:
Do I need overlaps if I’m only grabbing some features like mel or bark? I don’t even necessarily need every frame of a STFT even one every second or so works. How I understand fft overlapping is that each overlap sums together after the final window after the inverse fft. However, for my needs I do not ev... |
I previously asked at which stage pulse shaping is applied in a WiFI transmitter and my question was closed due to duplication. The answers I was directed to have only served to confuse the issue for me. I have therefore reworded my question as suggested. I understand the need for pulse shaping of an RF signal (to re... |
What is this method in general?
What is the main difference between the quantile cross-spectrum (QS) approach developed by Baruník and Kley (2019) and the cross-correlation function (QS) developed by Han et al. (2016)?
Also, through which program can I use this method?
|
In the three figures below, $H(e^{j\omega})$, $H(e^{2j\omega})$, and $H(e^{4j\omega})$ are shown. I understand that up-sampling shrinks the spectrum. However, I need a few clarifications:
Why does up-sampling result in the creation of new copies (imaging)?
What is the periodicity of the images? In other words, how, fo... |
I have some intuition about bandwidths from these videos (1 and 2) explaining how the frequency response corresponds to the Z-surface.
The only thing I could find on the internet about calculating the bandwidths is this matlab tutorial:
The bandwidths of the formants are represented by the distance of the prediction p... |
I am running into consistent connection issues when trying to run a USRP N210 on a VirtualBox Ubuntu 22.04 VM. I built uhd from source on the ettus page, configured a static IP address in Ubuntu, connected via ethernet port, and successfully found/pinged and received WBFM in GNURadio from my USRP N210. However, after s... |
How is it possible to correct the possible phase distortion of a typical signal which has been received by an antenna and passed through different analog elements including an amplifier and a band pass filter?
From band pass filter data sheet, I do have the group delay information per frequency in Nano second.
For the ... |
I currently am trying to portray PSNR's ability to measure noise in an image. The PSNR is said to be high (around 40 dB) if there is a low amount of noise in the 2D signal.
In my experiment I have fairly noisy image (artificially added) and through 6 iterations I add more and more denoising strength with fastNlMeansDen... |
Greetings Forum!
I posted a question sometime back and got some relevant feedback on that question however I have now run into another issue.
I am using a radio to transmit a 2-GFSK waveform. I used GNU Radio Companion 3.10 to see the signal in time domain in a LoS channel condition. However, I am confused as to what... |
If the integral used to calculate the total energy for a continuous time real signal converges, its average power is equal to zero. But could the average power still equal zero if the total energy diverges?
Equivalently, could this limit evaluate to zero
$$P_\infty=\lim_{T \to \infty}\frac{1}{2T} \int_{-T}^{T}\left| x(... |
This question is a follow up on previous. As I am analyzing the sampled data I need to know if it is complex or real. This is important for me since I want to transform my two-sided to a one-sided PSD. Then I have to either divide the spectrum by $N/2$ (for real-valued signals) or $N$ (for complex-valued signals).
So f... |
I would like to perform a fixed sine test at my DUT's natural frequency after finding the nat. freq. from a sine sweep test. I can see from "fixed sine" softkey under "Source" that I could have a fixed amplitude. But it is possible to have a fixed frequency instead of defining a range?
|
Window functions are commonly used to control spectral sidelobes. Typically there is a trade between supressing sidelobes, and maintaining the resolution of the mainlobe. A window with lower sidelobes has a wider mainlobe.
But another characteristic that is desirable in some situations is to control where the first (or... |
In the Fourier series, I knew that the coefficient $a_0$ represents the DC value, shifting the signal up and down by the amount.
