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I try to work out the simplest system, whose response is $H(z) = 1 + cz + c^2z^2 + c^3z^3 + \cdots = 1 / (1-cz)$
Now, Z-transform done quick (Fourier connection) and 4.5 Transfer Function, Poles and Zeros say that we just need to plug $z => e^{iw}$ to get the frequency response. I am getting single pole at $w = i \ln ... |
Suppose we know the DFT of a discrete limited sequence, some $X[k],\ k = 0, 1,\dots ,N-1$. How can we calculate the Fourier Transform of the same signal for a random frequency $\Omega$?
EDIT:
How about this, we can get the starting signal with inverse DFT, like: $$x[n] = \frac{1}{N}\sum\limits_{k=0}^{N-1} X[k]e^{j\fra... |
I'm having 2 signals which have different sampling rates i.e., 1ms(A) and 4ms(B) and I've tried to upsample/downsample either of the signals based on the code snippet Resampling based on FFT which I believe is equivalent of what is descriped in How to resample audio using fft or dft
% FFTRESAMPLE Resample a real signa... |
What would be a decent formula for approximating the connection between voltage and decibel output for a typical set of unpowered in-ear headphones?
I'm generating sine waves from the computer: an array of floats between -1.0 and +1.0 gets sent out through the speaker:
X_out = h * sin( f * 2π * t ); // h is the amplitu... |
I am new in matlab and signal processing. The time series that have been used are obtained from accelerometer in a building. As far as I understand both the time series' length and window function length(hamming or hanning) should be same. But for smoothing the fourier amplitude spectrum, optimum window length can vary... |
I've just come across a paper that interrelates the covariance matrix of time discrete signals to their autocorrelation function (or a time-delay, respectively), i.e.
$$\mathbf{C} = E\{\mathbf{x}\left(t\right)\mathbf{x}^H\left(t-\tau\right)\}.$$
Coming from the image processing field, I have usually just worked for the... |
MATLAB's filtfilt does a forward-backward filtering, i.e., filter, reverse the signal, filter again and then reverse again. Apparently this done to reduce phase lags? What are the advantages/disadvantages of using such a filtering (I guess it would result in an effective increase in filter order).
Would it be preferabl... |
If I understood it right, a keypoint is a tuple $$(x,y,\sigma, r),$$ where $x,y$ define the position of the keypoint and $r$ an orientation (given by the most domiant gradiant vector around the keypoint). I'm still not sure about the intention of $\sigma$.
What is done in the first part of SIFT is to consider $$L(x,y,\... |
I have a project in which I have to create chorus effect to an audio file. I have tried this code and i don't know weather its right or wroung. Anyway the code is below
[x,fs]=wavread('ShortWAV.wav');
f=0.25;
delay_in_sampels=0.05*fs;%50 ms delay
index=1:length(x);
sincurve = 1*sin(2*pi*index*f);
y = zeros(length(x):1)... |
i am working in digital image restoration field, recently i have studied image degradation model which is like that.
g(x,y) = h(x,y)*f(x,y) + n(x,y)
where
g(x,y) is the degraded image
f(x,y) is the image that we wish to capture or its is latent image
h(x,y) is image degrading function, and * is convolution operator
n(... |
Hi I'm trying to separate sound from music using wfilters and heard each part separately using its name scale3 function
function s = scale3(q,n)
p = length(q);
if p==n
s = q;
else
s = resample(q,n,p);
end
the program
s = wavread('K:\Khaleel_3aaza (1).wav'); % whatever here
[l,h] = wf... |
Is there a way to make an inverse notch filter block in matlab simulink?
I have found the peak-notch block, but I need to amplify the signal instead of attenuate it.
Thank you.
|
In trying to answer another question, I got stuck applying what I thought would be a valid way to answer it: using the system function. Here's what I did:
First, I put the question
Y[i] = Y[i-1] + ALPHA * ( X[i] + Y[i-1] )
into DF I:
Y[i] = ALPHA * X[i] - (ALPHA-1) * Y[i-1]
This tells me that the system function is:
... |
In order to get more detail when encoding the blacks, a log LUT is going to be applied to the pixel values. Like gamma, the logical place to do it in the image processing pipeline is after demosaicking, so the same LUT is applied to the R,G,B components.
