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Both Correlation and Convolution are displacement function, i.e they are used to slide the filter mask across the image. Convolution is same as correlation except that the filter mask is rotated 180 degree before computing the sum of products. We mostly use convolution. why it is widely used? why convolution is prefer...
In the below screenshot I was unable to understand how the equation 4.3 is derived from the above equation.. How $x(nT)$ has become $x(n)$ ? Please clear my doubt and suggest some link to understand DFT. I am sorry for the poor image quality.
Apologies in advance for what I imagine is a simple question, however I can't work out how to do this. I have two signals of the same type but with different sensor types, the figure below will give you an idea of some of the data I've got: I very simply want to calculate the average percentage difference between the...
Is there any relationship between the FFT and linear predictive methods? Can an FFT result or the input to an FFT be modified to do non-circular prediction/extrapolation from the FFT results.
I'm looking for a good resource (book, tutorial, lesson etc.) that explains the usage of Kalman filtering in image processing applications. I'm aware of the fact that Kalman filtering is an optimal estimator which is a tool not a concept in image processing because of that I didn't find a book that explains the usage o...
In an audio file I would like to find calls of a specific bird specie, one that makes wobbling sound with rising notes. Can you recommend what would be the best audio feature to calculate in time or frequency domain to capture its essence? Among others I tried MFCC, but the results were not promising (or perhaps my im...
One definition of the Morlet wavelet function is given by: $$\frac{1}{\sqrt{\pi f_b}}e^{\frac{-t^2}{f_b}}e^{j2\pi f_c t}$$ The Fourier transform of this equation is: $e^{-\pi^2 f_b(f-f_c)^2}$ (is it right)? First I attempted to plot the FFT of Morlet function by FFT function in Matlab, then I've plotted the Fourier t...
Given the noise spectral density ${N_0\over 2} = a$, what can be deduced about the variance? In the case of a two-symbol canal(/cable), and when $T_0 = 1$. Is it that, $\sigma_X^2 = {N_0 \over 2} *T_0$?
I am reading Digital Communication, 3rd edition by Barry & Lee, Book's Webpage: enter link description here This is an excerpt taken from this book (mentioned above) from Chapter 3, Page # 57, 1st Para, 1st Line. It says, Although modulation and demodulation are deterministic,the information to be transmitted ...
In the DSP class I'm taking, a lot of the questions ask me to list the multiplicity of a pole or zero in a rational Z-transform. For example, the multiplicity of the zero at $z=1$ in :$\frac{(z-1)^2}{(z-5)}$ is $2$. What knowledge do we gain by knowing the multiplicity of some pole or zero? Of course, if we know the m...
Let $|A|$ denote the power of the analog signal $A$. Is that true to assume $|A+B|=|A|+|B|$, for two arbitrary signals $|A|$ and $|B|$? That is the power of the summation of two signals is -at least roughly- equal to the summation of the powers of the two signals. What if $A$ and $B$ are random signals?
I am not able to understand the formula, What is $W$ (window) and intensity in the formula mean, I found this formula in opencv doc http://docs.opencv.org/trunk/doc/py_tutorials/py_feature2d/py_features_harris/py_features_harris.html
I am doing OCR for Kannada and English. What are the best features in image? Please also tell me some tools to perform OCR before implementing it as code (like Rapiedminer for machine learning).
How to determine the frequency of a signal with a value of the sample and value of the time instant the sample was taken ?
If I have spectrum $X_k$ of the signal evaluated at frequencies $k \delta f$ and I want to know what is the value of that same signal at some other frequencies , i.e., $k' \delta f'$.I know that $\delta f$ and $\delta f'$ are somehow reelated (e.g. $\delta f = 12 \delta f'$) how can I express spectrum of the signal wit...
I am working on decimation of signal, and I want to know which is the best way to understand if the downsampling is well done or not. At the moment I am comparing the FFT of the source signal and the downsampled signal and I observed a downward shift of it (I think it is due to the lesser quantity of samples), I had al...
Please consider the following excerpt (from Rodrigo R, Zouqi M, Chen Z, Samarabandu J.: Robust and efficient feature tracking for indoor navigation): Robust feature tracking is a requirement for many computer vision tasks such as indoor robot navigation. However, indoor scenes are characterized by poorly localizable f...
would you please tell me what does LPCC exactly do?How can we calculate the coefficients of LPCC? I need to know clearly what is it so i can implement it. I dont know if i am asking this question correctly but whats the different between LPC and LPCC?do they even have any similarity that gives us the ability to talk ab...
