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In many of my readings, whenever some author speaks about working in the frequency (transform) domain (of a digital signal), they often times take the DFT, or the DTFT, (and of course their corresponding inverses). Different authors will tend to work with one or the other. I have not been able to really ascertain a pa...
I have been studying about exponential average. There are enough explanations about this at Internet, but they do not explain about the time constant. I have one channel with a $T$ seconds time signal with sampling frequency $f_s$. If I want to do average of this time signal, we need to use either linear or exponentia...
I have been doing some research into modelling acoustic spaces, and (with help from this site) I've managed to take the frequency response of fibrous layer model, convert it to a reflection response as a target for an IIR, and then find the polynomial coefficients for it using FDLS. All good so far. I performed the us...
I need to get the lines from a photographie of a sheet of paper. While recherching, I came along this, where one answer referred to this blog entry. Now I am having trouble with this approach. I have implemented it as follows (with the c++-interface of OpenCV: // the image has to be grayscale if (image.channels() != 1)...
I'm trying to compute the the interference that one OFDM user with beamforming causes to another user. The two systems are fully overlapped in frequency and time. I know how to generate an OFDM signal in MATLAB and compute its BER in AWGN and fading channels, I just don't know how to implement beamforming. The beamform...
I would like to calculate the total SINR of a IQ modulation system in time domain. $$ SINR = \frac{P}{N+I}$$ The blocks of the system are. Bit generation QAM modulation pulse shaper (sq root raised cosine f) IQ mod IQ dem filter (root raised) QAM demodulation I made a simulation of a IQ modulation system in time ...
I would like to implement a HOG descriptor in C++. I found an implementation of this code in Open Pattern Recognition Project - Histograms of Oriented Gradients (HOG) Feature Extraction. An extract of the code follows: //extract HOG features from a group of key points void HOGExtractor::extract(ublas::matrix<float> &ds...
After investigating signal analysis methods based on empirical mode decomposition (EMD), I found that recent developments are mostly related to the Hilbert Huang transform (HHT) and the Local Mean Decomposition (LMD) method. I have been reading a few articles on the subject and I would like to have your opinion regard...
I have to apply some kind of adaptive filter to my function $f(x).$ I present each point of my signal as a Gaussian, whose bandwidth depends on its location (not the point of observation $\textbf{x}$) as $h(t),$ which is a known pre-calculated function. The final output function $s(x)$ is a superposition of the influen...
I am detecting the object from the live camera through feature detection with svm , and it read every frame from camera while predicting which affect its speed , i just want that it should select the frame which contain the object and ignore other frames which have no object like empty street or standing car's , it sho...
Is a dirac comb a Linear and a shift Invariant System? I could prove the linearity, but I do not think it is shift invariant. How do do I prove it is shift variant?
Is it possible to make one IIR filter from various cascading IIR filters of different orders? I know it is possible to do it from filters of the same order, but what about filters with different orders?
Please excuse me if this is not the proper place to ask this. I just need some general advice. I have set up a very basic homemade motion capture system where I put three TrackIR cameras from NaturalPoint in a three fixed positions. I want to capture movement of a single object in three axis. So I laid out two of the ...
Excuse my ignorance... Given a digital image of a label I'd like to Requirement: Match it to the closest pre-existing image in a datastore Optional: OCR it, if possible (only necessary if step 1 fails) I'm a programmer but not an engineer. I have no idea where to begin or where to look into solving this problem. I do...
I know about Harris corner detector, and I understand the basic idea of its second moment matrix, $$M = \left[ \begin{array}{cc} I_x^2 & I_xI_y \\ I_xI_y & I_y^2 \end{array} \right]$$, edges and other unstable points can be removed via $M$. But about Hessian detector, it uses Hessian matrix to detect key points and re...
I recently received some scanned pictures which are in a pretty good state, with OK resolution, but they are randomly spattered with a bunch of black dots. These cover everything (clothes, faces, walls, windows, and so on) except for a few very bright areas. I also have no idea how they got there. Here is a sample: Fr...
