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I have an ambisonic multitrack with several spatialized sound effects (I'm using 1st order ambisonics). However, I'd like to add some mono sources (e.g. narration) to my project. My initial thought was just to add the mono source to each channel with the same amplitude, but that doesn't seem reasonable when I see how t... |
For my master's thesis I collected CSI with 4 WiFi network cards (Intel AX210) for three different persons for 102 positions (spaced roughly 30cm apart).
The covered area was around 10m x 7m and in each corner {(0,0), (10,0), (0,7), (10,7)} I placed a desktop computer with an AX210, collecting CSI from a router that wa... |
I'm wondering what's an algorithm I can use to segment shapes on a binary image. Below is an example of the wanted result:
Thanks
|
My question is about an example in Adaptive Filter Theory, Haykin 4th ed (refer figures 1.8 and 1.10 of Haykin). We want to determine the region for asymptotic stationarity of an AR process in the 2D space of $\rho_1$ and $\rho_2$. Process model (characteristic equation): $$1+a_1z^{-1}+a_2z^{-2}=0$$
For asymptotic stat... |
So I have a fundamental question (for which I can't find a good answer anywhere). Say we have two images (A and B) and their mixed image (C) as follows:
Now say that I've been given Image (A) and Image (C) or Image (B) and Image (C). What are some algorithmic methods that will allow me to separate Image A from Image C... |
When running the MATLAB QPSK transmitter and receiver example (found here: QPSK Tx and Rx Example) you can see on the plot of the spectrum that the bandwidth is ~80kHz.
If we double click on Bits Generation we can see how each frame is constructed.
For a symbol rate of 50kHz we get $\ T_s = 1/50000 $
From the setting... |
have a distorted (blurred image) and I perform several deconvolution algorithms on it to sharpen it (Richardson lucy, regularized inverse filter, tikhonov miller). And i want to measure each performance by calculating the MSE (mean square error). However I don’t have a reference or an ideal image where I can compare my... |
I have an experimental PSF (point spread function) of an optical system for remote sensing images.
However, it is asymmetrical and has some sidelobes as below.
And I want to perform deconvolution to sharpen the images.
Is it possible to use this PSF as it is to sharpen the images?
Or should there be some modifications... |
I am studying the DSSS method and I have a question. For example, the emitter and the receiver use a $c_1$ sequence for spreading of length $L$. An eavesdropper uses a $c_2 $sequence of length $L$. We have $c_1 \neq c_2$. The transmission parameters are known to the eavesdropper: modulation, bandwidth, carrier frequenc... |
I want to transform the continuous signal $$u(t)=u_0+\hat{u}\cdot \cos(2\pi \frac{t}{T_0})$$ to a discrete signal. The discrete signal should contain $N$ values and exactly $m$ periods (starting with $T_0$). Furthermore is given a (yet) undefined sampling period as $T_s$. How long must I sample this function? How can I... |
I have a mathematics problem that can be related to Signal Processing. Let $f$ be a real function, striclty increasing, striclty convex on $(-\infty,0)$, strictly concave on $(0,\infty)$. We furthermore assume $f$ to be smooth and integrable w.r.t gaussian kernel.
My problem is the following, can one show that for $\si... |
I am looking for the most precise way to determine the onset time of a
very fast pulse sampled at 5 GHz coming from a physics detector.
Here is an example of my signals:
I already tried different things using the unfiltered and smoothed waveforms including:
Different ways of looking at the second derivative in the ho... |
I have panchromatic images taken from a remote senescing device, and it is downloaded as a raw which is dark/ underexposed (as 1st pic), hence it needs enhancement in brightness and contrast to be readable. I usually enhance it visually using auto level then modify the parameters to be not that highly contrasted. (as 2... |
It seems I've forgotten much of the details of signal processing theory when attempting to reconstruct algorithms for extracting FFT-like amplitude/phase information from examples online. I was particularly lost when I could not reconstruct the negative sign of the amplitude/phase of one of the sinusoids I had summed ... |
I have a drivetrain setup where I have a ~3Hz vibration which I can feel quite strongly in the seat of my pants, but can't seem to see in accelerometer data.
