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I have implemented and tuned a PID block in a simulink model and now i want to convert this block to C code to use on my micro controller I have taken the discrete equation of the PID block and the tuned coefficients and did inverse Z transform to get the difference discrete time domain equation but when i implement it... |
I am using MATLAB's built-in function to compute maximal overlap discrete wavelet transform (modwt)and want to know the frequency bands at each scale level. I am looking for a mathematical equation that I can use to compute the bands in each level. Given the sampling frequency $f_s$, the length $N$, and the levels.
|
I have a root-raised cosine pulse signal of bandwidth $500$ MHz.
If the signal is modulated at the frequency $6$ GHz, the result should apparently look like this :
If I understand well, to modulate the root raised cosine, it needs to be "multiplied" by a signal like $h \left( t \right) = \cos \left( \omega t \right)$ ... |
Consider a signal $s(t) = \sin(2\pi ft)$ where $f = 1$.
Let's now discretise it by replacing $t$ with a vector of $n$ sampling instants equi-spaced by a sampling interval $T_s = \frac{1}{201}$ s, thus obtaining $s(nT_s) = \sin(2\pi nT_s)$.
In the following figure is depicted a plot of such a discretized signal:
We see... |
I have never fully been able to wrap my head around Fourier transforms, so I apologize if what I am trying to do is trivial or violates basic theory in some way.
What I have is a "made up" frequency spectrum (well, it comes from somewhere but not relevant for my problem). The spectrum has the form:
$$|G(\omega)| = \fra... |
I am attempting to design a custom filter for equalizing an audio track using the DFT. I am newish at this, but my understanding from the DSP Guide is that you can do this with the knowledge that multiplication in the frequency domain equals convolution in the time domain. To make a custom filter you:
Create an array ... |
I'm implementing a bandpass filter to isolate the amplitude of a UHF FM signal with a 12.5Khz bandwidth. I'm having difficulty isolating the correct bins and bin spacing, given my DFT to isolate the signal such that neighbouring channels don't interfere.
My code is as follows:
int16_t* samples = (int16_t*)transfer-... |
Just for starters: I am very new to audio processing
I was looking for audio data that is in 1-bit format and I stumbled upon DSD. From what I was able to read it is old but kind of controversial format among audio engineers. It samples in binary format but with very high sampling rate.
But when I tried to see if there... |
I am new to MEMS digital microphone world. Currently, we are using built in PDM to PCM converter available in microcontroller.
The problem that is troubling me is how to choose the PCM buffer size, for the below configuration. Once we get the PCM data, I would like to store it into an output file using some terminal pr... |
I was reading the answer to this question provided by Phil Karn.
In the answer, it has been said:
Ensure that the impulse response of your lowpass filter is shifted to the front of your time domain buffer AND properly windowed to M samples before you take the forward FFT to get the frequency domain representation of y... |
Crossposted at Electrical Engineering SE
A very naive question: How do we use Fourier transform for real world signals - for which you have the information only up to the present instant (and the present time keeps moving continuously)?
The Fourier integral is defined from $-\infty$ to $+ \infty$. The standard approac... |
I have a question concerning time series.
I have a Dataframe with several Columns, each containing a time series.
I need a Covariance Matrix of the different columns.
However, my columns are autocorrelated and thus (as far as I understand) the normal covariance won't work.
Is there any way to calculate the HAC variance... |
Despite using the same channel for convolution, each time I get a different estimated channel. Could you please tell me what I'm doing wrong? I want to demonstrate that I can estimate the same channel for different modulated signals (dataIn). The code is:
Channel data moved here.. It comes from here.
M=4;
nfft = 512; ... |
I'm struggling to solve the following question. I've solved it partially, but I can't get complete it.
We have the given information about a signal of the form x[n] = Acos(Bn+C)
The signal is even and real.
Its period is N=10.
The eleventh coefficient of its Fourier series is 5
$$\frac{1}{10}\sum_{n=0}^9 |x[n]|^2 = 50... |
In my DSP class a few lectures ago my professor shared a basic algorithm with us for computing wavelet coefficients when the wavelet basis belong to the Haar system. He also shared a circuit with us which looks something like this:
(I made it in tikz rather than having to post a picture of my scribbled notes)
Here is ... |
Assume we have a LTI system which has poles in the half left plane of the s domain.
