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SIFT algorithm provides a 128 dimensional feature vector that is used for image classification.When all the interest points(key points) are taken together and K-means clustering is applied,the image is then clustered into many clusters.My question is how does this clustering/grouping of feature points together help in ...
I repeatedly see two representations of the general transfer function in the literature. The first is the following which is factorization of the numerator and denominator polynomials: $$\textbf{G}(s) = k \frac{ (s - a_{1})(s - a_{2})(s - a_{3})\ldots }{ (s - b_{1})(s - b_{2})(s - b_{3})(s...
I can not understand the reason why the following sentence is true: If we have a Right-Hand signal(RHS) x(n), X(z), the Z-transform of x(n), has the same number of poles and zeros except at z = infinity can any one explain why!?
If I have a discrete real-time signal $x[n]$, with its latest value $x[i]$ and all its past values $x[i-t]$, how can I estimate the derivative at $x[i]$?
I am currently working on a race car project where steering and braking is to be done by a fully electronic x-by-wire system. To roughly determine the minimum bandwidth of the controllers, I took user + car data from an existing race car. I though I would do an FFT of the input signals to see what the 'user' bandwidth ...
Euclidian distance Cosine similarity Histogram Intersection Or any other? Consider the importance of the method in case of images. Eg, Cosine Similarity matches angle of two vectors... What impact comparing just the angle or distance has on accuracy of BOW? Detailed explanation is welcome!!!
I am trying to sample a sine wave and plot it's frequency components, but I am having problems implementing it.The result of taking 65536 samples of one cycle of a sine wave with max amplitude 1 and a frequency 100 can be seen below.Where the Y-axis this the magnitude of the complex Fourier sum, and the x-axis is the s...
Given the frequency response $H (jω) = csc(ω)$ how would I go about finding the difference equation? I know I can find $h[n]$ from here but after that I feel like I'm missing something.
i am trying to understand how does the image rotation occur? i studied and came to know that we just multiply the image coordinates with rotation matrix which gives us the value of new coordinates. ok that is fine i get this. But main thing which is confusing me is the pixel values. i read that the pixel coordinates wh...
In image processing, how do region growing and clustering differ from each other? There are many papers like clustering based on region growing. I couldn't able to draw a fine line between these two concepts.
I have a raw EEG signal of length 2,50,000 samples. I band passed it from 0.05-10Hz (sampling frequency=500). Then I down sampled it at 20Hz. Now question is: What will be effect if I vary down sampling from 20Hz to say 5Hz or 60Hz? And why 20Hz is appropriate?
I need to know what is iteration and divergence in anisotropic diffusion filter technique. Isotropic diffusion $$\frac{\partial I(x, y, z)}{\partial t}={\rm div}\left[c\cdot \nabla I\left(x, y, z\right)\right], \quad \text{where } c \text{ is the diffusion coefficient}$$ Anisotropic diffusion $$\frac{\partial I(x, y,...
I'm trying to program a spectrum analyser in C++ but I'm faced with a conceptual problem : how to map the results of the FFT to an image ? Let me detail : I'm doing all this in a dynamic way. I have chunks of audio data as float [-1;1] arriving to my class. I put them in a circular buffer which has the same size as m...
I know that all periodic continuous time signal have discrete spectral representations, but are all discrete spectral representations periodic in continuous time? Also, can all periodic signals be represented by a fourier series?
I have a periodic term $V(x) = \sum_K \exp(iKx) V_K$ where $K =2\pi n/a$ where $a$ is the periodicity of the term and $n =0,1,2,3....$ Now I want to find the Fourier coefficient $V_K$ corresponding to a particular $K$. Suppose I have a vector for $V(x)$ having $10000$ points for $x = 0,0.01a,0.02a,...a,1.01a,....2a.....
I'm trying to do time-series prediction with dynamic recurrent networks in Matlab (will try at least NARX and LRN architectures, if not one or two others). The question I have is, what is the functional difference between giving a network multiple inputs, versus giving it an input vector with multiple elements? My inpu...
How does the Non Local Means Filter different from other approaches such as: Gaussian Smoothing. Total Variation Denoising.
I came across a paper that is trying to classify and detect object(s) present in what they call a natural scene data-set. They have images containing objects like cars, bikes and people. Can anyone tell me how this is different from other data-sets that contain the images of cars/bikes/people? The site with the data-se...
