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I'm trying to reproduce this !!!WARNING LOUDNESS!!! sound in C. In the website, it says to use Carrier: 500 Hz, modulator frequency: 1 Hz, modulator index: 500, however I'm getting lots of clicks and fuzzing (along with the right sound) with this code:
#include <stdio.h>
#include <math.h>
#include <string.h>
#define D... |
Say you want to compute the least squares estimate of $w$ from a data-set:
$$
\begin{bmatrix}d_1 \\d_2 \\\vdots\\d_N \end{bmatrix} =\begin{bmatrix} x_1 \\x_2 \\ \vdots \\x_N\end{bmatrix}w + \begin{bmatrix} q_1 \\q_2 \\ \vdots \\q_N\end{bmatrix}
$$
Note: $q$ is white noise.
This can be compactly written as
$$
\mathbf{... |
I have an irregular shaped 3D image that I want to do frequency domain filtering on. Currently I simply zero-pad the data to make it a cube, then do an fft.
I was wondering if this was a valid approach and what effects this may have. Are there any alternate (better ways) to do this, and if so could you point me in the... |
during Zimmerman trial, each camp claimed the yell for help recorded belonged to their side. given that Zimmerman is still aside and the recording still exist, could there be some scientific way to tell which way?
|
Savitzky-Golay smoothing filters of polynomial degrees 4 and 6
have frequency response with ripples of roughly opposite signs.
Thus averaging the coefficients of two filters of degrees d and d+2
is an easy way to attenuate the stopband:
Can anyone explain the roughly opposite ripples —
or are they just a lucky coincid... |
I have three orthogonal accelerometers that give accelerations in x,y and z dimensions. I'm trying to detect directional vibrations between certain frequencies.
In other words, my input is 3d vector and I would like detect signal coming from unknown direction, filter out noise from other directions and turn the signal ... |
Hello I am trying to computer the Complex Cepstral Transform of an audio signal. I am using wavread to get an array of data type double precision. However, when I try to use cceps to do the Complex Cepstral Transform, I get the following error:
Error using cceps>chkinput (line 157)
The input signal X must be real.
Er... |
I'm looking for literature or other information that covers iteratively estimating the auto- and cross-correlation vectors and the auto-correlation matrix.
Initially, I can assume the signal is wide-sense stationary, but I also have to deal with changing signals and statistics.
For what it's worth, I looked at two sim... |
Hello I am working on audio watermarking in MATLAB and am trying to choose specific frames of the audio signal where the watermark will be embedded. Through research, I have read that the best way to do this is to choose parts/samples of the audio signal that are the loudest, or above a certain threshold. By using wavr... |
After calculating the principal component analysis (PCA) of a given data set, we are normally left with a matrix containing the eigenvectors sorted in order of the size of the eigenvalues. Now, in pattern recognition which eigenvector should we choose: the first eigenvector or do we have to do further processing of the... |
I'm analyzing speech signal for identifying voiced and unvoiced regions. Voiced regions are supposed to have a "pitch", which can be estimated using auto-correlation function (ACF). Basically, one estimates ACF for each frame of speech (say 20ms) and then finds the time lag between peaks in ACF output. If all the signi... |
I have gigabytes worth of seismic ground data, that I wish to upsample by a factor of 2 or 3. I wanted to ask what efficient way might there be for me to upsample such a big file? Is there a way to break it down and do it in mini-files? (I am using MATLAB for my software, I think the resample command might take too lon... |
I am not sure if this question suits this site, but I want to convert humming into notes or relative values to analyse melodies, word of "Signal Processing" looks suit for this. I am planning to use Flash AS3, and Tortoise and Hare for Cycle Detection. I need very simple script implementation. All I want is find notes ... |
I am currently frequency filtering a 3D image in matlab, with x,y,z dimensions of different lengths. The matlab filter takes normalised frequency as an argument. What does this mean for an image and how do I calculate it?
