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I would like to compare the positions of the same object in 2 different pictures and get the difference between the positions in in terms of x and y coordinates.Suppose I have a picture with a square figure on it at the centre and then I reposition the square to the top right corner.I would like to find by what distanc...
I am reading The Wavelet Tutorial Part III MULTIRESOLUTION ANALYSIS & THE CONTINUOUS WAVELET TRANSFORM by Robi Polikar The author explains about the following fig and says; Note that boxes have a certain non-zero area, which implies that the value of a particular point in the time-frequency plane cannot be known...
Our current project requires us to do some analysis using Wavelet Transform. Can anybody suggest me a practical book, preferably with MATLAB or C examples. I am currently reading some tutorials, but it is not giving me a feeling as I have for Fourier transform. I need a book having many practical examples with source ...
The problem I have can be seen in the context of DoA estimation or blind source signal separation and similar fields, where several source signals are observed by several antennas (or by an antenna array, respectively). The basic model is something like $$\mathbf{y}(t) = \sum_{i=1}^{n_s} s_i \mathbf{a}\left(\theta _i\r...
I have an image as attached The blue to yellow to red points are the points of interest the algorithm found, while the pink points are the radius per point. This maps pretty fine, but the issue I am having is that I need to find the area covered by this, but there are usually small holes, or when the direction changes...
Assume there are 11 signals. These signals are very similar to each other in both time and frequency domains. (And the similarity between them is not random; they are similar because they have somehow a common source. Actually they are well-correlated.) All these signals have the same number of samples. And I have acce...
From the IIR filter flow graph below i don't understand how the transfer function is calculated in every node:
I doubt because you have 9 multiplications and you are adding 9 numbers together, when using a 3x3 convolution matrix. Adding 9 numbers: 1+2+3+4+5+6+7+8+9 We are performing 8 additions as we repeat the operation of 'adding' 8 times (8 times a '+' sign). We can say there are 9 additions as well, if you would accept a d...
The uncertainty principle is usually stated as a relationship between a continuous signal and that signal's Fourier transform, and says that $$ \int_{-\infty}^{\infty} \! x^2 f(x) \ \mathrm{dx} \int_{-\infty}^{\infty} \! \xi^2 \hat{f}(\xi) \ \mathrm{d\xi} \ge \frac{1}{16\pi^2}. $$ Apparently it is somewhat difficult to...
I want to achieve something like this https://soundcloud.com/user7421656/sonicworx-isolate-preview . I've researched it and a few papers mention the use of "descriptors" on the audio and use thresholds on the descriptors to determine singing sources. But before I dive in and go hard out on how these people achieved it,...
For the preliminary part of a lab I had to do, I was asked the following question: I get how to do 1.1, but with 1.2, I really don't get what's going on; I really would've thought that the resolution bandwidth would be equal to the bandwidth of the low-pass filter. Furthermore, I'm pretty sure a spectrum analyser has...
I have some (well lots) of data signals which are measurements of current pulses. The current pulses are the result of spark breakdowns (20 kV + 700 pF capacitor so quite energetic) and my problem is that noise, and displacement current signals are clouding the underlying current pulse. The noise and the displacement c...
I have designed 9 IIR bandpass filters (chebyshev type 1, order 4) using fdatool in Matlab. Then I use a and b filter coefficients to apply it on differential equation. So, my question is how to apply (real-time) gain of each bandpass filter using sliders? ( how to add gain or loss to certain filter?) Here is my code i...
I want some data (images that have only shift) to test on xuvtools or Microsoft ICE and my own software. Is there any available?
I'm new to wave problems, so excuse my simplified words. I have a signal consists of high range of frequencies . I applied a Band-pass filter to this signal and I have been able to separate it to bands of specific frequencies. The problem that I want to solve is as follows: Each band have a lot of individual waves of V...
I have a Virtex 6 FPGA running at 200MHz with ADC/DACs on it. I have been converting a WiFi signal (2462MHz) down to a more reasonable IF of 25MHz, sampling, running the signal through taps, tweaking them, and mixing back up. This all works fine. My issue is that since I am mixing down and sampling at a lower rate, I...
I wish to use trainCascadeObjectDetector to train on new objects (other than faces); I have the Computer Vision Toolkit installed; but this function trainCascadeObjectDetector seems to be missing from my system. Did I miss something during installation? Or has this been removed from R2012b?