Then, what's the actual meaning of the amount of frequency $0$ in the Fourier Transform? I saw many people refer to it as the DC component. But I thought the DC should be the average value an... |
I am trying to create an audio program that can do a variety of spectral processing techniques on an input signal. I am currently stuck on implementing an fft program that runs quickly enough. I tried a recursive and iterative version, neither of which worked quickly enough, so I have begun searching for a reliable and... |
I learned from my signal processing course that PSD is the Fourier transform of the autocorrelation function.
$$\mathscr{F}\Big\{\mathrm{E}\big[x(t)x(t+\tau)\big]\Big\}$$
Today I took my statistic course but my professor introduced spectral density function, which is the fourier transform of autocovariance function.
$$... |
I know that an IIR filter exhibits non linear group delay (so different frequencies can be delayed more) and that this is related to the fact that it is asymetric (due to considering previous outputs), but I don't have an intuition for why this would affect different frequencies by differing amounts.
|
I'm working on frame synchronization, and I don't fully understand the probability of miss detection and probability of false alarm in evaluating the performance of the synchronization block. As far as I understand:
Probability of false alarm is the probability when the correlation peak is greater than the threshold, ... |
I am learning about radar and signal processing. I try to run a simple code snippet provided on Mathwork's official website
It is just about FREE SPACE, which I think you know very well, but I am new to the topic and cannot understand the MATH behind it. For your convenience, let me put the code snippet below:
fs = 8e3... |
I need to detect the Direction of Arrival of a speech signal using an 8 microphone array as part of a project I've been working on. Currently, I have a working solution in the 2 microphone case:
$$G(\omega ) = X(\omega ) \cdot X_{delayed}^{*}(\omega)$$
$$g(\tau ) = {F}^{-1} [G(\omega)] $$
$$\tau _{max} = argmax\; g(\t... |
During setup of a USRP N210 on MATLAB's Communications Toolbox Support Package for USRP Radio, the initial test failed on "Receive the signal and verify the frequency of the tone" field. Afterword, MATLAB crashed and my USRPN210's led's swapped from D&F to B & F. I also can't get a successful ping or find my USRP with ... |
I am trying to calculate manually the phase of the frequency response of an LTI system as a function of frequency and plot the results. The original system is described as $y[n]=0.1(x[n]-x[n-1]+x[n-2]$, which I was able to write the frequency response $H(e^{j\hat{\omega}})=0.1(1-e^{-j\hat{\omega}}+e^{-j\hat{\omega}2})$... |
I am trying to find the PSD of a signal by taking the DFT of its auto-covariance sequence.
My code is as follows
% define constants
m = 256;
k = 0:m - 1;
a1 = -0.9;
b1 = 0.8;
sigma = 1;
% compute ACS for positive lags k
r = (sigma^2 / (1 - abs(a1)^2)) * ( (1 + abs(b1)^2) * d(k) + b1 * d(k - 1) + conj(b1) * d(k + 1));
... |
I create spectrogram of a sine wave that changes its frequency from 60 MHz to 70 MHz over a period of 1 millisecond.
can someone help me understand why I see those jumps? how can I make it smoother?
import numpy as np
import matplotlib.pyplot as plt
from scipy import signal
def wave_create(duration: float,
... |
Suppose we use a DSSS system such that each modulation symbol is spread by a different code. On reception, the signal has a frequency drift and a phase-locked loop (PLL) must be used to correct it. However, this loop is set up during the demodulation step. But when we despread the message received by the different sequ... |
Hi I'm looked to graph a Raised Cosine Bell Curve according to ones similar presented in the linked article. I have no experience with MatLab, and have found little luck scowering this and other sites along with the paper cited.
