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I am currently working on a project which involves measuring the sound produced by the onset of a vapour bubble in boiling water. All the experiments went perfectly well and I'm now trying to obtain a frequency spectrum from the signal. The sound-pulses produced by the bubbles only last about 40 ms max, simply too sho...
I'm using Octave and trying to use the "instfreq" function from the time frequency toolbox which requires an analytic signal input. To get this analytic signal I'm using the Octave function "hilbert" on my real valued signal thus: analytic_sig = hilbert( my_real_valued_signal ) ; [ instf, t ] = instfreq( analytic_signa...
I created a white noise signal. I am trying to use the array I have created with the white noise to play in 2 channels. When the left channel I was trying to have 5000 samples and the right will have 0 and when in the right channel it will have 5000 samples and the left will be 0(which means no sound from this channel)...
I have signal, periodic with amplitude 10 with frequency about 10 kHz. It is hidden in Gaussian noise with standard deviation 100. Some idea how to extract periodic signal? Thanks in advance.
I am trying to understand the usage of Hermitian symmetry in OFDM systems and have a couple of questions regarding this. What is the reason of using the Hermitian symmetry in OFDM? How can we arrange the data in terms of Hermitian symmetry? In OFDM system where do we arrange our data as Hermitian symmetry? Is there an...
I was looking at performing a Stockwell transform on a matrix (or an image as it has two directions - the matrix I am interested in has a spatial dimension and a time dimension, but this should make no difference) instead of an array. Whilst there are a couple of codes that perform a Stockwell transform on an array (se...
I want to find the fundamental frequency for human voice in an Android Application. I want to use autocorrelation method. My code is this: public double calculateFFT(double[] signal) { final int mNumberOfFFTPoints =1024; double[] magnitude = new double[mNumberOfFFTPoints/2]; DoubleFFT_1D fft = ...
I know this topic is not clearly relevant to the forum, but I think it's fun and interesting. I'm looking for computer vision riddles - fun and interesting questions that have a clear and precise solution. For example the following: We have a binary image with paths (contours) that are one pixel wide. Come up with an a...
By Gabor uncertainty, I mean the principle of uncertainty as applied to signals — with the result that you can't have arbitrary time and frequency localization. By way of background to my question, I wrote about this recently in a blog post. In the last paragraph, I stated: Signals do not have arbitrarily precise tim...
Can I assume that the R.O.C. from a Z-Transform will always start from or end in a pole or 0/infinity? Using an example: $$H(z)=\frac{(z+1)(z-1)(z+j)(z-j)}{z^4}\space,\space\space j=\sqrt{(-1)}$$ So we have 4 zeros in the unit circle equally spaced by a $\frac{\pi}{2}$ angle and 4 poles in the origin ($z=0$). If not sp...
Good Morning! I am new in signal processing and I am trying to do a work in noise control of an electronic steering lock device (ESL). My aim is to calculate the loudness (Zwicker Method- ISO 532 B) of this device. To do so, first I need to obtain the 1/3 octave spectrum of a time signal that I measure with a microphon...
My input signal is phase vector. I want to differentiate it to get frequency vector. My input signal is somewhat noisy. Here is the input signal. This is the derivative of the input signal as calculated using diff(inputSig) When i differentiate this signal, I am not getting a smooth curve. The output looks 'spikey...
I'm currently working on a library to generate synthetic fingerprints using the SFinGe method (by Maltoni, Maio and Cappelli) link :http://biolab.csr.unibo.it/research.asp?organize=Activities&select=&selObj=12&pathSubj=111%7C%7C12& One of the steps requires me to apply different gabor filters to an image, each pixel in...
I am trying to recover my baseband signal that was AM modulated. I am using the filter command in MATLAB to perform the low pass filtering ...the code is given below: Fsig=10; Fc=100; Fsamp=10.*Fc; t = 0:1/Fsamp:2*pi; Tx = cos(2.*pi.*Fsig.*t); % Transmitted Signal Carrier ...
Please Help, I have a white noise signal that I created and need help to apply the Short-time fourier transform to my code so that it can do the bandpass filtering. I am trying to put it in the code so that I don't have to use the FDA tool. Also be able to plot the graphs from the output of the STFT being applied. L = ...
