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I am trying to implement the algorithm described in this paper. However, I am having trouble understanding the last part where I need to find the energy differences of bands.
In another paper, this process is described as shown below. Also, I am not sure I understand that either. What exactly I should do in this part ... |
I have trouble to distinguish these two concepts namely ergodic process and strict-sense stationary process. I look at one of the books about the signal processing that says an ergodic process must be a strict-sense stationary one. So how to show this property?
Can anyone help??
Thanks,
Victor
|
I would like to clarify one doubt regarding autocorrelation function $R_x (\tau)$ of a process $X(t)$. Suppose $R_x (\tau) = \Pi (\tau)$, where $\Pi (\tau)$ is the rectangular function. Then is the process $X(t)$ deterministic.
|
I'm trying to implement a normalized cross-correlation algorithm but I don't get what in fact is this measure. What confuses is the wikipedia definition:
$\frac{1}{n} \sum \frac{(f(x,y)- \overline{f})(t(x,y)- \overline{t}) }{\sigma_{f}\sigma_{t}}$
Which result is an scalar (AFAIK)
But then adds other way to measure it:... |
I need to find the timedelay between two signals. I guess MSE is the best approach. How to find that MSE? I am writing the code and I have access to the autocorrelation matrix of one signal and the crosscorrelation of both signals plus the power of the signal. Is this enough information to calculate the MSE? Any ti... |
I am currently toying around with the Discrete Fourier Transform (DFT) in Matlab to extract features from images. I like to fully understand the concepts that I use. I have read several explanations, such as this, but so far, none really explained the meaning of the "DC term". All I know is that the k'the term of the D... |
I am trying to filter a signal using a given FIR band stop filter. I have to implement a band pass filter. Assuming I want to get the signal between normalized cutoff frequencies [0.2,0.3] - can I use two band stop filters over the same signal - one for the range [0,0.2) and second for the range (0.3,1] ?
|
I have a low-voltage amplifier with an analog LPF of ~106 Hz and a HPF of 0.482 Hz (both first-order RC). I can choose to apply a calibration pulse that will shut off the input and send a 500uV, 5Hz square wave across this analog filter band. The amplifier is followed by an ADC and I'm receiving all values in code.
W... |
I have sucessfully implemented a STFT (+inverseSTFT with perfect reconstruction of original signal, with overlap add, etc.) in order to work on audio files.
Using this STFT / iSTFT framework, I have tested :
Basic "noise reduction" algorithm that works by spectral subtraction of a noise template -> it works very well ... |
Can somebody provide me with the mathematical equation to calculate relative and absolute power of a signal (continuous) from FFT.
|
Contextual knowledge (CK) is used in some AI application, Can someone give me a concrete definition of this term.
I am working on ontologies( a set of concepts and relationships among them) and computer vision. I basically aim at integrating knowledge (ontologies) and visual features extracted from images to leverage m... |
I'm trying to design a MATLAB audio player that includes something like a 31-band equalizer. Right now I have the filters working by using the FIR1() function to create an FIR bandpass filter at each frequency specified on the bands. This is great except that it's not exactly what I'm wanting to do. I want to either bo... |
Assuming I have a set of frequencies $\mathbf F$, and I pick 2 frequencies $f_1$ and $f_2$ randomly from the set, generate sine waves for them, and pass the signals through a non-linear distortion process while adding them.
Is there any way for the harmonics or intermodulation distortion products to have a magnitude ... |
Suppose I have an instrument which is out of tune. For example lets say, as an example, the notes it plays are as follows: C+20 cents, C#+20 cents, D+20 cents D#+20cents etc. In other words, it plays all its notes 20 cents too high.
I then record this instrument, having played polyphonic music on it.
What algorithm ca... |
I am new to the wavelet field, and I am working with ultrasound signal. For my study, I needed to measure material using the pulse echo reflection method.
Basically, when the transducer transmits the signal, the signal will reflected back to the transducer. If material has a thickness, the signal will reflected for tw... |
I a recent topic, it has been told that zero-padding in the STFT can improve it, and avoid some circular convolution related things.
Here is my Python code for STFT :
fftsize = 8192; overlap = 4; hop = fftsize / overlap # ie hop = 2048 here
w = scipy.hamming(fftsize) # hamming window
# x is the input signal,... |
I want to take a real signal on my FPGA and create a complex one. I have FIR core implemted for a Hilbert Transform, and Xilinx's datasheet on it shows coefficients of: (-819,0,-1365,0,-4096,0,4096,0,1365,0,819) as an example.