Below is a diagram of the system. There is a 4-cycle 4-cylinder Rotax 914 engine which drives a 2.45:1 gearbox which in turn drives a driveshaft which in turn dri... |
I'm trying to compute a basic spacecraft link budget, and could use some help with understanding/sourcing minimum SNR requirements for demod of GFSK encoded signals. I'm familiar with the DVB-S2X standard and using the information in "Table 1: Performance at Quasi Error Free FER=10-5 (AWGN Channel)" to perform link bud... |
I have some noise density that consists of 1/f noise and white noise like so:
I then select an arbitrary sampling frequency $f_S$ and data length $n$ and compute a time domain noise data vector like this (for $f_S=$12.5 Hz and $n=$1e6):
Next I compute the Allan deviation for a bunch of different $\tau$ values (this i... |
As a math student interested in signal processing, particularly with wavelets, I am seeking recommendations for introductory books on the topic.
My current knowledge is limited to Fourier transforms covered in a standard calculus course. I would prefer books that include a significant number of practice problems.
|
I am modelling radar scenarios in python and am trying to make a model which retains phase information. To do so, I'm envisioning something which works like this:
Generate samples of an LFM waveform at RF in a numpy array
Calculate the distance to the target (two way)
Divide the time take for the echo to return by the... |
In the following equation:
$$B_{\rm rms}^2 = \frac{\displaystyle\int_{-\infty}^\infty f^2\lvert G(f)\rvert^2df}{\displaystyle\int_{-\infty}^\infty \lvert G(f)\rvert^2df}.$$
It is not clear why do we divide the RMS of the signal by its energy, I tried looking for answers in several books but did not find any explination... |
I have a problem with a DSSS communication. On reception, if I use another sequence for despreading, I have a BER equal to 0. However, this is impossible. Do you know where my mistake is?
This is my code:
clear all;close all;clc;
%% Data
Ba=20e3; %bandwidth
fe=100e3; %sampling frequency
R=fe/Ba;
SF=32; %spreading fact... |
I am working on a project writing a realtime FFT / phase vocoder based pitch shifting algorithm in C++. In development, I notice that shifting down has much better audible pitch tracking and resolution than shifting up, using the same overlap and bin settings. In particular I notice pitch up starts sounding very AM / r... |
I'm reading Signals and Systems by Oppenheim, and in the section 10.7.2 about stability there's a conclusion I don't understand:
For the impulse response h[n] of a discrete time system he summarizes:
BIBO stability $\iff$ h[n] absolutely integrable $\Rightarrow$ Fourier-Transform of h[n] exists $\iff$ Unit circle is wi... |
I am completely new to signal processing! Please explain as if speaking to someone who knows nothing more than linear algebra and what a Fourier transform is defined as.
I am working with an 8-month timeseries of data, sampled at 15-min intervals (at least for the present variable of interest). The data is from a real-... |
currently I'm measuring an oscillator with a laser vibrometer. The oscillator is excited with a shaker and a burst chirp. The vibrometer's output is the time signal of measured point on the oscillator. I then use matlab and fft() to get the spectrum.
My goal is:
Characterizing the oscillator with the peak amplitude and... |
Inspired by this post and this video I have been playing around with different methods in Matlab to do spectral analysis. In the past I have simpy used digital fourier transforms and played around with different window types, window lengths, step sizes and averaging.
Now I compared for different signals that approach w... |
In order to downsample a signal sampled at 48KHz, I implemented an anti-aliasing filter. An Elliptic LPF with a cutoff at 16KHz and order of 10. Everything looks OK until the input to this filter is a clipped signal, say a clipped sinewave. In that case, the output of the filter has a larger magnitude than the input (l... |
I am removing low frequency noise from a signal using Matlab using the DWT and then reconstruction at a specific level:
I am removing the approximation signal at level 10 which approximates the low frequency noise
signal sampled at 500 hz
[a,b] = wavedec(signal,13,'db8'); % DWT to max level
lfnoise = wrcoef('a',a,b... |
Possible a duplicate but I could not find a direct answer. I used a sound card (Focusrite, at 96 kHz sampling frequency) connected to a calibrated hydrophone to record an ultrasonic signal, with a known frequency at known power. I have all specification of the hydrophone including the frequency sensitivity. The gain on... |
I would like to simulate the process of transmitting a binary phase shift keying (BPSK) signal from start to finish, going through each step over additive white Gaussian noise (AWGN) channel. I know how to start, but I'm not sure how to do some parts in the channel and receiver
To start suppose I have these parameters ... |
Lets say I have 2 vectors (1D signals that are sigmoids): $s$ and $m$, both related through the relation: $m = s * r$, my goal here is to recover the vector $r$ (should look like a gaussian).