However, it occurs to me that a very similar effect can be ach... |
Which measure should be considered better and when ?
I tested both the measures on some data that I have and I got mixed results i.e. some are showing better results with Pearson and some with the Normalized Squared Error.
By similarity I mean similar shapes of the signals that I have.
I normalized the data before appl... |
I want to find all occurrences of a specific phrase in a set of audio files (it's a catch-phrase in a TV show). The phrase is pronounced by different people with different speed etc. What is the best approach to this problem?
Additional details: It's a one time thing. The end result should be a program that takes 30 ho... |
This is simple i thought, but my naive approach led to a very noisy result. I have this sample times and positions in a file named t_angle.txt:
0.768 -166.099892
0.837 -165.994148
0.898 -165.670052
0.958 -165.138245
1.025 -164.381218
1.084 -163.405838
1.144 -162.232704
1.213 -160.824051
1.268 -159.224854
1.337 -157.383... |
I am not very experienced in signal processing, and it seems like this should be a simple problem. I have two computers, one plays a tone a 19 kHz at a sampling rate of 44.1 kHz, and the other listens with a sampling rate of 44.1 kHz and tries to find if the 19kHz tone is present. The transmission of the signal is in a... |
I'm trying to get MFCC from speech audio data. I'm using 26 Mel Filter on 0-24kHz spectrum. My MFCCs doesn't look correct to me, cause always first bins have high values and values getting smaller through the high frequency bins, even with high frequency audio data.
Then I realized that the cos((PI/N)*(n+0.5)*k) part i... |
i am interested if where can i find matlab code for Prony method for frequency estimation.there is pdf article about prony
https://www.google.ge/#site=&source=hp&q=prony+method+for+distorted+signal&oq=prony+method+for+distorted+signal&gs_l=hp.3...2187.8079.0.8365.0.0.0.0.0.0.0.0..0.0.ernk_timediscountc..0.0...1.1.... |
I am beginner and I want a help to know how to detect hand in Matlab.
First, I need code which detects skin and then draw a rectangle around a region so it show to me face and hand.
When I try to make rectangle only around hand and the hand is away from the face, the the rectangle is drawn around the face instead.
What... |
I am currently in need for filtering accelerometer data for an Android application.
First of all, I used a simple low-pass filter (simple infinite impulse response filter) as follows:
for i from 1 to n
y[i] := y[i-1] + α * (x[i] - y[i-1])
This helped me achieve a smoother result.
Then I decided to play with FFTs. I... |
i'm looking for a free/opensource solution for digital image processing with reasonable performances processing quite large images (e.g. actual smartphones have resolutions of 8-10Mpx) and being ready for fast-prototyping/research.
Octave would be a good solution as it provides a lot of statistical and matrix manipula... |
Im trying to cross correlate two signals in matlab and get the phase difference between the signals.
For cross correlation (the idea is to do it without xcorr) I used:
Cxx=fftshift(ifft(fft(x,N).*conj(fft(y,N))))/(norm(x) * norm(y));
I get the result and not sure about the reference point I have to take for phase calc... |
Can anyone explain what are Autoregressive Coefficients? What is their meaning that is.
Consider a method:
public double[] calculateARCoefficients(double[] inputseries, int order)
When this method is called on 256 values, lets say these values represent some kind of signal, I will get an array of n numbers, where n=ord... |
I want to implement a HPF which will pass all frequencies above 150hz.
Fs = 1000; % Sampling frequency
T = 1/Fs; % Sample time
L = 1000; % Length of signal
t = (0:L-1)*T; % Time vector
% Sum of a 50 Hz sinusoid and a 120 Hz sinusoid
x = 0.7*sin(... |
For the purpose of controlling DIY disco lighting, I'm looking at implementing beat detection in a microcontroller with limited computing power (i.e. Arduino Uno). While detecting all beats would be a plus (for one, it would allow measuring BPM, which could be used to adjust the light to the style of music), it's not a... |
This code was working fine when I asked about it in Filter is too large but when I tried it again on another PC it gives me this error:
Warning: Direct access of structure fields returned by a function call (e.g.,
call to Untitled2) is not allowed. See MATLAB 7.10 Release Notes, "Subscripting In... |
I am working on a university project in which I need to find a periodic net element in an image.