I am running a wavelet transform (cmor) to estimate damping and frequencies that exists in a signal.cmor has 2 parameters that I can change them to get more accurate results. center frequency(Fc) and bandwidth frequency(Fb). If I construct a signal with few freqs and damping then I can measure the error of my estimatio...
I am starting with a typical $\ell_1$ basis pursuit problem: $$ \min_{\mathbf{x}} \Vert \mathbf{x} \Vert_1 \quad \mathrm{s.t.} \quad \Vert \mathbf{ERx} - \mathbf{y} \Vert_2 \leq \epsilon, $$ where $\mathbf{R}\in\mathbb{C}^{M \times P}$, and $\mathbf{E}$ is an $M' \times M$ row selection matrix that randomly chooses $M'...
I have a time series that has been measured after convolution with a moving average filter. Knowing the parameters of the moving average filter, is it possible to reconstruct/constrain the values of the original? When considering deconvolution, there is much discussion in the literature of the form and magnitude of mea...
what is 'parm' means when you set the name of wavelet function in cwtft or icwtft. wave = {wname,[7.6]}. also can I change Fb and Fc when I use 'morl' function in cwtft or icwtft transform? and If not, then how can I reconstruct my signal with cwt transform? cause cwt let me to select optional value for fb and fc (cmor...
I'm trying to get an understanding of the relationship between an FIR filter designed from "first principles" using a filter kernel with convolution, and a filter designed in one of two ways using FFT (see below). As far as I understand, the impulse response of an FIR filter is the same thing as the filter's convolutio...
A classical technique in still images (e.g. fluorescence microscopy images) to remove uneven illumination and isolate bright blobs is to use morphological operation such as the top-hat transform. For instance the Rolling-ball algorithm[1] uses a ball as a structuring element and performs the top-hat transform[2]. The s...
Ok, this is leaving me increasingly frustrated. This builds on a previous question here but I am still not getting the indexing, although I learned from that question all the same. Case 1: First convolution equation example: $$ h[k] * h^*[-k] = \sum_{n=-\infty}^{\infty} h[n] \ h^*[n+k] $$ The way I got the right hand s...
For an image $I$, its first order derivatives can be computed using several oprators, such as $$K_{sobel} = \left[ \begin{array}{ccc} -1 &0 &1 \\ -2 &0 &2 \\ -1 &0 &1 \end{array}\right]$$ $$I_{x} = I * K_{sobel},\ I_{y} = I * K_{sobel}.T$$. And to compute the second order derivatives $(I_{xx}, I_{yy}, I_{xy})$, I can ...
Having found some unofficial sources on Complementary Filter (Thousand Thoughts Sensor Fusion and The Balance Filter by Shane Colton), I wish to work out its rigorous mathematical proof. The Complementary Filter, $$y=\alpha \times y+(1-\alpha) \times x$$ where $\alpha$ is the filter parameter, usually chosen to be ~0.9...
I know that a function $f_a\in \text{L}^2(\mathrm{R})$ is an analytic signal if its Fourier transform is 0 for negative frequencies, i.e. $$ \hat{f_a}(\omega)=0$ if $\omega<0 $$ We can characterize an analytic function by its real part $f = \text{Re}[f_a]$, and the Fourier transform of this function is $$ \hat{f_a}(\om...
By using Python (numpy/sklearn/opencv) I'm trying to process images with hand-written digits, and after all my main problem turns to be connected digits. For example: If I crop the digits by hand I get expected output, by taking advantage from NIST dataset, and sklearn SVC classifier, but segmenting the digits seems l...
I have 6.4 sec recording of a discrete signal on 40Hz frequency (256 values). How can I compute the variance of the signal? Please, share an algorithm in pseudo-code or in an actual programming language.
Given that $|a|<1$, then which of those inverse-Z-transform equations are we to use? I am leaning towards the first because (as I understand it), $z$ is merely a complex number that is evaluated on the unit circle, and thus $|z|$ must always equal 1? I guess my question is, when is $|z|=1$? Always?