Note : the distance between each pixel is 2mm
|
I was reading a couple of these threads and I was wondering:
Is it possible for an FFT signal to include frequencies less than $1/N$?
Consider a time-series of length $N$, and I perform an FFT on it. In the zero-th bin of the FFT is the DC signal, and first bin is frequency $1/N$, in the second bin is frequency $2/N$ ... |
Hi: I'm new to this list but I had a question that's not exactly related to dsp but maybe in a slight way and I didn't know where else to send it. So, here goes and my apologies if this is totally the wrong place to send it.
The following describes the background as far as number generation.
Suppose have a series of ... |
I have an application that displays several signals from different sensors at various sample rates. In order to display the data, I "stretch" the signal by repeating samples to match the highest sample rate. For example: if I display a 500 Hz signal and a 100 Hz signal at the same time, every sample in the 100 Hz sig... |
I am reading a paper on SIFT by Lowe.
In it, gradient magnitudes and orientations are calculated using these formulae:
$$$$
I don't know why m(x,y) is calculated above. I heard that gradient magnitude is usually calculated like the following.
$$\sqrt{f_x ^2 + f_y^2}
$$
|
i want to design a filter that can effectively detect the QRS complex in ECG signal(whose Fs=360Hz) and reduce high frequency noise components. i want to know as to how the center frequency is detected so as to design a digital lowpass differentiator filter with bandwidth equal to the center frequency of the QRS comple... |
Suppose I have some amplitude modulated carrier wave $x$. If I have no access to the original message wave nor the unmodulated carrier, is it some how possible to calculate what is the modulation depth of $x$?
If I have understood this correctly, amplitude modulation depth is given by
$M/A$, where $M$ is peak amplitude... |
This is the first time that I post a question here so excuse me for asking such a beginner question.
what free open source 3rd party libraries which have good documentation(because I'm new to using 3rd party libraries) would you suggest me for fourier transform,inverse fourier transform , meshgrids , dynamic arrays an... |
I am trying to transfer text data via sound wave between two android phones. I am a newbie and would like to know how to proceed with it. I am stuck at the very first step on how to modulate text data as FSK or PCM raw data ? As I shall be implementing it on Android any Java pointers will be of great help
|
The best standard may be varying from PSNR to human's visual feeling. Any one is OK.
I have tested J.Weickert's tensor diffusion and P&M's, they all can't satisfy me.
Is there any anisotropic diffusion method now is best for denoising, especially in tensor diffusion, others like the KAD, MAD and so on is ok as well.
Th... |
Is a radix-4 implementation faster than a equivalently well coded radix-2 FFT? And if so, why would it be faster?
|
Question:
I want to kick up a discussion on how people set algorithmic parameters when no validation against groundtruth is possible (maybe because groundtruth just cannot be obtained or is very hard/tedious to obtain).
I have read numerous papers and implemented the underlying algorithms wherein --- a set of paramete... |
I'm trying to implement a QPSK burst modulation and demodulation through a channel and I'm now considering the synchronization part.
I've read that Barker sequence were excellent option for this as their autocorellation looks like a dirac.
So far I know that they're composed of 1 and -1 and that the longest found is 13... |
I just have a quick question. I'm a second year computer science major with a background in C# and C++. I'm interested in studying neuroscience after graduation, and I've been researching some subjects on my own in my free time. I'm thinking of purchasing a Neurobit Optima 4 and writing some applications for it (specif... |
I want to check the effect of this channel model on my signal transmitted with "impulse response-Ultra Wide Band" transmission (use very short pulses ~0.5 ns each), but i don't know where to find this channel impulse response or how can I generate it?
if there is only one line of sight (LOS) path and one non-line of si... |
For this question purposes a function, $ f \left( t \right) $ is said to hold the Partition of Unity if:
$$ \forall t, \ \sum_{n \in \mathbb{Z}} f \left ( t - n \right ) = 1 $$
Using the Poisson Summation Formula the Partition of Unity is equivalent of $ F \left( 2 \pi k \right ) = \delta \left ( k \right ) $ where $ F... |
I need to compare FFT result from audio files,
I have 2 FFT result from 2 audio files..