I wanted to add more taps, so as practice I used the matlab fdesign function:
d=fdesign.hilb... |
For example, I have an image
There are 7 people in the image and one apple and a bird.
How can I extract every person and every object from this image and put the 7 people and 2 objects into 9 different new images? How can I also identify which is person and which is object?
|
I know that the Fourier Transform of the autocorrelation function is
the Power Spectral Density. But how can we arrive at such a result intuitively?
Is it just a theorem?
|
As people acquainted with Compressive Sensing would know, incoherence and sparsity are two main principles. I've been reading about compressive sampling and developed an interest into the topic. What I'm about to ask might be quite basic so sorry for that in advance.
As incoherence principle states, sensing basis and s... |
I'm trying to calculate the short term energy of a digital signal
in python, using a rectangular window, but don't understand
the basics of how this works, a pointer to any good explanation would be great.
Specific things I'm confused about:
Where do I initialise the window? (at position 0,window_length,
or -window_le... |
I have the RLS algorithm implemented in two different ways - transversal and lattice.
I was expecting them both to behave the same way, but they don't! Transversal seems to converge faster than lattice does. I'm not sure why. I plotted them together here:
Lattice is shown in red, transversal is in blue. This is MSE.
... |
I have a time series file which starts with 1 and then quickly tends to oscillating about 0.001. The time series has a resolution of 0.01s and has 10^6 sequential points starting from t=0. The goal is to study the divergence of the fourier transform at Zero frequency. Is there any trick that can be employed to improve ... |
Given a blob of an image (representing an object), according to Wikipedia, we can compute the co-variance matrix using the image moments.
I understand that the eigenvectors of that matrix can be used to represent the principal axes of an oriented bounding box (OBB), but what are the guarantees for that particular orien... |
I have a video (in webm format), and I want to produce a graph that shows the degree of motion in this video as a function of video time. I'm sure there are many different ways to define "degree of motion", but I envisaged something that could be based on the average difference between pixels in consecutive frames, or ... |
Why, when I have to calculate some types of noise (for example shot noise and others which are in the denominator of SNR ratio) or attenuation from turbulence, I have to use variance of the of the density probability and not the mean value? In all of the formulas they use variance (the second moment) and this is not c... |
I know the basic working principle of sinusoidal oscillators. But I have a doubt on this matter when I came across Gram-Schmidt orthogonalization procedure. I have read that the source for sinusoid signal output from an oscillator is the noise. Due to controlled positive feedback a single frequency component is selecte... |
We as a group are new to signal processing and speech recognition. We have stored .wav sounds of three different words and we are trying to record a new .wav sound and have the system compare the sound to these three pre-recorded sounds to see which one correlates the most to the newly generated sound.
We have altered ... |
Is there a rule of thumb for how many coefficients to choose for a fractional delay filter? I figure there might be a trade off based on the signal frequency as well as my sample rate (this is running on an FPGA), but I can't seem to find a good rule of thumb.
|
How would you reliably detect scene transitions in a video? The simplest case wouldn't be too hard, but sometimes it might be a fast moving scene, or there might be an effect like lightning that would throw it off.
|
Let us say we have a signal $y(k)$ that is of length $K$. I can compute its PSD, also of length $K$. Let us say that this length was not good for finding peaks, so I computed a PSD that was of length $10 \ K$ instead, by doing a zero-padding in the time-domain. I understand that this is equivalent to interpolating the ... |
In signal processing, when you sampling an analog signal,
the sampling theory says that the sampling frequency needs to be at least twice the maximum frequency in the analog signal content.
I understand that if the sampling frequency is smaller, then it can cause
a data loss, but what if the sampling frequency will be... |
I was reading a paper about finding optimal morlet wavelet function for CWT, and they were minimizing Shannon entropy.
what is Shannon entropy and it uses?
(specially on continuous wavelet transform)
|
I am doing research on images using gabor filter.
I have read the following post:
Apply a Gabor filter to an input image
There must be some formula to determine the value for gamma,sigma,lambda,psi...can i know how to determine it?
|
I'm struggling to validate that my C++ implementation of a filter I designed in Matlab is correct.
My first issue is to make sure I'm implementing the sections in the right order. This is the output of the SOS matrix from fdatool:
0.5961 -1.1815 0.5961 1.0000 -1.9792 0.9884
0.5793 -1.1475 0.5793 1.... |
I am trying to calculate shannon entropy of CWT. I am not sure if I am doing it right.