I tried to use Fourier transforms but it did not work very well (see here: https://dsp.stackexchange.com/questions/88801/convolu... |
I am using DSSS (Direct Sequence Spread Spectrum) communication. The mathematical expression for the spreading sequence is given by: $g(t)=\sum_{i=0}^{L-1} c_i g(t-lT_c)$ where $c=[c_0,...,c_L]$ is the spreading sequence and $g(t)$ is the shaping filter.
My question: If for each symbol of the message, I use a different... |
Use of "Audacity"for teaching undergraduate labs of digital signal processing for electrical engineering students?
My above realted linked question has got some good answers.
Traditionally MATLAB and python are used for signal processing teaching but i want to know for case of basic/entry level so as to motivate studen... |
I'm trying to implement a procedure from a paper.
There the authors use two equations to transform a fft transformed music audio file to a cent-spectogram.
Equation 1:
$$ f_{cent} = 1200 \log_{2} \left( \frac{f_{Hz}}{440 \left( \sqrt[1200]{2} \right)^{-5700}} \right) \tag{1} \label{1}$$
Equation 2:
$$ X \left( k \right... |
Intro
My question is related to a previous one linked here.
I am interested in non-linear phase FIR filters with a specific desired phase response.
After I tried the options in the linked question I also stumbled upon the idea to separately define weights for phase and magnitude responses developed in Matt's PhD thesis... |
This is a repost from Mathematics Stack Exchange due to lack of answers.
A minor is a determinant of a square matrix formed from a bigger square matrix by removing rows and columns. A minor having value zero means that its associated submatrix is non invertible.
The amount of zero valued minors of a size n discrete Fou... |
I have an UFF file which consists of Vibration data. The objective is to convert the timewaveform signal to frequency domain.
I have written a python script to convert the timewave form signal to frequency domain using scipy.fft. I have an UFF file. I am importing that uff using pyuff. I am validating the output of my ... |
I am implementing the exponential/logarithmic sine sweep approach to impulse response measurement (as presented by Farina - http://pcfarina.eng.unipr.it/Public/Papers/134-AES00.PDF).
My query is regarding amplitude correction of the inverse sweep. I have found a lot of differing information about what correction should... |
How to convert s-parameters in Matlab's frequency domain to the time domain with code example?
|
Consider the design of a Savitzky-Golay filter of window length 101 and (high) polynomial order 20. Using scipy version 1.10.1, the filter coefficients can be obtained in python as:
from scipy import signal
h = signal.savgol_coeffs(window_length=101, polyorder=20)
leading to
For reference, in Matlab r2022b
order=20;... |
I was trying to think of all possible constraints or factors that needs to be taken into consideration while designing an OFDM-based wireless communication system (For example, LTE, 5G technologies etc).
Assume the bandwidth is $B$. The coherence bandwidth and the coherence time of the channel are $B_c$ and $T_c$ r... |
I am using CWT to obtain scalograms of ECG signals as the one below:
In my research, most articles contain a pre-processing step before performing CWT (that usually includes signal normalization and digital filtering).
The filtering usually consists on a bandpass filter (for example 0.5 Hz - 100 Hz for the ECG).
My qu... |
I am trying to understand the spectrum of the sound of a cuckoo. It is 13-second audio clip and its download link is
https://file.io/d4nqddrb1NNY
Actually, I have a confusion, I have attached a snap of research gate where sound levels are mentioned with + db sign. I have also attached a snap of the spectrum of this aud... |
I want to achieve OFDM modulation combined with a chirp signal (for example a LFM signal). Is it correct to do as below (this is a global scheme):
Mapping
Spreading with chirp signal
OFDM Modulation
Channel
OFDM Demodulation
Despreading with chirp signal
Demapping
|
Pointing a microphone to a connected loudspeaker creates a feedback path and usually results in acoustic resonance, which manifests in a sustained oscillation: https://youtu.be/_XTtjZ8aZbc. A similar effect can be observed with guitar feedback, where the guitar coils pick up the sound coming out of the amplifier.
