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I have 2 arrays of 800000 input and output data samples. The system is a kind of oven that works from 0 to 10 volts. The sample time is 0.001s. I have to identify the model of this system, but first of all, given that the data are clearly dirty, I would like to filter the noise. How can I do it with the System Identif...
I would like to emulate a signal with bandwidth (like BPSK) for some testing, but the two signal generators I have access to do not have any BPSK options installed. If I know the frequency (2462MHz) and the bandwidth (16MHz) I want, is there a way to mock this up easily with two signal generators? I feel like this is...
I am trying to record a sound two times and compare the two recordings. I am playing exactly the same sound both times (for example, play a small .wav file twice and record it each time), so I hoped that my attempt would not rise to the complexity of Speech Recognition... I have tried computing the DFT (using FFT) and ...
I read a paper about directional filter bank (Improved structures of maximally decimated directional filter Banks for spatial image analysis) in image processing and I don't understand clearly about the process. It says they use '2-band structure' like this image: Could someone explain it to me how this process works? ...
I have an fft function in matlab/octave that works great on an imported mono wav/audio signal that is 1 second long, but when the signal imported is more than 1 second the frequency values become incorrect. To check this I created a 2 second signal using audacity the first half of the 2 second signal (1 second of it) ...
According to JOS on stiff-string synthesis, stiff strings (like on a piano) introduce inharmonicity (i.e. the harmonics of the tone are not all in tune) due to different frequency components of the wave travelling at different speeds. This inharmonicity is called "dispersion". According to another page on the same site...
Let $x[n] = \cos(wn)\ \ $ be our signal. Let the other signal be $x_2[n] = \cos(2wn)\ \ $ . The spectrum of $x[n]$ and $x_2[n]$ are well defined and consist of impulses at the relevant frequencies with weight $\pi$. However, if we interpret $x_2[n]$ as the decimated version of $x[n]$, the spectrum of $x_2[n]$ must have...
Let's start with a simple temporal law, that I want to analyze in the frequency domain : $x(t)=\frac{1}{2}(A+Bcos(2\pi \nu_0 t+\phi_0))$, with : $A=4$ $B=10$ $\nu_0=10^8$ $\phi_0=10^8$ I made a simple DFT (in Ruby) with $N=256$ samples at $f_s=6.10^8$ and got the following results, that are self explanatory, hopefu...
This question is a little theoretical. Say I have 3 points that are bound within a certain 3D volume of some shape (i.e. box, cylinder, cone, etc). By using PCA, I find the most two dominant directions and reduce the data dimension to 2D. All of this is done by the following: Compute 3x3 covariance matrix of the 3D d...
I was looking over some DSP and came across the following signal: $x[n] = 1/n$. So I wondered whether it had a z-transform but I soon realized that it fails to meet two conditions described here: ...(1) the signal starts at a finite time and (2) it is asymptotically exponentially bounded... So I modified it to meet...
I have been struggling to find a well-cited approach to this. Essentially, I am experimenting with different background subtraction algorithms, but none seem to perform well when there are significant lighting changes (say, from a window in the scene getting significant glare at specific times). The question is simple...
I have a periodical function $f(x)$ that is sampled in time, at a rate of 1 millisecond. Together with this, I also have a second function $t(x)$. I want to find the correlations between periods of the first function and the second function $t(x)$. How can I approach this issue?
I am trying to make sense of FFTs and binary data. Say I have a series of random binary data, which is measured with a repetition rate of 400Hz (interval time of 0.0025s). I have a total of 12489 points, which corresponds to a total measurement time of about 31 seconds. I would like to be able to learn more about what ...
I have two audio signals, both with supposedly matching frequencies at each time window. What I would like to do is to verify that the frequencies are identical. However, one was created at a higher power than the other, so simply checking for a match at each sample won't work. I've already cross-correlated the sign...
Is there any good documentation about training your own classifiers(with examples included) different from this one provided by MathWorks. Thanks in Advance
For the differential equation $$\frac{d^2y(t)}{dt^2}+\frac{3dy(t)}{dt} + 2y = x(t),$$ I was able to find the frequency response as $$H(j\omega)= \frac{1}{-\omega^2+3j\omega+2}.$$ However, I am not sure how to find the impulse response, which is the next part. Please help, and thank you in advance.