FFT 1 (sample)
-0.16314493488504767 - 0.103707391105263i
0.07863935536550609 - 0.30111206509352917i
0.06753127619272284 + 0.02837438103569126i
FFT 2 (sample)
-0.006953384463464084 + 0.0291883094944081i
-0.09447919915711694 - 0.2267... |
I'm decoding a pulse amplitude modulation encoded digital signal. The goal is to detect the individual pulse against a background noise floor. The amplitude of the pulses varies depending on the signal strength and quality.
My sampling rate is 2x the bit width such that one bit fits into sample. Thus a single bit is th... |
I'm having a lot of problems with my project:
I added salt and pepper noise by inbuilt function
I DWT function with Haar family
Now I want to remove noise from original speech signal.
I don't know how to do this
|
I need to design a moving average filter that has a cut-off frequency of 7.8 Hz. I have used moving average filters before, but as far as I'm aware, the only parameter that can be fed in is the number of points to be averaged... How can this relate to a cut-off frequency?
The inverse of 7.8 Hz is ~130 ms, and I'm worki... |
Given a frequency response obtained with FFT, I would like to apply a 1/n octave smoothing. What filter should I be using and how? Maybe someone could point to a good reference (a paper or book on the subject).
|
White noise has infinite variance, whereas Gaussian random variable must have some finite-variance to exist. How do two disjoint phenomena co-exist?
Help will be highly appreciated.
|
I'm looking for a way to visually inspect the amplitude of frequency components in a 3D image that I have taken the fftn of in matlab. Does anyone have any idea how to go about this?
|
I am doing research in audio watermarking and have come across the paper that uses the complex transform to embed the watermark in order to improve the robustness and inaudibility of the embedded watermarks.
This is done by embedding the watermark into the complex cepstral components of the audio coefficients transform... |
I realize that this might not be the most well-formulated question, but I figured I'd throw it out there as an open-ended discussion that could turn up some interesting input.
I'm relatively new to speech processing, and came across the statement "invariance to transformation is important in many speech engineering tas... |
I've been studying sift very hard for two weeks.
I found much materials about scale space.
It is very hard to understand scale space depthly.
What I've found and confusing things are that
'scale' in scale space means the sigma of gaussian function.
'octave' is used to represent images that are reduced.
In paper by Lowe... |
I'm trying to implement a speaker recognition system and want to make sure I'm aware of the latest trends in speech segmentation. I've read a number of very different methods at a higher level but I'm not sure which one would be best for my particular situation. I'll list what I found so far and I'd appreciate either (... |
I am new learner in the field of speech analysis. I am wondering there are so many resources which explaiins principles of speech recognition. Most of them are advanced topics. I am curious to know the methods behind the speech recognition. If any one can explain it or if any one have easy helpful resource expaining it... |
I presume we are all here well aware about foldback aliasing when sampling signals above the Nyquist frequency; i.e. half the sampling rate.
By contrast, the phenomenon of beating occurs when sampling a signal with a frequency slightly below the Nyquist frequency. You can see and hear some beats on this educational sit... |
The following image is from robert collins' ppt from http://www.cse.psu.edu/~rcollins/CSE486/
How to calculate sigma ?
It shows [1 4 6 4 1] is corresponding to sigma=1.
I learned that usually 3*sigma is best to represent. So kernel size is 7*7 is suited. But its size is 5*5.
I don't know why [1 4 6 4 1] is matched to s... |
I am working on Pedestrian dead Reckoning using Particle filtering. I am using mobile accelerometer sensor to detect number of steps and stride length and calculating the total distance traveled.
I want to use particle filter to enhance accuracy in my results. I have read the theory of particle filter but stuck at the ... |
I'm reading Schaum's DSP book, and in Fig 1-3 they demonstrate why shifting and reversal are order-dependent, showing a couple of simple systems, 1) delay followed by reversal, and 2) reversal followed by delay.