What criteria should be used to safely decide if it is an IIR filter or FIR filter from a difference equation? FIR filter are always stable, meaning all poles are inside the unit circle AND have pole-zero cancellations i.e have equal number of poles and zeros at the same point in the z-plane. IIR filters are conditiona...
I know that for linear phase filters, all frequency components have equal delay times. That is, there is no distortion of signal due to the time delay of frequencies relative to one another. However, for filters generalised linear phase, where the phase is of the form $a\omega+b$, and $b$ is not zero. Will the same be...
Are there any modulation schemes (or other situations) where sampling at or very near twice the carrier frequency would be better than IQ or quadrature sampling using a local oscillator at or near the carrier frequency for SDR demodulation?
For antisymmetric FIR windows, the phase response is of the form $a\omega+b$, where $b$ is $\pm\frac{\pi}{2}$, so it is not strictly linear. Does the waveform of the signal gets retained?
$X(jw) = 1 + \frac {jw+3}{jw+4}$ what is the IDFT of this signal?
I am searching for a good thesis topic on DSP with the use Matlab/Simulink/LabVIEW for my master, however, when I contact my department (we only have 1 DSP professor), he asks me to try to propose one but I cannot think of any. I wonder if someone can help me with some ideas
Assume I have a (single real float channel) biquad IIR filter implemented in C, and some coefficients for a bandpass filter (right out of RBJ's cookbook). If I convert the input, output, data path, and internal arithmetic of this IIR filter to use complex data types and multiplies, can I use the resulting code to filte...
What are the differences between KLT and SVD? It looks more or less similar to me. Did I misunderstand the concepts? Explain me more about SVD and if possible provide me the link for SVD (Image Processing). Thank you.
Sound are digitalized in computers as an archive with its intensity amplitude at the given time. I want to ask a very simple question about frequency filtering. How it's done, if there's no frequency information stored in the digital audio? It's all intensity. I cannot understand.
Compressed sensing (CS) guarantees exact object recovery (or with high probability) given a) sufficient measurements are taken in a sparse basis which is b) incoherent vis-a-vis a given object representation basis. Does CS theory state anything about reconstruction of successively less sparse signals? That is, if I don...
I'm afraid this might be a bit of an elementary question, but I'm afraid I'm a little new to this. I've got a system that measures a frequency spectrum to find resonant frequencies. It does this by applying an impulse, and then performs a linear sweep through a range of frequencies measuring the amplitude of the respon...
I am trying to compute the accuracy of the HMAX model. I am using the Face category (containing 435 images) from the Caltech101 database. I split it into x training and y testing. At each time, when x increases, the accuracy also increases. Furthermore, I heard that the number of training should be equal to 80% by comp...
Over the past few weeks, I have been working on a problem of segmenting an audio file containing some bat samples. I have now successfully come up with a moderate algorithm to do this. I am trying to calculate the herts, or, the Htz of each of blocks that I have deemed to contain interest. Currently, for the signal I...
The big disadvantage of a DFT is that it has only frequency resolution and no time resolution. This means that although we might be able to determine all the frequencies present in a signal, we do not know when they are present. So we go for Wavelets. My question is, What are the applications (other than music & speech...
I am a beginner in digital image processing, recently I read a paper on color image contrast enhancement, In that paper authors have implemented their method in HSV color model. In my work I have implemented that paper in MATLAB using RGB color model. I just want to know that whether it will make any difference or not?...
May i know what does this function do in matlab? num = fir1(80, [5Hz, 50Hz]); I know that 5 Hz is the low cut-off freq and 50 Hz is the high cut-off freq. I'm not sure why it is 80 for the first parameter. What will be the output of this function?
I'm trying to do some analysis of a curly human hair on an iPhone app. I've been using the openCV library. Particularly I want to be able to estimate the length of the hair, and locate its end points (a measure of curliness is stretched end-to-end length vs. relaxed end-to-end length). My approach so far has been to us...