Before I learnt Routh's stability condition I had imagined that this was enough to decide whether a LTI system was stable or not. Is it possible for a LTI system to have its poles only in the half left plane of the s domain but not be s... |
I analyzed a voltage signal and plotted the PSD. However, I need to produce a power-spectral-density plot from the data in terms of non-dimensional frequency. How do I calculate the Strouhal number for the frequency range and generate an SPL plot for it?
nfft=2048;
window = hann(nfft);
overlap = 0;
[PSD, freq... |
I have been reading The Scientist and Engineer's Guide to DSP to learn how to create filter kernels with an arbitrary frequency response (I design the magnitude response by hand). The method proposed in the link is:
Create an array that describes how you want to multiply the magnitudes of each bin in the frequency do... |
Suppose we have a closed loop system controlled by some microcontroller $K$
First we take the open loop gain which is $\frac{K}{s(s+6)}$.It has 1 pole at the origin and at $s=6$ and 0 zeros.
So we would except to have 2 regions when $Re(s)<0$ where our system is stable.Something like this:
But according to the diagra... |
I have a signal that I am trying to analyze via Matlab's CWT functions. The signal has two distinct oscillating components and I would like to extract the decay factor from each of them. Here is what the raw time-domain data looks like:
and the CWT using Morse wavelet (which gave me the best frequency resolution):
I ... |
I have a tiny problem with understanding a thing here.
for the 1st the phase shift is 0 degrees, so on its modulated wave we put 180-degree phase shift.
for the 2nd 0, the oscillator puts a 90-degree phase shift in the carrier signal and as it's a 0-degree we have to put 180-degree shift.
When we sum up the two 0's in... |
For first-order LTI systems, the gain-bandwidth constant is often discussed. I've seen the claim that in general, gain and bandwidth don't directly trade off with each other as much as delay. For an example of a discussion where this claim is made, see this Caltech lecture, starting at 6:39 (timestamp in link). Althoug... |
One of the definitions of the cyclic spectrum is
$$S^\alpha_x\left(f\right)=\lim_{\Delta f \rightarrow 0}\lim_{\Delta t \rightarrow \infty}\frac{1}{\Delta t}\int_{-\Delta t/2}^{\Delta t/2}\Delta f X_{1 / \Delta f}\left(t, f + \frac{\alpha}{2}\right)X_{1 / \Delta f}^*\left(t, f - \frac{\alpha}{2}\right)dt$$
where $X_{... |
I recently came across this post stating that the continuous ideal LPF is BIBO-unstable since the impulse response is not absolutely integrable, and this post stating some examples.
I have been trying to understand this using the frequency response. For example, take a system with bounded input $x(t) = \sin(2\pi B t)u(... |
I am working on analyzing the total harmonic distortion (THD) vs. frequency curve for an acoustic system. I am aware of the traditional method of exciting the system with a mono-frequency wave and measuring the fundamental and harmonic frequencies, but it is too time-consuming for my requirements.
Instead, I am conside... |
I'm looking for some good resources to learn audio processing for a machine learning task based on classification of users as either 'COVID-19 positive' or 'COVID-19 negative' based on their cough data. I'll need to extract features from audio files containing cough from a bunch of volunteers before using those feature... |
I want to find the resonant frequency of specific end-blown flute called Persian ney, Using LMS in arrangement of system identification. Two signal is needed for algorithm:
system excitation (x)
desired signal (d).
The plant here is flute. The adaptive is coefficients describing flute behavior in time and frequency.... |
I am trying to implement fractional-octave smoothing of amplitude response data with approximated Gaussian filters, as briefly discussed in this AES paper.
Unfortunately, no implementation details are given and I have trouble getting things to work as intended.
The "room eq wizard" software seems to do the exact same t... |
For a given digitised voltage trace, I want to avoid any quantization error introduced via an insufficient precision of the float data type – and at the same time reduce the memory cost. However, I am not sure that I have understood all the concepts correctly. The digitised signal is given as a 32-bit integer which nee... |
Consider some record with Enrico Caruso singing, anno 1904 or so. Back then, records did not employ pre-distortion such as with RIAA curve to then later, during playback, invert the curve, to get a normal amount of high (and low) end and reduce the hissing sound from the needle scratching the material.