I have all of the data gathered along with the values of A, B1, and C1 for input in the rai... |
Pulse amplitude modulation in general is often explained using plots like this one from wikipedia:
There is a sinusoid analog signal (red) to be "imprinted" on a train of pulses. The result is a train of pulses with varying amplitudes. (the blue signal)
When looking at the waveform and eye-diagram of a PAM-4 signal we... |
Determine if the following sequence is time invariant or not:
$$y[n]= n^2x[n-2]$$
|
I am trying to implement an FFT in a shader to run on the GPU. I grabbed the algorithm from here:
https://github.com/jbouny/fft-ocean/blob/master/js/shaders/FFTOceanShader.js
However, when implemented and compared with other FFT objects (which I'm doing in max/msp) the results of the above algorithm don't seem to match... |
I'm studying the full synchronization chain, and it's not clear to me at which domain each block works on in terms of sample vs. symbol domain. I'm particularly confused between frame synchronization and symbol timing synchronization. Coarse frequency synchronization operates on the samples, the fine frequency synchron... |
I originally made this flow diagram to process signal in real time from cubesats that have a cent-freq of 437 MHz, but I want to record signals to anaylize later the wav file so . However, I want to know if the process to record a wav file is correct. Can you give some advice to know if my flow graph is good? Do I ne... |
I ask this question because it seems to me that if pulse shaping has been applied in the transmitter, the signal that is actually transmitted is not phase modulated at all, rather, it is amplitude modulated. I have made some pictures to illustrate my thinking, which goes as follows:
Most literature about BPSK illustra... |
Following the answer in this post: Phase noise and frequency noise relationship, I can calculate the phase noise spectrum density with the unit from $\frac{dB_{c}}{Hz}$ to $\frac{rad}{\sqrt{Hz}}$. However, when I want to convert it to frequency noise spectrum density, I am confused about the frequency I should use to m... |
I use a spread spectrum communication with a DQPSK modulation. Each modulation symbol is spread with a different spreading sequence. At reception, I use a sliding window for despreading with the different sequences.
Is this reception method feasible in a real case? Wouldn't there be a recovery timing problem in practi... |
My question is a follow up of original question, specifically the answer provided by @Dan Boschen Dans answer.
The question is about converting floating point coefficients to fixed point signed numbers.
I am comparing Dan's answer to Surf VHDL Fixed Point which does the same thing differently.
I understand Dan's answer... |
I am transmitting an OFDM signal. According to 802.11a Standards, I am using Short and long training Preamble each has length of 160 samples for time and frequency synchronization.
According to Essam Sourour, Hussein El-Ghoroury and Dale McNeill: Frequency Offset Estimation and Correction.
Equations no. 4 and 5 for coa... |
Adobe Audition has a very interesting notch filter that have been very useful to me. It has the following parameters
According to their documentation, the chosen frequency is the center frequency and the width is either a second, fourth or sixth order, depending on the "Notch Width" option selected.
I want to recreate... |
Consider a MIMO system equipped with $N_t$ transmit antennas and $N_r$ receive antennas.
The received signal over $L$ snapshots are given by
$$Y = H X + Z,$$
where $X$ is the $N_t \times L$ transmited signal, $H$ is the $N_r \times N_t$ channel matrix to be estimated, and $Z$ is the $N_r \times L$ Gaussian noise with z... |
Different sets of eigenvectors for the Discrete Fourier Transform (DFT) are well-established.
Are there any potential practical implications associated with deriving explicit formulations for these DFT eigenvectors? Moreover, is there a specific advantage in identifying a real eigenbasis for the DFT?
|
I am reading from the internet and my professor's course on PAM. This course is not well-organized, and I would like to ask for book recommendations. I know that this modulation is divided into two parts, and the one that interests me is baseband modulation. It is based on sampling a modulation signal and then multiply... |
I have tried to evaluate the analog microphone (IMP23ABSU) built into the STEVAL-STWINKT1B with the FP-SNS-DATALOG1 (HS-Datalogger). The values were exported as "raw values" (this is an option that can be selected when executing the data logger script). According to the manufacturer, the unit of measurement is dB. Belo... |
I need a way that when I get voice and chunk it with librosa Library when I do this my voice change to Typeless type .
And I can not do any thing or do any progress in my voice .
Other wise I don't want to save each step of processing to have less I/O , instead I want to find a way to save the final step and keep whole... |
I wanted to understand why this text talks about applying the Fourier transform on H(f) to obtain h(t). I view Fourier transform as moving from the time or spatial domain to the frequency / spatial frequency domain.
Is it normal for Fourier transform and inverse Fourier transform to be used interchangeably?
the first i... |
Why does my dft plot of $\sin(x)$ look so strange? It goes to negative and then inverts closer to the edge.
x=0:0.1:10;
N=max(size(x));
y=sin(x);
plot(fftshift(real(fft(y)/N)));
|
I need to transfer data from A to B using audio as the medium, with a speaker at one end and a microphone at the other. Sometimes simply through the air, but it must also work through legacy phone lines and in environments where the bandwidth is low (phone lines using cables, old wireless networks such as 2G, etc).