I have to design an 8th-order IIR filter with the following transfer function magnitude response as a specification: I use "ellip" function from matlab to calculate the unquantised coefficient set: Then i implement the filter using a cascade of second-order sections using "tf2sos" matlab function: And then i quantiz...
My signal graph is as follows I have < 10 reputation, thus cannot post the photo... The graph are simply roughly the series of -5, 0, +5, 0, -5, 0, +5, 0.... but with sawtooth-like small peaks around the prominent peaks. I am going to do the peak detection. But what I need to count is the "prominent peaks". i.e. The "...
"Using a state-of-the-art atomic force microscope, the scientists have taken the first atom-by-atom pictures, including images of the chemical bonds between atoms, clearly depicting how a molecule's structure changed during a reaction. Until now, scientists have only been able to infer this type of information from spe...
I've read many questions on SO, and frankly, each one of them is not describing any particular way to go about it. Some say "do FFT" and some say "zero crossing" etc. But I've only gone as far as understanding that the digital audio input consists of an array of amplitudes for a particular frequency and well.. Don't re...
The primary source lists Butterworth polynomials in s-domain and provides a link to bilinear transform for digital implementation. But, who needs analog specification in our digital world? Why should everybody get involved into the quite laborious bilinear transform, which, additionally requires frequency pre-warping? ...
I need a tool or code for extract not blurry images from video (it can be used for panorama or SFM building from video) something like http://research.microsoft.com/en-us/um/redmond/groups/ivm/sharppanoramas/ (It is worth noting that software for building panorama from video is already exist, but I don't know why they ...
I'm using a set of features extracted from a signal for classifying the data window with KNN algorithm. Since the features have different value ranges, their influence on distance calculation is different when you use euclidean distance in KNN. To equalize the influence of these features on classification: I can norma...
related to Filtering voice signal from randomly added frequencies I'm trying to create a simple low pass filter but i'm not sure what is wrong. my code: [y,Fs]=wavread('sound.wav'); m = length(y); %30586 Fc = 500; %% frequency cut n = [0:m-1]; %% kernel length / samples Fs = Fs; %22000 ...
Can you tell me whether the following signal is a energy or power signal? $$ x(n) = e^{j (n \pi/2 + \pi/8) } $$ I've solved it and found it as it was neither power nor energy signal was that right?
I went through this material on Sampling theorem. http://www.wescottdesign.com/articles/Sampling/sampling.html It uses a powerline signal example and concludes that if a signal is cyclic and repeating over a period N then you could sample the signal at lesser than N samples per second.In the example he shows for a 60Hz...
I want to sample two signals, S and R and later do IQ demodulation on these. I am multiplying S with R and I also want to multiply S with a 90 degree phase shifted version of R by shifting the samples in R so I get a 90 degree phase shift. The problem I encounter with this is that unless I have a sampling rate that is ...
suppose we have a cascade realization of a transfer function of a higher order. What difference does it make if the order of the transfer function is an Even or an Odd number?
I have a set of recordings sampled at 16khz, but some of them are upsampled from 11khz and some from 8khz. Quite some of the recordings are noisy. What is the best way to classify the recordings according to the original bandwidth? I know some people are using GMM classifiers for detection. A paper on the subject is DE...
Do you guys know good DSP related websited and blog posts? I know that for example there is this site called Minutify: http://www.minutify.com which is blog posts regarding all areas of electrical engineering including DSP. Do you guys know of similar resources?
Background I am working to identify and isolate each human limb of a known person, in pose, in an image which is relatively noisy. It should also be noted that the image is a snapshot extracted from a video (which can be utilized in this task). The end result ideally would be a bunch of polygon estimates of each body ...
@Michael C. Grant wrote that "the cost of FFTW isn't an easy formula based on log_2 anymore." But the Wikipedia article says that the FFTW "can compute transforms of real and complex-valued arrays of arbitrary size and dimension in O(n log n) time." So can it compute 1 forward FFT with the size 1920*1080? in kNlog(N)/l...
In the filter graph below: where circle with 'X' inside is a multiplication circle with 'Σ' inside addition and the triangle is a delay element What could be the values x(k) in this type of filter? There is a rule? There is a range? I can understand the matrix of coefficients but i don't know about the x(k) values.
If you want to compute a convolution with a DFT/FFT, you will have to do a DFT on the image and a DFT on the kernel. And you will need to multiply the spectrum of the image with the spectrum of the kernel. Is it correct that the multiplications in the fourier domain are complex multiplications? So, that those point-w...
