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I'm reading Computer Vision: Algorithms and Applications wich is available online as a PDF and on Chapter 3.2 page 111 it introduces convolution operators for images giving the formulation: $g(i, j) = \Sigma_{k,l} f(i − k, j − l)h(k, l) = \Sigma_{k,l}f(k, l)h(i − k, j − l)$ since $(i,j)$ are pixel coordinates $f$ is th...
What does High frequency and low frequency component signify in an image. How to separate out the high frequency as well as low frequency component of an image using Bilateral filter.
I'm wondering why the following code for GNU Octave appears to be incorrect in the frequency domain by a magnitude of 2. For example, in Figure 1. (Sorry, I'm apparently not allowed to post images!) it can be seen that the -3dB points of the bandpass butterworth filter are at 150 and 300Hz instead of the design require...
Sorry, two questions in one day! I'm struggling to understand what I'm doing wrong in this very simple filter design. I want to design a simple, single pole low pass filter and implement it as an IIR in an embedded system. I was looking at the DSP book online at recursive filters, and using their single-pole example to...
I was confused about the double hat notation $\hat{\hat .}$ and my teacher and I thought it probably means an estimate of an estimate most of the time, but we could not be 100% sure of its meaning. One hat $\hat .$ is an estimate but what does this double hat usually mean? Does it have many different meanings in DSP? C...
I'm following a computer vision course and I have this exercise: write a program that, given an hand image, it can recognize if the hand is open, closed, in a punch, or holding an "ok" posture, using only the techniques provided till now (pixel 4/8 connected, connected region, contour finding, holes finding, blob prope...
I have asked this question before in the sense that what does a high frequency and low frequency component signify in a image and i got satisfactory answers now i want to know that how i can get high frequency and low frequency component separately from the image. I mean what changes i have to make in the equations of...
I am working on developing an autonomous control system for toy IR remote-controlled helicopters (like the Syma s107, for example) using a webcam and an Arduino with some IR LEDs. I am using C++ with OpenCV on a laptop running at about 2Ghz and I need to keep the system as close to real time as possible. I am unsure of...
I need to calculate frequency response, phase response and apply to signals the Butterworth, Chebychev1 and Chebychev2 band-pass filters. I'm developing in C++ with Qt, and I'm looking for algorithms or a (free) library to do that. I have my plotting solution, I just need the formulas. The input values I have are the t...
I encountered a problem when I tried to generate by a program an sinus audio signal. My parameters are Sample Rate: 44100 Hz Data Bits: 16 bits Channels: 2 I need to create a WAV file that is generated on the fly in and play in real time directly to a audio out port. There are two more requirements. I need...
I got the concept of Convolution in signal processing from a video lecture that it is method to get the area overlapped between two signals when one signal is flipped over and traversed over another signal and that it allows for getting combined effect of two signals during that traversal. But how this concept is appli...
Is there an accepted way of determining the noise floor of a signal by looking at it in the frequency domain? Is it a matter of averaging all the bins, or median, or some more complex calculation like those described in the question below? What's the best criterion for determining a frequency peak? I want to determine...
I have an PCM (pulse-code modulation) signal. If I want to work out the frequency of a sample $ x $ long, I count the number of peaks/troughs of the sample, and divide by $ x $. The equivent dB of frequencies $ 20Hz > 20 000 Hz $ is then just the equivelent dB of the signal as a whole However, I would like to create a ...
By iterating or nesting a sine curve like in this question I get curves like these: that seem to tend to a square wave. The eight case of these looks like: which here on purpose was chosen for it's roundness. The Mathematica code for these plots can be found here: http://pastebin.com/6UK1u1uX I don't know much about ...
In "A Computational Approach to Edge Detection", John Canny explains that the optimal "detector" for finding step edges profiles in an image via convolution is a non-closed-form function closely approximated by the first derivative of a Gaussian. He derives this by defining "optimal" as the function which maximizes the...
I have a sound signal, and I would like to implement a simple realtime level control (for info, I am using Pure Data, but I'd like to know about the general case). For this I thought of using the envelope amplitude (in dB) of the input Vin_db, calculated over a window of N samples ; use it to calculate the gain G to ap...
I've been reading bits and pieces online but I just can't piece it all together. I have some background knowledge of signals / DSP stuff which should be enough prerequisites for this. I'm interested in eventually coding this algorithm in Java but I don't understand it fully yet which is why I'm here (it counts as math,...