I have one and only one frequency signal, a sine signal at f = 10 kHz that is sampled at Fs = 320 kHz. I have read about applying a simple analog anti aliasing (AAA) filter then oversample, do some digital anti aliasing (DAA) and decimate to get the correct signal frequency. I understand how this is useful if signals o...
Sorry if this sounds elementary but I am struggling to grasp the physical idea behind ARMA (auto-regressive, moving average) process. The "AR" part is intuitive and so is "MA", but put together? If I model my time series using "AR", I can predict the next sample using linear terms of the previous samples and the differ...
I'm not a signal processing person at all so hopefully I'm not asking an obvious question (if I am, I'd appreciate any resources that would help give more context). I have a 1D vector where the values give temperature at some position. I want to find out where the regions of greatest change are and then mark off those ...
I'm trying to wrap my mind about what happens in Matlab/Octave when you compute the fft of a sine function. Theoretically, the Fourier Transform of a sine is given by two delta functions multiplied by the imaginary unit (one placed at the frequency of the sine and another at minus the frequency). I can see that in the ...
I'm looking for advice or references on how to implement a "type I" compensation network of digitally. An analog example is shown in the following diagram: I am familiar with the basics of digital signal processing, and have read about digital integrators. Would it be possible to use something like a first order IIR...
I have a vector of signal x(t) with its time vector. I want to obtain a frequency representation of the signal, in particular the energy spectrum of x(t). Can someone please show some light into obtaining the energy spectrum ? Is energy spectrum and energy spectral density ,ESD produce the same result ? Any help is m...
Laplacian of Gaussian formula for 2d case is $$\operatorname{LoG}(x,y) = \frac{1}{\pi\sigma^4}\left(\frac{x^2+y^2}{2\sigma^2} - 1\right)e^{-\frac{x^2+y^2}{2\sigma^2}},$$ in scale-space related processing of digital images, to make the Laplacian of Gaussian operator invariant to scales, it is always said to normalize $L...
I am studying my Oppenheim and Shaffer book, (New Edition), and the last chapter deals with something called homomorphic filtering. I have read the wiki and some other websites about it, but they do not really delve into details on applications. Moreover, I have almost never heard of it be brought up in discussions or ...
The impulse is given as a sequence: $x: {10; -3; -1}$. To determine if it is minimum-phase I compute the $Z$-transform (the geophysical case, positive powers of $z$) of it: $X(z)=10-3z-z^2$. Then I factor $X(z)$ into dipoles $(a_0+a_1 z)$: $X(z)=10-3z-z^2=(5+z)(2-z)$. Then I observe that all the dipoles satisfy the $|a...
For the 2D discrete Gabor transform, why is it that we cannot use a set of orthonormal basis for its representation, instead we have to use frames for representing it?
I am trying to filter a sinusoid ($f_{sample}=50 Hz$) ,, i need to remove all frequencies above 10 Hz with accepted transition region 2 Hz (stop band freq=12 Hz) the question is why after filtering this signal is the 20 Hz still appearing in the FFT? is there something wrong ? I know results should be the same in fre...
For a signal sampled at $f_s$, the frequency resolution (or "bin width") for an $N$ point FFT is $f_s/N$. Does this mean that the $k$th bin will contain energy from sinusoids within $0.5f_s/N$ on either side of the center frequency of that bin (ignoring spectral leakage for now)? Is this energy averaged? Is it the same...
I've been using Thinning recently on line-like structures in an attempt to generate skeleton-like chains of pixels connected through their Moore neighborhoods. The description of Thinning says that the algorithm "finds the skeletons of foreground regions in image by applying morphological thinning until convergence"....
How could I know the best size of the board for my camera? My camera is Canon D-100 It's 18 megapix I have created a board consist of 14*10 squares the length of each square is 7*7cm I need to know what the step of choosing the number of squares ? and How could I know the best dimension for each square ? Thank you ver...