Say I have an exponential smoothing for certain $\Delta t$, $t_{i+1} = t_i + \Delta t$. In this sampling, I choose a particular $\alpha$ to filter signal $z_i$ like $$ v_1 = z_0 \\ v_{i+1} = \alpha\:z_i + (1-\alpha)\:v_i $$ and we have some sort of agreement as to what $\alpha$ to use. Now, someone comes along and te...
I'm trying to generate sounds as described in a scientific paper that I found on the Internet. What I need to do is to generate simple tones in a certain pattern, however I have some issues understanding how to do this correctly. The attached image contains the full description of the method. I'm stuck at "Thirteen to...
I have two signals sampled at $100\textrm{ Hz}$ during $3\textrm{ seconds}$. So I have $300$ sample points at each signal series. I want to calculate coherence values between these two time series and would like to know whether this coherence value is statistically significant. I found two resources online. In the f...
i know that texture refers to surface characteristics and appearance of an object given by the size, shape, density, arrangement, proportion of its elementary parts. But i am more interested in dynamic textures. Can anyone give me information about what are Dynamic Textures? also in papers it is written that they ex...
I am trying to design a low pass filter to apply before resampling the sensor's output at a lower rate, however I am not quite sure how to choose between the IIR and FIR fitlers. Matlab by default, filters the data with an 8th order Chebyshev Type I lowpass filter with cutoff frequency .8*(Fs/2)/R, before resampling. b...
What is the transfer function of a line, can i ignore it? After simplification i got the diagram: I don't know what to do with the empty feedback
How do I set the frequency for a low pass filter illustrated below?
This is the first time I'm posting a question on here so I hope I'm doing it right. I've tried searching on here for an answer but haven't found anything particularly relevant. I have some data collected from a line array of 32 hydrophones. I'm trying to estimate the power spectral density of the underwater noise at di...
Assume I have a random process $X(t)$ that is white gaussian noise with psd $S(f)$. Now, let's filter that noise through a LPF of bandwidth $B$. How would I evaluate the following expression: $$E[S(f_1)S(f_2)]$$
I am trying to use the example from here to detect a stop sign. But when I change the image to search for stop signs in there are a huge number of false positives. Am I doing something wrong?
This is an example from my text book of a continuous signal: $$x_{in}(t)=\sin \left( 2\pi \cdot 1000 \cdot t\right) + 0.5\sin \left( 2\pi \cdot 2000 \cdot t + \dfrac{3\pi}{4} \right) $$ So to perform a fourier transform on this signal, how to do that, isn't it a bit funny, since it has two sine components. Shouldn't co...
I have read many articles about DTFT and DFT but am not able to discern the difference between the two except for a few visible things like DTFT goes till infinity while DFT is only till $N-1$. Can anyone please explain the difference and when to use what? Wiki says The DFT differs from the discrete-time Fourier trans...
Can FFTW perform the FFT on data which is not uniform in time? I can't seem to find a straight-forward answer to this question anywhere online.
I am aware that this a simple question, but I haven't been able to find a single reference that explains this situation. The system function is: $$\frac{1}{1-(0.3)(1+j)D}$$ How can one draw a block diagram with only real components to represent something that is complex? The system function is $$\frac{Y}{x} = \frac{1}...
Suppose I have two time series $A:=[0,0,0,4,5,6]$ and $B:=[1,2,3,4,5,6]$. I implement the following filter (with initial value at the first element): $$ f(t) = f(t-1) + \alpha*(f(t)-f(t-1))$$ with $\alpha \in [0,1]$. Is there any way I can use filtered values of $B$ to recover the filtered values of $A$? For example su...
I am trying to calculate the autocorrelation function for the telegraph process, but I somehow don't get the right results. I am calculating the correlation function of a signal X which can take values 0 and 1 by calculating the spectral density $S(\omega)=|FFT(X)|^2$. The autocorrelation function is then given by the ...
I understand how to relate bin numbers of FFT output to frequency when dealing with a uniform FFT. How does this process work with a non-uniform FFT?
I have an difference image produced by substracting two images of a certain location taken a two different time steps. The resulting image of this operation shows a histogram with a very large peak centered a 0. Since this naturally can be identified with no-change between dates I interpret this as a difference image s...
Let us say i have some digital data placed in a wav file by lpcm method. At a sampling rate of 44.1khz i can place 5.5kb of data per second of audio by choosing two amplitudes (quantas) for each sample: 0 and 255. I can now save this audio file on an analog media and can retrieve the original digital data easily later....