I'm finding some difficulties to understand the difference between Filters, and Windowing. accoriding to my understanding both filters and window functions will make changes to the input signal parameters, according to wiki, Filter: a filter is a device or process that removes from a signal some unwanted component or f...
I have three vectors, $x[n]$, $w[n]$, and $z[n]$. I need to compute: $y[n] = z[n] * \Big(x[n] \cdot w[n] \Big)$. This is easy to do. However, is there a way I can do this by 'opening' up the bracket? For example, let us say that I am already given $z[n] * x[n]$. If this was the starting point, what could I do with th...
I have one question concerning phase rotation of signals and I am not sure wether it works or not: Let's assume we have a zero-phase wavelet x(t) i.e. Ricker wavelet. Unfortunately, in our experiment, the wavelet y(t) we use is not zero phase anymore but rotated by an phase angle $\Psi$. Yet, we still can clearly iden...
I have frequency response data. It is output from an acoustic simulator. I convert this into an impulse response, and get: In that picture the blue line is the impulse response. Mostly it is hiding behind the green line (which is the impulse response if I halve the frequency resolution). I'm sorry for an unclear pictu...
I heard from someone online who said: "Only random signal has PSD, determinate signal does not have. For example, period signal does not have PSD." I am very astonishing about this statement, but it seems he is quite sure about his idea. In my opinion, both random signal and determinate signal have PSD, and it is the F...
Assume the following: img = 255*rand(512); %Generating a 2D matrix with random numbers between 0 to 255. fftimg = abs(fft2(img)); %FFT of the image. The frequency transform of this image is a single high value at the top left corner of the image. This means the 'energy' or the frequency response is the highest and co...
I have a continuous signal,xc(t) which is bandlimited to =pi/T , and x[n] = xc(nT) are samples of this signal. I need to show there is a Parseval's relation between the original signal and the sampled version.That is I need to prove that energy will be conserved! Can anyone please help?
I'm trying to obtain the overall LTAS of a large batch of audio (wav) files. To do this, I should obtain the LTAS of each file individually, then take the average value across all LTASs for each frequency. I also need the standard deviation for each frequency. I'm doing this in Matlab, but am rather a noob. Can anyone ...
We often hear that "convolution in time is the same as multiplication in frequency", and vice versa, that "convolution in frequency is the same as multiplication in time". So in a typical windowing operation, we do a point-wise multiplication of a signal $x[n]$, with a window $w[n]$. This means that in the frequency d...
I may have a potential application where maximizing the mode (as opposed to typically minimizing the variance) would be useful for state estimates. The situation may arise from skewed distributions where the mode is of more interest than the mean. I was wondering if anyone has seen a filter like this before? Thank you....
I have a raster image (A) of size 20x20 and I want to scale it down to a size of 10x10 (B). Naturally in the resulting picture B one pixel will represent 4 pixels shaped 2x2 from A. Is it possible to give a canonical answer to how the pixel values of B have to be calculated if no neighbouring pixels (surrounding the 2x...
I am trying to find the fundamental frequency of a bearing fault.This is my code c2=cceps(abs(Signal_fft_hilbert(2:250))); plot(freq_s(2:250),c2) And this is the cepstrum plot in the image. I have plotted the frequency instead of time. Is that right? And I am not able to identify which should be the fundamental freq...
I need to implement a script for generating features from an input image by using the Gabor filter. I have no past experience of wavelets and I'm just learning Fourier analysis (I understand the basic idea behind Fourier analysis and transform) so they can't help me to understand Gabor filter, because I need to have an...
I would like to get the bpm of a song analyzing the spectrum of the volume. Doing a fft what I get is a peak at the origin and of course that can't be the frequency corresponding to the bpm, so I do the following: $\overline{h} = h - \frac{1}{l}\sum_0^l h$ where $h$ is the fft of the volume and $l$ is the length of the...
I have a dataset of images representing human skin. How do I plot the skin color distribution in HSV color space and YCBCR color space. However, I'd like ignore the "value" or "brightness" in both color spaces and analyze rest of the layers. edit The below is the code, I found to plot the distribution for ycbcr but don...
Suppose we have a communication channel that has a bit-error rate of 50% and the bits are flipped at random such that the receiver has no efficient way to tell which bits have been flipped and there are no error correction bits transmitted with the signal. How do you calculate the information loss (in bits) imposed by ...
I'm currently trying to decide what classifier to use for my snore detection algorithm. Which uses some features like Zero cross rating, frequency band energy, first formants... The classifiers which have been pointed to are: Support Vector Machine Naive Bayes Classifier Knn Nearest Neighbour Linear or Quadratic Classi...