I need to count the number of 'patterns per second' in a signal. The problem is that the signal changes 'shape' a lot and unpredictably. It is still easy for a person to see the the pattern, but I don't know how to do this with code. EDIT For extra clarity: say the first sample is 1 second. There are 17 copies of the p...
I retrieve contours from images by using the Canny algorithm. Is this enough preprocessing to have a descriptor image for input into an SVM to find similarities? Do I need other features like elongation, perimeter, area? I am asking about this because I was inspired by this example. I have given my image in greyscal...
I'm playing with Python & numpy to filter audio. It is relatively straightforward to design and us low-high-bandpass/stop FIR and IIR filters, stuff can be done in a couple of lines. I'm implementing a small software defined radio software, and I need an all-pass filter that has 90 degrees phase shift everywhere, and i...
I want to learn computer vision and so I downloaded image processing slides from my university archive. I'm on histogram equalization. Learning about local histogram equalization points me out to this procedure: The procedure is to define a square or rectangular window (m*n, n,m odds) and move the center of the wind...
I'm having troubles trying to use one relatively simple noise estimation method, here it is:A Fast Method For Image Noise Estimation Using Laplacian Operator and Adaptive Edge Detection My problem is with their proposed noise deviation formula (page $2$, ($3$)), I can't seem to comprehend it. What I want to know am I ...
I would like to implement edge detectors for creating a feature vector for archeological signs. Which are best features, algorithms to retrieve? That could be loaded in a Support vector machine for classification. If there are in the image only signs with a uniform background it's better converting an image in a binary...
How should buffer value at N/2 get initialized prior to performing an inverse FFT on a buffer, size N, that contains a complex frequency response in only values 0 to N/2-1? It is pretty clear the buffer needs to be made symmetrical for prior to calculating the inverse FFT per table 12-1 of http://www.dspguide.com/ch12...
I have a wav file containing a series of chords, each containing 3 or 4 notes, played concurrently. I need to find the frequencies of each note in each chord, and I can use matlab to do any computation. Currently I think I need to find the spectrogram of the wav file and convert the frequencies from normailsed frequenc...
I = imread('13.jpg'); Ir = I(:,:,1); Ig = I(:,:,2); Ib = I(:,:,3); %# Create the gaussian filter with hsize = [5 5] and sigma = 2 G = fspecial('gaussian',[10 10],1); %# Filter it Ig = imfilter(I,G,'same'); %# Display imshow(Ig); PSF = fspecial('gaussian',7,10); Irc = conv2(Ir,PSF,'same'); ...
I have an EEG signal that I want to extract features to apply multilayer perceptron (MLP). What I should use, Fourier or wavelet coefficients?
I have multiple trials of physiological data. I am doing a frequency based analysis to analyze power (amplitude) in certain frequencies of interest. Is averaging multiple trials of equal length and then taking a single FFT of the averaged signal vs computing FFT for each trial and then averaging the frequency bins the ...
i am simulating OFDM transmitter in matlab....As u all know OFDM consists of different sub systems such as FEC coding, Bit interlever , Modulator , IFFT block and Adding cyclic prefix.... i have integrated all the subsystems.. now i have to check the datarate of the complete OFDM transmitter...how to find the datarate ...
I'm currently designing a receiver which has to determine whether a signal contains specific frequencies or not. The frequencies are at constant 215Hz difference: {16,351 16,566 16,781 16,996 17,211 17,426 17,641 17,856 18,071} I sample at 44100kHz and I have 1024 samples each time. Therefore my resolution i...
I have images where there is a lot of "black" background (few shades of black). (In many images at least half the pixels are background). I need to get interest points from the image, but because the background is not smooth (although black) , I get a lot of "false" key points in the background. What filter/s will fit ...
Hello I've been trying to understand the grayscale dilation operation using the following example, but my results are different. $I = \begin{pmatrix} \bf{5} & 2 & 4 \\ 1 & -3 & 1 \end{pmatrix}$ is the image to be dilated, and the structuring element is $S = \begin{pmatrix} 0 & x & \bf{0} \\ 0 & 2 & -1 \end{pmatrix}$ I ...
I have a block of n bits that I'd like to encode into m bits that have an equal number of zeros and ones. I looked at protocols like 8b/10b but it wastes 25% of the data to achieve balance. As an example, it turns out that 128 bits have ${128 \choose 64}$ combinations where there is an equal number of zeros and ones. ...
