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I am programmatically generating a CPFSK encoding, and between shifting from two frequencies, there is extra noise generated by my speaker. For instance, going from 18375hz to 18900hz, there is extra noise. I am changing frequency at the point where the 18375 hz sinusoidal is at 0, and beginning at 0 for the 18900hz si...
I'm making peaking equalizer in C# using NAudio (EQ-Cookbook). Can anybody tell me what is wrong with this code? It gives me noise. Sliders class Equalizer : WaveStream { float[] sFc; double a0, a1, a2, b0, b1, b2; public Equalizer(WaveStream source) { sourceStream = source; sFc = ne...
Prewitt operator is a common operator to compute image gradient in edge detection, and I knew that to apply this Prewitt operator to an image is just to convolve them together, say, the image is represent as $A$, and as the wikipedia page lays out, the Prewitt operator for *x-gradient is $$G_x=\pmatrix{ -1 & 0 & 1\\\ -...
I am trying to implement a Weighted Matching Pursuit algorithm. Essentially I would like to weight the atoms of a Dictionary (redundant matrix) with a frequency masking function, like the one specified for MP3. I have read some of the papers, and theory of frequency masking, but I am still having trouble figuring out h...
I am trying to smooth my discrete-time data points using the method of WMA. Currently, I am using n as the window size and the weight array, {n/(n(n+1)/2), (n-1)/(n(n+1)/2), ... , 1/(n(n+1)/2)}. If the y-value of each point is irrelevant, I can just simply randomly choose my size n. However in my case, I hope to reserv...
I have a complex data series, and I would like to fit a fixed number (in this case, two) of complex exponentials of the form $Ae^{jBn}$, where A, B are complex. Not interested in the phase (i.e. arg of A), just its magnitude, and mainly the complex frequency. Target length is maybe 100 samples. I would like to be able ...
I know that both SIFT and SURF descriptor vectors are normalized with (unit vectors) to deal with illumination changes, but as far as I can see, the SIFT descriptors are also normalized with the major orientation of the keypoint. Does anyone know if the SURF descriptor does this, as I cant find anything about this in t...
I need to compute the eigenvalues and eigenvectors of a 3D image Laplacian. I'm trying to evaluate the heat kernel on the 3D uniform grid (the uniform structure generated by the voxelized image) at different time values, to implement a Volumetric Heat Kernel Signature (please see the "Numerical computation" section). T...
I'd like to do a deconvolution of image. For example for convolution I'm using a $3\times 3$ mask with all elements $= 1$: $$\begin{bmatrix}1 & 1 & 1 \\ 1 & 1 & 1 \\ 1 & 1 & 1\end{bmatrix}$$ For a pixel $X$ as follows: $$\begin{bmatrix}X_1 & X_2 & X_3 \\ X_4 & \ \mathbf X & X_5 \\ X_6 & X_7 & X_8\end{bmatrix}$$ which i...
I hate this kind of questions, but I'm really stuck. I'm trying to implement Lucas-Kanade algorithm as described in this paper (see pages 4 and 5). Unlike most explanations I've seen they don't assume optical flow, but instead use affine transformation for most examples and in general refer to warp function just as $W(...
In A3E transmission, I wonder how a single antenna is used to transmit a carrier wave, upper sideband and lower sideband altogether at the same time?
I have the original signal, a .wav file and by blind source separation, i have reconstructed the signal. How can I calculate the SNR in this case? or is it good to use any other performance parameters? What should be an acceptable value of SNR?
I have the impression there is little added value for more than 100 dB of attenuation in a stopband of significant width (i.e. not just a specific frequency). What is the point at which there is little practical application for more attenuation (80 dB, 100 dB, 120 dB, ...) over a stopband of significant width? Also wh...
I read various papers about the log polar transform and its application on template matching with images and have some questions: In "Image Registration Using Log Polar Transform and Phase Correlation to Recover Higher Scale" the authors say: "In this algorithm first the sense image is downscaled by the factor of 2. L...
I've looked at various resources about Delta-Sigma modulators, and I find them fascinating. However, I'm confused on a particular point and and am looking for an explanation. In my understanding, the Delta-Sigma modulators operates by sampling at a very high sampling rate (much higher than the ulitmate rate) an input ...
How exactly is a wavelet used digitally? Wikipedia states: For example, a wavelet could be created to have a frequency of Middle C and a short duration of roughly a 32nd note. If this wavelet was to be convolved with a signal created from the recording of a song, then the resulting signal would be useful for de...
I'd like to create an image classifier that takes a set of labeled images and creates a classifier to be used on some unknown images. Most of the examples I see assume all images are the same size (m x n pixels), but my training set will have pictures of various sizes. This is a problem because training my classifier w...
I'm implementing a very basic forward and inverse DWT, based on Mallat's book. Nothing fancy, simple inefficient matlab code (proof-of-concept more than runtime necessary, so I'm not focusing on optimisation). It almost works. I'm getting a distortion along the top and left edges. I'll post link to (need 10 points to p...
Define an $n\times N$ sensing matrix $A$ by $A_{ij} = 0$ with probability $p$, and $A_{ij} = 1/\sqrt{n}$ with probability $1-p$. Does $A$ satisfy the restricted isometry property? For reference, the symmetric case is answered in the following paper: R.G. Baraniuk, M.A. Davenport, R.A. DeVore, and M.B. Wakin, "A simpl...