In the following question and answer I kind of get it how it is calculated at surface but I don't get it fully. Can someone please explain the answer in depth?
Say for instance, face detection algorithms - I see researchers testing their algorithms and table their results under TP and FP. What are these?
I have a sound source that generates an ultrasonic pulse. I have two microphones seperated by a distance which are at a distance from the source. I want to find the time delay of arrival of the ultrasonic pulse at the mics. I am filtering out all the frequencies other than my source frequency from the outputs of the mi...
I have a doubt. We calculate metrics such as PSNR, SNR, and MSE to prove that the proposed algorithm is better than its counterparts. To do this we need Original image (Noise free), Noisy image (we add this noise) and then Filtered image. Now we compare original and filtered output to estimate these parameters. My ques...
Lets say I have a matrix x=[ 1 2 1 2 1 2 1 2 3 4 5 ]. To look at its histogram, I can do h=hist(x). Now, h with retrieve a matrix consisting only the number of occurrences and does not store the original value to which it occurred. What I want is something like a function which takes a value from x and returns number ...
I am having trouble wrapping my head around digital filters with different orders of numerator and denominator. Let me know if any of these points is wrong: All (digital or analog) transfer functions have the same number of poles and zeros, if you include the ones at infinity. So $H(s) = 1/s$ has a pole at the origi...
So, I am reading a paper about ORB: An Efficient Alternative to SIFT or SURF and it happens to calculate the direction of corner from it center . According to Rosin the author of the paper;moments of a patch is given by and the centroid is of the patch from the center is calculated using: The direction of the patch...
How to calculate the Q value for a high pass filter ? Ps: I'm a beginner in signal processing, so please put it in simple terms.
I know that the frequency response of a filter is described by equation: H(w) = 1-1.176*e^(-jw) + e^(-j2w) , where w is the angular frequency. Then in time domain I have a filter that is described by difference equation: y(k) = x(k)-1.176*x(k-1)+x(k-2) Now I'm asked to explain the connection between these equations an...
Hi I am working on one project and I have some trouble with comparison audio files. I have read a lot of articles about this problem but I couldn't find a good one working. In my project I need to compare two audio events like gun shot, explosion, crying, breaking of glass, ... These files have max 5sec duration but it...
I am working on an audio system where I need to implement an equal power crossfade between two signals over a specified number of samples. Initially I implemented a linear fade but this creates a dip in the middle of the crossfade which I don't want. I tried a few other methods that I found on Google but they either ma...
In the below link: Active Contour http://en.wikipedia.org/wiki/Active_contour_model Internal Energy of the snake is $$E_{internal}=\frac{(\alpha\,\!(s)\left | \mathbf{v}_s(s) \right \vert^2 + \beta\,\!(s)\left | \mathbf{v}_{ss}(s) \right \vert ^2)}{2}$$ Why it is divided by 2 ?
I am stuck in a situation here while calculating received SNR in AWGN channel. When I performed calculations directly in decibels, I got very high SNR value for instance 140 dB. I am using the formula: SNRreceived = [power transmitted(dBm) + FSPL(dB)]-[No(db) * B(hz) + 10log(KT) + NF(dB)] Could anyone point out where...
For a question on my homework, I am given f(t) = e^(-a*abs(t)) and asked to find the Fourier transform of the function. I understand how to do that part, but then the answer requires me to "Compute the amplitude and phase spectra" of the function. My question is, what are these spectra and how do I obtain them from the...
I'm quite new to the subject and am having fun playing around with the FFT. What I am currently doing is trying to sample an audio signal and display its frequency spectrum at the same time. This works alright with what I am doing so far, but there are still a few missing puzzle pieces. This is what I have implemented ...
I am working on bearing faults detection. I am doing the following steps: signal->mean zero->high pass->hilbert->y->fft->cepstrum of y. my code is: analy = hilbert(Signal); y = abs(analy); N = 2*2048;T=N/Fs; sig_f = abs(fft(y(1:N)',N)); sig_n = sig_f/(norm(sig_f)); Signal_fft_hilbert = sig_n;...
I am creating an Android app which records sound for t seconds at a sampling rate of 44.1kHz with a buffer size of 8192 (16 bit mono samples). I need to plot a graph of amplitude against frequency, and so first I need to run an FFT over the buffer each time it is full (which happens multiple times every second, more so...