Are there any general assumptions for the calculation of the Hurst exponent? Does the signal need to be stationary, for example? Does it depend on the method? What about the length of the time series, is longer better? I am interested in using implementing it for the analysis of an EEG signal. I have read many artic...
I am developing software for transmitting data from a computer to a mobile device using audio. Specifically, from the computer's speaker to the device's microphone. (I am developing both the sender and the receiver software.) This is one channel, and the sample rate is a standard 44.1Khz. It needs to be VERY resilient ...
I want to make any song sound like it's being played on a gramophone record in the 1920s. What specific filters would I apply to make this happen? I'm looking for technical details, not some magical program that does it for me. I'm trying to create a programming interface that does this, so a user interface doesn't wo...
Let me start by clarifying the terminology that I will be using in the following. A codec packet is the compressed data which an encoder returns when it is called once. An RTP payload can consist of one codec packet or a sequence of codec packets. My questions: Does the RFC for SILK allow the RTP payload to consist of ...
I am working on building speech recognition software for my company. In the meantime however, we will purchase some off-the-shelf commercial speech recognizers until the in-house one is ready. My question: 1. How are speech recognizers evaluated? 2. Are there any standardized tests available that can be used to test ...
I would like to evaluate frequency of some unstable periodic signal coming from a detector: The signal is registered continuously and may or may not be present (i.e. be periodic). The frequency and amplitude of the periodic signal can change but stays in some certain range. The shape of the periodic signal is more or...
There are other shape properties scale-rotation invariant except HuMoment? There are example that show me how can i implement them? I find this example in c++ https://stackoverflow.com/questions/8675878/opencvc-calculating-moments-from-contour but i prefer working in python Which is the difference between WeightedNorma...
I'd like to compare 2 speech similarity and show it in terms of 0-100%. Here's my approach: 1: Convert 2 audio waveform into audio spectrum by Fast Fourier Transform (FFT) 2: Use Dynamic Time Warping to compare 2 spectrums Everything is fine when converting them into audio spectrum But how to show the similarity in ter...
I am struggling to find or derive the relationship between Doppler frequency and the associated phase shift. Let's say I have a signal transmitted at a given frequency $f_c$. I then down convert from $f_c$ to baseband at the receiver, but due to speed between the transmitter and receiver, I receive a frequency shifte...
What are the steps to built a system which recognizes hand written signs in an image and returns the most similar sign in Python? I know that depends on the image and many variables, but isn't there a sequence of steps, a main path to follow for recognizing images with at most 2 or 3 signs per image? I thought the ste...
can any one explain how actually IFFT generating sub carriers in OFDM.... when we are simulating OFDM transmitter in matlab, we are taking input data stream, encoding it and modulating the encoded data. then we are converting this modulated serial data stream to parallel data and giving it to the IFFT block. my problem...
I am in trouble with Viterbi decoder for a while. I have some questions. I need to decode messages from a continuous stream which is encoded without resetting the state metrics. Each message starts with a preamble, but since these preambles are also encoded as part of the stream, there is actually no message concept be...
I am trying to compare 2 speech samples and rate them on similarities. Think of someone trying to repeat a phrase, and then comparing those 2 audio files. I started by implementing the MFCC (http://en.wikipedia.org/wiki/Mel-frequency_cepstrum) algorithm. I calculate the MFCCs of both audio samples, which gives me roug...
Let's suppose a chain of 6 multipliers: $b = a_1*a_2*a_3*a_4*a_5*a_6*a_7$, where $b$ is the output, and $a_i$ the inputs with 16 bits. How can I optimize the word lengths of the output of each multiplier in order to minimize the fixed point error against a full precision implementation? Let's also suppose that the outp...
In the image processing world, I've noticed that a lot of the popular convolution filters are separable. Here's a quick list of common separable filters: Sobel Gaussian blur Box filter (all ones, for example) Not separable: Laplacian What other common filters are not separable? I'm a little bit worried about this ...
I am trying to analyze the musics as possible as precisely. Of course I tried FFT, but got some problems. I found low frequencies have very low resolution than human's hearing. I tried very long time FFT to resolve this problem, but even analyzing with 8192 samples/s in 44100Hz sample rate(Means lack of time resolution...
i have 2 audio signals, both available in time and frequency domain: A: Noise B: Noise(A with minor variations) + Additive sounds (C) i want to eliminate Signal A from B to leave only C. C is for testing almost zero, so after removing A from B silence would be the best. My signals to visualize the problem: Signal A: ...