So, it's usually... |
I am currently working on developing a strategy to measure a time-varying signal and would appreciate some input. In my signal acquisition process, the acquired signal $a(t)$ can be thought of as the convolution of the "true" signal, $y(t)$, with a kernel, $k(\tau)$.
\begin{equation}
a(t) = \int y(\tau) k(t-\tau)\, \ma... |
I am clearly missing something obvious here because I am trying to do something that ought to be very simple: compute the ifft of a continuous signal.
My understanding is that the ifft of a continuous signal should be a sinc function. Is this intuition correct?
MATLAB implementation:
N = 10000; %number of samples
fNyq ... |
I have a question regarding on how to select a particular range of interest for the dechirp-on-receive FMCW radar for application within Synthetic Aperture Radar signal processing. If RX01 is the minimum range and RX03 is the maximum range of interest, where should I start sampling and stop sampling based on the timing... |
I have understood the math behind CPM. but recently I came across multi h CPM. I haven't found any good document to explain about multi h CPM. Can anybody explain the math behind it. and how it reduces bandwidth. or can point some good document to implement Tx and Rx.
|
I'm having a problem to normalize my output signal (extracted EEG signal) after applying Independent Component Analysis (ICA) that may be as similar as the original source (EEG signal). I need do this work to subtract and determine noises. I simulate it on Matlab. Could you help me solve this problem, please? Thank you... |
So, I've been working on a signal described by the equation $$y(t) = \cos(2\pi \ |sin(2\pi t)|).$$ On paper, I've calculated a sampling frequency of at least 2 cycles/sec, since the frequency of my signal is bound between 0 and 1 cycles/sec. However, while working on it in MATLAB I need to have the sampling frequency b... |
is this the right way to mix two signals? (I am self learning Blind source separation)
% house keeping
close all,clc;
fe = 1000;
f1 = 10;
f2 = 35;
t = 0:(1/fe):1-1/fe;
s1 = sin(2*pi*f1*t);
s2 = cos(2*pi*f2*t + 180);
figure (),
plot(t,s1), hold on
plot(t,s2)
legend ("s1","s2")
A = rand(2,2);
s = [s1; s2];
obs = A*s... |
Apologies if this is a duplicate, but I can't find a good answer anywhere.
If I have a time-domain cosine signal of the form:
$$x(t) = A\mathrm{cos}(\omega_0 t)$$
Then this will result in an Fourier transform with spectrum:
$$X(\omega)=\frac{A}{2}\delta(\omega-\omega_0) + \frac{A}{2}\delta(\omega+\omega_0) $$
I can ge... |
I am trying to locate the sound source using MUSIC algorithm in a circular microphone array. The array that I am using is "ReSpeaker Mic Array - Far-field w/7 PDM Microphones", and the distances between microphones are very small, less than 10cm. Is MUSIC algorithm suitable for such a geometry?
If not, which other meth... |
I have a question regarding the implementation of FMCW radars that seems rather basic to me, but for some reason I am a bit unclear about this.
So I understand that in FMCW radar, transmitter and receiver are constantly, simultaneously transmitting and receiving chirp signals, respectively. The received signal is then ... |
I have some question about the function in frequency domain and I'd like to know its inverse fourier transform (IFT)
$$G(jw) = \dfrac{jw\cdot (jw+1)}{(2+jw)(3+jw)}$$
I know that:
$$\dfrac{d}{dt}x(t)\Leftrightarrow jw\cdot X(jw)$$
And easy to figure out:
$$G(jw) = jw\cdot \left(\dfrac{2}{jw+3}-\dfrac{2}{jw+2}\right)\\g(... |
I couldn't find such a system but I have also not been able to prove otherwise.
Firstly, I don't know exactly how to take the inverse Z-Transform of $z^*$.