I'm sorry if my question might be simple, but I don't have a strong background on signal processing. I have a csv file of a 8 MHz bandwidth, sampled with a resolution bandwidth of 100 KHz. What I want to do is to compute the PSD over the 8 MHz bandwidth, but I'm not sure if I am doing this correctly. For a "stable" sig...
I'm trying find authoritative references for [1] Which analyzing techniques (time/frequency domain) i should use for determining the quality of a speech signal. [2] Which algorithms i should run for improving speech signals( As "cheaply" as possible) There are no constraints for the first requirement as it will be don...
I need to take the wavelet transform of an image (for reference, I will be using Debauchies wavelets, and the test image will either be Lenna or matlabs default woman image),and plot the coefficients in order of their magnitude. This is as an example for a document on compressive sensing I'm writing. I'm a little unsur...
I'm confused about the difference between a wavelet transform and a wavelet decomposition is. For example load woman [cA1,cH1,cV1,cD1] = dwt2(X,'db1'); [c,s] = wavedec2(X,2,'db1'); What's the difference between these two matlab commands, and when would you want to do one over the other?
Actually i am in search of a Voice Activity Detection Algorithm which could distinguish between voice and non-voice Roughly speaking it must not detect even a bullet sound,even a foot stepping and other non speech activity should only detect people conversation or any one shouting in that search this question arises ...
I am working on speech segeration and each source I am reading , spectograms are being made by filterbanks which uses gammatone filters . I can't find a proper c++ source code which would make me understand process behind gammafilters, completely. Now I am curious what is its difference than 2d FFT, I mean why all sou...
I'm looking for a DSP, around 500MHz, with an in-chip ADC (preferably 2 but 1 will suffice) that samples around 20MSPS. Does any such DSP exist or do I have to use external ADCs?
I am trying to further reduce the following expression (if there's a nicer way to write it): $\sum_{k=-\infty}^{+\infty}\sum_{p=-\infty}^{+\infty} \sum_{l=-\infty}^{+\infty} w(k) \varphi_\mathrm{y}(l-p)\varphi_\mathrm{x}(k-n-l+p)$ where $\varphi_\mathrm{y}$ and $\varphi_\mathrm{x}$ are the autocorrelations of the signa...
If my signal is $$ f(t)=\exp[i\phi(t)], $$ how do I show that $$ \int_{-\infty}^{\infty} {\xi}\left|S_f(u,\xi)\right|^2 d\xi = 2\pi\int_{-\infty}^{\infty} {\phi'(t)}\left|g(t-u)\right|^2 dt, $$ where $$ S_f(u,\xi)=\int_{-\infty}^{\infty}f(t)g(t-u)\exp{(-i\xi t)} dt $$ is the short-time Fourier transform of $f(t)$?
Say I have a function, $C=C\left(x\right)$, whose fourier transform is denoted by $c=c\left(k\right)$, i.e. $C\left(x\right)=\sum_{k=-\infty}^{\infty}c\left(k\right)\chi\left(x\right)$, where $\chi\left(x\right)$ is some complex valued basis function. I know that the autocorrelation is given by $M\left(\Delta x\right)=...
I'm looking for a way to implement a Speech Separation from a .wav file. Let's say I have a .wav signal containing the following words: "The cat flew away" Now I don't really want to do the speech separation after I've split the signal inside blocks of size NxM because I feel as though I could miss parts of the word. ...
I am studying windowed Fourier ridges for my project. I know that it computes the instantaneous frequencies and amplitudes from the spectrogram. But I don't really know how it relates to signal processing, can anyone explain this concept to me?
Given a 2D image where each row of pixels is shifted by some amount horizontally in a sinusoid fashion, is it possible to recover the original image? Imagine a camera that can capture one row at a time, and is moved vertically to capture the whole image. While it is moved, the camera vibrates, resulting in noise simil...
I am trying to implement a Savitzky-Golay filter by following this Wikipedia page and in the first instance I have written the following Octave software code to create the convolution coefficients clear all m = input( 'Enter number of bars (must be an odd number): ' ) ; % check no_bars is odd if mod( m , 2 ) == 0 % i...