The equations they present as follows:
1) (Time delay -> Reversal)
$x(n) \rightarrow x(n-n_0) \rightarrow ... |
When we apply pulse-width modulation (PWM) on controlling the brightness of a LED or the speed of a DC motor, is it the lag in our eyes that is responsible for the feeling of a continuous powering or the inertia of the devices?
|
i will use matlab code just for convenience, i get the same behavior using a C++ code.
I have two signals:
Fp=5;
A=0.00025;
Phi=0;
Off=0.5;
KADC_5VOLT=5.0/32768.0;
tc=250e-6;
tmax=30;
t=0:tc:tmax;
Buf1= int32 ((A*cos(2.0*pi*t*Fp+ Phi*pi/180.0)+Off)/(KADC_5VOLT));
Buf2= int32 ((A*cos(2.0*pi*t*Fp+ Phi*pi/180.0)+0)/(KA... |
I'm a little confused on how frame based high dimensional features such as mfcc's and bfcc's are applied in the classification of musical instruments.
Are statistical measures of the features used e.g. using the standard deviation across the frames used as a feature? Or is each frame of coefficients used as independent... |
Does the following code properly implement a phase-neutral band-pass filter using SciPy?
freq_fraction = FREQ / (SAMPLE_RATE/2)
sharpness = 0.1
filter_length=128
fir = sig.firwin(filter_length, [freq_fraction*(1-sharpness),freq_fraction*(1+sharpness)], pass_zero=False )
filtered = sig.lfilter( fir, 1, data_to_filter ... |
I have to calculate the magnitude-squared coherence (MSC) between two signal. However, using a routine that uses only one taper (or no tapers at all) my result is always 1, despite the signals are clearly different. This doesn't happen if I use more than one taper. Searching a explication for this abnormal result, I co... |
making pink noise (1/f) using list of frequencies
I would like to see what type of noise I would get if I used just the frequency in my voice. I created a matlab array using fft to get the
[frequency,amplitude,phase] to reproduce my vocal signal.
I would like to take this file/data and use it to create pink noise (1/f... |
I am working in the field of digital image processing and currently i am reading this paper and I have two questions about it....
After applying a gaussian filter to a histogram, will the pixel values of the new histogram be changed or not?
And will the sum of pixels in new histogram be the same or not?
|
I am using an adaptive RLS adaptive filter for noise cancellation. My sampling freq. is 500 Hz, but I am interested in only frequencies of up to 60 Hz. I filter the input and the reference signal to the desired frequency range and then apply the adaptive filter. The adaptive filter does a good job at removing the noise... |
I have created a frequency domain image filter and would like to test it against some images where it is known that the frequencies only occur within a given band. Trouble is, I don't know how to go about creating such images.
I have though about making an image that consists of intensities varying about the x and y ax... |
My aim is to find cluster phonemes from a wav file. I would like to go through the following steps:
I am calculating spectrogram.
Calculate energy of each frame.
Cluster the frames using k-means clustering.
save the clustered frames as wav file.
Now I am in confusion, what should be the critarion to find the similari... |
I'm using cross-correlation to look for a Barker code in a BPSK signal. It occurred to me that if I send out a zero DC valued signal proceeding the barker code, it won't affect the correlation because a zero DC signal has zero correlation with any other signal.
This made me wonder if there's an alternative to cross-co... |
I am quite familiar with signal processing using Matlab. Now I am working in its open source alternative 'Scilab.' In Matlab, cconv() function is used to perform circular convolution between two sequences. I want to do circular convolution in Scilab. But I could not find an equivalent function for circular convolution ... |
I've been using F-K Filter, for a while, but I guess I never had good basic understanding about it Math. Someone asked me what is the cause of frequency wrap around in F-K Spectra plot ? I know it's because of aliasing. But if somebody please elaborate more on the cause of this wrap around? Simple Math explanation perh... |
This question is theoretical. Let's say you have three different highly compressed images of a same source image, each compressed with a different algorithm and thus "lossy" in different ways.