Secondly, I don't know the ROC associated with it so there has to be more than one discrete-time function that has the Z-Transform, $z^*$.
|
I have a spatio-temporal signal $f(x, t)$ that propagates at a constant velocity $v$. To represent that propagation I'm reading in multiple papers that we can represent it in the Fourier domain with the delta function, where the propagation is formalized by $\delta(x - vt)$. The papers say that it can be proven that th... |
The Fourier transform of a sinc function will result in a rect function. Suppose I have a discrete time-domain sinc function with a frequency of $\omega_0 = 0.1$ Hz and amplitude of $A = 10^{-3}$:
If I take an fft of this signal, I get an approximate rect function centered on zero with cutoff frequencies at $\pm\omega... |
I am performing some simulations for FMCW radar for target detection. I am working with a dechirp on receive system. Therefore, the target distance can be determined from the Intermediate Frequency or beat frequency.
$$F_{beat} = \frac{\text{rate}_{\text{chirp}} \cdot 2 \text{dist}_{\text{target}}}{c}$$
I realised that... |
I need to model measurement data of a frequency response with physically meaningful band-pass filters.
All Measurements happening in die range of 20Hz to 20000Hz
I have to work with python.
The Problem:
Imagine a rational transfer function as an undefined number of second order band-pass filters in parallel:
$$H(s) = \... |
I'm trying to filter data from the force sensor mounted on the end effector of the industrial robot arm. Mechanical vibration caused by the robot's motion (possibly by actuators, reductors, etc.) generates noise in the force sensor readings. It seems that the spectrum of the noise depends on the speed of the motion and... |
I have data like the following:
x = [0,0,1,1.98999999999978,2.99000000000069,3.98000000000047,14.0500000000002,15.0599999999995,15.8700000000008,15.8700000000008,16.9399999999996,17.8900000000003,18.8800000000001,29.7900000000000,30.8299999999999,31.8700000000008,32.8699999999999,32.8699999999999,33.8299999999999,34.85... |
Given a signal $x(t) = \frac4{10}\cos(800πt) + \frac12\cos(820πt) + \frac1{10}\cos(880πt)$ and knowing that the sampling frequency is $4000$ Hz.
How many samples are needed at least to represent the signal in a DFT representation?
A solution I've found states that you need to take the difference between the 2 smalles... |
I am experimenting with EEG dataset from this (title: "The Perils and Pitfalls of Block Design for EEG Classification Experiments") paper. They have applied various ML models on different window sizes like $440 ms$, $200 ms$ etc.
I want to know are they epochs of the data, or some hamming window. How to code them.
|
For OFDM, will the following relation always hold:
subcarrier_spacing * sampling_time = 1/N
N: number of subcarriers which is the size of IFFT/FFT
This will imply that sampling frequency is exactly equal to the bandwidth of the OFDM.
Thanks.
|
My question is not really from the field of signal processing but I think you are the most suited for answering my question.
I am willing to compute the gravitational potential which can be written as the result of a convolution product:
$$\Phi(x,y,z)=\iiint G(x-x',y-y',z-z') \rho(x',y',z') dx' dy' dz' = (G * \rho)(x,y... |
I'm studying PSF function and the way to get the most accurate phase estimation at the non-integer locations.
My professor showed me a slide in lecture.
Rotate some integer coordinate in fourier plane.
Then you usually get non-integer real coordinate.
The image above is the way to get point spread function(psf)
My q... |
A window metric called independent sample rate is given in a book that I recently read. It says that in spectral estimation, the variance of power-level estimates is inversely proportional to the number of independent samples in the analysis window, which is given by
$$M\eta_w/N$$
where $\eta_w$ is the independent samp... |
I am using an ADC (ltc2378) with a resolution of 20 bits.
Based on the datasheet this resolution corresponds to an LSB calculated by the equation ($2 V_{\text{ref}}/2^{20}$), i.e., 9.536 µV.
In order to achieve a better resolution of 24 bits as described in this document Silicon Labs Oversampling I have to oversample a... |
In theory of FT(Fourier transform) and STFT(Short Time Fourier Transform) it is said that "A narrower window gives good time resolution but poor frequency resolution".
My question is, can we increase the frequency resolution without pooring the time resolution, by taking the samples in the narrow SFTF window, and repea... |
I have a piece of python code that creates a chirping signal and a downchirp signal. I would like to "dechirp" the chirped signal and my expectation is that doing so, I'll be left with a signal having just a single frequency.
I'm trying to establish what is happening in this video from the provided timestamp
https://yo... |
I recorded myself saying "Ah" in the tone of the note C3 which is about 130Hz, this is what it looks like:
Then I applied an EQ filter to it, so that there would be no frequencies above 1000Hz, and suddenly I hear "o" instead of "ah".
I moved the filter to 700 and got "u".