I am working on camera motion detection.How do I decouple translation and rotation? I have obtained rotation by using sensors. I understand we can do something as simple as X'=RX+T , where R-rotation, T-translation and X,X' are the points involved. These points are 3D points(either world or camera coordinate system). H...
I've got an array of different values representing the y-values in a plot: y-values = [0.07, 0.09, 0.04, 0.15, 0.14, 0.05, 0.02, 0.2, 0.18,... ...0.06, 0.2, 0.17, 0.19, 0.21, 0.3] I want to split it in 3 intervals: the first one should return the values to the first big amplitude (here: [0.07, 0.09, 0.04, 0.15]) the...
I'm trying to do some simple filtering for an audio signal using a window-sinc low pass filter. Supposing that my buffer has M values and the filter kernel size is N, after convolving these two arrays I would obtain an output of m+n-1 values. Which part should I take into consideration? I'm asking this because, if I re...
For audio classification we are using the cepstral coefficients and their derivatives. Which I calculate using the Matlab libary voicebox. I am looking for some literature for how to choose the frame length and overlap of frames. Most literature that I know of are considering speech recognition, but for this applicatio...
I am fairly new to both Matlab and FFT. I would like someone to explain to me the relationship between the frequency of the wave being processed, the number of point (usually called NFFT), the Nyquist Fs and frequency vector. Basically how do you find the peak frequencies? how does FFT represent the frequencies? Any an...
I am trying to implement Wavelet transform in an embedded environment. The problem I face goes like this: Two vectors, A and B. A is input samples. B is vector co-efficients. I take convolution of A and B, call it C. Now I go back to B and up-sample the co-efficients. Multiple C with new up sampled signal of B. Prob...
I'm trying to implement some watermarking algorithm found in a paper*. This is a line of the paper: Do the H-level DWT for all the renumbered segments. Then in simulation section the author explains the wavelet used for experiments. DWT transform adopted the common wavelet "Daubechies-1" and level H = 3. I just d...
I've a scatter plot with x- and y-axis. Now I want to calculate a rectangle which displays where 80% of the points in this plot are. I'm not very familiar with statistics, so I don't know the method I have to use. Read something about percentiles. Is that correct? I know I have to calculate where 80% are on the x-axis ...
I have a source that emits $M$ equiprobable messages, which are assigned signals $s_1, \dots,s_M,$ that are equidistant by $a$. That is, if we plot the $s_k$ signals in a horizontal axis they are dots separate by the same distance $a$. And so I would like to determine the optimum receiver and the corresponding error pr...
The question is revised to make it more specific and constructive. I am doing imtransform, yet the xdata and ydata are very much different from that of size(output). [ output, xdata, ydata ] = imtransform( imgDouble{1}, tform ); Real data test: K>> size(output) ans = 854 1230 K>> xdata(2) - xdata(1...
I'm looking for a kind of error correction code or solution that can correct my codeword in this case: My message holds $k$ bits, and $2k$ bits codeword (rate is $1/2$) is produced by the generator matrix, e.g. $k=5$, so it yields $10$ bits codeword $\rm 1001100111$. If the codeword is erased for some reason and it bec...
I am using IIR filter for smoothing $$y[n] = ax[n]+(1-a)y[n-1]$$ My question is, if I add another IIR filter, will it be the second order of IIR filter? If not, what it can be called? My second filter is $$y_2[n] = ay[n] + (1-a)y_2[n-1] $$
I have computed the MFCCs for two audio files and applied the DTW to both sets of coeficients for both files. Now each such pair the two files is described by one single value (computed by the DTW). Let's call this value Diff. If the audio files differ more, then Diff is greater, which is great. (so Diff is proportiona...
I have created a Low Pass filter (I am trying to understand the topic of DSP, and it's hard.. But moving along with samples, books and a nice post on Stack Overflow). I am saving the 'output' to a file, and I hear a constant clicking noise (very rapid, medium-pitch). The volume (intensity) of the 'clicking' gets louder...
The choice of frame length for calculating the Mel Frequency Cepstral Coefficients of a signal, depend on the how much the signal change over a small time scale. I guess that human speech change rapidity while some animal sound change more slowly, but what is the measure of small time scale of an audio signal?
I would like to create white noise in the frequency domain, and then transform it to the time domain using python. To understand the problem, I simply generated white noise in the time domain, and transformed it to freq domain: import scipy.signal as sg import numpy as np import matplotlib.pyplot as plt e = np.random...