This ... |
Can anyone recommend a good C library for SDR? To be more specific I just need a function in C to compute coefficients for CIC compensation filter (I need to do it in real time, so Matlab is not an option)
Thanks.
|
We've been working on a machine learning project for pattern recognition, using time-domain features such as kurtosis, mean, standard deviation, variance, skewness, and peak-to-peak values.
Background:
Initially, we trained our data after applying a high-pass filter at 1 kHz. The results were satisfactory.
Upon perform... |
I want to generate an ECG signal with PQRS signatures by coding. The signal generated must be repeating in nature. Although I have come across various toolboxes that generate ECG signals, I want to generate them using mathematical equations. Any help would be highly appreciated.
|
Objective: Finding the BPM of a short drum loop.
Approach:
% Step 1: Get audio signal and sample rate
[audio_signal, Fs] = audioread(".\90-bpm.wav"); % Drum loop recorded at 90bpm
audio_signal = audio_signal(:,1); % Only use 1 channel
short_signal = audio_signal(1:1e5,:); % Get short version of sample
% Step 2: Get th... |
I have an issue regarding a sampling process of a pulse train in an image sensor based on events. Basically, these are a family of image sensors in which each pixel outputs a train of pulses, and the frequency of the pulses encodes the illumination of the pixel. The higher the frequency, the brighter the light hitting ... |
I want to make an equalizer set to similar frequency cut-boost according to the results from the graph from hearing test, where an audiometrist will play tones of various frequencies at various volumes to check the hearing levels for various frequencies and then an 8-band equalizer graph will be programmed in the heari... |
If I have a signal that is lower than the noise power, I may want to use signal averaging over time to increase SNR.
However, when the signal has been obtained after digitization, there is quantization noise and signal averaging is not possible anymore.
https://en.wikipedia.org/wiki/Signal_averaging#Correlated_noise
Ar... |
Lets say I have 2 vectors (1D signals that are sigmoids): $s$ and $m$, both related through the relation: $m = s * r$, my goal here is to recover the vector $r$ (should be a gaussian $\rightarrow$ gives the $m$ sigmoid a smoother/steeper slope depending on $\sigma$).
I tried to use Fourier transforms but it did not wor... |
Say, I have $x[n]$ and $y[n]$ be periodic sequences of period $N$. Let us define $z[n] =x[n] \ast y[n]$ where $\ast$ denote the circular convolution operation.
What is $x[n+\delta] \ast y[n+\delta]?$
My answer:
Taking FFT: $Z[k] =X[k]Y[k]$. Now, FFT of $x[n+\delta] =e^{j2\pi\delta n/N}X[k]$ and $y[n+\delta] =e^{j2... |
In ssqueezepy, by finding a certain scale is it possible to do an icwt only for the scales above (or bellow) ? For exemple if I have a double sin (with $f_1$ and $f_2$), by taking the frequency $f_c$ such as $f_1$ < $f_c$ < $f_2$, going back to scales ($a_1$, $a_2$ and $a_c$), can we integrate the scales from $a_c$ (or... |
I need to produce a signal of $\sin(2\pi\cdot 60\cdot10^6 t)$, where $t$ in sec, with sample rate of $2^{28}$ Hz.
Since the signal has period of $\frac{1}{60\cdot 10^6}$ sec can I somehow sample only few samples of 1 period and continuously play it many times ?
My problem is the transfer of memory from computer to devi... |
I have a time series - let's say I'm given one new sample (a real value) every second.
I could take a moving average of the last 50
samples or I could use exponential smoothing of the samples with a time constant of 50
. The resulting graphs of the "output" would be fairly similar. The "current output" of the algorithm... |
I need to oversample my OFDM signal, however I have issues understanding which OFDM oversampling method is the recommended one.
First method :
Before taking the IFFT, the OFDM subcarriers are padded with $(\text{OSR} - 1) \times n_\text{subcarriers}$ additional null subcarriers, then the IFFT is taken.
Therefore, we ha... |
The below is a graph my designed constant current source.
You can clearly see that there are some noise in the values.
But when I take the FFT, the noise is not actually showing in graph.
