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Also, in Huffman Coding, why do we use fewer bits for high frequency data. I don't get the intuition. Shouldn't it be the opposite or does this have to do something with the psychoacoustic model of the human hearing which says that our cochlea is sensitive to lower frequencies more than higher, hence we use up higher b...
I am doing the Coursera course on DSP by Prandoni and Vetterli, and was excited to learn that we can use the tools of vector spaces to analyze discrete signals. At this early point in the course the professors did mention that since the signals live in Hilbert space, we can naturally talk of orthogonality, for one. I ...
This paper by Edward Rosten and Tom Drummond 2008 describes choosing interest point via machine learning approach using ID3 algorithm. I understand the concept of the ID3 they are using. But my question is how are they fitting this whole training and testing idea of machine learning, while are analysing real time video...
I am reading sensor data with my Raspberry Pi from the MPU-6050 sensor, which has an accelerometer and an gyroscope. I am using an adjusted version of the i2cdevlib and I'm taking the measurements from the FIFO buffer with the DLP (digital low pass) enabled. I don't know if this is important, but the sensor output is t...
What type of signal analysis (if any) might be performed on a recorded music vocal track to determine if pitch correction might have been applied to a performance?
If I have the following spectrogram: And only wanted to have the signals that are in a particular frequency range (2627 - 3939htz).. Can I run a FIR filter to only allow these signals through, and therefore, it will be much easier to segment each of the calls? If so, which FIR filter would be more appropriate for this...
Branching off from the question [Mapping the value of a sample in a 2-D DFT to cycles/pixel], I have a few more questions related to 2D FFT of images. I correct in assuming that the fundamental frequency is the image itself? And one copy of the image is equivalent to 1 wave cycle? The sampling frequency is the number ...
I'm trying to get the correct FFT bin index based on the given frequency. The audio is being sampled at 44.1k Hz and the FFT size is 1024. Given the signal is real (capture from PyAudio, decoded through numpy.fromstring, windowed by scipy.signal.hann), I then perform FFT through scipy.fftpack.rfft, and compute the deci...
Please help me in solving the following problem: Find the pulse transfer function and weighting sequence of the system described by the following difference equation: $$y[k+2]+ y[k+1]+(0.89)y[k]=r[k] \ ,$$ where $y[k]$ is the input and $r[k]$ is the output.
I am quite new to Java and Signal Processing, but I was assigned a project dealing in audio processing. My topic given to me is a game that has a component of analyzing a song (any given song) and outputs data depending on the frequency currently being played (real time). ie: as the song plays, it outputs the current f...
I'm trying to estimate the height of my device using an cheap integrated Freescale accelerometer. First, I estimate the earth direction when measured acceleration is close to 1g, then I project the measurement on that direction and integrate twice. Very quickly the estimated height drifts. I was thinking maybe to use s...
Why do we always characterize a LTI system by its impulse response and not by another response, like the step response? What does the impulse response have that is so special?
MATLAB's implementation of cwt uses the complex morlet wavelet as: psi(x) = (pi*Fb)^(-0.5)*exp(2*1j*pi*Fc*x)*exp(-(x^2)/Fb); However, it is possible to implement the cwt in the Fourier domain (cwtft). In this case, MATLAB uses a slightly different version of the Morlet wavelet: psi = (pi^(1/4)).*exp(2*pi*1j*t)*exp(-(...
For the application of Gabor filters in feature-based (e.g. lines and edges in images) digital image processing, it is desired that the DC component is 0. Why is this so? What effect does it have mathematically if the DC component of the filter is not 0? Thanks in advance!
I am working on Despeckling Ultrasound images. I have been trying out some basic filters and their hybrid combination.. Now I would like to try Diffusion Filtering techniques to despeckle the ultrasound images. I don't know anything about the diffusion filtering techniques (Isotropic diffusion or Perona Malik Anisotrop...