What would be the possible techniques to recover the best possible image, using the three different compressed versions ?
|
I've been trying to model some radio signal properties of wireless networks wrt its usefulness. For example, most WiFi network adapters would decrease their transmission rate if the overall quality of the signal is bad so as to avoid re-transmission, errors, etc.
What is the relationship of the received signal strength... |
Here is the problem. I have a 2D array of data, first column represents the time data and the second column represents the sinusoidal response data, based on the time data. I apply fft and I get my frequency (that I started with) in a specific bin as I expected and I find the amplitude and phase angle from that bin. No... |
I have asked the same question in stackoverflow ,https://stackoverflow.com/questions/17794772/steps-for-successful-image-recognition/17840975?noredirect=1#17840975, but since I am not getting considerable responses , I'm going to ask it again here .
In order to do successful image recognition what are the steps one ha... |
What I'm trying to do is, given some experimental data and it's (magnitude normalized to 1) FFT, match the FFT with one of say 1000 other (magnitude normalized to 1) FFT plots. Please note that the experimental/generated plots all have the same bin width/count.
Initially what I'm doing is taking the normalized cross co... |
I'm not sure if I'm on right place to ask a simple suggestion, so please be kind and don't downvote my question. I'm not asking someone to do it for me, I just need short guidance how should I proceed.
Currently, I'm writing a study of Blind source separation using eigenvalue decomposition. Since I'm not good at writin... |
I've read that OFDM normally uses a one-tap equalizer. It seems that a one-tap equalizer could only scale and delay a signal. Why is this useful?
I thought multipath was handled through the cyclic-prefix and by choosing a symbol width that is large compared to the delay spread. Why is an equalizer needed at all?
|
I'm currently using MATLAB's fdatool for filter design. Using that tool, I can easily design different kind of filters. For example, let's take a band-pass FIR filter with 10-40 Hz passband, and 5-10 Hz and 40-45 Hz transition bands. Usually, I design the filter with the selection "least-squares", which, if I understan... |
I've implemented a gaussian blur fragment shader in GLSL. I understand the main concepts behind all of it: convolution, separation of x and y using linearity, multiple passes to increase radius...
I still have a few questions though:
What's the relationship between sigma and radius?
I've read that sigma is equivalent ... |
As the title suggests , I'm looking for good books on medical imaging . It would be a great help if you can provide a link to the books website .
Thanks.
|
If I have a signal that has 740 samples, the sampling frequency is 1000 Hz, and I take an FFT of length 1024, are the frequencies I get now 0, fs/1024, 2*fs/1000, ..., 511*fs/1024? I mean, do I get this correctly: the frequency resolution does not depend on the number of samples I have, only FFT length and sampling fre... |
I have a thirty-second speech signal that was sampled at 44.1 kHz. Now, I would like to show what frequencies the speech has. However, I'm not sure what would be the best way to do that. It seems sometimes one calculates the absolute value of a Fourier transform, and sometimes power spectral density. If I understand co... |
I' trying to implement a QPSK modulation/demodulation.
At one point I have to decimate my signal by 4 for an easier implementation. I've read everywhere that the best way to do it is to use a CIC filter followed by a decimator. So I've read some more and I decided to use the following:
CIC Filter Length: 2
CIC Filter ... |
I have a problem with SIFT that I do not understand. Lowe [1] proposed in his work the s=3 levels of scale are enough for one octave. Afterwards, he mentioned that you need to compute s+3 levels.
Why there are 3 and not 2 additional levels required. I understand that you require one additional level above and one addit... |
I'm gonna implement multiple frequency estimation by summing harmonics amplitude!