I am working with speckle noise reduction in ultrasound images. I have used wavelet transform for removing the noise by eliminating certain frequencies in order to eliminate any existing noise. Since in an image HH,LH,HL components contains most of the noise. We can eliminate noise by eliminating those components. s...
I am new to Matlab. For me the task is to perform data classification on some data set. The given data set consists of numeric values. One thing that bothers me is I have to perform all KNN, ANN, and SVM. In performing KNN, I faced a problem of how to find the value of K. If it is user defined? From which value we have...
Statement 1 : When we take DFT of an image matrix, we get DFT co-efficient matrix. What does the DFT Co-efficients tells about the image? (For example, take the 1st co-eff in the DFT matrix, what does it tells us ?) Statement 2: In a DFT matrix, we have low frequency components on the top left corner of the matrix and...
I am working on a project regarding 3D reconstruction of 2D images. I have understanding of finding and Mathcing the keypoints and its algorithms; but i am not able to replicate the theory in MATLAB code. Is there any Book or paper which is basic for this problem?
There is a transfer function $G(s)= \frac{-2(s+4)}{(s+3)(s+2)(s+1)} $. Find its gain $K$. I'm not sure, what does this question mean. Is there any konstant gain for this system? I tried to find its frequency characteristic. $$G(s)= \frac{-2(s+4)}{(s+3)(s+2)(s+1)} = \frac{-2(j\omega+4)}{(j\omega+3)(j\omega+2)(j\omega+1...
Is there a better way to increase Recall Rate when using SIFT features? I am thinking a way to replace the NN1/NN2 ratio to account for slightly distorted objects. Moving towards clustering and using BOW(Bag Of Words) seems a way but I need to do one-on-one match of objects in images rather than training and learning. ...
Will using Cosine-Similarity or Histogram Intersection as distance-metric, for matching SIFT descriptors, help to get better results? Anybody tried on that ever?
I am using a SIFT algorithm to extract features from an image. I understand that the SIFT descriptor first finds the extrema and then finds the gradient and direction of each of these interest points. What happens after this part?
I have a function which looks like so: I would like to know if there is a measure for a "weak type run length". How can I know that a function is approximately constant in some specific neighborhood? How can I differentiate this function from random noise?
In theory of FT(Fourier transform) and STFT(Short Time Fourier Transform) it is said that "A wide window gives better frequency resolution but poor time resolution. A narrower window gives good time resolution but poor frequency resolution" Can somebody explain why a narrower window gives poor frequency resolution?
Suppose that we have some sample: y = 31.2241 -5.9830 22.6058 -2.2309 -42.8272 29.2850 -6.2652 20.8502 -2.4639 -44.7525 30.1104 -6.0180 21.4476 -1.5089 -45.1826 29.2744 and somehow I know that it is obtained via a sinusoidal method which contains two sine functions at dif...
I have searched this question but couldn't find the answer in this network. I know this is very confusing question for DSP beginners. Both DFT and Z-transform work for Discrete signal. I have read that "Z-transform is the general case of DFT, when we consider unit circle then, Z-transform becomes Discrete Fourier Trans...
I've found a material explaining the sample and hold operation pretty well, but there's one thing I can't understand. http://www.springer.com/cda/content/document/cda_downloaddocument/9781848001183-c1.pdf?SGWID=0-0-45-484012-p173780368 Next, we would like to take this ideally sampled signal and hold the values between ...
I'm trying to create a chirp/swept signal that has it's frequency going down exponentially using matlab/octave. I can't use the built in chirp function because I'm trying to create a chirp/swept signal based on an equation/formula. I tried creating some code below but as you see from the spectrogram the frequency doe...
I am doing my first steps in embedded systems and there is a big deal of fixed point code and of course DSP optimized code.I would like please if possible to recommend any book/material related to that topics.
I'm using portaudio for I/O audio streaming, and I wanted to apply some equalization on audio input from microphone. I found that most people refer to fft for task like that, so I use FFTW library in my project. My first idea was to select frequency bins in some range, from fft resulting complex array, and increase th...
I am not very good in statistic neither in probability theory, and I do not completely understand the mathematical background of Monte Carlo simulation or SIR sampling, but I'd like computer vision and I'm experimenting with it hoping to get some nice results with already known techniques (I don't want to reinvent the ...