SO i have to calculate the PSD of mentioned input. here is my code. (every little thing written wholly and solely by myself) x = 100; % number of packets of size 1kB P_final = 0; %final PSD while(x ~= 1) % loops over 100 packets b = rand(1,8192); %1kB packet with equally likely 1s and 0s maaped to -1s and +1s ...
The following pictures describe my situation. Any thoughts on an appropriate algorithm I could use to solve this problem?
A key result in compressive sensing states that, with high probability on the random draw of an $\ m × N$ Gaussian or Bernoulli matrix $\ A$, all $\ s$-sparse vectors $\ x$ can be reconstructed from $\ y = Ax$ using a variety of algorithms provided $\ m ≥ Cs\ln(N/s)$, where $\ C > 0$ is a universal constant (independ...
I am doing my Bachelor's project on Road extraction. I have detected the line segments(edges) in the image using hough transform. I need to detect the curved lines in the road and link them(edge linking). help me with matlab coding.
Suppose you have a signal $s(t)$, corrupted by Gaussian noise of $\eta(t)$ with zero mean and standard deviation $\sigma$. So that the measured signal is $f(t)=s(t)+\eta(t)$. I now want to calculate the spectrum of the signal using FFT. For my particular case, I have found that for wavenumbers $q$ above a given cuttoff...
In my project I am extracting the lines of an image with openCV2 by firstly creating a skeleton: I convert it to a binary image, invert it (to make it work with morphological operations) and then perform a bunch of morphological operations. Then I apply the canny edge detector on the skeleton and perform a Hough transf...
I have the following aerial image (apparently taken from an airplane): I also have a map of the area shown in the image: On the image and the map I can mark the corresponding streets, street intersections and landmarks (I have shown only three here, but I can provide more points for precision). I would like to skew ...
I am trying to perform the fast fourrier transform to non uniformly sampled data . The output is in the following image It was obtained using the following code in MATLAB %% x is the sampled signal %% t is the sampling instances L=size(x,1); t_new = linspace(t(1),t(end),L); % Performing interpolation to obtain a uni...
I want to know that why large laplacian spatial filtering is done on EEG signals? I tried a lot but could not find material on LLSF. And what is meant by spatial filter ( in context of Digital Signal Processing)? Why is it called "Large" and "laplacian"?
this is a crosspost from stackoverflow as I figured there may be more who can help me here... I've been writing a basic C program to generate a sine wave and write it out to STDOUT for piping into a filter which I am also going to write but I'm having issues. I am able to generate what sounds like a good clean sine wav...
Calculating the power of AWGN should be equal to infinity as PSD is constant and its integration is infinity over all frequencies. Hence for any signal with finite power mixed with AWGN, SNR should zero. But this doesn't seems right??
I am playing around with FFT and cant seem to get the expected results. I have a sound clip that i recorded from a mic placed on an analogue clock. So it has a sound every second. [wave,fs]=wavread('clock.wav'); t=0:1/fs:(length(wave)-1)/fs; plot(t,wave); n=length(wave)-1; f...
I want to know that what features might be needed for recognizing a word i a sentence exactly? I guess the routine for recognizing the word in a sentence is ganna follow this: 1-calculate voiced and unvoiced and silence frames 2-determine a threshold 3- when ever we get to a silence frame,that could be an end point of...
i have made a subroutine to calculate 1d fft of a series of size 2^n. I need to calculate discrete sine transform of a series using the fft subroutine. I am not sure what to do.
Firstly, the context: I've been given a signal in a wav file, the sampling rate and carrier frequency along with the fact it is Manchester encoded and has been ASK Modulated. Five periods of a sinusoidal wave at half frequency of the carrier denote the end of a 'row' and a parity column/row is added. From this I am to ...
How a mean filter is called as linear filter and a median filter is called as non linear filter? I understand how a mean and median filter operates, but I was not able to relate with the term linear and non-linear. Please explain me with an example.
For many papers which talks about using wavelet transforms for feature detection in image processing, it is stated that it is advantageous for the wavelet to have zero mean? Why is this so? Thanks in advance.