I moved the filter to 500 and got "E"
I move... |
* Spyder version: 5.4.3 (conda)
* Python version: 3.10.9 64-bit
* Qt version: 5.15.2
* PyQt5 version: 5.15.7
* Operating System: Linux 5.19.0-43-generic
* scipy: 1.10.0
I'm attempting to replicate the control system example in this video starting at about 17:00.
sys = ct.tf([3.553e-15, -.1642, -.1243, -.00161, 9.121e... |
I'm trying to understand an analysis of a sampled continuous time system in the Laplace domain. The source analysis is here (PDF page 6, slide marked 11); I'll explain further below. Suppose I have a system which takes the difference between its current input and its input sampled from half a clock cycle ago, so that:
... |
I'm wondering why evaluating a sinusoid that matches one of the frequencies of the DFT basis functions has a magnitude of $N / 2$. Using this definition of the Discrete Fourier transform, it looks like this holds (I guess for non-zero $k$ and $N \geq 3$):
$$ \left| \sum_{n=0}^{N - 1} a \cdot \sin \left( 2 \pi \frac{k n... |
What are some possible approaches to address the challenge of correlating two signals when the time delay ($T_d$) from the source to the microphone is greater than the microphone's frame time ($T_f$)? Given that the samples from source $s_1$ arrive significantly later than the samples from source $s_2$, making correla... |
The space-variant impulse response can be circumvented by collecting SAR data over a larger angular aperture than is strictly needed to achieve the desired azimuth resolution. From Carrara's text, it mentions that the azimuth frequency extent can be trimmed at both ends after stolt interpolation of the Range Migration ... |
I am exploring the possibilities of IQ modulator. I want to do analog FM modulation (range $0$ to $5 ~ kHz$) using IQ modulator.
So far, I can modulate the signal digitally inside of the STM32 processor by the following equation
$$I = \cos \left( - M_{i} \cos \left( 2 \pi f_{m} t \right) \right) \tag{1} \label{1}$$
whe... |
So say I'm given two image datasets (Image 1 and 2 - attached). I want to be able to find the point spread functions that are the same between the two images (so parameterize the PSFs based on some features) and be able to say that PSF 1 in image1 is PSF 2 in image 2. Also note that intensities of these PSFs (so the he... |
The pHash algorithm for computing perceptual hashes of images is as follows:
Convert the image to greyscale and scale to 32x32.
Compute the discrete cosine transform of this scaled image.
Discard all components of the DCT except the upper-left 8x8 portion.
Find the median value of the reduced DCT.
For each bit in the ... |
I am provided with the following information:
aCxyz: [3×10000 double]
raw: [30000×10000 complex]
Bandwidth: 3.6000e+09
PRF: 833.3333
Fs: 25000000
fc: 3.4800e+10
chirp_rate: 3.0000e+12 [Cr]
c: 3.0000e+8
The number of pulses is 10000 and the number of range samples is 30000
I... |
Haar wavelets are defined as:
$$\phi_{0,0}(t) = \begin{cases}
1, & \text{ for } 0<t< 1/2\\
-1, & \text{ for } 1/2<t<1 \\
0, & \text{ otherwise }
\end{cases} $$
Where the mother wavelet is $$\phi_{k,n}(t) = 2^{k/2} \phi_{0,0}(2^k t -n)$$
I know that this wavelet set is orthonorma... |
We have RSA algorithm for encryption key exchange for example for e-commerce.
If we consider the case of wireless telecommunications. Is it possible to implement an encryption key exchange algorithm (public key, private key) directly from the transmitted waveform, for example a chirp or an OFDM signal,...?
|
I'm learning about OFDM Transmitter and see that output of IFFT Block produces complex digital signal and we need DAC to convert it to analog and then use RF Frequency mixing to send it out of transmitter.
But this RF Freq mixing is not simply multiplying signal with high frequency, rather I see something called as inp... |
I am trying to compensate phase rotation due to carrier frequency offset (CFO) for Bluetooth Low energy (BLE) Angle of Arrival (AoA). In BLE AoA the adv. packets carry a Constante tone extension @250Khz over a BLE Channel (@2.4Ghz).
As the clock of reciever and transmitter are not perfect there is a CFO of around 15 KH... |
I’ve created an upsample and a downsample functions (x2) using polyphase FIR filter.