I am trying to introduce a fractional delay in my input vector x[n] so the output is something like y[n]=x[n-M/N]. I am not supposed to use the MATLAB fracdelay function.So, I used sampling and delay. The three steps I followed are: 1)Upsample the input by M. 2)Delay the upsampled version. 3)Downsample the output obtai...
As a follow-on to this post, how would you force linear phase for a FIR filter you synthesized using Berchin's FDLS?
When energy is referred to as the square summable sequence, is that energy a representation of the power of the signal? How is the energy of frequency determined? E.g. what if an audio signal has high energy on the higher frequencies and another has high energy on the lower, bass, frequencies, how is this determined? ...
I would like to know if $$ \text {Z-Transform ( } G(s)H(s) \text{ )} = \text {Z-Transform (}G(s) \text{)} \text { Z-Transform (} H(s) \text{) } = G(z)H(z) $$ where G(s), H(s) are the Laplace transform representations of g and h, and G(z) and H(z) are the Z-transform representation of g and h. Is this relation true?
Why do we need an oversampling in OFDM signal? and how to do it? If around half of the Sub-carriers are only used to send data, then do we need to perform oversampling to the signal? Is it necessary to have LPF in OFDM transceiver?
I have been looking to implement feature extraction and matching in MATLAB. For this, I've read many algorithms, but i'm not able to understand how to implement these in MATLAB. Is there any Book related to this topic which is based on MATLAB? i.e. gives exapmles and sample codes in MATLAB? I have read Gonzolez and the...
I have to detect speech intervals in large pre-recorded files in my project. I think, there won't be so much noise in the background (audio will be recordered in the room or even in the studio), but it still can be. So, what are good VAD algorithms that I can use (algorithms, not libraries)?
In below image, we have scaling property of DFT, how the final equation is obtained from the above equation. That is how we are getting the scaling factor , $ \frac{1}{|ab|} $ in the final equation ? Please explain how we are getting $ \frac{1}{|ab|} $. Thank you Equation 4.12 is given as
Does anyone know how to find information on the derivation (rather than the definition and application) of Teager's Energy Operator, both for the continuous and sampled case. I have Kaiser's early papers "On a simple Algorithm to calculate the 'Energy' of a signal" and "Some useful properties of Teager's Energy operato...
I'm designing a bandpass filter in scipy following the cookbook. However, if I decrecrease the filtering frequencies too much I end up with garbage at high order filters. What am I doing wrong? from scipy.signal import butter, lfilter def butter_bandpass(lowcut, highcut, fs, order=5): nyq = 0.5 * fs low = lowc...
i've written a source code in visual dp ++ in c language. the platform is sharc processor adsp 21062.after the build is complete. while running the program it is displaying a message Instruction timed out with PC at: 0x20a5a . what may be the error in my c code. can you please resolve?
I know that many books and papers talk about the DC offset/DC component of a filter. How do we define the DC offset mathematically, for the case of discrete filters?
I have a speech file that was recorded with a sampling frequency of 8kHz. I read the file in Matlab as [x,fs,nbits]=wavread('speech.wav). Now I play the file in Matlab using soundsc(x,fs). But if I use the command soundsc(x,fs/2) and soundsc(x,2*fs) the speech changes. What is the reason?
I am interested in analyzing full length songs for general features of the track which are not otherwise apparent. For e.g., I want to look at various parts of the song with time-frequency plots/wavelets/other signal processing tools for, lets say - How does the frequency content of the song during the guitar solo com...
Can you please explain or point the right way to think about how to implement two sound effects in a .wav file. So I read a .wav file and converted it to int values (in numpy array) array([59383, 58290, 60629, ..., 52606, 51982, 56274], dtype=uint16) First, I am not sure I am clear about what these values in array re...
A state space representation is given by: $$\dot{x}= \begin{bmatrix}0 & 0 & 0 & 0\\ 1 & 0 & 0 & 0 \\ 0&0&-2&-4\\0&0&1&0\end{bmatrix}x+\begin{bmatrix} 1\\0\\1\\0\end{bmatrix} u$$ $$y=\begin{bmatrix}0 & 1 & 0 & -1\end{bmatrix}x$$ I need to find a transfer function of this system. I don't know how, but I've tried it this ...
In general any form of enropy is defined as uncertainty or randomness. In a noisy environment, with increase in noise, I believe that entropy increases since we are more uncertain about the information content of the desired signal. What is the relationship between entropy and SNR? With increase in signal to noise rati...