I've read Anssi Klapuri paper but i can't understand him.
first of all he said spectral whitening i did it by i don't know whether it's true! after that i have to implement this formula:
$$s(\tau)=\sum_{m=0}^Mg(\tau,m)max_{}|Y(k)|$$
i ca... |
I want to estimate the direction of arrival of sound (angle in a 2d plane) using a microphone array. I'm hoping to follow a time-delay based method using a 4 mic array. The method requires me to find the time delay of the signal upon arriving at the different microphones. For example, in the 4 mic array, I'm required t... |
Suppose you have a transmission line carrying a digital signal and the line is picking up random noise such that half the bits in the original signal are randomly flipped before they reach the receiver. What techniques are available to recover such a corrupted signal?
|
for the (7,4) Hamming code, I know how to decode a codeword using the syndrome decoding method, by just using the 3 parity bits, and it is correctable if there is just 1 error.
However, for the (8,4) extended Hamming code, I know how to decode the codewords correctly when there is no error or 1 error respectively. Howe... |
The following is from Lowe 2004 paper ( http://www.cs.ubc.ca/~lowe/papers/ijcv04.pdf ).
One obvious approach would be to sample the local image intensities around the keypoint at the appropriate scale, and to match these using a normalized correlation measure.
However, simple correlation of image patches is highly sens... |
I'm trying to create two tones at different frequencies. So I have one tone at 15000 Hz and one at 15400 Hz using Audacity. The problem i'm running into is there is a third tone created at a much lower frequency I can tell exists from both my ears and a spectrogram.
How can I get rid of the lower tone? Where does it co... |
There is a Dirac pulse train following the scheme of the Shah function (or $\delta$-cumb function) with its Fourier series of the form:
$$\varsigma(t,T)=\sum_{n=-\infty}^{\infty}\delta (t-nT)=\frac{1}{T}\sum_{n=-\infty}^{\infty}\exp\left(\frac{i 2\pi n t}{T}\right)$$
here $n\in \Bbb Z$; $t$ time and $T(\in \Bbb N)$ the... |
If I have a sound recording with a sinusodial frequency with a constant amplitude and I do an FFT on the time domain of this recording to get the frequency back:
If I were to ignore the imaginary part and just watched the amplitude (real part) would there be anything different about it? Would it also be constant?
I'm t... |
In Harris corner detection, as I see, the goal is to find out those points $(x,y)$ which makes $S(x,y)$ a large value for any direction of shifting, $$ S(x,y) = [\Delta x , \Delta y]M\left[ \begin{array}{c}
\Delta x \\
\Delta y \\
\end{array} \right]$$, while $$M = \sum_{(x,y)}\left[ \begin{array}{cc}
I_x(x,y)^2,& I_x(... |
I begin to study how codec works, and I found a "summary-diagram" for DCT-codec (DCT stands for Discrete Cosine Transform). On this diagram, there are of course two steps: the encoding step and the decoding one. I try to make a reproduction of this diagram herebelow:
ENCODING STEP Image (8*8 blocks) -->| FDCT |-->| Qua... |
I have a set of 3D data points, indicated by the blue color in the picture below.
I then project them onto the x-y plane, i.e. setting z values of all the points to 0, shown by the yellow color below.
Then, I perform PCA on the 2D data points (the yellow ones).
I end up with the 1st Principal Component being the green ... |
I have a set of songs for which I extracted the STFT (Short-Time Fourier Transform) and used the magnitude spectrum $|S|$ to calculate the mel spectrogram by using a mel filterbank matrix $M$, so $X=\log(M\times |S|)$. I want to know is there any method to reverse this process, i.e. convert from the mel spectrogram bac... |
I am looking for a way to completely describe which kind of digital filters I used to process my data, although it seems only few publications in general report all the details.
I think I should describe at least
Which type of filter was applied (low-pass, high-pass, etc).
Cutoff frequencies
IIR / FIR
Phase response? ... |
For a current project I need to implement an algorithm that needs the autocorrelation sequence of the channel, as part of a key step.