Consider filtering square nn images by square,separable mm filters. What are the general computation time for the following approaches 1) FFT 2) by 2-D convolution 3) by two 1-D convolutions. (Capital letters to indicate implementation-dependent constants) p.s what does it mean for a 2-D filter to be separable? ...
I have four .wav files and each contain bird chirps. Given an input .wav file, I need to find out which bird it is. I have some basic understanding of signal processing. I have used spectrogram to analyze my audio files and now I don't know how to make the frequency spectrum comparison to find out the solution. Can so...
What algorithms or methods can be used for the tracking of audio pitch contours? (Such as in musical vibrato and glissando, or for tonal vowels in languages such as Chinese?). Will a PLL work even for pitched tones that have a weak or missing fundamental? Or is there a more suitable tracking method?
I am training a classifier of my own, specifically for detecting body arms. According to the training process I have to provide positive samples and negative samples, wich will be used in the different stages of the cascade classifier. As Mathworks says here, there is a tradeoff between having a higher false positive r...
I am trying to get the fundamental frequency of a signal that only has a single pitch. I coded out the autocorrelation function using FFT and already got the autocorrelation result. Unfortunately, I don't know how to get the fundamental frequency from the autocorrelation result. Can someone help me? My JAVA code for ge...
I have been trying to implement image classification by extracting facial features. I tried starting with some already established methods like Multi Kernel Approach, using Independent Component Analysis etc. Is there a very basic algorithm for image classification with which I can proceed the implementation ? The dat...
I need to divide an audio signal into frames in which the signal can be assumed to be stationary. How can I measure whether an audio signal is stationary in a frame?
I already have an algorithm for making panoramic image mosaics by image registration and then blending images in the overlap regions. The following image from Wikipedia illustrates several images under translational movement stitched together by finding optimal dividing lines in the overlap regions: Now I would like t...
Is there anyone with Image Processing background who can suggests which one is better for Image Processing algorithms among CUDA and FPGA. I'm specifically looking to write code for Wavelet Transform and Discrete Cosine Transform and eventually writing code for Quantization and Arithmatic coding, but i am confused whic...
What's an extremely efficient way of detecting when a user has stopped talking into a microphone? When I use systems like Siri it can detect when I've stopped talking almost immediately, even when there's background noise. My initial guess was to get the average volume of the background noise first but it seems there i...
I'm trying to segment a specific part in a gray-scale image and I want to achieve this by applying the OSTU algorithm, but after using this function: im = imread('ready.jpg'); oim =otsu(im,10); figure,imshow(oim, [1 10]); I could show the output image, but I do not know how can I save the result to be able to use it l...
Take: $$ (uv)(k) = \sum_{i=-\infty}^\infty u(i)v(k-i). $$ The $k$ is there, it's because you want to define $$ \ldots\ldots, (uv)(-3), (uv)(-2), (uv)(-1), (uv)(0), (uv)(1), (uv)(2), (uv)(3), \ldots\ldots >$$ etc. The number in the parentheses is $k$. Thus, for example, when $k=4$, we have \begin{align}...
I hope this ultra-noob question is not entirely out of place here. I'm trying to make a super simple ringtone: 0.5s silence + 1s "pure tone" + 0.5s silence + 1s "pure tone" + 0.5s silence. By "pure tone" I mean something like a sinusoidal or square wave with a fixed frequency (let's say 350Hz for concreteness). This ...
I'm having trouble implementing OFDM with a raised cosine (RC) window in Matlab. I know how to generate an OFDM signal and how to show its spectrum, I just don't know how to generate the window extensions. Each OFDM symbol is extended by $TW$ samples at both ends to smooth the transitions between successive symbols, th...
What are the options for recovering the phase of a BPSK signal in software? The only resources I can find online contain circuit diagrams – there seems to be no one willing to explain this to people unfamiliar with analog circuits. I'd like any explanation of how carrier recovery works in theory, and would also love ps...
I am trying to understand the periodicity of the DFT. How can this property (both in time and frequency domain) be used and can be helpful while developing on a DSP? It would be good to see some source code or pseudo code, in MATLAB preferably, where this property is exploited/demonstrated.
Consider filtering square $n\times n$ images by square, separable $m\times m$ filters. Give general equations for the computation time for the following approaches to this: Filtering using FFT 2-D convolution Two 1-D convolutions Use capital letters to indicate implementation-dependent constants. For particular ...