Could some tell me why is it so?
I am newbie here. Please spare me if I am wrong.
Below is my python code for FFT.
<
import numpy... |
I'm reading a paper on how construct isotropic laplacian filter, and perhaps because it's an old paper, the notation in it really bothers me a lot. So can someone please explain it to me? For example, the notations I have highlighted in redline in the screenshot, such as what G and U mean and how to interpret these equ... |
I read quite a few posts regarding the autocorrelation matrix and although some of them were of the LMS adaptive algorithm application, I didn't find an answer to my question.
How do I calculate R; that is, the expected value of the matrix ( x' * x ) shown above, with x being a column vector. If x(t) is a sine wave, ... |
I have two squarewaves of the same frequency but with different dutycycle. How can I calculate the phase difference between the two?
I know that for sinewaves one can use the correlation coefficient, but I am not sure how to do it with squarewaves, especially if the duty cycle is different.
I was thinking maybe it is p... |
Consider a discrete signal $x$ of size $n$, and write $c$ its discrete Fourier transform. Unless I'm mistaken, the relations between the two look something like:
\begin{split}c_k = \sum_{j=0}^{n-1} x_j \exp\left(-2i\pi k\tfrac{j}{n} \right) \\
x_k = \frac{1}{n}\sum_{j=0}^{n-1} c_j \exp\left(2i\pi k\tfrac{j}{n} \right) ... |
Here's some code.
Fs = 10000;
t = linspace(0,10,10*Fs);
f = 100;
i = zeros(size(t));
imp = [0 0.12 -0.2 0.1 1 -0.7 -0.2 0.1 0];
impTime = 0.2;
nSamp = impTime*Fs;
impFull = linspace(0,impTime,nSamp);
impUse = interp1(linspace(0,impTime,length(imp)),imp,impFull,'spline');
plot(impFull,impUse)
rate = 2;
nCycles = 10/... |
I'm trying to understand how to make Sample rate change by a non integer factor say $K=1.0012$
In real hardware I will be using cubic polynomial but for now I have chosen a linear interpolator. I have calculated the Lagrange coefficients:
Let's simplify my case and assume that $K=\frac{4}{3}$
So $K_{inv} = \frac{1}{K}... |
I am wondering if anyone knows of situations where one would need to accurately estimate the power spectrum of a signal at a frequency where the power is known to be very low, much lower than at other regions of the power spectrum? Suppose the spectral dynamic range of the power spectrum was greater that 100 dB or so, ... |
I am studying a pair of AC signals for voltage and current. In a long run, I found a distortion happening as shown in the attached picture. I want to understand what could possibly have happened in the signal from a mathematical point of view.
The base signal is $y = A\cos(100\pi)$
What kind of other signal added to th... |
I'm reading "Understanding Digital Signal Processing, 3rd Edition" by Richard Lyons. Chapter 9 derives Hilbert transform impulse response by defining it in frequency domain first and then taking inverse Fourier transform. I believe I could understand most of the derivation except one part.
It is stated that in order to... |
I am working on a problem that requires interpolating between two music files. One music file is very similar to a delayed version of the other.
I would like the 'average' of the two signals to sound like the music delayed an intermediate amount, with mixing between some other properties (like the frequency profile)
If... |
I need to perform an FFT on a signal sampled with 20 kHz and a measurement time of about 10 seconds. The signal contains frequencies of up to 2 kHz but I am mainly interested in the bandwidth of 0 to 300 Hz.
Just applying Matlab's fft function gives me an unnecessarily high frequency resolution that results in a rather... |
If this is not the right StackExchange group, please advice.
I seem to have the same problem as 1 (2019, on the miniDSP forum, unanswered) with the NotWired NW-AUD-ICS52000 boards. I have had no other boards with which to test the TDK InvenSense ICS-52000 mics.
The NotWired NW-AUD-ICS52000 board is at 3, and the TDK In... |
In the MC-CDMA (to simplify with $1$ user), the spreading sequence is a sequence of -1 and 1. I would like to know if it is possible to replace this sequence with a signal $s(t)$ of bandwidth $B$ to do the spreading.
|
Antoni uses the following definition of white process of order $p$:
a process whose all cumulants up to order p are such that
$$\text{Cum}\left[X(t),X(t-\tau_1)\cdots,X(t-\tau_{r-1})\right]=C_{rX}\delta(\tau_{1})\cdots\delta(\tau_{r-1})\forall
> r\le p$$
Where $X(t)$ is a random process of time.
