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let us consider following code clear all; clc; f1=10; f2=40; fs=100; ts=1/fs; t=0:ts:2.93; x=19*sin(2*pi*f1*t).*((t<0.25)+(t>1))+20*cos(2*pi*f2*t).*((t>=0.25)+(t<1))+1.5*randn(size(t)); plot(t,x); axis tight title('Signal'); xlabel('Time or Space'); output of plot is given ...
I use FFT to do filtering in frequency domain, but when I use big kernel I got shift on border, I think it's due to nature of cyclical convolution and I need to do more zero padding. But how to calculate required padding size? should it be power of 2 in matlab? Here is my matlab code: tic I= rgb2gray(imread('im.jpg')...
We know that Fourier Transform of a signal exists if it is absolutely integrable and it exists for periodic signals if impulse functions are allowed. If we consider the fourier transform of $\text{rect}(t)$ , we get $\text{sinc}(f)$ in frequency domain. Which has got zero frequency component as $1$. But we all know, D...
Should I apply some window before FFT if I want to find 5 frequencies in signal? Signal is pure 5 sinusoids + some noise generated during recording on analog cable.
The Shepp-Logan phantom is a standard test image used in reconstruction algorithms as well as image denoising. Does an equivalent image sequence exist for testing video denoising algorithms? I know that there are plenty of "natural" videos out there (one of two table tennis players springs to mind), but I'm looking fo...
I'm trying to plot 2D gaussian function using "imshow" in MATLAB. However I could not see any white points in image. I tried to change sampling values and limits of x and y values. In addition to this, I tried z.^2 and higher orders but nothing changed.Thank you for any help.What can be the problem? My code is below: s...
in regular discrete-time (1 dimensional) kalman filter, it is assumed that we have white gaussian noise affecting the transitions and the observations: $x(t+1) = Ax + w$ $y(t) = Cx(t) + v$ assuming $w$ follows Gaussian(0, $\sigma_1$) and $v$ follows Gaussian(0, $\sigma_2$) distributions. will the standard kalman fil...
this thread asks when a discrete time Kalman filter is better/different from a simple moving average of the observations: Why use a Kalman filter instead of keeping a running average? there's no definitive answer. can someone give a definitive example where the kalman filter, ideally in simple 1D case, does something d...
I have to make a project college. I have to come with some application of signals that I can implement in MATLAB. Is it possible to separate two voices from an audio signal, say a .waw file ? I mean, using techniques from a first course in digital signal processing (DFT, Spectogram, Cepstrum).
Is there a simple technique to watermark .wav audio files? So that a third person that receives them could check the authencity of the file. I'm trying to come with ideas for a school project. So I should mainly use techniques from a first course in digital signal processing (DFT, Spectogram, Cepstrum).
I am writing an open source microphone array simulation program:SharpEar. When I give uniform weight to microphone array elements, the result in the angle is a sinc function as expected. There are 41 elements in array, all equally weighted. Room is 20*20 meters. Red point in the picture below is a 1000Hz sin wave. ...
I am trying to answer the following question: Is the system described by equation: $$y[n]=0.5y[n-1]+x[n]-0.5x[n-1]$$ an IIR filter? My answer is yes. Thank you
I assume this is a relatively simple DSP task, but I'm having trouble finding any information on how to approach my problem. Hopefully someone with more experience and brain power can help me out: I have many (thousands) audio records of bird calls. I am looking for a technique that will identify the timestamps withi...
I have to do FFT in MATLAB on the digital sinusoidal data that is stored in excel. please find the amplitude values of the sine wave discrete signal stored in the Lookup table. Number of points : 256 Maximum amplitude : 1024 Numbers per row: 1 Numbers are in decimal 512, 525, 537, 550, 562, 575, 587, 600, 612, 624, 6...
Hi all Is there a generally accepted rule as to which order to perform filtering in? for example if I have three iir filters, a low pass filter a notch and a high pass filter, which sequence would it be best to perform these in? or does it not make any difference at all? cheers, Graham.
I know that frequency mixer $y(t)=x(t)cos(\omega_ct)$ is a nonlinear system since its output has frequency components that are not present in the input. To prove its nonlinearity, it shuldd be shown that the system either violates scaling or additivity. However, I could not do that. Can someone please help me?