I'm looking for two signals embedded in very noisy data. I know they are close to periodic and integral with the run, but are rather close (30s and 21s during a 210s measurement), and our sampling rate is low (1.25s). We have multiple observations to work with per sample. I am wondering how I might deal with the proble...
I am trying to understand the logic behind unsharp filter in matlab. If I type: image = imread('nameoffile'); filter = fspecial('unsharp', 0.5); resultImage = imfilter(image, filter); resultImage is sharper than image. I opened fspecial.m file and saw that unsharp filter is created as: h = [0 0 0;0 1 0;0 0 0] - fspec...
I sent a square wave signal (red) through a IIR filter (butterworth in this case), to make it more "realistic". Then there's a slight group delay in my output signal (blue). Is there a way to find out what the group delay is? I only need the total group delay, not spectrum of group delay. I need the number so I can shi...
I'm trying to detect the presence of a sinusoid in 1/f noise conditions using the STFT and the DWT with Haar wavelets. I find this interesting phenomenon that I'm not able to explain (see plot at 1) My sampling rate is 1200 Hz and the graphs show the one sided response (0-600 Hz). I am using 2 scales for the Wavelet Tr...
I am trying to understand where does the convolution fit in Discrete Fourier Transform. I know that convolution is producing a third signal from two other signals. I also know that DFT transforms one signal from time to frequency domain. So, what is the relationship between the two (if any)?
i got that system $x_n \to x_n-x_{n-1}$, so $h_n=[.....,0,1,-1,0,...]$, with $h_0=1$ and $h_1=-1$, so the transfer function given by: $$\sum_{i=-\infty}^{\infty} h_ne^{-jwn} = h_0e^{-jw(0)}+h_1e^{-jw(1)}=1-e^{-jw}$$. How to knwo the type of filter, if it is HPF, LPF, etc. I appreciate your help.
I know that while taking pictures from a camera, noise may appear in the image. It is often described as Gaussian, but what about the other noises, when do they occur in real world? Please help!
I have an image that I am trying to process: I want two seamingly contradictory effects: I want to remove "holes" inside "the blob". That is I want all pixels inside the blob to be solid red. I want to move the boundary of the blob inwards. From what I gather the morphological operation dilatation would do 1. and er...
i got a system with transfer function given by: $$H(\omega)=1-e^{-j\omega}$$ I already plot it, and that seems to be a periodic function with $H(0)=0$, $H(\pi)=2$, , is that enough to show that this is a FIR bandpass filter. thank you very much
What is the role of complex exponential $ e^{jθ} $ in Fourier Transform? Is it different in the continuous and in discrete time domain?
Is it possible to generate the z-domain transfer function from a pole-zero plot diagram?
If I have, for example a 16-QAM signal, and the constellation points are (3,3),(3,1),(1,3),(1,1),...(-3,-3). Now I have a simulated in coming signal with arbitrary power, how can I normalize the signal to this constellation so I can make correct decisions? The first thing I could think of is to calculate the average po...
Using the Fourier transformation I can translate a signal from the time-domain to the frequency-domain. If I divide the frequency axis into intervals, how can I calculate the audios signals energy for each of these intervals?
I have a sensor that reports its readings with a time stamp and a value. However, it does not generate readings at a fixed rate. I find the variable rate data difficult to deal with. Most filters expect a fixed sample rate. Drawing graphs is easier with a fixed sample rate as well. Is there an algorithm to resample fro...
I want to learn how to design structures that could model difference equations. I don't know how we call this things but i can show a picture: For example i want to understand the procedure that using addition, multiplication and delay operators like the above i can describe equations like: $ y(k)=1/2(x(k)+x(k-1)) $ D...
While reading a description on FFT from this book, I found the following statement: ...diluting the time domain with zeros corresponds to a duplication of the frequency spectrum. How can I prove this intuitively or otherwise? I have read some properties of Fourier Transform, such as a circular shift in time domain...
I'm working on an OFDM-QPSK based system and I'm having some trouble to match the BER of my system with the theoretical curve. My OFDM implementation fills all the subcarriers and have t_symbol s duration and t_cp cyclic prefix duration. I'm trying this to validate my system implementation before go on with my system d...
Is there a verb for removing the mean or DC component of a signal? I am trying to name a flag for doing just that. "Normalize" can mean many things, so I don't think that would be a good choice.