I just study signal processing and I have some questions. How to find impulse response $h[n]$ sequence of the system if I have $x[n]$ and $y[n]$ (input and output) sequences? Hope there is a formula to do that. How to find $x[n]$ input sequence if I have $h[n]$ and $y[n]$? Is it possible to find $x[n]$ if I just have ...
Reading the signal literature I often come across the expression $k \delta (t)$ where $k$ is a constant. I presume this is a notation to suggest we are referring to the area or strength of the Dirac function since the multiplication of a constant, $k$, by $\delta (t)$ seems to make no sense. My question is what does $...
I have a digital signal which may be represented as a pulse noise source filtered with an FIR (finite impulse response) filter. Suppose that the noise consists of discrete pulses (nonzero samples where most samples are zero), and there is an equal probability of a pulse at any sample (so for example the number of pulse...
In 'Digital Filters' by Hamming there is a cryptic section where he describes how the Gibbs phenomenon can be viewed as the displacement between the centers of two functions as they are convolved together. This is on pages 112 - 113 of the 3rd edition. In the process of this he shows that truncating the Fourier Series ...
I am somewhat new to DSP. I'm trying to implement BPSK by following this Python QAM tutorial. My code seems to work, but I really just ported the tutorial's Python code to C++, and do not understand it. This is the part that confuses me the most: # Demodulate in "real" and "imaginary" parts. The "real" part # is the co...
I am pretty new to DSP. I need a computer to transmit data to a receiver through audio signals in a loud outdoor environment. (I am writing the software for both sender and receiver so I can design the signal however I want.) I am using cheap speakers and microphones operating at 44.1Khz. I would like for the sender to...
My understanding of I and Q channels is as follows (please correct me if I am wrong): I = In-phase, or real component Q = Quadrature (90° shift of real component) Where do these two channels come from in the first place? Is one the electric field and the other the magnetic field of a EM wave? I was under the impressi...
One thing puzzles me about the state space representation and that is why is it claimed that such systems are linear and time-invariant but particularly why are they claimed to be linear? The right-hand side of $dx/dt = A x(t) + B u(t)$ does not seem to obey superposition? Is it because the solutions to these systems ...
I have a signal X(n) of length 24. But I want to find 1024 point FFT of this signal. I can do this by padding 1000 0s in x(n). But this requires huge memory which actually will be having only 24 valid data points. Is there any way I can avoid this extra padding and yet be able to perform FFT. To be very specifica I a...
I have a skeleton image. I am able to detect branchpoints, endpoints and able to label image using BWMorph, BWLabel. Now I want to differntiate between skeleton center line and branches. In short I want to detect branches in skeleton. I want to perform operations on skeleton branches and I do not want to consider cente...
my question is quite straight-forward to be grasped and handled. I have an 1D array, F_11 representing a target power spectrum. It is sketched in the following figure Then, I start generating a time series u which embodies the properties of F_11, as follows nn = complex(normrnd(0,1,[1,1000]),normrnd(0,1,[1,1000])); u ...
I have to design 2 IIR bandpass filter with the following specifications: Sampling frequency 1000 Hz Pass-band 50 Hz to 200 Hz Order 6 using bilinear transform and impulse invariance with Butterworth prototype analogue filter. I'm confused, because the two methods I've got for producing the filters have different resul...
My understanding of the scalogram is that, for a particular row, the scores of the projection of the input signal with the wavelet at a particular displacement is shown. Across rows, the same thing applies, but for dilated version of the wavelet. I thought that scalograms can be defined for all types of wavelet transfo...
If I have a average normalized power spectrum of noise is like this $W(f)=N_0$ while frequency is $-B$ to $B$. Why the average normalized power is $1/4 N_0 (2B)$ insteat of $N_0 (2B)$?
Suppose my model is reversible in time (e.g. GPS + accelerometers for a vehicle), so that I can run Kalman filter forwards or backwards. Kalman filter, of course, cannot be symmetric, because it is not casual. But will Kalman smoother give the same results when run forwards and backwards (up to numeric errors)? If no, ...