Hi I am using zedgraph to plot an FFT of a signal using c#.The FFT spectrum gives coefficients vs the frequency bins.I am confused whether the plot should consist of spectral lines(ie verticle lines) corresponding to each frequency bins or data point values? Thanks in advance.
Recently in Image Processing I've been studying image clustering (or segmentation) methods and one of them is Color Image Segmentation Using Competitive Learning. I understand the main idea behind it from which you start with one cluster with its cluster representative then pixel values get assigned to this cluster (al...
I came across this confusion in Chapter 16,"Book - The Scientist and Engineers guide to DSP" and Topic is "Windowed Sinc-Filters". I am doing some calculations for designing the sinc filter and came across these doubts: M = 4/BW where "M" is the filter kernel length and "BW" is the trans...
I'm looking for a fast way to do a 2D discrete Fourier transform of an image at many arbitrary frequencies. I know the Goertzel algorithm works for 1D, but is it possible to generalize it in 2D? Or any other useful algorithms? Two solutions that I know, but which I'm afraid might be too slow, are directly using the DFT...
How can I generate non-stationary Gaussian noise (noise uncertainty) using Matlab? where the noise power fluctuate between $(1+p)^{-1}$ to $(1+p)$ where $p$ is the uncertainty factor in dB. Best
How can I add noise to a recording to obtain a SNR value provided by the user? I am tryin to test the effects of noise on my algorithm. Is there any function or app for doing that kind of task? Thank you.
I am trying to get transfer function $H_{xy}$ from this graph. I managed to get the other one but I am struggling with this one. Any help or guidance? Thank you
I have samples of a signal that contains an AM signal added to an FM signal with the same carrier frequency $f_c$. The AM signal is a sinusoid with frequency $f_{AM}$ and the FM signal is a sinusoid with frequency $f_{FM}$. I know how to recover $f_{AM}$ and $f_{FM}$ when I have only an AM signal or only an FM signal, ...
I've plotted the response of a high-pass filter x=[1 -1]; freqz(x) gave me where as the magnitude plot for the fft gave me Why are the two plots different? Both are frequency responses, then why are my plots are diffferent. What does freqz() do?
How can I measure the frequency response range of a piezoelectric disc below. I don't have an oscilloscope but I do know how to program in matlab / octave I was thinking of creating a swept signal in matlab and playing it through my laptop or tablet and playing it through the disc. But how do I measure the frequency ...
I'm trying to decimate given picture Vertically. Horizontally. Horizontally + Vertically. I know that I should use low-pass filter* and then downsample** it not to see any aliasing in image. To this my code is below: Horizontally filter: filter_hor = [0.25,0.5,0.25]; filtered_hor = imfilter(big_shirt,filter_hor...
Since a couple of months I started working on the extraction (estimation) of signal frequency and amplitude components by means of two different time-frequency approaches, namely the Hilbert transform and the Teager-Kaiser energy operator. I tested both methods on standard signals, such as chirps, sine, cosine, etc. Th...
I have recorded an image of a large macromolecule and after several types of image analysis, have obtained a rotationally averaged image, that is not quite perfect, but appears to show 6-fold symmetry. With the origin on the axis of rotation, I compute a transform and unlike most of my colleagues, if I actually look at...
I am currently doing an electronics design for a basic 2.5W speaker amplifier for an embedded system based on COTS parts (e.g. codec and amp), similar to what one would find in a tablet. Aside from a wide range of parts to choose from, there are other engineering trade-offs, such as the filter on the CLass D amp for EM...
I have implemented an IIR low pass filter in direct form 2, the input to the filter is a unit impulse $\delta[n]$. I have calculated the output of the filter for the mentioned input using the difference equation of direct form 2, the equations are given below: $$\begin{align}w[n] &= x[n] - a_1 w[n-1] - a_2 w[n-2] \\ y[...
I have a 4096-point FFT of a continuous pipe organ sound (no windowing). I wish to extract the relative amplitudes of the various harmonically-related sinusoidal components for voicing analysis. I can easily find the bins with the local maximum power for each component, but since the frequencies are not exactly in th...
I stumbled upon the following algorithm for Gaussian noise generation and I just can't figure out how this is supposed to work at all. The algorithm is as follows: First of all: $j = \arg\min_i (q_1 - p[i])$ does not depend on $q_1$ and therefore yields the same result every time - I assume $\arg\min_i (|q_1 - p[i]|)...