The net is a set of diagonal lines in a specific angle ( which I don't know in advance ) on a noisy image. They are hard to detect so I would like to use all the information I can get my hands on.
My current algorithm is b... |
I am trying to make a frequency domain adaptive filter in matlab. It uses matlab adaptfilt.fdaf to create the filter parameters like step size and initializing initial filter weight values. Then I have tried to implement the overlap - save frequency domain adaaptive filter algorithm from the paper "Frequency-domain and... |
For an FSK modulation with coherent detection, in order to ensure orthogonality, the frequency separation between symbols is $ \Delta f=f_2 - f_1 =1/2T$. However, in case of non-coherent demodulation, which must be this value $1/2T$ or $1/T$?
|
Can we compare the size of two objects appearing in different images with different resolutions?
|
In this code I try to decompose music from singer sound with wfilters but when I hear it it is still mixed where Iam I wrong ?
function s = scale3(q,n)
p = length(q);
if p==n
s = q;
else
s = resample(q,n,p);
end
the program
s = wavread('K:\Khaleel_3aaza (1).wav'); % whatever here
[l,h] = wfil... |
One of the problems in Proakis's signal processing book using Matlab asks to plot the histogram of
x3(n) = x1(n) + x1(n-1)
where x1(n) is a random sequence.
To create x1(n), I do the following:
n1 = 1:100000;
a = 0; b = 2; % interval over which to pick random numbers.
x1 = a + (b-a)*rand(100000,1);
N1 = hist(x1,100);... |
I'd like to know what the parameters for the two-dimensional gaussian distributions are, which create this (I think) quite common image-processing filter:
$$
h_\sigma(x,y)=
\left[
\matrix
{
1&2&1\\
2&4&2\\
1&2&1
}
\right]
$$
The two-dimensional gaussian equation looks like:
$h_\sigma(x,y)= A\cdot e^{-\frac{x^2+y^2}... |
Although I have a solid experience in designing audio engines and such, I am fairly new to the realm of Digital Filter Design, particularly IIR and FIR filters. In other words, I'm trying to learn as much as I can on how to design filters and derive their difference equations. I'm starting from the basics, so please be... |
I am trying to calculate an actual frequency response from a desired frequency response.
fftSize = 128;
fs = 8000;
I have 2 input arrays: f[] and aDesired[], both of which are Real and length fftSize+1 (DC + Nyquist included). The f[] array is linearly spaced between 0 and fs/2 (0:4000). The aDesired[] array is either... |
Can one in theory achieve as high accuracy as needed by sampling a sinusoidal signal over n Periods? Is there a formula that connects the signal frequency (f), sampling frequency (fs), periods (P) and accuracy (Acc) or can someone explain how accuracy rise with number of periods? Let's say I want to know the signals am... |
Is there any method of decoding a Manchester encoded signal (IEEE 802.3) that is independent of the bit period?
Suppose the data I want to send is $\rm 10110$. The Manchester code of this data will be $\rm 0110010110$ (according to IEEE standard). We can see that in the $\rm 0101$ part, we need to consider the $\rm 0 ... |
As a rule of thumb, the bandwidth of a communication system with a complex signal must be equal to the bit-rate or symbol-rate?
Besides, after reading this post, it is not clear for me how a band-filter or low-filter signal can influence the maximum data rate.
|
I need to visualize some data as gray-scale images after contrast enhancement. The processing algorithm must be built on few samples, and then it will be applied on unseen images (suppose, the [min, max] range is given).
I plan to:
Use sample images to estimate actual histogram H1.
Define desired histogram H2.
Build a... |
What is the significance of the first zero up-crossing of a cross-correlation? The signals being correlated are cosines signals with some phase difference.