Thi... |
New user to dsp.stackexchange here. I'd like to know what's a quick and effective way to take 2 impulse responses and create one that's equivalent to both of these "in series" for audio applications. Basically i want to write a function that has:
inputs: impulse responses ir_a, ir_b
output: impulse response ir_out
su... |
I am trying to understand scipy.signal.kaiserord example. The example says
We will use the Kaiser window method to design a lowpass FIR filter
for a signal that is sampled at 1000 Hz.
We want at least 65 dB rejection in the stop band, and in the pass
band the gain should vary no more than 0.5%.
We want a cutoff freque... |
I use a DBPSK modulation. Each modulation symbol is spread by a signal $c_i(t)$ which is different for each DBPSK symbol. The transmitted signal is given by: $s(t)=\sum_{i=1}^{N_s}d_ic_i(t-iT_s)$ where $N_s$ is the number of DBPSK symbols and $T_s$ is the symbol time.
My question: At reception, how to do the timing rec... |
Let $x_\mathrm{a}(t)$ be the analytic signal for real signal $x(t)$. I want to find an expression for $\mathscr{F}\{|x_\mathrm{a}(t)|^2\}(f)$ in terms of $x(t)$. The analytic signal can be written as $x_\mathrm{a}(t) = x(t)+j\hat{x}(t)$ where $\hat{x}(t)$ denotes the Hilbert transform of $x(t)$. So we have:
$$\begin{al... |
Suppose I have some real-valued signal $x\mapsto f(x)$. The amplitude of its Fourier transform $\mathcal{F}[f]$ then looks like a peak around the DC-term, decaying as we move towards higher frequencies. If however we pre-multiply with $g(x)=f(x)e^{i N(x)}$ where $N(x)$ is random noise, then the Fourier domain has infor... |
We know that the array manifold vector, also called the steering vector, for an M element array is given by:
$$ \mathbf{a}(\theta, \phi) = [e^{-j \omega \tau_{1}(\theta, \phi)}, e^{-j \omega \tau_{2}(\theta, \phi)}, \ldots, e^{-j \omega \tau_{M}(\theta, \phi)}]^{T} $$
Or
$$ \mathbf{a}(\theta, \phi) = [e^{-j \mathbf{\ka... |
Let's say I am generating pure sine oscillations with a laboratory instrument on which vibration frequency and amplitude are tunable (there are uncertainties +/- 1 % with my inputs), and I get my output data from accelerometers (also with white noise).
The question is how can I estimate the transfer function of the afo... |
Suppose:
$$
s(t) = \sin(2\pi{f_0}t)
$$
Suppose I'm sampling the signal with a sample frequency $f_s >2f_0$ .
However, every $M$ samples there is a dead-time of $L$ samples.
Traditionally, the (continuous) sampling function is just a delta comb function:
$$
p(t) = \sum_{n=-\infty}^\infty \delta(t-nT_s) , \quad T_s=1/f_s... |
EDITED AND UPDATED:
My signal is an impulse response. In the following figure:
The frequency spectrum of the impulse response (original signal) is shown by blue line.
The original signal is corrected using an amplitude correction term shown by orange line.
The corrected signal is represented with a green line. The ... |
I am currently using the output of a polynomial regression to match a pressure sensor's A/D output from pressure and temperature input with a control's pressure output. This approach, with terms up to the 4th degree, works well enough to match the control's output with a reasonable enough error level for my purposes; h... |
I've been studying a kind of "introduction" for a few days now to get to talking about equalizers and ISI, and I would like to know if I have understood the things correctly or if I got confused:
For example, in PAM modulation we know that if we choose to use a rectangular pulse as the channel input (and if we do not f... |
I am creating a signal consisting of a single-cycle sine wave in Python. This signal has 32 samples. I am using a sampling rate Fs of 48 kHz so the signal's frequency is 1500 Hz. The signal looks like this:
If I take the 32-point FFT (no zero-padding, no windowing) of this real time-domain signal, and I normalize the ... |
In this question we have a mathematical proof that the DC component of normalized discrete Fourier transform should be the same as the signal's arithmetic mean. However, in the following example I print the signal mean and magnitude of DC component, but they are different. What is the reason behind this?