I want to create an adaptive filter. Its coefficients have this general shape: When the input signal for the filter is a sine wave, the filter behaves the desired way if the look-back window is set to a length equal to 1/4th the period of the sine wave. I can somehow understand why this is so because a sinusoid is ess...
I would really appreciate if I can get answers to following Fixed point queries(part-1): Converting a real number x to fixed point fraction X(m,n) [1] says: m = ceil(log2(max[abs(xmin,xmax)+1])) { assuming signed x>1) or m = floor(log2(max[abs(xmin,xmax)+2])) { assuming signed x<1} (1.1) and xmin,xmax defi...
I would like to calculate a long correlation function of length say, 1e6 points. I have a prior knowledge that the correlation peak will be in point k*1000. Is there an efficient way to apply this prior knowledge and perform a shorter calculation. I don't want to calculate correlation by definition and implement a 1000...
I am almost done with design and implementation of my first FIR filter using C++. Its a Sinc-Windowed FIR filter designed for a cut off frequency of 100 Hz with sampling rate of 500, with a filter order M = 100. All the operations have been carried out in discrete time domain. So basically my question is regarding test...
Can anyone provide an insight to select a filter structure out of multi-stage and single stage FIR filter?(it will be great help if references are also provided) Let's say I have to make 50 tap filter. So, is it better to design it using two cascaded filters each with 25 tap then 50 tap single filter only or both are s...
I'm looking for a way to pan the PCM audio I have. At the moment the audio is in a stereo format and I would like to pan it to the desired side (or more/less to a side). I've recently figured out how to do mixing here so I guess it would be something similar (adding stuff to each other and such). What I was thinking is...
I am working on speech recognition for Hindi language. My approach is to map speech to text based on phones. For this, first split the speech into the words and then map each word to text by detecting phones. Till now I am able to split the speech audio file into separated words audio files (I used energy levels for th...
I have a project that requires detecting and processing letters in a given input image. I am also new to subject of image processing. There are lots of subsubjects in image processing. I believe I don't have to learn all the concepts of image processing to achive my purpose. Question above; how can I detect letters in ...
Can anyone mention me techniques to hide an image in an audio file ? Thanks
In digital communications textbooks (e.g., Digital Communications by Bernard Sklar), the relationship between the "received" signal power $P_r$ and the "transmitted" energy per bit $E_b$ is given as follows: $P_r=E_b R_b$ where $R_b$ is the bit rate. What confuses me in this equation is that the channel attenuation is ...
Suppose I generate chirp signal with the following python code: t = np.linspace(0, 10, 512) f0 = 2 f1 = 5 fs = 2*f1 w = chirp(t, f0=f0, f1=f1, t1=10, method='linear') result = rfft(w) phase = np.zeros(257) mag = np.zeros(257) for binnum in range(len(result)): mag[binnum] = abs(result[binnum]) phase[binn...
I have a mini project to do about homography, but I am really a beginner in Matlab. I have calculated the homography matrix easily by taking a set pixels from both images but the problem is how to apply this matrix to all pixels of the original image (I think it's difficult to treat each pixel individually). Here is th...
Consider a simple first-order RC filter with AWGN input noise with single-sided density of No. We know the output noise power is No/(4RC). What is rarely noted is that this is true after steady-state has been reached. I simulated the noise from such a filter starting from rest at t=0 and found that the ensemble average...
At first, I want to say that I'm totally beginner in DSP and I don't have enough math education to understand Fast-Fourier transformation. I'm a programmer and I want to archieve something like this: My example file is in 8kHz, 16-bit samples, but currently, I'm trying to do with my own raw data - one sin(x) functio...
Have following questions about Fixed point conversion format. Suppose I want to change Fixed Point format as X1(m1,n1) (→) X2(m2,n2),m1+n1=m2+n2[Word length same], n2>n1[ Higher Fractionlength] Suppose X1=00001100 and suppose I do X1<<(2^3) Q1) If I interpret X1 as pure Signed integer, then X1= 12 and above op...
I am working on an image processing project in java where we use anisotropic diffusion for scale space generation. In the below equation, after finding the diffusion value.using the second equation, I do not know how to find the divergence of diffusivity and gradient. Since I am getting the diffusivity as a float val...