I was reading a paper related to privacy where they perform fourier transformation of a time series data and retain few coefficients of the FFT. Then they apply some noise to these coefficients. These noisy coefficients are zero padded. Finally inverse DFT is applied to obtain the original data which is not exactly equ...
I've built a simple device, which sends readings from a pulse sensor to a PC. I get these data: What technique or algorithm can I use in order to calculate the pulse rate (beats per minute) from these data? Update 1 (03.06.2013 21:01 MSK): The time series represents my heart beat during approx. 3 minutes. According to...
Someone wrote "The algorithm that Cooley and Tukey presented in their classic paper (Math. Comp. 19 (1965), 297-301. http://dx.doi.org/10.1090/S0025-5718-1965-0178586-1) can be applied to any composite length. The performance advantages are greatest for highly composite lengths, of which powers-of-2 are one example, a...
I know there are 4 types of FIR filters with linear phase, i.e. constant group delay: (M = length of impulse response) Impulse response symmetrical, M = odd Imp. resp. symmetrical, M = even Imp. resp. anti-symmetrical, M = odd Imp. resp. anti-symmetrical, M = even each with its traits. Which of these types is most co...
The wavelet transform informs what all derived wavelets (from mother wavelets) with different scales at different point of time , are present (along with the associated strength) in my original signal. I am not able to understand how do I now correlate it with FFT that simply gives what all sine and cosine waves are...
I have an audio buffer with 1024 samples per second. The sample rate is 44100. I am trying to identify the frequency tones. Using the Gorzel algorithm, I can get the frequency but it's not that accurate. I was trying some other scope program and I got better results with the same hardware. How can I decrease the id-ban...
I have seen many times, including in a post on this forum, that it is recommended to have unity gain at DC for a lowpass filter. I would like to now the mathematical explanation.
I am implementing a simple LPF in Java. But I have problems in choosing the value of ALPHA. public class LowPassFilter { /* * Time smoothing constant for low-pass filter 0 ≤ α ≤ 1 ; a smaller value * basically means more smoothing See: * http://en.wikipedia.org/wiki/Low-pass_filter#Discrete-time_rea...
I have this confusion. I was reading this paper. It says lets say I have a n dimensional vector X and I take its DFT i.e DFT(X). Then the n Fourier coefficients have the same L2 norm as X. How is it true? I tried on a vector X = ones(100,1). Applied DFT to X i.e, DFT(X). But norms are not equal
I have more general question about the extended Kalman filter usage. What is not clear to me is why the EKF uses non-linear functions $f$ and $h$ for state prediction and estimate, while in other places the Jacobian of these functions is used. Why the following is never used? First calculate the linearized state and ...
For seismological application, I am playing with an elliptical filter, and once I have both sets of feedback and feedforward co-efficients, how do I make it so that I can plot the frequency response of the system? Can I also use the same technique to make the frequency response for a bessel? I know that I can take the ...
I am watching the following video on software defined radio and at the beginning the narrator points out some "whistlers" on the spectrum display and says maybe they are caused by an "annason". He goes on to say that he knows "commercial annasons scan at a slower rate than the whistlers". What is an "annason". I don't ...
The frequency domain is allegedly preferred because it replaces convolution of complexity $n^2$ with a diagonal matrix multiplication. Yet, I see that in z-domain we have multiplication of polynomials, $(a_0z+a_1z^{-1}+a_2z^{-2}+\ldots)(b_0z+b_1z^{-1}+b_2z^{-2}+\ldots)$, which is, despite its name, is nothing more than...
I am working in digital image restoration field, recently i am studying a paper Blur-Kernel Estimation from Spectral Irregularities, in this paper i have got some terms, which i am unable to understand, can anyone please provide me specific description of these terms. [i have already google it, but i didn't get satisfa...
What does it mean by leakage in case of digital filters? My specific question is about the frequency domain adaptive filter function provided in the Matlab DSP toolkit, accessed as adaptfilt.fdaf. It has a parameter called LEAKAGE, but I am not sure what exactly does it represent or how it affects the filter response. ...
I'm trying to do global registration algorithm for panorama like described here. But I don't understand how they deal with false pairs. Also if expand the system u1*I*P1=u1*I => P1=I u2*I*P2=u2*A21*P1 =>P2=A21*P1 u(i) coefficients are reduced? Or I interpret system of equations wrong?