I am detecting roadline pattern in an image. I have applied delaunay triangulation. I get 160 triangles in that. Now I need to find the area for each triangle. Can you kindly help me. Only after this I should proceed with my project. Please help me...
I am trying to use Matlab to compute the spectrogram of a signal but I can't seem to figure out how Matlab computes the spectrogram.Consider the case where I have a signal x(t) with M samples and I want to compute it's spectrogram using hann windows of length N and using an overlap of K samples. Because there are M sam...
Calculating the coherence (sometimes called magnitude squared coherence) between two signals indicates the presence or lack thereof of a linear transformation between the two signals. Is there an analogue for signals which are related by a non-linear transformation? Let's assume that the expected non-linear transform...
I am student of network security, and very unfamiliar with the domain of signal processing. Do pardon my naivety at asking very basic stuff. My work deals with detecting malicious activity in network logs -say, differentiating between botnet traffic and benign Internet traffic. I came across this interesting article ti...
I am working on bearing faults.I did the high pass filter and then calculated the envelope of the signal in the time domain. I then found its fourier. The fft spectrum is still not clear. My fundamental frequencies are not visible clearly. Do i need to filter it further?
I'm trying to write a simple script that should plot the spectrum in scilab, to test it I use a sinus function with 440hz so that I get my dirac in this position , my problem is that it doesn't work and I don't understand why? here's the code : Fs = 8000; f = 440; t= 0:1/Fs:1; y = sin(2*%pi*f*t); nf = 1024; // numbe...
I am recording a 17khz signal at a sample freq of 44.1khz. I want to perform cross-correlation between the received and transmitted signals for calculating TDOA. But when I do xcorr, the results are inconsistent. Is it because my sample freq is too low for my signal freq? If so, how can I overcome this? Will upsampling...
I referred the below papers to understand the working of Phase Encoding,but the explanation is not proper.Just a few steps have been given in brief. ijcsi.org www.ijarcce.com Can someone please explain the concept of how Audio Steganography using Phase Encoding works?
I have a situation where a sampled signal is passed through a DAC and re-sampled by a switched capacitor circuit. I can't see a model for this in the literature. The standard model of a DAC is the ZOH. But when it gets re-sampled by the switched cap circuit it seems to me that ZOH effect of the DAC disappears because o...
I am trying to get a frequency response vs dB SPL plot for recorded sounds. I have a microphone with a flat response, a sensitivity of 4mV/Pa, and 1 Pa = 94dB. To get the dB SPL, I have the following formula: $dB SPL = 94 + 20 \cdot \log_{10}(AI/0.004)$ where AI in the input voltage from the microphone, and assuming no...
I have an on going input signal, most of the time with no activity. Once in a while there is an activity (oscillations in different freq). I need to take good 205 samples out of the signal. what i mean by good? really oscillating signal for 205 samples (not just crossing threshold). here is an image to clarify: bub...
I have I and Q samples of a time domain signal, sampled at a rate T, both of length L. I create the complex envelope, C = sampleI+j*sampleQ. Then take the FFT of C based on Matlab's FFT help algorithm: Fs=1/T; NFFT = 2^nextpow2(L) Y = fft(C,NFFT)/L fvar = (0:NFFT/2)*(Fs/NFFT) plot(fvar,2*abs(Y)) The frequency content ...
We know that when we window a signal, we increase the main-lobe width. Let 'main-lobe-width' here be the null-to-null bandwidth of the main lobe. Let us further more say that the main-lobe width of a square window is '1'. What I would like to know, is given a particular symmetric window, is there a closed form solution...
first of all, I am not a technician, but I would really appreciate your helping with one (I hope simple or trivial) issue. I recorded a signal (skin conductance level - SCL, uS unit) with my device - 16 bit, range of the measured SCL is 1-20 uS. In a recording software I changed digital upper range - "digital maximum"...
I have a small input signal of 60Hz sine wave from signal generator, which is corrupted with 50Hz mains supply frequency. I want to measure the amplitude of the 60Hz signal using FFT because it is very small to see in the oscilloscope. The Matlab FFT code: y = data; Fs = 2048; [r, L] = size(y); NFFT = 2^nextpow2(L)...
I want to create a project that will recognized the chords that i played in guitar. I recorded a raw sound to be compare to the one that the user will input. Example I have a recorded sound chordA then I will ask the user to play a sound in the guitar and check if the signal is similar to my sound chordA.. Getting the ...