As I know if the impulse response is symmetric around sample zero phase response should be entirely zero. The code below just set a rectangular window for vector "in" const int N = 10; // in = 0 , 0 , 0 , 0 , 1 , 1 , 1 , 0 , 0 , 0 , 0 std::vector< std::complex<double> > in (N); std::vector< std::complex<double> > ou...
In basis Pursuit, L1 minimization is done to perform compressed sensing. In the literature there is a $ \lambda $ parameter used as a regularizer. What is its significance?
Consider the cost function $$f(X,\lambda) = \|AX-b\|_2^2 + \alpha \|LX\|_2^2$$ $A:$Measurement matrix($R_{m\times n}$,$m \ll n$), $b:$observation vector($R_m$), $L:$Laplacian operator($R_{n \times n}$), $X:$vector form of an image($R_n$) (The above cost function can be read, non-mathematically as minimize $AX=b+error$...
How to check if a continuous signal is Periodic or not? My notes says $\cos(\frac{3}{2}\pi t)$ is periodic because $T=4/3$. And $\cos(\frac{3}{2}t)$ is not periodic as $T=4\pi/3$.
I would like to make a mathematical comparison between two methods in my field. One smooths with a Gaussian filter, and one smooths using a second-order Savitsky-Golay filter. Let's say the window size is the same in each case. Is there any way to formally compare these two approaches in terms of their impact on a func...
When impulse response is symmetric around sample zero phase entirely zero. in = 0 , 0 , 0 , 0 , 1 , 1 , 1 , 1 , 0 , 0 , 0 -> This gives zero phase result As I know if I apply a delay in time domain this should result a linear phase in frequency domain. So if I delay my previous vector by one; in = 0 , 0 , 0 , 0 , 0 ,...
Could anyone suggest the best algorithm for a real-time Car recognition (say in a parking space)? I am planning to implement the same on FPGA as well. Kindly suggest. Thank you.
I want to design a 2D IIR filter (that is, created line-by-line progressively, in this case from left-to-right). What is the best way to go about this? I would like the response to look something vaguely like this: The actual shape in the x-axis (time) isn't important - what's important is that the response's width f...
I'm a newcomer to digital signal processing, so I'm looking for some general advice and direction on this problem I have. I have a signal like that shown below. Starting around x=40000, you can see a signal which has a periodicity of about 1 day (you can't tell it's 1 day from the figure), although there is various noi...
I have two signals Sa and Sb, both affected by the same noise. I don't know how much noise is mixed into each one, and the signal+noise is transformed using some (unknown) nonlinear function before I get it. If the original signals are Sa and Sb and the noise is N, what I actually have Ta = f1(Sa + k1*N) and Ta = f2(...
I saw this question on one of the sites related to DFT: The analog signal $x(t)$ is band-limited to $40\textrm{ Hz}$. Suppose the signal is sampled at the rate of $100\textrm{ samples/second}$ and that at this rate $200\textrm{ samples}$ are collected. Then $200$ zeros are appended to the $200\textrm{ samples}$ to for...
I have implemented DFT from Vision & Graphics Group at the Faculty of Informatics and Information Technologies STU - Frequency domain filtration. Tried taking dft for the three Channels(R, G and B) and reconstructed the original image by taking inverse dft for all three channels and merged them together, but the image...
I have one question about digital correction of signal after RC filter which is high-pass. Let me explain detail. I have one simple signal conditioner. It has RC filter on the signal input with cut-off frequency equals 1.3 Hz. It means that after this RC filter my signal has lower amplitude from 1.3 Hz to approximately...
Let us take any image like Lena or cameraman. I know that when Fourier transform is applied on image,spatially global features like amplitude and phase are obtained locally. My question is whether it is possible to reconstruct an image from its magnitude only keeping phase value zero or different than actual value? Can...
I would like to analyse a gaussian smoothing kernel with a set standard deviation and support (let's say, in MATLAB, fspecial('gaussian', [5 1], 1.3) so sigma is 1.3 and support is 5) in the DTFT domain to determine the stopband. My understanding is that the stopband is when the magnitude of the impulse response is bel...
Can anyone help me to improve the degradation of the following image....i am working on OCR system in Amharic language and what is the best image filtering technique that can help me to separate text data from background noise?
If I sample a signal with a harmonic higher than half my sample frequency I can predict the aliased frequency easily as $$|F_h-kF_s|$$ But what if after sampling the signal, I re-sample at a lower rate (decimate)? Does the lower sampling frequency simply trump the original higher frequency or will there be mixing or mo...