Assume that $W(a_i,t), i=1;2;...;M$ is a set of wavelet coefficients. The Shannon wavelet entropy is calculated by:
$E=-\sum_{i=1}^{M}d_i log(d_i)$ $\rightarrow$ where $d_i=\frac{|W(a_i,t)|}{\sum_{j=1}^{M}W(a_j,t)}$
I am confused ... |
If I understand what LPC does, I think. It predicts the next sample by taking into account previous $p$ samples. I don't understand why it is useful in DPS and speech recognition.
I also came across LPCC that can be derived from LPC and are related to sound frequency. I also find it hard to follow the idea why cepstrum... |
I have been working on an assignment for university in which I have to compress an 80K message consisting of only two symbols, 'a' and 'b', having probability values of 0.6 and 0.4.
I have successfully written my code in Matlab and it is working efficiently for 10-20 symbols. However when I pass my 80k symbol message t... |
I was making a plot to show side lobes for different spectral windows and I noticed if I take a simple fft of a rectangle window, I don't see any side lobes. I think I'm just making a simple mistake.
Here is the Python code I ran:
from matplotlib import pyplot as plt
import numpy as np
fAxis = np.linspace(-1/2,1/2,100)... |
I have written a bit of code to upsample and interpolate a sample waveform in MATLAB. It's at the point where I have taken a signal, upsampled it 2x and filled in the gaps with 0's, and then I found the fourier series coefficients. I know I need to choose which coefficients to multiply by 0 and which to multiply by 1 i... |
i have to implement a function in matlab,for labeling silenced,voiced and unvoiced frames in signal.
I mean i want to specify each frame as a voiced or unvoiced or silenced frame by using
pitch frequency and zero processing rate and energy.
how can i calculate pitch frequency?
where should i start from?
Thanks
|
I would like to compute the minimum sampling frequency for the signal $x(t)$ given below:
$x(t)= sinc ^3(300t) * sinc ^5(600t)$ and $ *$ stands for convolution.
where the sinc function is defined as $sinc (w t) = sin (\pi wt)/(\pi wt)$.
I have tried like this : $sinc^3(300t)$ will correspond convolution of 3 rectang... |
I am trying to make a spectrogram viewer without using the spectrogram command. For this purpose, I used a sin function. I broke up the input signal into 256 segments, multiplied each segment with the Hamming window, took the FFT of each slice and then stored them to obtain spectra of the signal.
My problem is that I ... |
I am a little unsure about the generation of a additive white Gaussian noise, so I would appreciate to hear from you if I`m doing it correctly.
My goal is to generate the time-domain noise correspondent to a noise floor of -135 dbm/Hz
The procedure I`m adopting is the following:
1 - Convert the noise floor in dbm/Hz to... |
I have a question regarding an alternating pulse train signal, $p(t)$, and a system which looks like this:
Immediately, I'm supposed to be able to extrapolate that
$$
P_t = P_1(t)-P_1(t-\Delta)
$$
How do you obtain this?
How would you get the Fourier transform of this pulse train from there?
|
I need compare two signals. Signals have simliar shapes but completely different values (as on picture):
So I think cross corelation in frequency domain should gave me a number that shows me how the signals are similar. But I have no idea how to do that in matlab.
Where I can find code to do it?
|
Reference
Second post on EdaBoard.com
Time response of a system is the time evolution of the variables. In
circuits, this would be the waveforms of voltage and current versus
time.
Natural response is the system's response to initial conditions with
all external forces set to zero. In circuits, this would be the
respo... |
Does anyone know how to write arg min code in matlab with explanation?
I am referring to a paper for my research. I am not sure whether I can paste the paper link here. At first, I try to do feature extraction of finger using Gabor Filter. According to the paper, after feature extraction level, I need to do the followi... |
I am using IPP for FIR filtering over 1D data. I want to use the same input and filter it using two band pass filters. This is how I do it:
output1 = doFIRBandPassFilter(input,low1,high1);
output2 = doFIRBandPassFilter(input,low2,high2);
I want to combine those outputs so that the end result will show BOTH (low1,high1... |
I have applied a complex wavelet transform on two signals :
a first one
a second one that is a concatenation of the first signal and another one
For the 4th coefficient (as an example, i've got the same problem for all of them), i get the following graph for its module (1- is blue, 2- is green) :
What could explain t... |
I want to separate out the components that I need to evaluate the pronunciation of a user, so I am dealing with the energy/stress/volume and intonation or pitch.