What is the reason that a periodogram is not a consistent method for obtaining information about frequency response of a system? To be more precise, why even after increasing number of samples doesn't the variance of the estimator decrease?
I have some data that was collected at irregular intervals, and I want to estimate the values on a regularly spaced time grid, so that I can take FFTs etc. I've not had to do this much before, and I'm wondering if there is an "industry standard" or best algorithm to use? I'm currently using the Matlab function pchip, b...
Consider a static AWGN channel $$ Y = hX+N $$ where $h$ is a constant channel gain and the noise $N(0,N_1)$ is independent of the input $X$. The input is from a 2-PAM constellation, i.e., $X \in \ \left\{+√p,-√p \right\}$ with equal probability where $X= +√p$ is used to send a bit at 1 and $X=-√p$ is used to send a ...
I have some misunderstandings with Q format. Please help me in understanding this. If I say Q8 [unsigned] then there will be $8$ fractional bits. Is it ? How can I represent a $1$ and $1.5$ in this case? If I say Q8 [ signed ] then there will be $1$ sign bit and $7$ fractional bits. Here how can I represent a $1$ and ...
The idea is to get three classifications for crowd: High, low and medium based on an image captured inside a train compartment. Something like this: This gets classified in the 'High' category http://www.spiceflair.com/wp-content/uploads/2012/05/14a.jpg This perhaps in the 'Medium' & so on.
I have a piece of code that I am translating and I am trying to understand the reasoning behind it. The application here is thresholding medical (cellular) images. The function in question takes a histogram and returns the lower threshold value. The bit of code that I am translating mutates the histogram in a way tha...
In a direct sampling RF front-end, we can get real samples. But when we sample the signal with two ADC with a sampling delay $ \frac{4n+1}{4fc}$ where $n$ can be $0,1,2 \dots$ Then we have In phase(I) and Quadrature phase(Q) samples outs of two ADC. But there is a problem when we sample "I" before symbol transition and...
The LP-PSOLAR algorithm uses a the Linear Prediciton Coding (LPC) to calculate an error vector from a signal. This error vector is used to calculate the pitch markers. What do these pitch markers represent in the original signal in generall? Do they represent a change in fundamental frequency or a change in phoneme (I ...
Consider the measurements of two modulated AC signals (voltage and current) which may have noise, harmonics and a DC bias. Each measured signal consists of a list for the time and a list for the corresponding values. My goal is to find out the phase shift in degrees between these two signals in a robust way. I would li...
I have floating point data signal. I apply lossy control to it and I need some kind of metric to determine its quality. In image processing, there is PSNR (or MSE). I tried to use the same for it, It gives me some kind of result. But is there anything better (or any standard used for this kind or problem) ?
What I am attempting to do is take a signal (.wav audio file) which contains calls from different Bat species. Here is a graph representation of the signal: What I have done so far is: 1) Emphasise the signal 2) Compress the signal This is the output I got plotted: I need to remove the signals that do not warrent con...
I know that in many edge detection algorithm the first step is choosing a scale to work on. We actually operate on the chosen scale by applying a gaussian blur on the original image. I wanted to know , what is the relation between the detected edge's width to the sigma. e.g. If I want to detect lines 5 pixel wide , w...
I just studied the Algorithm of ANPR (Automatic Number Plate Recognition) from the chp 5 of book Packtpub.Mastering.OpenCV.with.Practical.Computer.Vision.Projects in which they detect the vehicle licence plate , but their are some restriction in that algorithm , i call it restriction because its specific for one countr...
I am trying to find out how can I count the number of people in a room using the feed coming in from a CCTV camera fixed in one of the top corners of the room. The number of cameras can be 2 at max but 1 for now. The objective is to only be able to tell how many persons are present in the room, in real time. The precis...
I am reading a nice book on DSP The Scientist and Engineer's Guide to Digital Signal Processing It mentions In electronics, the mean is commonly called the DC (direct current) value. How can a signal have no DC component if the mean is the DC component? Surely every signal's values can be averaged?