I can tell very littl... |
I'm trying to wrap my head around the process of selecting DS-CDMA spreading codes, and I'm having trouble understanding the tradeoffs of longer codes. For all scenarios described below, I'm using a Walsh matrix to select the orthogonal vectors, and we assume that I'm transmitting on a 1MHz channel and I'm using BPSK w... |
If I've got a bunch of short .wav files, 16 bit linear pcm, and an incoming stream of audio that's been run through a voice codec (g.722, g.711, or others) and then converted back to pcm, what's the right algorithmic tool to detect when one of the wav samples is present in the audio?
The context for this is automated t... |
I'm new to the field of Machine Learning and currently watching some lectures on speech recognition. I'm attempting to write a simple class implementation for a speech recognition model using Hidden Markov Models with Python's hmmlearn library. I have a few questions:
If I understand correctly, should I consider the hi... |
I know that for a given signal $s(t)$ and a given delay $ \tau $, by the shift theorem:
$$ \mathcal{F}\{s(t-\tau)(f) \} = e^{-j 2 \pi f \tau} \mathcal{F}\{s(t)(f) \} \tag{1} $$
However, when I try to derive the Fourier Transform of a delayed LFM chirp, written as:
$$ s_{LFM}(t-\tau) = A e^{j (\omega_o (t-\tau) + \alph... |
Suppose I have the $N\times 1$ vector $\mathbf{x}$ that contains binary phase shift keying (BPSK) symbols each with power $P_s$ Watts.
At the transmitter: I want to convert these symbols into samples using rectangular pulse shaping, where each symbol is repeated $J$ times. My question for this part is: the symbol powe... |
I've got a set of measured data (amplitude and phase responses) of a filter. And I want to normalize its gain to be unit (equal to 1) for further processing. I know that we can convert to time-domain coefficients and normalize by dividing to its sum of all coefficients.
The question is could we normalize the gain using... |
This is related to this post: https://engineering.stackexchange.com/questions/56050/obtaining-the-open-loop-gain-estiamte-the-gain-and-phase-frequency-response
Using python, I would like to write a method which estimates the transfer function similar to the MATLAB tfestimate(...) found here: https://www.mathworks.com/h... |
I have a fundamental question about the discrete-time Fourier transform. I used two methods but got two results.
Background knowledge
The discrete time signal is given by:
$$x[ n ] = x[ {n{T_s}} ] = x( t )\sum\limits_{n = - \infty }^\infty {\delta ( {t - n{T_s}} )} = x(t)\delta_s(t)$$ where $T_s=\frac{1}{f_s}$
The de... |
I have ECG signal sampled at 100Hz & using Python to remove noise, and generate a template signal from the collection of ECG from a total of almost 4700 signals. I have removed the noise from ECG using PCA and now looking to 'stack' together signals on top of each other to generate a template signal which will serve as... |
(I will be referring to info from these slides in this post, for your reference.)
Consider first for comparison a typical CT low-pass filter, without any sampling, for example implemented by the RC circuit below.
Here the CT input source $v_n(t)$ with transform $V_n(s)$ is the noise generated by the resistor. The nois... |
In Fundamentals Of Statistical Signal Processing: Estimation Theory page 19, Kay mentions a biased mean square error estimator for $\mu$ where the samples $x\sim\text{N}(\mu, \sigma^2)$.
The suggested estimator is $$\hat{\mu}=\frac{a}{N}\sum_{i=0}^{N}x_i$$ where optimal $a$ is given by $$a=\frac{\mu^2}{\mu^2 + \sigma^2... |
I am following a paper on computationally modeling optical phenomena within a lens. (https://resources.mpi-inf.mpg.de/lensflareRendering/)
At some point they use the Fourier power spectrum of an aperture image to generate an image of the refractive phenomenon known as a "starburst". From the paper:
Assuming a uniform ... |
Once upon a time desktop PCs had a 2D graphics card and a 3D graphics card.
A pass through cable connected the 2D into the 3D card, which combined and outputted the two signals over layed.