We're measuring the frequency utilisation in our local area - simply be scanning the spectrum (from 440 MHz to 780MHz) with an SDR and collecting the power at each frequency. The goal o... |
I am new to signal processing and im trying my best to figure out how to build a high pass filter then later on implement it. (in c#)
Edit: I rewrote the question with math notation,
if someting is still wrong i will try to correct it. It is my first time to write a question in math notation.
This is to learn and for m... |
I've tried googling and wikipedia-ing it, but I haven't gotten any answers beyond 'it's because the frequency of the input signal is sitting between two bins'.
I understand that this is the reason, but what I can't understand is why the leakage seems to extend to several adjacent bins rather than only one adjacent bin.... |
I have seen this be mentioned a couple of times in some books I read, so I want to make sure. Is the complex envelope simply the summation of the real and quadrature components of a signal, whereby the absolute value is the (real) envelope? I have read this wiki page but I am not sure I completely understand. Is the co... |
I am running Morlet continuous wavelet transform. I have got wscalogram of signal and now I want to plot freq-magnitude like the following picture., but I don't know how to do it:
I've used scal2freq MATLAB function to convert scales to pseudo frequencies. Also I have some frequencies in my signal that have a large d... |
I'm trying to write a perona malik filter in HLSL
texture2D Input0;
sampler2D Input0Sampler = sampler_state
{
Texture = <Input0>;
MinFilter = Point;
MagFilter = Point;
MipFilter = Point;
AddressU = Clamp;
AddressV = Clamp;
};
struct VertexShaderInput
{
float4 Position : POSITION0;
float... |
I want to simulate a direct sequence spread spectrum (DSSS) system and the receiver works with matched filter. I want the received power to be different for each user, in other words, SINR shouldn't be 0. Is there any idea how to do that in matlab?
|
I'm currently implementing a voice recognition algorithm using FFT and now some questions came up to me:
Spectral Leakage: I know what it is, why it appears and how to solve it using a Windowing Function, but is there another work around for this issue?
Phase: I know what it is, but where can I use it and how?
Getting... |
In general, correlation-based image matching can be referred to as
$$c(I_1,I_2) = \sum_{x \in patch} f(I_1(x), I_2(x))$$
, I do think normalized cross-correlation like
$$ncc(I_1, I_2) = \sum_{x \in patch}\frac{(I_1(x) - \mu_1)(I_2(x) - \mu_2)}{\delta_1\delta_2}$$.
is a good indicator to show if 2 patches match or not.... |
How do I calculate the Spectral Entropy of a signal in MATLAB ? I know the basic steps but it would be nice if someone can help,
Calculate the power spectrum of the signal using FFT command in MATLAB.
Calculate the Power Spectral Density using the power spectrum or using any other technique.
Normalize the Power Spectr... |
As a follow up to this question -
Is there a way that I can calculate the amount of spectral leakage in a given bin, i.e., given an input amplitude, what percentage of the amplitude is present in the adjacent bins?
Or given an input frequency and the frequency range of each bin, can I calculate the amplitude present ... |
I'm working on Cognitive Radio in TV Band Spectrum. I need to sense the presence of signal in a particular spectrum. I have to sample the incoming signal (say 1000 samples) and I have to find the auto correlation strength for a smoothing factor L(means my sample will be 0 to L-1 samples delayed). I have assumed a smoo... |
Is there a simple or potentially intuitive explanation for, with the DFT, vector multiplication in one domain being equivalent to circular convolution of the transforms of the vectors in the other domain?
Since a DFT is just multiplication by a (special) square matrix, what about this matrix and matrix multiply allows ... |
I am working on a project that employs a linear array sensor that provides data from the same object at two different energies. Collected in time, I end up with two images (16-bit sensor values, MxM image), call one HIGH the other LOW. My engineering background is Material Science and metallurgy, so I am coming up to... |
I read the following on Wikipedia:
Power spectral density:
The above definition of energy spectral density is most suitable for
transients, i.e., pulse-like signals, for which the Fourier transforms
of the signals exist. For continued signals that describe, for example, stationary physical processes, it makes more... |
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