I'm trying to do something similar to Shazam (paper here) in Python. Basically the idea is to use FFT with sliding window to transform both WAV files and recorded audio to the same spectral representations for recognition purposes. The only problem is the spectrograms I extract from certain songs just seem to generate...
In image processing, scale-space is a technique to represent an image at different scales. But this really confuses me, since I always think that the word scale just means the size of an object, so the scale-space representation of an image shall have different sizes of the same image. However, it turns out that the s...
We can describe a BPSK signal, x, as: $$x=cos(2\pi f_c t+n\pi),\;n\in \{ 0,1 \}$$ The BPSK signal is then transmitted through a narrow band frequency flat Rayleigh channel, described by a single complex exponential $h=Ae^{i\theta}$. $A$ has Rayleigh distribution and $\theta\sim\mathrm{Unif}(0,2\pi)$. The received signa...
Section 13.2 of the HD Radio standard for AM defines a pulse shaping function that appears to be some strange variant on a raised cosine filter: More specifically, H(ξ) is a raised cosine filter with total width (1+α)T, and W(ξ) appears to be a "stretched" version. Why do they do this?
I was reading section 4.3.3 from chapter 4 of "A Wavelet Tour of Signal Processing", "...Let $\bar{f}(t)$ be a continuous time signal defined over $[0,1]$. Let $f[n]$ be the discrete signal obtained by a low-pass filtering of $\bar{f}$ and uniform sampling at intervals $N^{-1}$. Its discrete wavelet transform can only ...
So, in my class I have to learn how to derive the power relation in a Double Side Band, Full Carrier Amplitude Modulated Wave However, I have hit a wall somewhere. The basic equation is this : $P_{total} = P_{carrier} + P_{lower-side-band} + P_{upper-side-band}$ Then, $P_{carrier} = \frac{E_c^{2}}{R}$ where $E_c$ i...
I asked this question here: Audio Separation of .wav signal but it wasn't clear, so, here is my second attempt: First off, assume that I have a .wav file containing a sentence as follows: "My name is Michael" I would like to extract, from this, the following: "My" -> Phoneme (1) "Name" -> Phoneme (2) "Is" -> Phoneme...
This might be more of a statistical question, but let us assume that we take a spectrogram, or STFT, where time is on the x-axis, and frequency is on the y-axis in this matrix. What information might be lost - or 'changed', if we decide to normalize each column by its own standard deviation, so that the variance of eac...
Given this problem I know that the autcorrelation and power spectrum are fourier pairs, so when taking taking fourier transform of Rxx, one should end up with Sxx However, when I take the transform of $R_{xx}(m)$, I end up with $S_{xx}(\omega)=\frac{2\alpha }{\alpha ^{2} + \omega ^{2}}$, which also matches up with fo...
For a real signal, the power spectrum will have Hermitian symmetry. If I consider the one-sided spectrum (as negative frequencies, or anything above the Nyquist frequency are redundant) must I then multiply my power spectrum by a factor of 2? I have seen this done but I can't understand why the "redundant", symmetrical...
Consider the $m$-dimensional VAR process $${\bf x}_t = \sum_{l=1}^{P} A_l{\bf x}_{t-l} + {\bf e}_t$$ where the componenets of ${\bf e}_t$ are spatially and temporally independent and follow a generalised Gaussian distribution i.e. $e_{i,t} \sim GGD(\alpha_i, \beta_i)$. Given my data $X$, I am trying to estimate the VAR...
Say I have a signal that is not x-indexed. That is, the x-axis of the signal is the distance traversed by car and the y-axis is the heading direction of the car at the corresponding distance. Can I apply the Fourier Transform to this signal? If so, what is the physical meaning of this transformation? I believe that the...
I have gone through the below related posts of filtering noise from the signals MATLAB, How to filter a discrete signal? MATLAB: Filter a noisy EKG signal But the signal I want to filter contains noise which is not periodic as shown in the images below: Can anyone suggest a filtering technique for the noise labeled w...