For my source code witten in c ,in visual dsp++ 5.0 for a forward error correcting code on adsp 21062 sharc family platform. i want to plot a graph between different data sets. when i tried to use the plot window, it is showing the data sets which are in assembly.so how to plot a graph between the input and output in t...
I'm working on a Ultrasonic Transducers resonance tracking system. Where phase shift between Voltage and Current is the feedback signal for my system when phase difference is Zero that means system is resonant other wise depending on the phase difference frequency will be changed proportionally. I went through lots of ...
Is it possible to model the acoustic transfer function in MATLAB for a common set of headphones? i.e. I want to model the summation of a noise sample and the audio from the output of the headphone e.g. signal = exteriorNoise + Music Where signal would be the information the user would hear? I would imagine headphones w...
When I do a spectrogram in matlab / octave I can create a swept signal that looks like the RED plot line below. But how can I create a swept signal like the BLUE line in the 1st plot using the equation below. thanks to Daniel and David for getting me this far with the code is below startfreq=200; fs=44100; endfreq=20;...
I have a signal consisting of two sine waves of same frequency but have different phase (and also a lot of HF noise). Their amplitudes differ by about $120\textrm{ dB}$. How to I go about separating them? SNR is about $130\textrm{ db}$ but its frequency is far higher than my signal, i.e. $f(\textrm{noise}) \gg f(\text...
I am inquiring as to a practical way to solve a problem I have. Basically, I need to transmit a signal, $x[n]$, through a seismic transmitter. (It will go through a D/A, etc). The transmitter that we have has a certain frequency characteristic, in that, it passes certain frequencies out better than others. (Nothing sh...
Problem I wish to filter a discrete-time input signal, $x[n]$, so as the output signal's, $y[n]$, exponential moving average, $a[n+1] = a[n] + \alpha(y[n] - a[n])$, tracks around a constant reference $\mu_x$ such that $\mathbb{E}\{y[n]\} \approx \mu_x$, and also an approximately constant variance, $\text{var}\{y[n]\} \...
What does the Eigen values and Eigen vector of a signal or function represent? What is its physical significance? I know about basis vectors of a signal which constitute the orthogonal planes where signal projections are represented. Are basis vectors and Eigen vectors same thing? Can we reconstruct signal using these ...
I would like to experiment with some input from a microphone and am receiving a 2 channel, 512 samples buffer in real time I know the signal could be passed through a low pass filter, and then windowed before the FFT. The sample rate is 44100.00, what low pass filter is needed for this, does the analog to digital conve...
I would like to write a BPM detector. Does either of the domains provide more information for this type of analysis? What exactly is being looked for in the data to detect onset for BPM detection in either domain? Does tracking energy values in the time domain lead into rhythm detection? Obviously pattern recognition ...
I have been trying to understand a publication here where Short time fourier transform is applied over n samples in steps of m samples each (m is the size of the moving window). I understand that the computational complexity of the Short time fourier transform for m samples is mlogm. The operation also consists of extr...
It might be a very silly question, however I'm facing problem with it. Now, the Gaussian noise is a signal with zero mean and variance. I understand what the mean and the variance mean however, I'm facing problem with zero mean. What does that mean? Does it mean that it is only a signal with variance which represents t...
I am trying to get a working understanding of how to resize images using bilinear and bicubic transformations. I coded a resizing function based on things I've read here and Wikipedia, etc., but I get different results than what MATLAB and Mathematica give. A minimal working example is the image im=[100 50 0] in MATLAB...
Does FFT method input argument have to be power of 2, i.e, $2^n$ I just realized there are many algorithm for FFT implementation, is there any algorithm that takes arbitrary amount of samples as input argument? if yes, what is their advantages amongst each other? namely, what algorithm does Matlab apply?
I have an image processing application that is different than anything that I've done before, and am looking for suggestions on how to approach. Here is a basic description of the problem: Given a set of images taken of over time, an "intensity wave" results for each pixel in the image, i.e. if you plot the intensity...
First time studying image processing... I just don't understand what does fourier transformed image of an image describe? For example consider given following pictures, The first one is the image, and the second one is the fourier transformation of the image. Now my question is: By given the fourier transformation i...