EDIT 1 : Cross-correlation plot attached. The signals are cosines with phase difference and may be with some low frequency components.
Sampling frequency of the me... |
I am making a pronunciation Scoring system, and I need to know how to separate the pronunciation information of an MFCC of a given spoken word, so that later i can match this MFCC with the MFCC of the word. can that be done threw matlab or HTK ?
Second Question:
In the matching part, shall I used DTW or standard edit ... |
I have recently fallen into fallacy, considering pole s=1 as there is infinite response at frequency 1. Yet, response was only 1. Now, can you derive the frequency response, given the poles?
Secondly, the theory says that a system is stable when poles are in left s-plane and, thus, decay in time. But, wait. Does'n "po... |
Suppose I have an FM signal with modulation index mf which is passed through a frequency tripler, then what will be the modulation index of output waveform ?.
Thanks in advance....
|
I am going to start my research on digital image restoration, I just want to know that should i learn the course digital signal processing before starting my research.
|
I am dealing with images from biological slides and want to segment out the ROI (cells) from the background. I have tried k-means clustering, otsu thresholding, adaptive thresholding, edge detection etc. Running out of ideas now.
Which other segmenting techniques can be tried for such images? Keeping in mind that the c... |
Can anyone recommend the best references for independent subspace analysis (ISA). I have read 'Multidimensional Independent Component Analysis' by Cardoso and would like to know of any modern, solid (i.e. generally accepted to work) approaches.
If you have implemented ISA before how did it perform?
|
In a set of images taken with the same infrared camera, I have a noise signal presumably originating from the camera chip. How would I go about extracting said signal to be able to substract it from all images I have taken?
|
I am programmatically generating a CPFSK encoding, and between shifting from two frequencies, there is extra noise generated by my speaker. For instance, going from 18375hz to 18900hz, there is extra noise. I am changing frequency at the point where the 18375 hz sinusoidal is at 0, and beginning at 0 for the 18900hz si... |
I'm making peaking equalizer in C# using NAudio (EQ-Cookbook). Can anybody tell me what is wrong with this code? It gives me noise.
Sliders
class Equalizer : WaveStream
{
float[] sFc;
double a0, a1, a2, b0, b1, b2;
public Equalizer(WaveStream source)
{
sourceStream = source;
sFc = ne... |
Prewitt operator is a common operator to compute image gradient in edge detection, and I knew that to apply this Prewitt operator to an image is just to convolve them together, say, the image is represent as $A$, and as the wikipedia page lays out, the Prewitt operator for *x-gradient is $$G_x=\pmatrix{ -1 & 0 & 1\\\ -... |
I am trying to implement a Weighted Matching Pursuit algorithm. Essentially I would like to weight the atoms of a Dictionary (redundant matrix) with a frequency masking function, like the one specified for MP3. I have read some of the papers, and theory of frequency masking, but I am still having trouble figuring out h... |
I am trying to smooth my discrete-time data points using the method of WMA.
Currently, I am using n as the window size and the weight array, {n/(n(n+1)/2), (n-1)/(n(n+1)/2), ... , 1/(n(n+1)/2)}.
If the y-value of each point is irrelevant, I can just simply randomly choose my size n.
However in my case, I hope to reserv... |
I have a complex data series, and I would like to fit a fixed number (in this case, two) of complex exponentials of the form $Ae^{jBn}$, where A, B are complex. Not interested in the phase (i.e. arg of A), just its magnitude, and mainly the complex frequency. Target length is maybe 100 samples.
I would like to be able ... |
I know that both SIFT and SURF descriptor vectors are normalized with (unit vectors) to deal with illumination changes, but as far as I can see, the SIFT descriptors are also normalized with the major orientation of the keypoint.