I need to draw waveforms for biometric data like ECG and EEG signals. When I have more samples than pixels at the X-axis, I need to draw a vertical line between the MIN and MAX sample-value for that period. Now I was wondering, suppose the signal is highly volatile, and the MAX for the next point is lower than the MIN ...
I want to do one of these common experiments to see the effect of increased noise on the performance of a feature detector. To do this add zero mean gaussian noise with fixed sigma to the image and look at the result. To vary the amount of noise I multiply the noise signal by different amounts before adding. Since the ...
There is a Matlab-based package called Chebfun that (among other things) computes polynomial approximations of functions. See here: http://www2.maths.ox.ac.uk/chebfun/. I don't know anything about signal processing, but it appears that this sort of approximation process is the foundation of some kinds of filter design....
I want to use morphology operator for filling holes of a 3d image. The 3d image consists of 116 2-D image slices of the same size (512*512 pixels). I tried using imfill in Matlab for the 3d image but the result is not I want. One of the slices of the 3d input image(a) and the corresponding slice from the 3d output imag...
Please bear with me, I know some of you will scoff but I have looked on Wiki and in various literature (see below) and can't quite get a handle on a few things. I am a general business programmer, in so much as I very rarely perform any complicated maths/physics related programming. Anyway, whilst at Uni and studyin...
I'm generating an 18kHz sine signal using a dsPIC microcontroller at 44100kHz (I'm not sure I can call it sampling rate, maybe generation rate, although the concept is the same). The dsPIC's DAC then feeds an LM386 amplifier and then an 8-ohm speaker. Since the generation method is digital, I get harmonics (36kHz) w...
So, I've just begun a speech and language processing course and have found the explanation of the process of getting the cepstrum of a signal and its properties a little confusing. The following is a description of my current understanding and an explanation of confusion it's causing me: start with the speech signal. ...
I have mp3s of radio talk shows in which the host interviews people on the phone. The host uses a microphone, and the interviewee uses either a landline phone or a cell phone. I want to save just the telephone audio to a different file. Is that feasible?
I have been asked to study an experiment which says "what is the effect of variation of duty cycle of sampling frequency on a reconstructed signal" I am not an electronics/electrical engineer I am CS guy.This experiment probably is a part of Analog to Digital Converion techniques, by now I have been able to understan...
I'd like to ask a theoretical question about power spectra. In MATLAB, by means of a particular approach, I'm trying to generate a Gaussian variable which embodies a particular power spectrum. The theoretical power spectrum has been calculated over an illimited range ([-Inf,Inf]), whereas the simulated one is obtained ...
I'm on histogram equalization process. I have quite well understood the math but i don't understand why an histogram on bimodal image fails in image enhancement and what techniques can be used in this case (some kind of local/adaptive histogram?).
I'm attempting to code Pisarenko's method for some time-series data that I have. I've had a look at a couple of papers, which explain in detail the theory behind it, but I'm having some trouble figuring out how to convert the math into an algorithm for either C or Python. Is there any information on how one can do tha...
I am implementing a digital image correlation algorithm, and I found that for some calculations the variance of the image noise would be very interesting. How can I determine that parameter? The only thing I have are a set of 1000 images of a metal deformation process and no prior information about the noise in the im...
What papers/articles/books are considered good for getting started with wavelets' application to cryptography? I'm primarily interested in irregular grid wavelets, but regular grid will do for starters. What I'm definitely not interested at is steganography. I'm posting this question here instead of Computer Science or...
In my studies of wavelets, there appear to be 3 different families of them: The Continuous wavelet transform The Discrete wavelet transform The Redundant wavelet transform They are all based on the same concept, but vary in how they are shifted/scaled, and/or decimated or not at every scale level. My question is, wh...
I'm looking for a broad classification on the kinds of techniques available. Something I can use to begin a literature survey. Some details: should be feasible for real-time implementation. Binary decision, I should be able to input the signal continuously, and the algo has to detect the start and stop times of the si...
There was an article a few years ago entitled "Analog is Not the Opposite of Digital". Film is not analog, period. I used to shoot film on a Canon AE-1 from the ’70s. Now I have a digital SLR from Canon, and they’re obviously extremely different. But we have to be careful not to confuse ‘old’ and ‘new’, with two very ...
I have following two related questions concerning unit step function. 1-I want to sample the following signal. What will be the sample value at t=0? The signal takes 0 time to change from 0 to 1. So I am confused what should be the sample value at t=0: 0 or 1? 2- This signal is having infinite bandwidth (This s...