Focusing first on the pitch, is there any way to get the average pitch of the wave file? I would like to get the average of the user's input with respect to ... |
I`m confused on how to calculate a signal power. For example, suppose I generate a complex: signal $$x[n] = (1 + j3)e^{j2 \pi k n/N}$$
The DSP theory says the average power of a periodic signal with period $N$ is:
$$ \frac{1}{N} \sum \limits_{n=0}^{N-1} \left| x[n] \right|^2$$
Does that mean the signal power is indepen... |
I have just started my first proper course in DSP, and am a little confused about a statement in my textbook. The book states:
"Notice that when the Nyqust-frequency is defined as $\pi/\Delta t$, we will with $\omega_m = 2 \pi m/N \Delta t$ where $m = 0, \pm 1, \pm2, ...$ have $\omega_m \neq \pi/ \Delta t$, unless $N$... |
I need help with iris localization. I have applied the imfincircle function to locate it, but it only locates the inner boundary of the pupil and does not find outer boundary.
How do I locate the outer boundary?
|
I'm asked to design FIR filter, an ideal low pass filter.
I'm given with pass band frequency (wp) and stop band frequency (ws). Now, I have to choose Ns points from stop band and Np points from pass band. Taking the inverse will give me impulse response.
The problem is, Ns points are equally spaced and Np points are eq... |
Let's say we have 2 signals of frequencies 4 kHz and 8 kHz.
If we sample both signals with sampling frequency 10 kHz and pass the signal through a low pass filter with cut off frequency of 10 kHz, what frequencies will be present in the output signal?
|
I have two wave files each containing a sample of a note. I wish to compare, using a program, each sample and present how loud the first sample is with respect to the second if each sample is played back to the listener with the same physical conditions (volume, distance from speaker etc.). For example, if the second s... |
Assume a signal $x[n]$ with $N$ samples and its $N$-point FFT $X[k]$. Assume also the simplest power spectral density estimator:
$$ S_x[k] = \frac{\left| X[k] \right|^2}{N} $$
With this estimator, the average total power can be evaluated as:
\begin{align}
P_x = \sum \limits_{k=0}^{N-1} S_x[k] \Delta F
\end{align}
where... |
I want to extract Audio Features using RBM (Restricted Boltzmann Machine). For this, I am giving the spectrogram (PCA whitened) as an input to the RBM. For each audio file, The spectrogram is a matrix with no. of columns fixed but with different number of rows for each audio file. My question how can I train my RBM, or... |
The concept of frequency spectrum exists because it is possible to sum sines and cosines of various frequencies together to make any periodic signal. The frequencies of these individual sinusoids make up the spectrum of the final signal. Now when it comes to image processing we treat image as a signal similar to sound ... |
I have an image of size 160x160 of pixels. I am trying to convolve this image with two Gabor filters of different scales (7 and 9 respectively) but with the same orientation. I red in an article the following phrase :
The maps (the results of the two convolution operations) have the same dimensionality but they are the... |
This is just a question out of my curiosity, normally we do not do this
I have a x(n)=[1 2 3]
y(n) = fft[1 2 3 0] = 6 + 0i -2 - 2i 2 + 0i -2 + 2i %zero padding
y'(n) = fft([1 2 3]) = 6 + 0i -1.5 + 0.87i -1.5 - 0.87i
I used Matlab to generate these results.
My question is there way to obtain y'(n) from y(n)?
|
This is the spectrum of the signal:
And this is the signal:
On the FFT spectrum you can notice the high amplitude peaks and I think these frequencies are the main ones in the signal:
I think that the "low" harmonic must be clean, but I have the problem that low amplitude components are placed in every frequency inte... |
I saw a lot of use of tensor in signal processing. What is the intuition behind it? Is it simply a common representation for audio (1D), image (2D) and video (3D) signals?
|
A question asks for the energy and power of the following signal;
$$x[n]=\text{sinc}\left(\frac{n}{5}\right)$$ given that $$\Omega_c=\pi/5$$
Energy is $$E=\sum_{n=-\infty}^{\infty}{\left |\text{sinc}\left(\frac{n}{5}\right)\right|^2}\rightarrow \frac{1}{2\pi}\int_{-\pi}^{\pi}\left|X_c(\Omega)\right|^2d\Omega$$
My resul... |
I have got a signal in frequency domain. This is a frequency response function from software, so I can do nothing about it and have to leave it in frequency domain. I want to transfer the data to 1/3 octave spectra and I need some help on this.