The $\frac{d_1}{d_2}<0.6$ gives a large number of false matches and it assumes the nearest neighbour is a correct match. Anybody knows any good method to reduce number of false matches or filter them ? Can we use k-NN matches instead of $\frac{d_1}{d_2}<0.6$ ratio for matching the SIFT features and then choose the bes...
How can I generate an NxM DCT matrix? In Matlab dctmtx function generates only square matrix (NxN).
I tried the following code for modulation of two analog waves. Can anyone point to how to modulate and demodulate for an analog and a digital wave? I tried by replacing the second signal (x2) with a digital pulse but I am not able to recover the analog wave after demodulation. T = 1; Fs = 44100; t = 0:1/Fs:T; Fc = 1700...
I am reading some paper, and I am having some trouble with some Fourier transform, Suppose that $F(\omega)$ is the Fourier transform of $f(x)$, i.e. where $$F(\omega)=\int_{-\infty}^{\infty}f(x)e^{-i\omega x}\,dx.$$ What is $$\frac{1}{2\pi}\int_{-\infty}^{\infty}\frac{F(\omega-a)}{i\omega}e^{i\omega x}\,d\omega ?$$ I w...
I have project in which I'll work on audio files using Matlab, I'll be listing to different machine and should get thei rotational speeds, which is kind of periodic, my question how should I analyse this should I work with the spectrum and its magnitude using the fft or should I use the PSD , which is the result of p...
There is a recursive way of finding threshold in the image using histogram that are biomodal, and is as follows: An automated procedure for bimodal histograms : Select an initial estimate of T Compute the means of the two regions determined by T Set the new T as the average of the two means Repeat steps 2,3 until th...
let's assume I have two time series: one with a delta peak at position $x$ and another time series with a minimum phase wavelet that starts at time $y$ with a maximum peak at position $z$. Now, when I convolve both time series, is the beginning of my minimum-phase wavelet at position $x$ or is the maximum amplitude of ...
kernel A = [ 0 -1 0 -1 4 -1 0 -1 0 ] and B = [-1 0 -1 0 4 0 -1 0 -1 ] i have found the frequency spectrums with freqz2 function of Matlab. A's spectrum has the highest magnitudes in the corners (1,1),(-1,-1),(1,-1),(-1,1) and B's in (1,0),(0,1),(-1,0),(0,-1). I know that the center of the frequency spectrum has th...
I applied Fourier transform on a sound (PCM generated digitally with sin function) and frequency detection is accurate. However, if I play the sound and record it, and then apply a Fourier transform to detect frequencies, my results are nowhere close to original results. The window size I have chosen for Fourier transf...
After reading this question: PSD (Power spectral density) explanation I am still a little confused as to what extra information the PSD gives us over simply taking the magnitude of the fourier transform. So my understanding is that the PSD is equal to: $$|X(\omega)|^2$$ but what does that tell me that simply $$|X(\omeg...
Lets say we have a picture with 6bits/pixel and we use discrete cosine transform with 8x8 blocks of pixels and each coefficient is 8 bits. We can use 2 8x8 masks, a zonal mask A and a zonal bit allocation mask B. A = [1 1 1 1 1 0 0 0 1 1 1 1 0 0 0 0 1 1 1 0 0 0 0 0 1 1 0 0 0 0 0 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 ........
How do you measure the SNR of a sinewave in the frequency domain? (Assuming no filtering.) Suppose: $$x(n) = s(n) + n(n)$$ Create the signal (nCycles of a sinewave) in MATLAB...: sampleRate = 1024; f0 = sampleRate/8; nCycles = 11; time = 0:1/sampleRate:nCycles/f0; signal = sin(2*pi*f0*time); Create the noise and sc...
This is an extention to my other question. My sampling frequency is only about 2x my tone frequency. I want to estimate time delay of arrival using this signal. I am generating the tone signal so I know its properties. Can I use compressed sensing to reconstruct my tone wave received by the sensors and get a good time ...
I need to generated a high frequency sinusoidal signal for modulation in MATLAB. But it turns out to be something weird. This is the simple code snippet I used... t = 0:0.001:100; A = 1; s = A*sin(2*pi*1e9.*t); %1 GHz signal plot(t, s); title('Modulating Signal'); xlabel('Time'); ylabel('Amplitude'); Instead of the s...