I try to work out the simplest system, whose response is $H(z) = 1 + cz + c^2z^2 + c^3z^3 + \cdots = 1 / (1-cz)$ Now, Z-transform done quick (Fourier connection) and 4.5 Transfer Function, Poles and Zeros say that we just need to plug $z => e^{iw}$ to get the frequency response. I am getting single pole at $w = i \ln ...
Suppose we know the DFT of a discrete limited sequence, some $X[k],\ k = 0, 1,\dots ,N-1$. How can we calculate the Fourier Transform of the same signal for a random frequency $\Omega$? EDIT: How about this, we can get the starting signal with inverse DFT, like: $$x[n] = \frac{1}{N}\sum\limits_{k=0}^{N-1} X[k]e^{j\fra...
I'm having 2 signals which have different sampling rates i.e., 1ms(A) and 4ms(B) and I've tried to upsample/downsample either of the signals based on the code snippet Resampling based on FFT which I believe is equivalent of what is descriped in How to resample audio using fft or dft % FFTRESAMPLE Resample a real signa...
What would be a decent formula for approximating the connection between voltage and decibel output for a typical set of unpowered in-ear headphones? I'm generating sine waves from the computer: an array of floats between -1.0 and +1.0 gets sent out through the speaker: X_out = h * sin( f * 2π * t ); // h is the amplitu...
I am new in matlab and signal processing. The time series that have been used are obtained from accelerometer in a building. As far as I understand both the time series' length and window function length(hamming or hanning) should be same. But for smoothing the fourier amplitude spectrum, optimum window length can vary...
I've just come across a paper that interrelates the covariance matrix of time discrete signals to their autocorrelation function (or a time-delay, respectively), i.e. $$\mathbf{C} = E\{\mathbf{x}\left(t\right)\mathbf{x}^H\left(t-\tau\right)\}.$$ Coming from the image processing field, I have usually just worked for the...
MATLAB's filtfilt does a forward-backward filtering, i.e., filter, reverse the signal, filter again and then reverse again. Apparently this done to reduce phase lags? What are the advantages/disadvantages of using such a filtering (I guess it would result in an effective increase in filter order). Would it be preferabl...
If I understood it right, a keypoint is a tuple $$(x,y,\sigma, r),$$ where $x,y$ define the position of the keypoint and $r$ an orientation (given by the most domiant gradiant vector around the keypoint). I'm still not sure about the intention of $\sigma$. What is done in the first part of SIFT is to consider $$L(x,y,\...
I have a project in which I have to create chorus effect to an audio file. I have tried this code and i don't know weather its right or wroung. Anyway the code is below [x,fs]=wavread('ShortWAV.wav'); f=0.25; delay_in_sampels=0.05*fs;%50 ms delay index=1:length(x); sincurve = 1*sin(2*pi*index*f); y = zeros(length(x):1)...
i am working in digital image restoration field, recently i have studied image degradation model which is like that. g(x,y) = h(x,y)*f(x,y) + n(x,y) where g(x,y) is the degraded image f(x,y) is the image that we wish to capture or its is latent image h(x,y) is image degrading function, and * is convolution operator n(...
Hi I'm trying to separate sound from music using wfilters and heard each part separately using its name scale3 function function s = scale3(q,n) p = length(q); if p==n s = q; else s = resample(q,n,p); end the program s = wavread('K:\Khaleel_3aaza (1).wav'); % whatever here [l,h] = wf...
Is there a way to make an inverse notch filter block in matlab simulink? I have found the peak-notch block, but I need to amplify the signal instead of attenuate it. Thank you.
In trying to answer another question, I got stuck applying what I thought would be a valid way to answer it: using the system function. Here's what I did: First, I put the question Y[i] = Y[i-1] + ALPHA * ( X[i] + Y[i-1] ) into DF I: Y[i] = ALPHA * X[i] - (ALPHA-1) * Y[i-1] This tells me that the system function is: ...
In order to get more detail when encoding the blacks, a log LUT is going to be applied to the pixel values. Like gamma, the logical place to do it in the image processing pipeline is after demosaicking, so the same LUT is applied to the R,G,B components. However, it occurs to me that a very similar effect can be ach...
Which measure should be considered better and when ? I tested both the measures on some data that I have and I got mixed results i.e. some are showing better results with Pearson and some with the Normalized Squared Error. By similarity I mean similar shapes of the signals that I have. I normalized the data before appl...
I want to find all occurrences of a specific phrase in a set of audio files (it's a catch-phrase in a TV show). The phrase is pronounced by different people with different speed etc. What is the best approach to this problem? Additional details: It's a one time thing. The end result should be a program that takes 30 ho...