In my opinion, I just need a couple of band pass filters which in the frequ... |
I am having difficulty in calculating the SNR for the following scenario:
In a cellular network
Transmit power=36dBm
Noise spectral density=-174dBm/Hz (AWGN)
Bandwidth=10MHz
I have some other parameters such as pathloss and Rayleigh fading, but I think my biggest problem is the conversion of dB to power.
|
I am generating a sine wave from a DAC at a constant frequency. If each full cycle is specified by N data points what is the relationship between the accuracy of the rendered sine wave and N?
N=1 will obviously be pulses. N=4 (peaks and zero crossings) will yield a triangular wave of sorts. Is there some equation that ... |
I have an unusual low-pass filter problem that I would like to realise with a passive RC network (Note 1).
It's easy enough to find prototype filters for passive RLC networks, or for the various active RC filters.
I am having difficulty finding prototype filters for the classic filter types (e.g. Butterworth, Chebychev... |
I have a dataset which records the arrival time for a series of photons.
I am using lomb-scargle method of scipy.signal to try to find some kind of period.
Since the arrival time is random,how to express the intensity of the signal?
If I bin the dataset and let the photon counts in every bins be the intensity , the da... |
The Auto-Correlation function of Walsh-Hadamard codewords does not have a good characteristics.
It can have more than one peak and thus, the Walsh-Hadamard codes do not have the best spreading behavior or correlation property.
The cross-correlation function of the Walsh-Hadamard codewords can also be non-zero for a num... |
There seems to be a number of different ways for telecommunication. What is the most recent (and maybe standard) technique used today? It seems the simple AM/FM modulations are definitely somewhere in the past, then there are these spread spectrum systems... All hints welcome :)
|
I read the DAFX book by Udo Zölzer about the distortion effect at pages 124-125
and it says that suitable simulation of the distortion is given by the function:
$$f(x)=\frac{x}{|x|}\left(1-e^{x^2/|x|}\right)$$
Can someone explain this formula and what kind of signal we get?
From what I understand 'x' is the sampled si... |
I'm trying to suppress the sidelobe of OFDM but having troubles showing its spectrum in Matlab. After IFFT, I get a time-domain signal, but in which way I can get its spectrum(or PSD)? I want to analyze the out band power emission, with rectangle window in the time domain the out of band power curve decrease slowly, so... |
My understanding if a frequency in my signal doesn't line up exactly with a bin, it is smeared over a few bins to the right and left.
How can the instantaneous magnitude be determined? (do I have to worry about the windowing function?)
|
Based on a paper in here, they have proposed a method to optimize morlet function parameter. I have tried to implement their technique, but I can't get rational results. any idea?
In here you can see what they have done briefly:
Imagine you have a signal with following frequency and damping pairs :
$(f_i ,\zeta_i), (f_... |
To get a panning effect I know that there is a need for a Circular Matrix.
The angle is can be set as a default or I can let the user to select the angle.
So if the angle is $\theta$, the circular matrix will be:
$$
\begin{pmatrix}\cos \theta & \sin \theta \\ -\sin \theta & \cos \theta\end{pmatrix}
$$
and the gain i... |
I want to know if there is a generalized way to select a signal from a set of signals. Let me clarify it: I have a number of signals $S = \{S1, S2, S3,...\} $.
Some of the signals are not important to me like, all having all zeros for data, or all same values, or only one datum comes in at the first occurrence then re... |
http://astarte.csustan.edu/~tom/SFI-CSSS/info-theory/info-lec.pdf compares a bacteria DNA vs a random string:
My first step was to generate for myself a "random genome" of
comparable size to compare things with. In this case, I simply used
the Unix `random' function to generate a file containing a random
sequen... |
When you plucked on a string, the string is vibrate and except the fundamental frequency,
there are an integer multiple of the fundamental frequency, which are the harmonics.
I know that the HPS algorithm is a tool to find the fundamental frequency, but
I don't understand the downsample part. How can I know how many... |
Is there any speech enhancement algorithm library?
I need to process a sound to voice recognition, and I want to remove noise from audio source.
|
I don't know if this question is related to DSP, but I'll give a chance.