How does one quantify how "white" some noise is? Are there any statistical measures, or any other measures (FFTs for example) that can quantify how close to white noise a particular sample is?
Suppose x(t) is a continuous-time signal and x(n) is a discrete time signal obtained by sampling the signal x(t) at a Nyquest sampling frequency Fs. The signal x(t) requires a bandwidth of B1 Hz for faithful transmission through a communication channel, while the signal x(n) requires B2 Hz in the same communication ch...
I've read there are two admissibility criteria for wavelets, both of which are designed to preserve total power of the signal (source: http://en.wikipedia.org/wiki/Wavelet#Mother_wavelet, as well as various scientific papers) Condition for zero-mean: $$\int_{-\infty}^{\infty}\psi(t)dt=0 $$ Condition for square norm on...
Since the list of phonemes in English is fixed, it should be possible to come up with a sample text(s) to collect all possible biphones for text-to-speech synthesis. Does anyone have a sample text for this purpose?
this was a question i found online. given an image with values between [0,1]. how can I get the maximum contrast using linear gray-level transformation? something like I(x,y)=A*I(x,y)+B , where I is a histogram. we are also given MAX(I) and MIN(I). as far as i can understand. MAX(I) and MIN(I) will give you the value...
I have some telephone voice audio with occasional "blips" in the audio. The blips appear to come from an IP link buried in the PSTN (this is a conceptual explanation, so don't worry about things like packet loss concealment). When a packet is dropped, there isn't a gap In the audio, instead, the audio has been splic...
I'm implementing DBPSK (differential binary phase-shift keying) over audio (44.1kHz sample-rate) using software. Eventually I'll use multiple audio frequencies at once, so I'm using FFTs and inverse FFTs. As expected, as the window size (symbol length) gets smaller, ISI becomes more and more of a problem and the BER (...
Q and I seem to come up a lot in discussions of digital communications, such as the comprehensive answer provided to this question: FFT window phase syncronising It seems like the Q and I channels ultimately are combined into a sin wave of a given frequency, phase and amplitude. If the communications system is all sof...
I have been trying to reconstruct a random signal from its PSD and am running into trouble. I know that many different signals in the time or spatial domains can result in the same PSD-- I am interested in any of these signals. I am not the greatest at signal processing, so I've gotten a bit confused. I am reading this...
I've got a 3D image of non-isotropic voxels, to which I'm applying a general rotation. How could I go about determining an appropriate voxel size for the rotated image? I need to minimize loss of information, but avoid too much supersampling to keep the image from getting too big.
I'm trying to create a seamless loop using a "non-periodic" signal using interpolation to smooth out the beginning and the end but I'm still getting a click at the beginning when it loops and I listen to it. Can interpolation do this? Please Note that the equation is done this way to create a "non-periodic" signal to...
im currently comparing 2 signals: A = some music B = A + noise and ive implemented cross correlation in frequency domain via complex multiplication to determine the time offset, which works perfectly so far. Now what i need in addition to the time offset is the amplitude offset, means a faktor by which A should be scal...
I have a video and I want to calculate the SNR for it. I know how to do it with an audio signal, but with a video I don't have a clue. I'm using SCILAB for the analysis.
When the captured image is blurring, one way of obtaining a clear image is via image deconvolution technique. In order to perform deconvolution successfully, usually we need to pay attention to the following issues: Image content, which determines what kind of apriori information you will use (Total Variation, Laplaci...
EDIT: First off let me explain that I understand that frequency makes no sense for a single sample. What I am actually talking about is the power spectrum within a short time window around a specific sample. I have a time varying discrete signal: $\ s(t_i), i=0...199 $ with sampling dt = 10 ms. I want to find the powe...
I have a system that is forced by a sinusoidal function with specific amplitude, phase and frequency. I get a deformation of sinusoidal form as well, and I would like to find the phase lag. I use FFT and isolate the phase lag numerically, it says it is pi/2 however when I look at the plots, it is not pi/2 at all. What ...
I would like to improve a simple Harris corner detector by adding more scales. I don't need scale-invariant interest points, just to detect more points by processing more scales. The Wikipedia says there are two sigmas involved: a derivative scale $\sigma_{D}$ and an integration scale $\sigma_{I}$. So after computing d...