This is simple i thought, but my naive approach led to a very noisy result. I have this sample times and positions in a file named t_angle.txt: 0.768 -166.099892 0.837 -165.994148 0.898 -165.670052 0.958 -165.138245 1.025 -164.381218 1.084 -163.405838 1.144 -162.232704 1.213 -160.824051 1.268 -159.224854 1.337 -157.383...
I am not very experienced in signal processing, and it seems like this should be a simple problem. I have two computers, one plays a tone a 19 kHz at a sampling rate of 44.1 kHz, and the other listens with a sampling rate of 44.1 kHz and tries to find if the 19kHz tone is present. The transmission of the signal is in a...
I'm trying to get MFCC from speech audio data. I'm using 26 Mel Filter on 0-24kHz spectrum. My MFCCs doesn't look correct to me, cause always first bins have high values and values getting smaller through the high frequency bins, even with high frequency audio data. Then I realized that the cos((PI/N)*(n+0.5)*k) part i...
i am interested if where can i find matlab code for Prony method for frequency estimation.there is pdf article about prony https://www.google.ge/#site=&source=hp&q=prony+method+for+distorted+signal&oq=prony+method+for+distorted+signal&gs_l=hp.3...2187.8079.0.8365.0.0.0.0.0.0.0.0..0.0.ernk_timediscountc..0.0...1.1....
I am beginner and I want a help to know how to detect hand in Matlab. First, I need code which detects skin and then draw a rectangle around a region so it show to me face and hand. When I try to make rectangle only around hand and the hand is away from the face, the the rectangle is drawn around the face instead. What...
I am currently in need for filtering accelerometer data for an Android application. First of all, I used a simple low-pass filter (simple infinite impulse response filter) as follows: for i from 1 to n y[i] := y[i-1] + α * (x[i] - y[i-1]) This helped me achieve a smoother result. Then I decided to play with FFTs. I...
i'm looking for a free/opensource solution for digital image processing with reasonable performances processing quite large images (e.g. actual smartphones have resolutions of 8-10Mpx) and being ready for fast-prototyping/research. Octave would be a good solution as it provides a lot of statistical and matrix manipula...
Im trying to cross correlate two signals in matlab and get the phase difference between the signals. For cross correlation (the idea is to do it without xcorr) I used: Cxx=fftshift(ifft(fft(x,N).*conj(fft(y,N))))/(norm(x) * norm(y)); I get the result and not sure about the reference point I have to take for phase calc...
Can anyone explain what are Autoregressive Coefficients? What is their meaning that is. Consider a method: public double[] calculateARCoefficients(double[] inputseries, int order) When this method is called on 256 values, lets say these values represent some kind of signal, I will get an array of n numbers, where n=ord...
I want to implement a HPF which will pass all frequencies above 150hz. Fs = 1000; % Sampling frequency T = 1/Fs; % Sample time L = 1000; % Length of signal t = (0:L-1)*T; % Time vector % Sum of a 50 Hz sinusoid and a 120 Hz sinusoid x = 0.7*sin(...
For the purpose of controlling DIY disco lighting, I'm looking at implementing beat detection in a microcontroller with limited computing power (i.e. Arduino Uno). While detecting all beats would be a plus (for one, it would allow measuring BPM, which could be used to adjust the light to the style of music), it's not a...
This code was working fine when I asked about it in Filter is too large but when I tried it again on another PC it gives me this error: Warning: Direct access of structure fields returned by a function call (e.g., call to Untitled2) is not allowed. See MATLAB 7.10 Release Notes, "Subscripting In...
I am working on a university project in which I need to find a periodic net element in an image. The net is a set of diagonal lines in a specific angle ( which I don't know in advance ) on a noisy image. They are hard to detect so I would like to use all the information I can get my hands on. My current algorithm is b...
I am trying to make a frequency domain adaptive filter in matlab. It uses matlab adaptfilt.fdaf to create the filter parameters like step size and initializing initial filter weight values. Then I have tried to implement the overlap - save frequency domain adaaptive filter algorithm from the paper "Frequency-domain and...
For an FSK modulation with coherent detection, in order to ensure orthogonality, the frequency separation between symbols is $ \Delta f=f_2 - f_1 =1/2T$. However, in case of non-coherent demodulation, which must be this value $1/2T$ or $1/T$?