I have a signal which comes in chunks where sampling rate of each chunk can vary between 1x and 2x (e.g. 48000 Hz and 96000 Hz).
Is there a way to devise a test for a function that would resample the signal to 2x sampling rate (96000 H... |
In a proof for the Wiener-Khintchine theorem (See p. 572-573)
I have seen the following operation being done:
\begin{align*}
S_{xx}(f) &= \lim_{T \rightarrow \infty} \int_{-2T}^{2T} Rxx(\tau) e^{-j 2\pi f \tau} (1 - \frac{|\tau|}{2T})d\tau \\
&= \lim_{T \rightarrow \infty} \int_{-\infty}^{\infty} Rxx(\tau) e^{-j 2\pi... |
Hi everyone I need your help to understand this phase response would you please share your ideas?
|
In telecommunications the received signal $y(t)$ of a receiver is equal to $x(t)S+n(t)$ where $x(t)$ is the sent signal of the sender , $S$ is a constant which describes the total attenuation of the sender receiver channel and $n(t)$ is the noise.Assume that $n(t)$ is purely thermal it will follow as a consequence of t... |
I am trying to generate a sine wave of 440Hz frequency that lasts for 10 seconds using the following C code:
#include <stdio.h>
#include <math.h>
double sine(double amp, double freq, double time);
int main(int argc, char *argv[])
{
int sample_rate = 48000;
double freq = 440;
double amp = 1;
double dur... |
Background
My question comes from here, it's a response of 1st order LPF RC circuit from an arbitrary periodic input.
How to determine the transient response of a circuit to causal periodic inputs?
Problem
Suppose if I have input signal with period of $T = 10s$
$\displaystyle u(t) = 2t (\theta(t) - \theta(t - 5)) + ... |
I am getting very confused with the textbook I am reading through: Modern Digital and Analogue Communication Systems:
3.8.1 Parseval's Theorem says that:
The total power in a power signal is its time average energy.
It then goes onto section 3.8.2 where it says that the Autocorrelation function is:
In terms of a disc... |
I want to detect some targets where their range, velocity and angle information can be given before creating the radar cube. Then using IF signal for MIMO FMCW radar I want to construct the radar cube. Applying fft2 I can obtain the range doppler map, however I observe extra targets on the map. This reflects to the CA-... |
I understand that we have analog complex signal after it passes DAC after coming out of IFFT in OFDM Transmitter. After that we do orthogonal upconversion on it and send it.
I've question now for OFDM Reciever, suppose i recieved a signal then i separated I & Q component, for processing anything on the signal now I've ... |
I am working with antenna phased arrays for direction finding. According to the manufacturer of the antenna some elements of the uniformed rectangular array (URA) have a rotational difference. This makes some elements have a positive rotation towards the azimuth direction $\theta$.
The generic form of a single antenna ... |
I am trying to use Frequency-wavenumber analysis for estimating speed of sound using two-dimensional data obtained via channel of arrays. The traditional technique is to use FFT to transform the data from distance-time domain to wavenumber-frequency domain and detect the slope of the bright line in the FK plot which in... |
I'm in the process of re-learning DSP (not a subject I've visited since University) and in quite a few resources I see this general form of a DT-LTI difference equation:
$$y[n] + a_1y[n-1] + a_2y[n-2] + \cdots + a_Ny[n-N] = \\ \qquad \qquad b_0x[n] + b_1x[n-1] + b_2x[n-2] + \cdots b_Mx[n-M]$$
However I don't understand... |
I have an app which lets users download presets as ~10kB of data and share them. There's no web-service or anything, it's just posting files and stuff.
Bandwidth and storage space aren't so tight in the modern world, so (for fun, because it's a creative app) I'd like to provide an option to download the data as an ima... |
I wish to control a simple pitch tracker using my voice. So far I have experimented with fft and yin. I am extracting the frequency and amplitude in real time. I then feed these into:
a*sin(2 * pi * f * t)
where a is my amplitude and f is the frequency. I'm wondering though if I should also be calculating the phase of... |
In Chapter 1 of Oppenheim & Schafer's Digital Signal Processing, the authors state that not all discrete-time signals are the result of sampling continuous-time signals, and thus they are making the choice not to "force results from analog systems theory into a discrete framework" for the textbook.