Does anyone know if the SURF descriptor does this, as I cant find anything about this in t... |
I need to compute the eigenvalues and eigenvectors of a 3D image Laplacian. I'm trying to evaluate the heat kernel on the 3D uniform grid (the uniform structure generated by the voxelized image) at different time values, to implement a Volumetric Heat Kernel Signature (please see the "Numerical computation" section). T... |
I'd like to do a deconvolution of image. For example for convolution I'm using a $3\times 3$ mask with all elements $= 1$:
$$\begin{bmatrix}1 & 1 & 1 \\ 1 & 1 & 1 \\ 1 & 1 & 1\end{bmatrix}$$
For a pixel $X$ as follows:
$$\begin{bmatrix}X_1 & X_2 & X_3 \\ X_4 & \ \mathbf X & X_5 \\ X_6 & X_7 & X_8\end{bmatrix}$$
which i... |
I hate this kind of questions, but I'm really stuck. I'm trying to implement Lucas-Kanade algorithm as described in this paper (see pages 4 and 5). Unlike most explanations I've seen they don't assume optical flow, but instead use affine transformation for most examples and in general refer to warp function just as $W(... |
In A3E transmission, I wonder how a single antenna is used to transmit a carrier wave, upper sideband and lower sideband altogether at the same time?
|
I have the original signal, a .wav file and by blind source separation, i have reconstructed the signal.
How can I calculate the SNR in this case? or is it good to use any other performance parameters? What should be an acceptable value of SNR?
|
I have the impression there is little added value for more than 100 dB of attenuation in a stopband of significant width (i.e. not just a specific frequency). What is the point at which there is little practical application for more attenuation (80 dB, 100 dB, 120 dB, ...) over a stopband of significant width? Also wh... |
I read various papers about the log polar transform and its application on template matching with images and have some questions:
In "Image Registration Using Log Polar Transform and Phase Correlation to Recover Higher Scale" the authors say:
"In this algorithm first the sense image is downscaled by the factor of 2. L... |
I've looked at various resources about Delta-Sigma modulators, and I find them fascinating. However, I'm confused on a particular point and and am looking for an explanation.
In my understanding, the Delta-Sigma modulators operates by sampling at a very high sampling rate (much higher than the ulitmate rate) an input ... |
How exactly is a wavelet used digitally?
Wikipedia states:
For example, a wavelet could be created to have a frequency of Middle
C and a short duration of roughly a 32nd note. If this wavelet was to
be convolved with a signal created from the recording of a song, then
the resulting signal would be useful for de... |
I'd like to create an image classifier that takes a set of labeled images and creates a classifier to be used on some unknown images. Most of the examples I see assume all images are the same size (m x n pixels), but my training set will have pictures of various sizes. This is a problem because training my classifier w... |
I'm implementing a very basic forward and inverse DWT, based on Mallat's book. Nothing fancy, simple inefficient matlab code (proof-of-concept more than runtime necessary, so I'm not focusing on optimisation). It almost works. I'm getting a distortion along the top and left edges. I'll post link to (need 10 points to p... |
Define an $n\times N$ sensing matrix $A$ by $A_{ij} = 0$ with probability $p$, and $A_{ij} = 1/\sqrt{n}$ with probability $1-p$. Does $A$ satisfy the restricted isometry property?
For reference, the symmetric case is answered in the following paper:
R.G. Baraniuk, M.A. Davenport, R.A. DeVore, and M.B. Wakin, "A simpl... |
I have a signal which looks something like:
The 12 main round peaks is what I'd like to keep. However, the peak detection algorithm I apply is picking up the in-between noisy peaks. Any strategies I could use to get rid of this noise?
I tried smoothing the signal, but the noisy peaks are still there.
Thanks
|
I would like to combine two images A and B in the following way:
1) I want to take a Fourier transform of both of them
2) For image A I want to apply a weighted filter, which gives more emphasis for low frequencies
3) For image B I want to apply a weighted filter, which gives more emphasis for high frequencies
4) I wan... |
Ok I need to understand how some harmonic calculations work to apply some transforms.
If I have a periodic functiion f(x). And I have the values of that function for several values of x (say x goes from 0 to 359) so I have
f(0)=3
f(1)=5
... etc
So if I'm asked to take the max of those values without the first harmonic.... |
The problem
Take 50+ of Lena images, add gaussian noise to them to the point that you couldn't say it is Lena. I insist, gaussian noise, not gaussian blurring.