Can we compare the size of two objects appearing in different images with different resolutions?
In this code I try to decompose music from singer sound with wfilters but when I hear it it is still mixed where Iam I wrong ? function s = scale3(q,n) p = length(q); if p==n s = q; else s = resample(q,n,p); end the program s = wavread('K:\Khaleel_3aaza (1).wav'); % whatever here [l,h] = wfil...
One of the problems in Proakis's signal processing book using Matlab asks to plot the histogram of x3(n) = x1(n) + x1(n-1) where x1(n) is a random sequence. To create x1(n), I do the following: n1 = 1:100000; a = 0; b = 2; % interval over which to pick random numbers. x1 = a + (b-a)*rand(100000,1); N1 = hist(x1,100);...
I'd like to know what the parameters for the two-dimensional gaussian distributions are, which create this (I think) quite common image-processing filter: $$ h_\sigma(x,y)= \left[ \matrix { 1&2&1\\ 2&4&2\\ 1&2&1 } \right] $$ The two-dimensional gaussian equation looks like: $h_\sigma(x,y)= A\cdot e^{-\frac{x^2+y^2}...
Although I have a solid experience in designing audio engines and such, I am fairly new to the realm of Digital Filter Design, particularly IIR and FIR filters. In other words, I'm trying to learn as much as I can on how to design filters and derive their difference equations. I'm starting from the basics, so please be...
I am trying to calculate an actual frequency response from a desired frequency response. fftSize = 128; fs = 8000; I have 2 input arrays: f[] and aDesired[], both of which are Real and length fftSize+1 (DC + Nyquist included). The f[] array is linearly spaced between 0 and fs/2 (0:4000). The aDesired[] array is either...
Can one in theory achieve as high accuracy as needed by sampling a sinusoidal signal over n Periods? Is there a formula that connects the signal frequency (f), sampling frequency (fs), periods (P) and accuracy (Acc) or can someone explain how accuracy rise with number of periods? Let's say I want to know the signals am...
Is there any method of decoding a Manchester encoded signal (IEEE 802.3) that is independent of the bit period? Suppose the data I want to send is $\rm 10110$. The Manchester code of this data will be $\rm 0110010110$ (according to IEEE standard). We can see that in the $\rm 0101$ part, we need to consider the $\rm 0 ...
As a rule of thumb, the bandwidth of a communication system with a complex signal must be equal to the bit-rate or symbol-rate? Besides, after reading this post, it is not clear for me how a band-filter or low-filter signal can influence the maximum data rate.
I need to visualize some data as gray-scale images after contrast enhancement. The processing algorithm must be built on few samples, and then it will be applied on unseen images (suppose, the [min, max] range is given). I plan to: Use sample images to estimate actual histogram H1. Define desired histogram H2. Build a...
What is the significance of the first zero up-crossing of a cross-correlation? The signals being correlated are cosines signals with some phase difference. EDIT 1 : Cross-correlation plot attached. The signals are cosines with phase difference and may be with some low frequency components. Sampling frequency of the me...
I am making a pronunciation Scoring system, and I need to know how to separate the pronunciation information of an MFCC of a given spoken word, so that later i can match this MFCC with the MFCC of the word. can that be done threw matlab or HTK ? Second Question: In the matching part, shall I used DTW or standard edit ...
I have recently fallen into fallacy, considering pole s=1 as there is infinite response at frequency 1. Yet, response was only 1. Now, can you derive the frequency response, given the poles? Secondly, the theory says that a system is stable when poles are in left s-plane and, thus, decay in time. But, wait. Does'n "po...
Suppose I have an FM signal with modulation index mf which is passed through a frequency tripler, then what will be the modulation index of output waveform ?. Thanks in advance....
I am going to start my research on digital image restoration, I just want to know that should i learn the course digital signal processing before starting my research.
I am dealing with images from biological slides and want to segment out the ROI (cells) from the background. I have tried k-means clustering, otsu thresholding, adaptive thresholding, edge detection etc. Running out of ideas now. Which other segmenting techniques can be tried for such images? Keeping in mind that the c...
Can anyone recommend the best references for independent subspace analysis (ISA). I have read 'Multidimensional Independent Component Analysis' by Cardoso and would like to know of any modern, solid (i.e. generally accepted to work) approaches. If you have implemented ISA before how did it perform?
In a set of images taken with the same infrared camera, I have a noise signal presumably originating from the camera chip. How would I go about extracting said signal to be able to substract it from all images I have taken?