Can anyone think of ... |
I am currently working on blind source separation using sparse hypotheses and convolutive mixtures. For my project, I have compared three different methods and calculated the Signal-to-Distortion Ratio (SDR), Signal-to-Interference Ratio (SIR), and Signal-to-Artifact Ratio (SAR). Now, I would like to know what I can do... |
I have some governing equations of the form:
$$\begin{align}
\ddot \theta(t) &= \frac{MgL + mgl}{J} \theta(t) + \frac B J \dot x(t) - \frac \alpha J V + \frac {mg}{J} d - \frac{c_1}{J} \dot \theta(t) \\
\ddot x &= \frac{\alpha}{R(M+m)} V - \frac{\beta/R + c_2}{M+m} \dot x(t)
\end{align} \tag 1$$
And I would like to tra... |
The Hilbert transform of a function is defined as :
$$\mathscr{H}\big\{f(t)\big\} = \lim_{T\rightarrow \infty}\frac{1}{\pi}\int\limits_{-T/2}^{T/2}\frac{f(\tau)}{t-\tau}\, \mathrm{d}\tau$$
Okay but what does it do exactly? What information do we take from it?
|
I am new to EEG analysis, and I wonder what are the common pre-processing steps of EEG signals, for example, for motor imagery classification?
Currently, my pre-processing steps include
Epoch the data
Filter the data
Down-sample the data
Do baseline correction
Do common average reference
Do scaling
Are they appropria... |
I have a task in image processing, which need to detect vertical lines from matrix. For example, there is a pair of white and black vertical lines segmenting the picture below. I need to find it. So I want to find the lines labeled in red.
Actually, this matrix consists only -1(black), 0(gray) and 1(white), which is ... |
I am looking for a comprehensive explanation of Automatic Gain Control (AGC) for Radio Frequency (RF). I am unable to find any good resource online. Can anyone provide some pointers or an explanation of various blocks and types for AGC in RF?
Thanks.
|
My question is related to a theoretical part of the book: Digital Processing of Synthetic Aperture Radar Data, Algorithms and Implementation by I.G. Cumming et al. [Chapter 4. Equation 4.25 (for those who know the book)].
Basically what I undertand that is said there, is that if I have my transmit pulse waveform $s_{p... |
I am wondering about physical interpretation of energy / power for complex signals. E.g. for signal energy definition:
\begin{equation}
E=\int\limits_{-\infty}^{\infty}|x(t)|^2 \, \mathrm{d}t
\end{equation}
For $x(t)$ real, when representing voltage (or current) waveform, above definition would give energy dissipated a... |
I'm trying to understand how oversampling can help to reduce the signal to noise ratio, so I looked this document, and at page 6 it is said that:
Along with the input signal, there will be a noise signal
(present in all frequencies as white noise) that will fold or alias into the measured frequency band of interest (f... |
I have a real world system of a rotating back and forth motion and the time domain data suggests that the data is a random signal (not a deterministic signal). I recorded discrete data points at a constant sampling rate in my lab notebook from raw data recorded by a computer program at a sampling rate where I don't ha... |
In short, I am coding an optics simulation for an experiment I am running. There is a point where I perform a Fourier transform on my incoming signal and want to extract the phase; however, I get unexpected results.
More specifically, I want to unwrap the phase from the signal below:
To extract the phase I use the fol... |
For computing the Harris corner detector, I want to calculate $I_{xy}$ and $I_{xy}$. Obtaining $I_x$ and $I_y$ are clear for me. But what about the second order derivatives?
This and this and this one posts say I have to do convolution operation twice:
$$
\frac{\partial^2}{\partial x \partial y} f = \frac{\partial}{\pa... |
I am reading Digital Communication Systems by Simon Haykin and I am stuck at one point.
Consider a two-dimensional signal space that has a message constellation of four points, given by $s_1,s_2,s_3,s_4$. We need to calculate pairwise error probability in the presence of additive white Gaussian noise ($w$). The signal ... |
In this paper they preprocess a cent spectrum of an audio file.
For comparison of audio files they create the Logarithmic Fluctuation Pattern(LFP)(p.3).
For this I have to use a FFT on each frequency band of a block and then only take into account the amplitude modulations up to 600 bpm. (See screenshot)
I want to im... |
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