I am in a intention of Porting Voice Activity Detection algorithm in real time hardware,so i was successful in classifying speech and silence/noise but now the problem is replacement of silence frames with comfort noise(preferebly any stationary noise) so can any one suggest any opensource CNG algorithms or how a comfo...
I am new to signal processing and I am trying to get the basic concepts right. I have a signal and I performed the FFT of the signal in MATLAB. I also did the PSD of the signal in MATLAB using the psd command. So I am confused about: 1) When we get the FFT of the signal, It's basically showing what frequency components...
There are various types of ADC in used, like, ramp-compare ADC Wilkinson ADC pipeline ADC Flash ADCs Sub-Ranging ADCs Folding ADCs Pipelined ADCs Successive Approximation (Algorithmic) ADCs Integrating (serial) ADCs but Successive Approximation (Algorithmic) ADCs (ADC0808) is most commonly used. I do not understand ...
I have the following system: Where $ {H}_{1} \left( f \right) = {H}_{2} \left( f \right) $ and $ \theta \sim U[0, 2\pi]$ independent of any other factor in the system. Given the input is identical, Is this an LTI system? Could you prove it? Otherwise I don't have explanation why if the input is White Gaussian Noise wi...
I am dealing with a complicated optimization problem with two parameters $N, K$. Setting $K =$ constant and finding the minimum for different $N$'s lead to a set of function values. I can do this for $K = 2,3,\cdots,8$ and $N = 5,6,\cdots,40$. Larger $K$'s or $N$'s take more time to calculate (e.g. 1 hour). I've plotte...
i've read that it is possible to interpolate between two sets of filter coefficients (if they are close of course), but how to interpolate between two sets of coefficients which are of different orders? Is it possible ? If not what is the possible way to fade between two IIR filters ? thanks Jeff
Are there any known methods to use existing keypoint detectors (which generally work on grayscale images) on color images? of course, one could always add the results of the detector on all three channels to get $3N$ keypoints, but I was wondering if there exist other established methods. Specifically, I am looking to ...
I am working in image cryptography using an ICA algorithm like JADE. To evaluate my work, I am used the SDR and PSNR. I would like to know what the acceptable values of the SDR and PSNR in image blind separation are.
Which one is better for frequency detection? In what circumstances? Advantages and disadvantages of both? Which would be better for a Voice Activation Detection algorithm? Which would be better for an instrument tuner? How would one detect pitch user an HHT transform? Is it faster than FFT? How would one implement HHT?...
I work with images in gradient domain because I sometimes need to manipulate image gradients (i.e. Laplacian filter responses) rather than intensity values. However, I need to apply a geometric transform (distortion) to an image already in gradient domain. Is is then meaningful to reintegrate the distorted image back t...
Sorry if this is elementary. If I add a small sinusoidal signal to my data and then take FFT of both my original and the modified one and then inverse FFT the division of the two, I should then be left with the sinusoid. But when I do the above, I get a shift in my output, as if the phase is lost. What am I doing wrong...
I am trying to understand one simple thing in wavelets and I searched a lot, but did not get any idea. I need to perform Multilevel 1-D wavelet decomposition on my signal. I have arrays signal[] and time[] ( time[x] corresponds to the time at which signal[x] occurs ) . After I perform wavedec(signal,level=1,'...
I am doing my project on speaker recognition. One of my friends is doing his project on speech recognition. Both of us need to calculate the MFCC for feature extraction. Is the MFCC going to be the same one for both of us? Is there any difference between the algorithm of MFCC for speech and that for speaker recognitio...
Let the function be $$ f(t) = e^{-3t} u(t -1) $$ The criterion is set at 70% of the total signal energy. I know how to find total energy, but not how to find the the $W$ (or frequency) at which the signal is a certain amount.
I have worked with state space models in relation to Kalman estimation. Here I have always seen state space models with fixed state size over time, i.e. the state transition matrix is square. Let us for example define: $$x_{t+1} = A_t x_t + B_t u_t\\ y_t = C_t x_t + D_t v_t$$ Then $A_t$ is $n \times n$. I suppose I cou...