Might be a simple question, but I keep second guessing my algorithm, so a confirmation would be good. I first do bandpass filtering, then PCA, then FFT of the top principle components. I also do some sine wave fitting on the filtered data, and on the principle components, but it doesnt make sense to do it on the fft da...
I edited the initial question to make it general and thus applicable in more situations. If the Fourier Transform of $x(t^2)$ is some function of $\omega^2$. Then what is the Fourier Transform of $x(bt^2)$, for $b>0$? Is it $\dfrac{1}{b}$ times a function of $\left(\dfrac{\omega}{b}\right)^2$, or $\left(\dfrac{1}{\sqr...
I am trying to figure out speech features that can be inverted to recover the audio back with no loss or as less loss as possible. I am currently using the power spectrum which is completely reversible. I also read about MFCC and LPC as potential features but both of them seem to be lossy in terms of recovering the aud...
I wanted to know the meaning of "deep zeros" in the spectrum of the input data. I came across this terminology while going through the introduction of chapter 2 (Bussgang techniques for blind deconvolution and equalization) of the book named "Blind Deconvolution" by Simon Haykin. Thanks, JK
I am beginner in the image processing I want to know what is difference between Texture and Image? or whether they are same? can anyone explain it. Also,if i want to do texture segmentation,can i use same algorithms as applicable to the image segmentation?
I am reading a paper on frequency warping and I need to do a little manipulation of the Z-transform. Can somebody help me on how can I go about deriving equations $(3)$ and $(4)$ from equations $(1)$ and $(2)$. $$ \tag{1} \bar{A}(z) = \sum_{k=0}^{p}\bar{a}_k\bar{z}^{-k}(z), \ \ \ \ \ \ \bar{a}_0=1 $$ $$ \tag{2} \bar{z}...
I've been using a Gaussian Filter and know how to use it but I don't understand the physical significance behind using the filter. We use it but why? What is the significance? I'm just starting out in image processing, so please no high level responses. Thanks!
I was wondering how to calculate the autocorrelation of a deterministic signal $x(t)$ multiplied by a stochastic process $M(t)$, whose autocorellation $R_M(\tau)$ is known a priori. In my case, $x(t)$ is a truncated monolateral exponentially decaying function. I suppose that the result of such multiplication $y(t) = x...
I am working in image processing. I am confused while getting into wavelet. I applied the fft2() function on Lena image and I got frequency information for the same. But when I applied bior4.4, a popular bi-orthogonal wavelet, I got edge information HORIZONTAL, VERTICAL and DIAGONAL orientation. May you plz. tell me ho...
in my attempts to synthesize a plucked electric guitar string (using supercollider's digital waveguide, which is based on the Karplus-Strong algorithm) I came to a dead end and decided to look that the Cepstrums of a sampled string and my synthesized string, to nail down the audible differences. They look like this (A-...
I attempted to implement a real-time audio filtering in a callback function using the simplest moving-average filter with coefficients [0.5 0.5] : void process(void* in_, void* out_, DWORD length, void* user) { //data is float, mono int n = length/sizeof(float); float* in = (float*)in_; float* out = (f...
An excellent answer to this post explains how to do $1/n$ octave energy smoothing and mentions complex smoothing can be done as well, but it's tricky business because of phase wrapping. How is phase wrapping solved in $1/n$ complex smoothing?
I'm reading this book "Applied Optimal Estimation" by A. Gelb. In the example 1.0-1, there are two measurements $z_{1}$ and $z_{2}$ with some noise for measuring $x$ as following $$ z_{1} = x + v_{1} \\ z_{2} = x + v_{2} $$ Now to design estimator for $x$, we need the estimated and the actual value of $x$, so we defin...
If a digital LTI filter has phase response on the form $arg[H(e^{jw})] = -\alpha\omega$, what could one say about it's causality (and consequently, the causality of it's impulse response)? Also is Linear Phase a characteristic of only Causal FIR Symmetric Systems?
Services like Siri and Google Now perform speech recognition (speech-to-text) on mobile phones by doing part of the computation on a server in the cloud. What do they upload to the cloud? Do they upload the raw audio? Or do they do some feature extraction and only upload features (e.g., MFCC coefficients or something...
I am using a DSP kit TMS320c6711. It has a fixed sampling rate of 8000 samples per second. As I have to do real time filtering of frequencies above 5oHz with sharp cut off, I must reduce the sampling frequency to around 1000 samples per second. Suggestions are highly appreciable.