I Would like an explanation of the parameters for opencv's HoughCircles function. I'm new to image processing in general and I have recently started using opencv. I'm struggling to find circles. I obtained the image below by converting a source image to greyscale then performing a binary threshold. To my eye, the left...
I'm currently at a loss for why my code seems to not be working. I'm doing a sine sweep on a filter to calculate its freq and phase response. The frequency response checks out, however the phase response doesn't look right when compared to MATLABs freqz function. Plus since I know the filter is a comb, I can tell it...
I am always confused between FFT and DFT. In both algorithm, it has been assumed that the signal is periodic (so my understanding is that, if you have a 20 sample point signal, and you do DFT or FFT, the algorithm assumes 20 sample points repeat). Now here is the question, someone told me that DFT can take odd number ...
Is it possible to design an LPF that has an output identical to the input at specific points in time domain (the rest of the input waveform can get filtered/distorted)? Is there a general name/technique for this kind of thing (assuming it is possible), so that I can search for more information on this topic?
How can I derive the fourier transform of g1(t) = {A cos(2πfc t),−T/2 < t <T/2 0, elsewhere and; g2(t) = cos(t), −∞ < t < ∞ my answer for g1 is; ∫ g(t)exp(-j*2*pi*f*t)dt = ∫ Acos(2*pi*fc*t)e...
How can I derive the fourier transform of g1(t) = {A cos(2πfc t),−T/2 < t <T/2 0, elsewhere and; g2(t) = cos(t), −∞ < t < ∞ my answer for g1 is; ∫ g(t)exp(-j*2*pi*f*t)dt = ∫ Acos(2*pi*fc...
The input space has artefacts that are already in a grid-like layout: I'd like to be able to fit/draw a grid over the artefacts so as to separate them, preferably with lines on average mid-way between neighbouring artefacts: A bonus would be to tweak the threshold to cluster artefacts that are in closer proximity: W...
Aim of my project is to differentiate between cases of silence, single active speaker and multiple ( simultaneous ) active speakers using as basic ( but effective ) logic as possible. Is it possible to achieve this without use of extensive machine learning algorithms ?? Actually, i don't need to do speaker "identifica...
I have implemented IIR filter in direct form, parallel form, and cascade form. The input in all the cases is a kronecker delta function. How to judge which implementation is better is in real time and why? What are the factors to be considered while selecting the best implementation? Please help.
I'm a newbeginner when it comes to signal processing. I'm reading up about anti-aliasing, Nyquist and band-pass filters. There's something I don't get: Anti-aliasing in analog filters. Lemme try to explain. I use a 8th order analog butterworth band-pass filter (centered at 17.5 Hz with a 25 Hz 3dB bandwidth) on the an...
How to determine $a$ and $b$ in this equation. I know how to determine $c$ and $d$ by looking at the intensity histogram. But what about $a$ and $b$? $$P_{out}=(P_m -c)\left( \dfrac{b-a}{d-c} \right)+a $$
I am implementing my own discrete Kalman filter to estimate velocity from acceleration and position measurements (using Matlab ). I think I managed to deal with the $R$ matrix (measurements noise covariance matrix) in this way: M = [x;x2dot]; R = cov(M); What I am not sure about is the matrix $Q$. In all the examples...
$ f(x, y) = \begin{cases} 1,\hspace{30px} x > 0 \\ 0,\hspace{30px} else\\ \end{cases} $ i.e. $f(x,y)$ is a bi-variate function which is zero everywhere to the left of the y-axis and one everywhere to the right of the y-axis.
Short question What are main stages (steps) of calculation frequency response of digital filter by their structure? Detailed question Let suppose that there is discrete FIR filter with known structure which is implemented in programming language (for instance with structure which shown in picture): +---...
I'm trying to implement a solution that would, to some extent, evaluate a picture telling what are the chances that it was upscaled from the smaller image. My approach is as follows: Take a picture of unknown origin (scaled or not) Perform sharpening using some technique which sharpens only evident edges (1) Extract...
By just looking at the 2D Fourier Transform of a signal, can it every be known precisely which values in the space domain are zero?
i have mini project to do about homography , but i am really a beginner in matlab . i have calculated the homography matrix easily by taking a set pixels from both images but the problem is how to apply this matrix to all pixels of the original image (i think it's difficult to treat each pixel individually ) . here is ...