I have been given a sinusoidal signal x in time domain,asked to investigate the frequncy responce of the signal using matlab and observing the result,and using grpdelay(x,1) command in matlab to observe the group delay of the signal .and taking y=conv(x,x);which define the convolution of x signal by itself and then ask...
I am trying to apply kalman filter for video processing , i am studying about it from different sources but it take me towards question that if i don't know that where my object come in frame mean i don't know about its angle (location in frame) then is it possible that Kalman filter automatically detect its measuremen...
Using under-sampled data, what determinations can be made about the Nyquist rate (correct sampling rate)? If my instrumentation limits me to sampling data at a fixed but insufficient rate what, if anything, can be determined from that data about the sampling rate that would be required to properly represent the signal?...
I have a sinusiodal signal x in time domain.I have done the convlotion as followings using matlab: y=conv(x,x) and y=conv(x,fliplr(x)). Observing the result by Matlab, both of the above signal giving the same result in frequency domain. Why do you think is that? P.S: FLIPLR(X) returns X with row preserved and columns ...
I have the following two graphs but I don't know how to recognize from the graph if they are IIR or FIR! Can you please guide me through? P.S.: Does MATLAB have a function to detect if an impulse response graph corresponds to IIR or FIR?
I have used the following command to slice a large image into blocks of 2x2 pixels: convert -crop 2x2 +repage Logo.jpg sl%d.jpg When I attempt to put the blocks next to each other in the order that they should be, I get a smaller size of the original image. I wonder if there is anything I should do to get the original...
I have a physical system consisting of a microphone, which feeds to a DSP, which drives a speaker. The microphone and speaker system have an amplitude response or gain, as well as a phase response, which is clearly lagging as the frequency increases. The aim is to produce an IIR filter in the hardware of a DSP that ca...
suppose we have an fir filter: t=[-50:50]; x=[sinc(.1*t).*cos(2*pi*.15*t),sinc(.1*t),sinc(.1*t).*cos(2*pi*.3*t)]; using matlab command grpdelay(x,1),we observe the groupd dalay of the system. then we have y=conv(x,x),again using the matlab command grpdelay(y,1) what changes in group delay would you expect to see in co...
TL/DR: I'd like to take the median of lots of images to make a single "better" image. The problem is, the camera is slowly moving, and there is significant lens distortion. On the plus side the motion is constant, I've got lots of images, timestamps on each image, and I've been able to track points across the images ...
I know that the complex wavelet transform is given by $$\int_{-\infty}^{\infty}f(x)\psi^*_{s,u}(x)\,dx$$ where $f$ is the signal, $\psi_{s,u}(x)=\frac{1}{\sqrt{s}}\psi(\frac{x-u}{s})$ is the wavelet. and $^*$ denotes the complex conjugate operation. If I have the wavelet $\psi_{s,u}$, which is represented by psi in MAT...
I am working with speckle noise reduction in ultrasound images. At present I am learning about anisotropic diffusion.I will explain first what I have understood about anisotropic diffusion. I learnt the difference between isotropic and anisotropic diffusion using the diagram below: In case of isotropic diffusion diff...
I am trying to detect a crosswalk pattern on the road by using Hough transform, and then selecting lines that separate between black and white regions in a binary image. The stripes are detected fine, but the problem is that a lot of other unwanted objects provide false positives, as you can see in the image: *You can...
Let's say I have a sound file, I then compute the Spectrogram and segment each part of it. Assume that it is a recording of someone saying the word "Hello" and therefore, each piece is given like so: 0 -> H 1 -> E 2 -> L 3 -> L 4 -> O Now I want to calculate the duration of each of the parts/phonemes. Is there a way...
For a time discrete signal, the rect-filter (for 1D signal [1,1]) is a perfect low pass filter. I was wondering, what is the best gaussian approximation of this filter? For a current problem, I cannot use binomial filters but have to use Gaussian filters for smoothing. Ths question is thus, what is the standard deviat...
I want to make/create this spectrum visualization on a physical Led_Bar_Panel(s). Here's what I think I should do: Take sample's in real-time( Sampling freq = Fs KHz, only voice) Do a FFT( As far I know we take (2 ^ n) samples at a time. Is there a mathematical relation between n and Fs ?) Take the averaged amplitude...