I'm currently trying to perform analysis of a spectrogram, with the use of PCA. I'm confused about what the dimensions of the result should therefore be. Currently, the dimensions are: 451x128. But I thought that it was a standard image, and therefore it should have 2 dimensions.
Hough transform is a very powerful tool to find dominant straight lines in a black-and-white image. A very common practice of analyzing Hough transform result is to find some local maximum points in the Hough transform image, and then try to analyze the straight line segments in the image domain that correspond to the ...
My code works in the following manner: 1.First, it obtains several images from the training set 2.After loading these images, we find the normalized faces,mean face and perform several calculation. 3.Next, we ask for the name of an image we want to recognize 4.We then project the input image into the eigenspace, and ba...
I should say which of the filters from the following image implements an ideal low-pass filter with cutoff frequency $\frac{\pi}{4}$. In the picture, Sa is a zero insertion filter, which inserts a zero sample after each input sample and Sb is a decimator, which extracts every second sample of its input. The first filte...
I am analyzing the spectral components of a time series using the continuous wavelet transform following Torrence and Compo (1998). I would like to partition the signal variability or spectral power across different scales (e.g. 50% of the variability is in the scales X to Y). As pointed out here the global wavelet sp...
I need to calculate $\frac{E\left[|x(n)|^4\right]}{E\left[x^2(n)\right]}$ in real time, where $x(n)$ is complex. For $E\left[|x(n)|^4\right]$ I plan to take the absolute, raise it to the power $4$ and do exponential averaging with weights $0.999$ and $0.001$. I have no clue on how to calculate $E\left[x^2(n)\right]$. ...
Can someone help me with this interview question? It is about the sampling frequency rate. Under what condition the available bandwidth is $f_s$ instead of $\frac{f_s}{2}$?
I'm experimenting with image resizing techniques and algorithms. Specifically, I'm significantly downsizing images, e. g. from 2048x1536 to 64x48 - 32 times. Now, say I'm using a 4x4 kernel. Right now I'm upscaling the image to be by a factor of power of 4 larger than the target (2048x1536 becomes 4096x3072), and then ...
I am currently reading SEGMENTATION AND RECOGNITION OF SYMBOLS WITHIN HANDWRITTEN MATHEMATICAL EXPRESSIONS. There is one sentence I don't understand: The preprocessing stage is subdivided into three steps: smoothing the input data by lowpass-filtering. In this context your data usually looks like this: [[(x1, y1),...
Say I've got a signal that I know is composed of periodic components. However, I'm worried that there are "spiky" events that are embedded in this signal, and potentially occurring periodically. I can't just look at the power spectrum of the signal, because the spiky event will show up across many frequencies in the po...
I wish to find the maximum frequency of $\sin(\varphi(t))$, where $\varphi(t)=u(t)$. We have $\omega(t)=d\varphi(t)/dt=\delta(t)$. What is the the maximum instantaneous frequency $\omega_{\max}$?
I'm new to dsp.stackexchange and digital signal processing. I'm required to develop an app to measure the similarity of two signals.for an example I need to compare breathing of a man whether it is abnormal or normal. I have recorded two breathing signals. Relative error of two signals I have compared two signals and o...
I have an exercise that gives me the transfer function of a system $$H(s) = \frac{3s^2+27}{s^4+8s^3 + 16s^2} $$ and an input $$x(t) = \frac13 cos(3t) $$ An ask's what is the output I don't whant the answer but the steps to calculate it. Where to start? I need to find the inverse or there's a shorcut?
I have 2 satellite transponders closely situated, transmitting the same signal. At the receiver I receive this composite signal. I have to equalize it in such a way that the demodulator sees only one of the two signals from the composite signal. So, I modelled this situation as a channel with input $x(n)$ given to a ch...
0 down vote favorite I am totally new to Image processing. I just started my Master thesis in the area of Computer vision and Machine learning. My background is Informatics. Now, my first task is to register the images of some fishes(Image registration) as they come out of water. I have got stream of images and I want ...
I very simply have a .wav audio file, but there is a note with it that says "This audio has been sped up by 10x". I am wondering, what does it mean exactly from a signal processing perspective for an audio signal to be 'sped up' by some factor, and secondly, how can I undo this? Is it possible?
I'm searching for a method that can smooth a 3D volume whilst preserving the edges in my volume. I researched anisotropic diffusion filtering and bilateral filtering, but I'm having trouble to evaluate if they can be efficiently used on +30M voxels volumes. So I need: Relatively fast algorithm even with 512x512x200 vo...