import numpy a... |
"I would like to extract 'n' frames and obtain the corresponding spectrogram for those frames. I'm unsure if there is a specific duration, such as 20 frames equating to 15 milliseconds or something similar, that should be considered for this process."
|
I am studying ARMA processes. At the end of the course the professor told us that estimating the next sample in an arma process using past of length $p$ (so performing a projection of $X_t$ on $\text{span}(X_{t-1}, ..., X_{t-1-p})$ is the same as finding the least squares solution to $X_t = aX_{t-1} + bX_{t-2}...$, whi... |
I am working with Zadoff-Chu sequences in a Python simulation of an LTE-like communication system and need to understand the effects of carrier frequency offset (CFO) on these sequences. Specifically, I'm looking to derive an analytical expression for the cross-correlation between the original IDFT transformed Zadoff-C... |
I using a microphone to determine (air) leakage characteristics. The signals need to be filtered to reduce the noise level and i thought of using the Spectral Subtraction Technique. Since im having 4 leakage-free sampled signals im thinking to correlate all of them to 1 signal that captures most of the overall noise le... |
Hey everyone, I'm stuck on this problem and I've got an exam in a few days. I think the filter H is used here so as to avoid aliasing artifacts, and afterwards went on to the second question.
I've got using my notebook
$g(n) = \int_{-\frac{1}{2}}^{\frac{1}{2}} G(e^{i2\pi v}) \cdot e^{i2\pi vn} \, dv$
$g(n) = \int_{-\f... |
I'm trying to extract PCM wave sequences from different files (files that are NOT audio files but contain audio data, and other data all in one file), and while this code does extract the expected audio, it seems to introduce a very audible hiss/noise into the sequence.
I've tried to find similar topics but the only th... |
Following the sketch from Wikipedia 1, I am able to carry out DCT-II with DFT by manipulating the input signal:
import numpy as np
from scipy import fftpack as fftpack
from numpy import fft as fft
n = 5
x = 3 + np.arange(n) #my input signal
xpad = np.zeros(2 * n)
xpad[0:n] = x[:]
xpad[n:] = x[::-1]
X = fftpack.dct(x)
t... |
For perfect reconstruction, it seems that $2\times$ oversampling allows a prototype filter that ensure the spectrums of the subchannels overlap correctly.
In the gnuradio reference on this it seems the output after reconstruction is at $2f_s$, which makes sense as the oversampling process has twice the data.
So the que... |
I was reading about digital transmission, there are many steps namely
Analog to Digital conversion
Source coding
Channel coding
Line coding
Pulse shaping
Modulation
Multiple access techniques
Out of curiosity, I tried to find information about what are the techniques used in ethernet (standard Gigabit) for 2nd to 7th... |
Suppose I have a signal $\mathbf{x}\in \mathbb{C}^{N}$ and a digital filter with impulse response $\mathbf{h}\in\mathbb{C}^L$, where $L<N$. If we pass the signal through the filter, the output will be $\mathbf{y}\in\mathbb{C}^{N+L-1}$. My question is: if I want to truncate the output vector to be the same length as $\m... |
I have some doubt about the upfirdn built-in function in MATLAB. First I generate the root raised cosine transmit filter as
tRRC = rcosdesign(0.25, 20,4,'sqrt');
which gives back 81 coefficients. I then pass a signal $\mathbf{x}$ of length $N$ through this filter using the upfirdn as
txFilteredSignal = upfirdn(x, tRRC... |
I have a program that allows me to record wav and get the FFT all at the same time.
However, the FFT I have I can't export it to work with. So I need to make a script to get it from the wav.
Can anyone give me a hand? The sample rate is 10kHz, the max frequency is 5000Hz, the resolution is 3.125 seconds, with a Hanning... |
In this article "Sampling: What Nyquist Didn't Say, and What to Do About It" from Wescott, the author shows in Figure 6, the plot of the frequency spectrum of a signal which goes through a LPF before sampling and its aliased component. He says:
You can see that this is not going to give good performance. In fact, if yo... |
I am trying to estimate CFO in LoRa chirp signal (preamble part). I have seen the discussion about CFO on this forum but it is mainly related to CFO estimation in OFDM.