I'm using a Hidden Markov Model in order to train an application for gesture recognition. Currently, I have gathered samples, and, using a library have trained these values (They have been passed through a Baum–Welch function in order to train the dataset. I then (in real-time) have computed the gestures of each model...
I'm trying to figure out how to use DSP techniques in order to both add artificial echoes to a sound file, but more importantly to be able to detect these echoes. I've been reading about it here: http://www.snotmonkey.com/work/school/405/methods.html#echo Can someone point me to possibly some example code using Apple's...
my low pass filter keeps distorting and I'm at a loss to explain why. It calculates the correct coefficients, but distorts everything. The code is as follows. It's driving me mad. What's weird is my high pass works perfectly and it's coded in a similar way, but with a negative b1 coefficent and the same a1 and a2 coeff...
I am an ECE student, and I am doing my project on underwater communication. The main concept of my project is to denoise the signal by using UFB and WPD algorithms, and giving that output to a matched filter. I have already generated the sinewave and added noise and given it to the matched filter. How do I write the c...
How we define frequency in an image? Does the repetition of a single gray level defines the frequency?(Example: can a black(single color/constant region) background image be called high frequency image?) I think it is not so. Or Does the frequency depends on number of changes in the gray levels in an image? If so W...
this is for an application in which passwords are stored in the form of audio inputs . In order to provide user access , the password(in audio form) will be matched with already stored password(audio only) in the database . I wanted to know how to measure the extent of similarity in two audio patterns . the audio patte...
There are couple of methods listed here about finding the value of k in k means algorithm. How much sound is each of these procedure when used to find out the value of k in unsupervised manner.
This is a pretty general question about how to compute derivatives of a digital signal $x[n]$. I would like to know what are the different approaches (from naive to complex) and how are they compared to one another? Is it possible with FIR/IIR filters? What are the pro's and contra's. Which are better for real-time app...
I came across various works in the literature of document analysis addressing the problem of detecting base lines in multi-oriented texts. But no amount of search could give me satisfactory answer to the following quest. Is there an algorithm to detect the base line which neither use a histogram based method or any kin...
I am working with wavelet transforms. When we apply a discrete wavelet transform to an image, it decomposes into four components named the approximation component, thevertical component, the horizontal component, and the diagonal component. These components are indicated as LL, HL, LH, HH respectively. I understand ho...
If a signal depends on only one variable then we call it one dimensional, and if a signal depends on two variable we call it a two dimensional signal. But when we represent an one dimensional signal, we use two axes(amplitude vs. time). In case of a two dimensional signal, we use two axes (for example, the x axis and t...
$f:\mathbb{R}^n\rightarrow\mathbb{R}$ is a signal and $g:\mathbb{R}^n\rightarrow\mathbb{R}$ is a known point-spread function (say, a Gaussian). A system samples $f\star g$ at a known sequence of irregular points, $(\mathbf{x}_n)_n$. Are there any methods, analytic or numerical, to deconvolve the samples from $g$ to rev...
Does anyone have suggestions for performing the above? Presently, in my job we send signals through geological formations; however, if the rock is too dense, then we are unable to receive any signals. The signals must be sent from a battery-powered transmitter. Here is an article I've found sending neutrinos, but I a...
I am currently doing foreground segmentation in image sequences. A DCT 8 by 8 feature is extracted for each pixel. Now i want to explore dynamic block size for different scenes. Is there any rules or criterion to decide optimal block size? Thank you!
Can you please help me with this problem?? I have been working on AM335x sitara evm and my project needs self diagnosis of the captured audio from the mic through the speaker on the same board (we have re-worked on the evm to add the mic and speaker). In the application it is preferred that the user only requests for t...
It's a very basic question, but I havn't found the question on google or dsp.stackexchange.com: How does integraion of a signal change the spectrum?
As see in the picture, there are high frequency noise in the signal. How to design a low-pass filter to eliminate the noise? How to apply this filter in Matlab?
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...