I understand, that image padding is necessary in order to prevent wrap-around effects due to circular convolution. My set of questions are related to different padding schemes. Zero Padding is easy to visualize in the context of reducing wrap-around effects. In the case of schemes like mirror reflection, I have a dou...
I'm learning some basics of image processing. Recently I've read about image filtering and two-dimensional Fourier transform, because I'm preparing for exam. And I have one question I don't know answer for: is it possible to do filtering in frequency domain (by using Fourier transform), which in spatial domain is imple...
Sorry if this is too damned long. I did what I could to abbreviate it. The question is about Least Angle Regression (LARS). I'm new to numerical work with matrices. I believe I have a way to compute Least Angle without explicit matrix inversion. I'm hoping someone with more experience can check my work to see if I'm do...
Given I have: a camera with estimated rigid motion 3x4 matrix $P = [R|t]$ that starts from the origin of the reference coordinate system an intrinsics 3x4 projection matrix $K$ a 3D point $X$ How would I find its projection in the second camera/image? Is the following correct? $x = K*[R|t]*X$
This is now a second time I am attempting to ask this very important but simple question here. What I want to know is can you do deconvolution by convolving a signal. It is often stated that, for example by cutting and boosting the same frequency on an equalizer the result is the original signal. Is that the case? Can ...
Are transmit diversity signal and the multi-path signal same? Multi-path contains LOS+delayed and attenuated/distorted versions of it and transmit diversity signal contains LOS1+delayed LOS2+ +delayed and attenuated/distorted versions of the two. Kindly, elucidate in terms of the impact of each type of signal on the eq...
I am trying to understand the basics of the cross/auto-correlation if the noisy signal $y(t)$ which is received from the channel at the receiver. Here at receiver we want to estimate the noise in the received signal and then get a clear signal of $y(t)$. My questions are below: How autocorrelation or cross-correlatio...
know that dynamic textures are spatially repetitive and temporally varying patterns. 1.I read one sentence in one paper i.e. They are sequences of frames exhibiting certain stationarity properties over time. They exhibit spatial self-similarity and temporal self-similarity. but i don't understand how it does? 2.Also ho...
I have $1200$ face images in my training set. There are $2989$ test face images. I am using eigenfaces (PCA) for feature extraction and $k$-means clustering. I even tried all $2989$ test face images as training set images, but no accuracy is achieved. Source code i tried: tic clear all close all clc % number of images...
I am trying to segment and label white blood cells from this image: Using K-Means I managed to segment most of what I wanted: But as you can see, there are some cells connected to each other which I would like to label as separate objects. So my questions is: Can you point me to some known algorithms that perform the...
I have a set of objects, which may be spherical or elongated, which may or may not have a tail like structure attached to one or two ends. The tail may be long or short. It may also have a curve to it. I'm currently looking over the information returned by regionprops and trying to see if anything jumps out at me which...
Here goes the problem I faced: Sig1= A*sin(w*n) Sig2= B*sin(w*n+phi) Length of both signal is L. Then, I put them in F(1:2:2*L)=Sig1 and F(2:2:2*L)=Sig2. Now we know what would be the frequency response of Sig1 and Sig2 but is there any way we can find frequency response of F? FFT of sinusoid is the delta function. Th...
In many applications such as estimation theory, when we need to estimate a parameter then we usually consider in presence of white gaussian noise of zero mean and some standard deviation. During Maximum likelihood estimation, we also use this assumption. So, my question is - Do we consider noise to be uncorrelated or ...
I am calling fspecial() in MATLAB to create a Guassian blur of an image. For example, %Create Gaussian Filter G = fspecial('gaussian', [5 5], 2); % Blur Image blurredImage = imfilter(nonBlurredImage,G,'same') The parameter [5 5] is hsize which defines the extent to which the Gaussian filter is applied ... which accor...
I want to know the characteristic differences between real and complex impulse responses. Thanks, JK
Some introductory questions: What is the unit of the group delay in a digital filter (is it samples?) What is $\Omega$ in the equations below (is it $\Omega=2 \pi\frac{f}{f_{sample}} ?) $ If I got any of these wrong, then there is no point reading the rest of this question. Just correct me. What I want to achieve I a...