I want to estimate the CFO in a chirp signal using FFT method. Simply taking the FFT of the signal and then taking the Frequency index with highest fre... |
I am trying to understand the following example:
I understand all of them except the R2 for NOMA. What is the value of h2? and what is the value of N2? Also do we use the freq BW anywhere in these formulas?
Thanks
|
I want to derive the PSD of $a \cdot x(t)+b$ if the PSD of $x(t)$ is known. Here $a$ and $b$ are constant. The signal $x(t)$ is a zero mean stationary random variable. Thanks in advance.
|
After reading a few posts, like this. I know that arbitrary LTI systems always have complex exponential eigenfunctions. And that for specific LTI systems you can also have other types of eigenfunctions. But for arbitrary LTI systems are complex exponentials the only eigenfunction that always exists? Preferably with a p... |
I'm looking for the inverse $\mathcal{Z}$-transform of a shifted Dirac delta function in the $z$ domain, i.e.
$$ x[n] = \mathcal{Z}^{-1} \{ \delta(z - z_{0}) \} = \ldots $$
Does an analytic/closed-form expression exist? I know that $ \mathcal{Z} \{ \delta[n] \} = 1 $ so from duality I suppose $\mathcal{Z}^{-1} \{ \de... |
I am a senior in high-school and am currently trying to conduct an exploration on Fourier Analysis, specifically using the Discrete Fourier Transform to analyze a chord played on my piano. Basically, I'd play a chord of 3 notes, deduce the different frequencies of the sound wave, and estimate the chord played. I am hav... |
the function c2d allows to convert a continuous laplace transfer function to a discrete z-transform transfer function. The base method is the Zero Order Holder. In example:
s=tf('s');
P=3/(3+s);
Pd=c2d(P,0.1); % 0.1 is the sampling time
Produces
$$ \frac{0.25918178 }{z - 0.74081822068} $$
$0.74081822068 = e^{... |
I provide a short scheme for forward explanation:
There are two sources: noise signal source and anti-noise signal source (2). Noise signal travels through primary path $h_{N_P}$ to microphone (4). At the same time noise travels to adaptive filter $h_N$. Then anti-noise $-y(k)$ travels through secondary part $h_{N_S}$... |
I have constructed a high pass filter using scipy:
l_freq = 0.1
fs = 500
normal_cutoff = l_freq / (0.5 * fs)
sos = butter(4, normal_cutoff, btype = 'high', output = 'sos')
scipy_iir_continuous_data = sosfilt(sos, data)
As well as a bandpass filter using scipy:
l_freq = 1
h_freq = 4
normal_cutoff = [l_freq / (0.5 * fs)... |
In LTE the Primary Synchronisation Signal (PSS) can be detected by taking a correlation with the known 3 zadoff chu sequences (roots: 25,29,34). Once the peak has been detected the receiver knows which sequence has been sent. The sequence is placed at the middle 63 subcarriers (DC with 0 in the middle). Assuming $x_{u}... |
I want to design a digital allpass IIR filter with the following transfer function.
$$
H(\omega) =
1 \textrm{ for } \omega < \omega_1 \textrm{ or } \omega > \omega_2
$$
and
$$
H(\omega) =
-1 \textrm{ for } \omega > \omega_1 \textrm{ and } \omega < \omega_2
$$
I assume that higher-order filters are needed for a stee... |
I'd like to start by saying that I'm pretty clueless about controllers, control theory, or really anything to do with this subject. I'm a programmer, but I'm part of a project involving a 6 degree of freedom robotic arm with Dynamixel MX-106 motors.
I've created a PID controller for each motor of the robot, using C++ a... |
Suppose $X$ is a real-valued N-dimensional Gaussian vector, $X \sim \mathcal{N}(\mathbf{0}, C_X)$. The discrete Fourier transform can be obtained by left-multiplying with the unitary DFT matrix, i.e. $\hat{X} = W X$ (where $W$ is defined as in this page). What is the distribution of $\hat{X}$?