Now imagine a cropping window randomly moving around Lena's face (like a video camera focused on this part).
Now you have a handful of noisy pictures containi... |
I want to simulate a multipath fading channel, different amplitude and delay is given. I want to use delay in padding style. I read that the different values for delay like [0 .6 3.1 ...] represent the delay between the first and last signal arrival? Giving the number of paths, How to simulate it?
regards
|
Having had little experience in PHP, I moved to C++ in-order to create a spectrogram, I wanted to create probably the simplest/easiest one for my first project, so easy infact I don't need reading/write to files or add settings to it.
But, as far as I have been learning C++ I was able to manipulate audio file, only to... |
In some papers, I read that the additive noise is band limited Gaussian white.
How can I simulate this type of noise use MATLAB?
|
I am looking for a method to find the region where a function is periodic. So, for example, I have a function that is periodic in some places and not in others, I would want to know the range of x values where it is periodic.
I'm trying to do this in matlab and I tried to use autocorrelation but I can't use it to find... |
Assuming I have samples of a colored noise.
I know the empirical Variance of those samples.
I also know the bandwidth of the filter which colorized the noise (Only the bandwidth, not the actual filter).
Now I want to estimate the variance of the noise if I filter those samples by a (Different) given filter.
Is there a ... |
I'm currently facing the following problem: I want to approximate the Fourier transform $F(\omega)$ of a (let's say, $L^2(\mathbb R)$) function $f(x)$ by calculating the discrete Fourier transform, using only a fixed number of sampling points. These points are allowed to lie arbitrarily far away from the origin. In add... |
I have a periodic series of 360 values. How can I use the fftw c libraries to get the amplitude and phase of the first, second, third and fourth harmonic?
If I do this for N=360
fftw_plan_r2r_1d(N,input_array,output_array,FFTW_R2HC,FFTW_ESTIMATE);
I can get the 360-point Discrete Fourier Transform (DFT) where output_a... |
I have implemented Dutt & Rokhlin's FFT algorithm [1] for non-equispaced data but for some reason I am getting very large errors ($E_2=0.04$ and $E_\infty=0.1$). I was wondering if there are any free available implementation of this method so I can compare my implementation.
[1] Dutt, Alok, and Vladimir Rokhlin. "Fast ... |
I am trying to make proper use of the circular convolution property of DFT.
I was taught that the DFT of x[n]*CircularConv*y[n], would be equal the product of the individual DFT's X[k],Y[k].
On the problem im trying to solve, the signal x[n] is convolved (Circular convolution) with the discrete impulse response y[n] to... |
I'd like to build a color difference metric that will be invariant to small changes in lighting conditions. In other words, I should be able to tell that a red shirt is not the same as a blue shirt, even if one image is lit differently (brighter / darker, outdoors / incandescent).
Moving to YUV space and ignoring the ... |
Hello i am working in digital image restoration field, recently i have studied concept of convolution, i studied that to find the impulse response/point-spread function of an LTI system, an unit impulse function is passed from the system and its output is considered as impulse response, My question is that, Why an unit... |
The task that I have is to remove the annual and semiannual oscillation from a set of temperature measurements, taken over several years, by means of least squares method.
I found the method described here and since I have to wait for my colleague to first make some analysis on the data and pass them on to me, I decide... |
I would like to use a .WAV file and convert it into a spectogram. Any idea how to do this with Naudio? Basically the flow should be like this, I think:
Convert WAV file to PCM stream
Convert PCM stream to double-array and applying FFT
Using the result from (2) and convert it to a spectrogram
|
Given a discrete signal $ \left\{ x[n] \right\}_{n=0}^{N-1} $ and its Discrete Fourier Transform $ X[k] = DFT \left\{ x[n] \right\}[k] $.
I'd like to effectively compute the norm of each sample of the DFT, namely $ \left | X[k] \right | ^ {2} , k \in \left\{ 0, N-1 \right\}$ as opposed to the norm of the DFT series... |
I am using HTK to recognize which word has been said.