In the development of Hilbert transform relationships, Prof. Oppenheim has chosen \begin{equation} \int_{-\pi}^{\pi}X_R\left(e^{j\theta}\right)\sum_{k=-\infty}^{\infty}\delta(\omega-\theta-2\pi{}k)d\theta =X_R\left(e^{j\omega}\right) \end{equation} in his book "Discrete-Time Signal Processing, 2e" Chapter 11 pp807, Eq....
It seems that the most advanced pure mathematics course most EE engineers take is Fourier analysis, and after that it's basically 'applied' courses. There's probably a good reason for this, but I'm not sure what it is. Functional analysis seems like it pops up all the time in signal processing. Would exploring function...
I am processing data which was put through a low pass filter at some point. Looking at the FFT, I can see that it is less than the Nyquist rate. I am trying to filter another set of data so that the spectral content will be better matched. Is there a good way to calculate the cutoff frequency or bandwidth of data? I kn...
Kevin Murphy's Kalman Filter toolbox (for Matlab) contains an example where it's the fact that the state space system in not identifiable causes problems. I include the example in it's entirety but you won't actually be able to run the code without installing the toolbox first. See the comments between the code blocks ...
I mean, does exists some state of the art algorithm able to extract music features, classify them and identify similar music or find lyrics? I know that there are many topics or steps involved on my question, like digital signal processing, feature extraction, machine learning, pattern recognition, etc. I'm a computer ...
Currently, I'm working on removing 50 Hz power line interference from an ECG signal. Before trying out notch filters, I decided to try out a simple lowpass filter with a cutoff less than 50 Hz. Here's the MATLAB code I used to make the filter: Fs=500; %Sampling rate in Hz Ast = 120; %Stopband attenuation Ap = 1; %Passb...
OK, I have tried to ask the same question in the math site on stack exchange but no real joy. Apologises I'd this is the wrong forum. I am trying to compute the PCA of a spectrogram.. It is 486x128 after I compute the covariance matrix, the resulting matrix is a square matrix containing 128x128 dimensions. Now, I want ...
How to get the conjugate symmetric of the complex matrix using matlab
I perform FFT of two real sequences using "two for the price of one" - array $x[n]$ and $y[n]$ is packed as a complex number - $z[n] = x[n] + j y[n]$ - using FFT2 and FFTW. I am able to decode the half-complex output format of FFTW and the result match with the output of FFT2. The output conversion goes like this. If F...
I'm trying to study a nonlinear system by sending a pulsed signal into the system and look at the response spectrum. The signal I send to the system is something like and the response spectrum from I get from the system is something like (note the spectrum is plotted log scale, ie, (frequency, log of absolute value of...
Brief introduction: I am working on my university project. My task is to develop application that captures sound from mobile phone and removes/reduces noise (this simpilied version of task). I never faced with DSP before, I spend a lot of time to figure out how to get audio signal from phone (no direct API is porvided)...
I'm trying to retrieve a melody from a signal (simple sine with 3 notes). I've windowed the signal and performed autocorrelation to get the fundamental frequency of each window and then converted frequency to note. So for a signal with notes "G A E" I am getting something like "G G G A A B E E E E F". It makes sense t...
i am working on object tracking. i want to use background subtraction. i have tried to find out papers on background subtraction but i am really confused to use which one. I have already done tracking using just gray scale subtraction and thresholding. but now i want to model the background and then use it for tracking...
I've been trying to solve the following problem, but I noticed that I get two different answers by means of two different techniques: Find the impulse response, $h(n)$, of the following system, described in frequency: $Y(e^{j\omega}) = e^{-j\omega}X(e^{j\omega}) + \frac{dX(e^{j\omega})}{d\omega}$ Try #1: I found the di...
Given a function $x(t)$ and a first order low-pass filter $\frac{dy}{dt} = \frac{1}{\alpha} (x(t) - y(t))$, is there a way to choose $\alpha$ analytically such that the derivative of $y(t)$ is bounded, i.e. $\forall t \frac{dy}{dt} \leq c$ for some given constant $c$? Thanks for your help. Note that this question is si...
Let’s assume that we have two audio signals, x(t) and y(t) affected by the noise as shown below. And we would like to cross-correlate these two signals and the cross-correlation plot is shown as below. In this correlation plot there is a peak value around -11 msec. I am trying to understand how we interpret this p...
I am learning image processing. I want to ask very basic question related to FFT topic Which information do we actually get from "phase spectrum" and "magnitude spectrum" about an image?