If I had, say, two outputs, both the composite signal for old analogue TV, can you just combine the two signals together and, as lo... |
How can one define Continuous White Noise in a coherent way?
Is there a way to derive it Mathematically?
Specifically, is there a way to define it which will works as the model in Signal Processing yet will obey all Mathematical properties of a Random Process?
|
I tried to extract a pure signal from the noisy signal using a lock-in amplifier with the help of python code.
The output is from a photodetector circuit.
These are the reference signal, expected voltage Vz noisy output and extracted signal.
As you can see from the output, the extracted signal is not having the same... |
I have a noisy signal which is the voltage output from a photodetector circuit. I have researched about extracting signals, and found that lock-in amplifiers are one of the main approaches for the photodetector circuit.
I have a laser which is on for 5s and off for 5s. I am measuring the voltage from a photodetector ci... |
Let $f$ and $g$ be two images. With two functions $a$ and $b$ the brightness difference of the images is to be estimated with the following functional
$E(a,b) = \int_{\Omega} (((a \cdot f+b)-g)^2 + \alpha a_x^2 + \beta b_x^2) dx$
How can I choose $\alpha$ and $\beta$ so that instead of the lokal affine brightness diffe... |
Let $x(t)$ a signal of duration $T$, the sampling frequency is $f_s$.
I know the signal $x(t+\tau)$ orresponds to a signal delayed by a delay $\tau$. In Matlab, the signal $x(t+\tau)$ is given by: $[0,...,0,x(1),...,x(N)]$ where $\texttt{size}([0,...,0],2)=\tau f_s$ and $\texttt{size}([x(1),...,x(N)],2)=Tf_s$.
My prob... |
I read one equation from a technical document, i.e., assuming the time-domain white Gaussian noise samples with variance $\sigma_{TD}^{2}$ in dB are sampled using ADC with sampling rate $f_{s}^{ADC}$, then if they are downsampled to the sampling frequency $f_{s}$ (e.g., with a downsampling rate of 2 and appropriate fil... |
The direct-digital phase noise measurement technique is described in "Direct-Digital Phase-Noise Measurement" by Grove (DOI:10.1109/FREQ.2004.1418466, PDF link). The basic architecture of this technique, for 1 DUT and 1 reference channel, is shown in the following two block diagrams.
In the 2nd image, the "Sample and... |
I am quite new to the VHDL domain and totally new to using its ip core. I'm trying to implement an 8 point FFT using the ip core FFT 7.1 but my results are a bit off and I cannot figure out why that is.
In python:
import numpy as np
x = [511, 1013, 310, 94, 890, 762, 20, 488]
[print(i) for i in np.fft.fft(x)]
Result:
... |
How do I reduce a block diagram with just a line as a feedback loop, I dont get how it adds K to the denominator.
The bottom equation is supposed to be the answer.
|
I am trying to implementation Cfar algorithm to our radar signal processing algorithm. I obtained some output but I want to be sure that is right. Cfar algorithm's output below like, I'm interpreting these outputs like this, where it should just specify something at the detected distance and speed, there's just a gap i... |
Consider the following image:
This is an image of a set of contours (the "contour image") that was drawn by a human using a mouse on another image (the "original image"). FWIW, the purpose of hand-drawing contour images like this is to create a dataset for eventually training a CNN to segment the images like the origi... |
I have some simulated outputs from a model of a sensor that a teammate wrote. I'm attempting to verify, using Allan deviation, that random walk and bias instability were modelled correctly.
To do this, I'm following this mathworks tutorial: https://www.mathworks.com/help/fusion/ug/inertial-sensor-noise-analysis-using-a... |
I am trying to get a better understanding of DFT of an image. Assume that we have an image of size $N\times N$, denoted as $f(m, n), m, n = 0, 1, \ldots, N$, and its Fourier transform $F(k, l), k, l = 0,1, \ldots, N-1$. It is often said that $F(N-1, N-1)$ represents the highest frequency. But what exactly does it mean?... |
It seems quite common to use np.diff() for removing DC components of signals when plotting spectrograms.
I noticed some unexpected behaviour during analyis which indicated that a numeric derivative may amplify higher frequencies and I've been trying to check myself since.
So, I decided to treat the derivative as an FIR... |
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