My attempt to answer this... |
Let the following T-periodic signal :
I found that
$$ x(t) = \frac{A \cdot t}{T} \qquad 0 \le t < T $$
and its Fourier series is :
$$ x(t) = \frac{A}{2} - \frac{A}{\pi} \sum_{n=1}^\infty \frac{\sin(n \omega t)}{n} $$
How do I determine the magnitude and phase spectrum?
|
So I am pretty familiar with FFTs and FFT windows. I know that in general there is a tradeoff between mainlobe width and sidelobe levels.
However, I recently stumbled across this discussion, where someone claims to have created a window or FFT method that achieves both very good mainlobe resolution AND very low sidelob... |
Consider that i have N signals of electricity energy consumption (from N different consumers) that ha 24 measurements over a single day each (1 measure per hour). These original signals are called x0 and labeled as class 0. So, a matrix with N rows and 25 columns (24 measurments + 1 class label)
Using these x0 signals... |
I'm new to GNU Radio, and I was watching this video However, I didn't quite understand the purpose of the FIR Filter. I know what decimation or downsampling is because I have the book "Digital Signal Processing" by John Proakis and Dimitris Manolakis. However, I didn't grasp how FIR Filters are explained in the book. ... |
Here is the image of a plugin distortion I have applied to a sine wave.
The sound characteristic is heavily distorted and digital but still good sounding bass wave.
Any ideas what kind of distortion this is and where to look for implementation? I currently implemented a wave shape distortion it squares the wave and doe... |
For an random signal $X$ (does not matter if it is correlated or uncorrelated), given lag $τ$. If white noise is added to this signal, then can the summation of them be a correlated signal? Also, even if these two are individually uncorrelated. Could somehow be related mutually?
|
I am a student working with digital circuits. I just started a term project with someone else working on an ADC. I need to design a IIR filter in verilog.
If the output of an ADC represents 1V by setting all output bits to '1' and 0V by all bits to '0', and I wanted to use a fixed point scheme for the filter coefficien... |
How does LBP work? To learn that, I decided to calculate by hand for the matrix but the problem is about the rotation pixel over the image based on the radius and neighbor of LBP. I do it clockwise and top most left then top most right, but the result is still different from the result in "skimage" lib.
With the examp... |
I am interested in the audio classification problem. You can think of the problem as ML, but my question is from a signal processing perspective. I am questioning the classification performance of MFCC in real-time (data is coming as streams) applications.
Let me summarize the process.After labeling the audio recording... |
Suppose I want to implement only via software on matlab a DFE.
I am studying the decision feedback equalizer but some slightly more practical concepts are not very clear to me from the theory.
Specifically:
How do I choose the number of taps (feedback and feedforward) to use in the equalizer? Are they chosen based on ... |
Sound attenuation can occur due to various phenomena. The most known are
Distance: For a point source the sound pressure decreases by 6dB with each doubling of the distance to the sound source.
Atmospheric absorption: Depending on the air pressure, humidity and tempreature sound attenutate increasingly faster if the f... |
I'm working on developing an audio device testing suite (for speakers and like). I'm looking for approaches to detecting two types of noise: static such as white or pink noise and artifacts such as crackling, 'wheezing' etc. I'm kind of new to the DSP field and was wondering how I should approach this task? The plan is... |
I am practicing Digital communication (Baseband demodulation/Detection) when I encounter this question.
An analog signal is PCM formatted and transmitted using binary waveforms over a
channel that is bandlimited to 100 kHz. Assume that 32 quantization levels are used
and that the overall equivalent transfer function is... |
I'm interested in identifying a square wave signal from recorded audio. This subject may be a complex problem so I would like to present the problem where I'm currently stuck at.
I recorded the audio at 44100 sampling rate, and applied FFT on the audio at the sampling rate, in ~0.1s intervals. For 10 seconds.
A 440Hz s... |
I don’t understand how this phenomenon appear in OQPSK carrier recovery loop. I found nothing on google. Is it a normal phenomenon in OQPSK loop or specific? Why does it get reversed I and Q? In the figure(b), why does the sampling point change?
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