I found an approximation of a 5x5 2D convolution kernel like this : Here, the sum of the elements is zero and this one was used for Laplacian of Gaussian! Another one here : This one has all positive and the sum is not zero and is being used for smoothing. Could anyone please tell me what is with the sum being zero i...
I am given 256 samples at a sampling rate of 48k and am asked to find the frequency of dominate tones and the amplitude of the largest tone. The data looks like this when plotted: I assume I must use a FFT transform to do this. I've tried this in matlab: Fs = 48000; x = dataSet; % 256 samples data set xdft = fft(x); ...
I am trying to write an image processing program to recognize bubbles in oil. It has been suggested I try computing the convolution of the image and an image of a typical bubble. i.e. : ifft(fft(a).fft(b)) will have high peaks near bubbles. where: ifft : inverse fourier transform fft : fourier transform a : the imag...
I read many sources about kalman filter, yet no about the other approach to filtering, where canonical parametrization instead of moments parametrization is used. So I would like to learn on examples how to use this filter with information matrix and information vector (not mean and covariance from KF). Can anyone he...
In an old paper "A New Approach to Geometry of Range Difference Location", there is a mathematical derivation that does not include any mention of the relationship to the underlying problem. In the problem, there are 3 receivers (at $(x_1, y_1)$, $(x_2, y_2)$, and $(x_3, y_3)$) and a transmitter at unknown position $(x...
I am learning DFT and I am not sure I understand it correctly. I am just going to write out the process of DFT and I am asking you to tell me if I am thinking the right way. Suppose we have a signal on which we run DFT to get the frequencies incorporated in the signal. We use a window of size N samples and calculate D...
I showed the question about flipping the impulse response to some math teachers and electrical engineers at an university. They read Dilip's answer and thought that it was hard to understand what Dilip meant and it was simply not true... Could someone confirm that it really isn't true what Dilip Sarwate wrote. And expl...
From what I have read about signal (de)modulation I understand that multiplying two sine waves, both with the same frequency, gives a two frequency component sine wave. One at DC and one at $2\omega$. Using a low pass filter will let us remove the $2\omega$ component. Now DC amplitude is proportional to the phase diffe...
We have several audio tracks, all of which are different versions of the same track under one type of distortion: the speed of the track is increased by a constant factor K (hence frequences are multiplied by K, timesteps are divided by K), K is relatively small (0.8< K < 1.2). I want to know the best way to get a good...
I am implementing a method known as trace transform for image analysis. The algorithm extracts a lot of features of an image (in my case features pertained to texture) using a set of transforms on the values of the pixels. Here is a paper describing the algorithm: (PDF link). The algorithm finally constructs a "triple ...
I'm trying to do a CT Fourier Transform of these two signals $$x_1(t)=e^{−a(t−1)} \cdot u(t−1)$$ and $$x_2(t)=e^{−a(t−1)} \cdot u(t)$$ Where $a$ is any real number, and $u(t)$ is the unit step function and that is multiplication, not convolution. Picture of my work and answers here. The answers listed should not conta...
I'm analysing several measurements taken with a SR830 lockin amplifier. These measurements look similar to this one. Since I'm interrested in the derivative of the signal I took the numerical difference. What I found was this. One can clearly see two bands in the signal. I'm no expert, but I'm guessing it might be rel...
I have a two data set files data1 and data2 each containing 5000 points having 3 columns representing x,y(coordinate (centroid)),angular movement(theta) resepectively for two robot movements obtained by numerical programming and solution of a circuit representing robot equations.I need to simulate a graph for coverage ...
Maybe a stupid question, but if there is a time-domain representation of audio signal, and also frequency-domain, so is there any other domain that signal can be represented in?
I want to fuse objects coming from several sensors, with different (sometimes overlapping!) fields of view. Having object lists, how can I determine whether some objects observed by different sensors are in fact the same object? Only then I can truly write an algorithm to predict future state of such an object. In othe...
I need to filter the image raster below with the following mask: 0 1 -1 0 and the Image Raster: 0 0 0 0 0 1 0 0 0 0 1 1 0 0 0 1 1 1 0 0 1 1 1 1 1 1 1 1 1 1 My results are: 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 -1 0 0 0 0 In what direction does this mask detect intensity change? What does the sign of the output signal me...
I have a bunch of simulated time series, for which I can compute the power spectrum. Generally, the simulated power spectrum can be sketched as follows: I now aim to calculate the features of the theoretical power spectrum. The theoretical power spectrum can be represented by: $$ S_{th}=\frac{4\left(L/U_{hub}\right)}{...