I am trying to learn signal processing and digital pre-distortion. http://www.altera.com/end-markets/wireless/advanced-dsp/predistortion/wir-digital-predistortion.html I am currently trying to auto correlate the input signal and from this I make a matrix. I then get the coefficients for this matrix and use them with a ...
I have two cameras, and I'm up to perform some matching stuff on videos captured by these two cameras since I'm a CV guy. The problem is that two cameras have very different performance in terms of color and brightness. The same object in the same illumination appears different in two cameras. Is there any academic wor...
I have a signal that contains two components. How can I separate it two its two components in matlab? Please note that the location and shape of components are different in each experiment and we do a lot of experiments. It should automatically detect the two components based on the frequency contents. I have added thi...
How can I do point by point multiplication of the quantized DCT coefficients with the quantization matrix before computing the inverse DCT for each block? I want to modify the reconstruction algorithm to approximately undo the quantization step shown here: for i = 1:Blocks for j=1:Blocks irange = (i-1)*8+1:...
I'm coding a phase vocoder in C for time-stretching audio signals. The only piece that is currently not working is the phase propagation. I take overlapping frames from an input signal, apply a window function, and perform FFT. Then i calculate the phase propagation (the one I'm stuck with) Finally I perform iFFT and ...
This is the logistic distribution of single random variable (taken from Wikipedia). x = random variable mu = mean of all random variables s = variance. Now, I want to do a Bivariate logistic distribution (having two random variables x1 and x2). My dataset is going to be image pixel values! When I find the covariance o...
I want to linearize a system to this form $$\begin{bmatrix} \Delta\dot{x}_1\\ \Delta\dot{x}_2\\ \Delta\dot{x}_3 \end{bmatrix} = A\begin{bmatrix}\Delta x_1\\ \Delta x_2\\ \Delta x_3\end{bmatrix}+B\begin{bmatrix} \Delta u\end{bmatrix}$$ $$\begin{bmatrix} \Delta y \end{bmatrix} = C\begin{bmatrix}\Delta x...
I got this simple Discrete time signal question on which i have to perform shifting and multiplication.But I have some Doubt regarding the answer this book shows. Please see the uploaded image.
I have no background in signal processing but I would like to know what is the actual state of the art for vocal elimination/reduction. Obviously, I am not talking about the infamous "stereo elimination", but other, more sophisticated techniques that not everybody is aware of. I understand that for many projects there ...
I have a noisy signal file in Matlab and I have to denoise the polluted signal using a discrete Fourier transform. I'm asked to perform the fourier transform, then take its absolute value. Then study/examine the absolute values to then implement a low-pass filter for the actual sound (and corresponding high-pass filter...
I have a project of recording human's sound via Android's SoundRecoder API, and then displaying the extracted note on The screen. It is specific for Android platform (can't use Jmusic or Jfugue API as javaSE environment is not supported). Now suppose I get to record a sound using this API . How can I then deploy FFT to...
I have this simple $x(t)$ graph, and I have some confusion regarding time shifting. What is the graph of $x(1-t)$ ?
Can you explain to someone with no mathematical background why is deconvolution used in telescopes, CT scans and seismographs? How do you explain that the same technique can be used for different things. The distortions in telescopes, CT scans and seismographs are all different from each other.
I have this wav file that I recorded using my laptop, It contains the sound of a G chord of a guitar. Since the guitar have 6 strings the sound should contain different frequencies I want to determine the top 10 most dominant frequency in the signal. [X, Fs] = wavread('chordG.wav'); freq = fft(X); plot(abs(freq)): I d...
I am trying to plot short time energy of an utterance with percent overlap as an input variable. As I referred to the short time energy equation it can be represented as square summation of $x(n)*w(n-m)$ where $w(n)$ seems to be window length. But I couldn't figure out how to write it as percent overlap.
I have to come up with an upsampling algorithm for a project and cannot use Matlab's built in upsample function. I came up with the following function but it just runs forever when I use a 13 second, 16-bit song sampled at 44.1kHz. Please let me know how I can improve the efficiency of this algorithm: x = desired_...
I need to identify two systems. Those are their Unit Step responses: I don't know, what order of the system that is. It could be second order system with zero in 0. Lecturer told me it was third order system, but then told me to solve it as second order system, so I'm a bit confused. More over we don't have materials...
I have two audio signal that I want to compare using Matlab, my problem is that I can hear the difference between both loud and clear, but when use the function pwelch to compare their PSD ,I don't see much difference between them,any idea how can I compare them with a different methode ! thanks in advance !