In my solution to a certain problem I use subsampling in the frequency domain. Amazingly, it produces good results. Is it really that awkward to see subsampling in frequency domain? What the consequences really are?
I read a wav file in MATLAB and it has a sampling rate of 16kHz. I want to pass it through a filter I designed which has a sampling rate of 44.1 kHz. So, I thought I would use resx=resample(x,44100,16000) first in order to make them have the same sampling rate. I can see that I will have more samples in time now, but w...
I am trying to read a paper and I found it difficult to understand. This is a quote: Standard Mel frequency cepstrum coefficient (MFCC) computation technique utilizes discrete cosine transform (DCT) for decorrelating log energies of filter bank output. The use of DCT is reasonable here as the covariance matrix of Mel ...
How to estimate if noise of sampled sound signal can be heard? I mean the noise that appear when an analog wave is sampled. Similarly if I use 16bits, 24 bits, and 32 bits resolution, how to calculate noise in dB. I found this eq. SNR = 6.02*NoBits but I am not certain is it correct.
I have a signal, s, given at the end of this message. I need to remove the square wave shape from it. The desired signal is actually very small and it has been contaminated by a strong square-shape noise. The desired signal is just couple peaks similar to an ECG or EEG but we have a very large square-shape noise that ...
Having some trouble understanding this concept, really could do with some advice. I want my STFT to have the following parameters: NFFT = 256 overlap/hop = 128 Now essentially, the algorithm will work as follows: 1) Split the signal into blocks of 256 This will result in around 72*265: 2) For each of these blocks, ca...
I was reading on windowed fourier transform and wavelet transform, and i was thinking that the windowed fourier transform is a subset of wavelet transform. Is that true?
A device is rotating with a particular wind speed and producing such pulse frequencies in a minute: 24.988 25.224 25.212 25.066 25.310 24.963 24.826 24.944 25.490 25.176 24.740 24.988 24.994 24.722 24.510 24.863 25.249 24.931 25.218 24.907 25.459 25.292 25.163 25.447 25.072 25.360 24.976 25.103 25.237 24.845 25.23...
I am trying to replicate the spetrum as given in Audacity and other commercial software... I am getting crazy with the correct scaling needed in order to have dBFS (with reference to a full scale sine wave). Suppose $x_t$ is the PCM sample at $k$ bit precision, first of all I rescale the PCM samples $$ s_t = \frac{x_t...
I'm doing pattern matching by means of normalized grey-scale correlation. My input image is noisy and so the peaks in the correlation plot cannot be found just by thresholding. The original correlation plot displayed as a surface, the peaks are visible, but are not the only local maxima in the picture, I need to find ...
If I have the discrete Hartley transform of a signal of length 2n, how can I use it to efficiently obtain the discrete Hartley transform of the same signal padded to length 2n+m for some m > 0?
I was reading a paper 'A Flexible Framework for Local Phase Coherence Computation' (paper URL) on using wavelets for local phase coherence. I just want to be sure. Is the gabor wavelet the same as the gabor function in the gabor transform (STFT)? If not, can we use a Windowed Fourier Transform instead?
I have a signal that consists of a sine wave, on which I apply a rectangular window and then an FFT. I'm only interested in the magnitude of the sine wave, and I want it to stay the same even if the signal shifts a few samples to the right relative to the window (of course assuming the sine wave has the same amplitude ...
I am new to DSP, so I wan't to know a couple of things. How does µlaw encoding function on a log scale such that you can input 16-bit signals and get the 8-bit µlaw signal output, after compression, and then use that as a building block to expand it back to 16-bit signals. I would love if someone can guide me through t...
I have a question regarding the fourier transform. If I plot the frequency against the fourier transform for a periodic signal and I get a peak, What is the physics behind it? I want to know the physical meaning of it. Why we get a peak and what does it signify? Thanks
I have been using convolution for finding outputs of various systems .I know how to use it.But I still don't know what does convolution exactly means? how one can define convolution ?
I have been experimenting with speaker independent automatic speech recognition. At this point, I understand that the signal is typically segmented into overlapping frames and MFCCs are extracted as features for each frame. A lot of the texts I read talk about reducing the dimensionality of the features using (for exa...