I want to use GMM for image classification. So, I have extracted SIFT features from each image in the corpus. Then, I apply EM algorithm to learn GMM parameters (I have coded it in matlab). I get very low Recognition rates for unseen images when I test the model on larger number of classes (>=4). Who can explain this ...
I have got a question concerning the definition of the frequency vector for an fft operation. Generally, I work with a frequency vector, f, with power of 2 elements (2048, 4096, 8192, ...). Given a certain simulation analysis time, time (e.g. 600s), I should define f as follows: % Frequency defition t = 0:dt:(time-dt);...
This is MR Modality Link MR Modality In this file where i find the LV Cavity in above link file. Also how to draw Tumor Marker from over it
The matlab "ellip" function can be used to design the unquantised coefficient set. From matlab website: $ [b,a]=ellip(n,Rp,Rs,Wp) , $ n: order of filter Wp: normalized passband edge frequency Rp: ripple in passband (db) Rs: db down from he peak value in the passband If i have an 8th order IIR filter with the foll...
Plotting the FFT of a signal with Matlab e.g. by the code: y_fft = abs(fft(y)); f = fs*(0:length(y)-1)/length(y); plot(f,y_fft); where y is the discrete signal and 'fs' is sample rate. creates a plot where the curve can be mirrored in the middle of the x-axis e.g. Why can the FFT always be mirrored in the middle of t...
I'm reading an IEEE standard and it says that the first channel needs to be 50dB below the highest power in a 6 MHz operating channel in 100 KHz bandwidth. It was mentioned several times that the power spectrum measurement needs to be done over a 100 KHz bandwidth. Isn't it sufficient to just say that the power of the ...
Suppose we have create an IIR filter with matlab function "ellip", and then we want to quantize the coefficients using: \begin{align} bq=Quantize('round',b,2^8); \cr aq=Quantize('round',a,2^8); \end{align} I have read that there are 4 major types of rounding: truncate round convergent rounding round-to-zero What i...
I am doing R&D over the MR Segmentation to draw marker over the Heart MR Dicom Images. My Question is that how I can draw a such marker over the dicom image that show in attach/link image. that Region of Interesr ROI read from the tag or its part of Pixel Processing.
1 2 1 1 2 1 2 4 2 (A) (1/16) * 2 4 2 (B) 1 2 1 1 2 1 Both matrices are the same except that the matrix B is multiplied by the sum of its elements. When convolved with an image, they give nearly the same blurred image. What is the exact dif...
I would like to ask what could be the easiest way to detect the presence and beginning (no more, no less) of under-damped harmonic signal in a collect of ultrasound data. I have some acquired data which is presented in 2-D image form, where vertical axis are related with time. Under-damped harmonic signal, which I need...
I have a GMSK modulator in MATLAB and wish to generate AWGN with a specific SNR, and I'm having some issues figuring out just how. I have found this step-by-step process to generate zero-mean Gaussian noise with a specified power spectral density W0: Form Gaussian distributed random variable: w = randn(1,N) Map top ze...
How can I obtain the wavelet filter coefficients ( i.e. lowpass decomposition filter, highpass decomposition filter, lowpass reconstruction filter, highpass reconstruction filter) for the Morlet wavelet? For convenience: the wavelet curve follows the equation $$\pi^{1/4} \exp(jwx) \exp(-0.5x...
I have values (plotted below) of expected RSSI values over time that I would like to compare with my measured RSSI values. What I was looking for was a way to quantify it so I can change parameters and be able to compare/contrast different approaches. It is a hard problem in my mind because I don't know how to compare...
I have some programming experiences but am really new to these kind of algorithms. I can find a few source code about implementing HHT, and most of them are written with MATLAB. What I am trying to do: Apply HHT algorithm in detection of image splicing. I want to find object edges in images. Implement this with C/C++...
I'm currently reading a paper and I can't seem to make sense of a certain part. A link to the paper: removed The part I don't quite get is on page 286 (after the part where they explain their filters). It starts with the sentence "Two common methods can be used ...". They explain how they calculate the interbeat interv...
I am a little confused as to how to compute the nyquist criterion for a sensor array that is not linear, and/or non-uniform. For the linear, uniformly spaced case, the spatial nyquist is delimited by finding the half-wavelength that fits between the space of two adjacent sensors. Thus, if the distance between two senso...

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