Introduction:
These days there are a Guitar Synthesizers which allow the guitarist
to play another instruments like: brass, drums, strings, pianos and other guitar models
from his guitar.
It possible to turn any guitar with the MIDI Pickup, that ... |
I am studying a deconvolution problem for which observations include noise. I am considering using one of several common regularization methods, including Tikhonov's solution, truncated Singular Value Decomposition (SVD), or Wiener filter.
A manuscript I am reading, Riel and Berkhout, "Resolution in seismic trace inver... |
I want to build a navigation system with gyroscope/odemeter and gps. I have konwn there is a famous method called kalman filter for this application. And now, I have done some experiment in matlab with kalman filter to get hide state vector which included vehicle's current postion/velocity/heading etc. But out of my ex... |
I have the following matlab code
N = 500;
M = 32;
K = N + M - 1; % Length of full convolution
x = randn(N);
y = ones(M);
% first one: convolution
r = conv(x, y, 'same');
% second one: multiplication
z = ifft(fft(x, K, K).*fft(y, K, K));
idx = round((K-N)/2+1):round((K+N)/2); % Indices of the central portion
z = z(id... |
Are there any differences if you use Maximum Likelihood or Maximum a Posteriori to estimate the Point Spread Function for image deconvolution?
|
I have an image with size $256*256$ then i get the Fourier transform of the image, If i only take the central region of the Fourier transform whose size is $64*64$ and get the inverse FT ,What is the effect on the output image$(64*64)$?
Down-sampling, Blurring, or both.
|
I'm trying to implement a QPSK Mod/DeMod system in C++. So far i have a bit generator that generates N bits and stores it into an array:
srand(time(0));
for (int i = 0; i < N; i++)
{
bitstream[i] = rand() % 2;
}
Now I parse through the array to group bits of two and send those two bits to get modulated.
for (int ... |
Norbert Wiener came up with the Wiener Deconvolution in 1949 or something like that. But what happened after that? When was Richardson-Lucy deconvolution developed for example? And what happened between that and Adobe's deblur demo in Creative Cloud?
|
I am having some trouble understanding how a digital signal gets converted back to an analog signal.
Why is the "sample and hold" effect a problem?
|
Why does image deconvolution still work with image without sharp edges? Take this image for example:
|
I capture signal (sound) in 44100hz and I need get response if there is peaks at amplitude, actually. Is this possible? How I can do this in real-time?
|
Is there an accurate way to create an aliased image from the Fourier transform of the original image?
in other words, i have the Fourier coefficients of an image and i want to make down-sample in frequency domain.
|
I am relatively new to signal processing techniques and Matlab and need a bunch of test data in the form of white noise, as defined on the Wikipedia page, with constant, flat power spectrum.
I tried the matlab built-ins wgn(m,n,p) and randn(m,n) to get vectors of white noise, and then used different combinations of th... |
My computer vision / image processing professor uses the term "Label", to represent part of an image, but cannot explain in a way that makes any sense to me what it is. She says this is a somewhat common term, but it is not clear what it is. Its somewhat similar to a pixel, but different. I can't find a definition anyw... |
I have doubt, please clear.
A transform is said to be unitary transform when the product of the transform matrix and its conjugate transpose (Hermitian) matrix is equal to Identity matrix.
DFT for N= 4 gives
c =
4 0 0 0
0 4 0 0
0 0 4 0
0 0 0 4
Thus, it satisfies the... |
I am trying to implement a Chebyshev type I low-pass IIR digital filter in C. I have got the SOS Matrix and scale values from Matlab.
What is the direct equation or algorithm to implement such a filter?
|
I am having a hard time understanding why my textbook takes the squared magnitude of the calculated DFT coefficients.
I know in DFT, you would normally take the magnitude to get the spectrum, how come it is the squared magnitude which gives me the spectrum when iam using goertzel?
|
If $x[n]$ is a discrete random sequence with uniform distribution having mean $\mu = 2$ and variance $\sigma^2 = 3$, if passed through a moving average filter to get an output of
$$\frac{x[n] + x[n-1] + x[n-2]}{3}$$
what would be the the variance of the new signal?
I am getting 2 answers like 1 and 3. Which is correct... |
As I understand it, in image processing, when it is desired to apply a filter of some sort to an 2D image, a kernel is applied to all the pixels of the 2D image in a process called convolution. Median, Gaussian, Bilateral, all are examples of kernel. These kernels are applied in a similar fashion to that shown in this ... |
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