I am working on 20 words and I have for each of them 10 .wav files.
Here are the steps that i followed:
1) get the MFC file for each .wav file for each word
2) for each word train it on the 10 MFCs files
3) add the grammar and dictionary and hmmlist in the HTK
4) t... |
I'm trying to come up with a formula for the average energy of the signals in a M-QAM constellation.
If $b$ is the number of bits represented by each point in the constellation, such that $M=2^b$. For $b$ even, I know the formula is:
$$ \bar{E} = \frac{(M-1)d^2}{6} $$
where $d$ is the distance between two points in th... |
I would like to build a computer vision application capable of doing robust pedestrian tracking in potentially crowded environments from a single video source. To this end I would like to find resources in order to learn as much about the current state of the art as possible. This could be:
recent survey papers
papers... |
First of all, I'm new to DSP so excuse my simplified words.
I'm testing the performance of a digital filter on a (partly) noncontinuous signal:
As you can see, the signal is not continuous at some points (like it is stopping and starting over again) When I apply the digital filter to it, I get this:
The filter shows ... |
What is the purpose of DFT & IDFT blocks which are placed after the Analysis filter bank & before the Synthesis filter bank.
The reason i am asking this question is because I tried to design Polyphase DFT filter bank & got the same output with & without DFT-IDFT block.
|
In the above image I have created a time series (in cyan) that is a simple addition of a major cycle (in green), a minor cycle (in cyan) and a linear trend (not shown separately). What I would like to do is filter this time series to remove the minor cycle so what is left would be the trend plus major cycle (in red). ... |
So I have the following data:
[[-344 13771 4600 ]
[-275.2 12478 6410 ]
[-206.4 19443 830 ]
[-137.6 69392 3830 ]
[ -68.8 143737 3780 ]
[ 0 189278 16870 ]
[ 68.8 184486 5090 ]
[ 137.6 188466 9380 ]
[ 206.4 185023 21680 ]
[ 275.2 128133 1460 ]
[ 344 51288 1950 ]
[ 412.8 10854... |
I have an x-axis which is in logarithmic scale representing frequency, I would like to have NumPoints on this axis that are equidistant on the log scale, I have the following variables:
Fmin = 100
Fmax = 4000 or 8000
NumPoints = 10
What formula should i use to calculate where the NumPoints-2 points between Fmin and Fma... |
Let $x$ be a $N \times 1$ vector in $\mathbb{R}^{N}$ where $M$ components are zero and the remaining $N-M$ components are standard normal random variables. $x$ may not be sparse e.g. $M$ may be small.
I am interested in bounding $||Ax||_{l_{2}}$ where $A$ is a $K \times N$ matrix $(K<N)$.
This made me think to look f... |
I have an FDM signal from 0 to 70 MHz. The total signal contains 4 FDM signals from (0-fs/4), (fs/4-fs/2), (fs/2-3fs/4) & (3fs/4-fs).
I used fvtool command in matlab to plot these signals.
Then i downsampled the total signal by 4. Downsampling is time is expansion in freq spectrum.
Thus on paper my new signal should o... |
I have a set of signals consisting of impulses at different instant of time i.e. its not a continuous signal. What I want is a mathematically intuitive way to distinguish between any two such signals?
I understand that the time of occurrence of impulses can be used to differentiate between the impulses. But what I need... |
I have several analog signals with constant amplitude in time. What I want is a way to convert these analog signals to an impulse signal i.e. a signal that consists of impulses occurring at different instants of time.
The conversion should be such that it should be possible to differentiate between different analog sig... |
I would like to estimate the peak-to-peak amplitude of a periodic signal whose frequency components are known. This is, I have the frequency spectrum (a peak in the fundamental frequency and other peaks in its harmonics) and I would like to compute the peak-to-peak amplitude. Is that possible without reconstructing the... |
Let's say you have an original image and a version of the same image that may have been convoluted with a Gaussian blur. How could you demonstrate that the Gaussian blur has been applied and calculate the blur radius?
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