I have a signal which looks something like: The 12 main round peaks is what I'd like to keep. However, the peak detection algorithm I apply is picking up the in-between noisy peaks. Any strategies I could use to get rid of this noise? I tried smoothing the signal, but the noisy peaks are still there. Thanks
I would like to combine two images A and B in the following way: 1) I want to take a Fourier transform of both of them 2) For image A I want to apply a weighted filter, which gives more emphasis for low frequencies 3) For image B I want to apply a weighted filter, which gives more emphasis for high frequencies 4) I wan...
Ok I need to understand how some harmonic calculations work to apply some transforms. If I have a periodic functiion f(x). And I have the values of that function for several values of x (say x goes from 0 to 359) so I have f(0)=3 f(1)=5 ... etc So if I'm asked to take the max of those values without the first harmonic....
The problem Take 50+ of Lena images, add gaussian noise to them to the point that you couldn't say it is Lena. I insist, gaussian noise, not gaussian blurring. Now imagine a cropping window randomly moving around Lena's face (like a video camera focused on this part). Now you have a handful of noisy pictures containi...
I want to simulate a multipath fading channel, different amplitude and delay is given. I want to use delay in padding style. I read that the different values for delay like [0 .6 3.1 ...] represent the delay between the first and last signal arrival? Giving the number of paths, How to simulate it? regards
Having had little experience in PHP, I moved to C++ in-order to create a spectrogram, I wanted to create probably the simplest/easiest one for my first project, so easy infact I don't need reading/write to files or add settings to it. But, as far as I have been learning C++ I was able to manipulate audio file, only to...
In some papers, I read that the additive noise is band limited Gaussian white. How can I simulate this type of noise use MATLAB?
I am looking for a method to find the region where a function is periodic. So, for example, I have a function that is periodic in some places and not in others, I would want to know the range of x values where it is periodic. I'm trying to do this in matlab and I tried to use autocorrelation but I can't use it to find...
Assuming I have samples of a colored noise. I know the empirical Variance of those samples. I also know the bandwidth of the filter which colorized the noise (Only the bandwidth, not the actual filter). Now I want to estimate the variance of the noise if I filter those samples by a (Different) given filter. Is there a ...
I'm currently facing the following problem: I want to approximate the Fourier transform $F(\omega)$ of a (let's say, $L^2(\mathbb R)$) function $f(x)$ by calculating the discrete Fourier transform, using only a fixed number of sampling points. These points are allowed to lie arbitrarily far away from the origin. In add...
I have a periodic series of 360 values. How can I use the fftw c libraries to get the amplitude and phase of the first, second, third and fourth harmonic? If I do this for N=360 fftw_plan_r2r_1d(N,input_array,output_array,FFTW_R2HC,FFTW_ESTIMATE); I can get the 360-point Discrete Fourier Transform (DFT) where output_a...
I have implemented Dutt & Rokhlin's FFT algorithm [1] for non-equispaced data but for some reason I am getting very large errors ($E_2=0.04$ and $E_\infty=0.1$). I was wondering if there are any free available implementation of this method so I can compare my implementation. [1] Dutt, Alok, and Vladimir Rokhlin. "Fast ...
I am trying to make proper use of the circular convolution property of DFT. I was taught that the DFT of x[n]*CircularConv*y[n], would be equal the product of the individual DFT's X[k],Y[k]. On the problem im trying to solve, the signal x[n] is convolved (Circular convolution) with the discrete impulse response y[n] to...
I'd like to build a color difference metric that will be invariant to small changes in lighting conditions. In other words, I should be able to tell that a red shirt is not the same as a blue shirt, even if one image is lit differently (brighter / darker, outdoors / incandescent). Moving to YUV space and ignoring the ...
Hello i am working in digital image restoration field, recently i have studied concept of convolution, i studied that to find the impulse response/point-spread function of an LTI system, an unit impulse function is passed from the system and its output is considered as impulse response, My question is that, Why an unit...
The task that I have is to remove the annual and semiannual oscillation from a set of temperature measurements, taken over several years, by means of least squares method. I found the method described here and since I have to wait for my colleague to first make some analysis on the data and pass them on to me, I decide...
I would like to use a .WAV file and convert it into a spectogram. Any idea how to do this with Naudio? Basically the flow should be like this, I think: Convert WAV file to PCM stream Convert PCM stream to double-array and applying FFT Using the result from (2) and convert it to a spectrogram
Given a discrete signal $ \left\{ x[n] \right\}_{n=0}^{N-1} $ and its Discrete Fourier Transform $ X[k] = DFT \left\{ x[n] \right\}[k] $. I'd like to effectively compute the norm of each sample of the DFT, namely $ \left | X[k] \right | ^ {2} , k \in \left\{ 0, N-1 \right\}$ as opposed to the norm of the DFT series...
I am using HTK to recognize which word has been said. I am working on 20 words and I have for each of them 10 .wav files. Here are the steps that i followed: 1) get the MFC file for each .wav file for each word 2) for each word train it on the 10 MFCs files 3) add the grammar and dictionary and hmmlist in the HTK 4) t...
I'm trying to come up with a formula for the average energy of the signals in a M-QAM constellation. If $b$ is the number of bits represented by each point in the constellation, such that $M=2^b$. For $b$ even, I know the formula is: $$ \bar{E} = \frac{(M-1)d^2}{6} $$ where $d$ is the distance between two points in th...
I would like to build a computer vision application capable of doing robust pedestrian tracking in potentially crowded environments from a single video source. To this end I would like to find resources in order to learn as much about the current state of the art as possible. This could be: recent survey papers papers...
First of all, I'm new to DSP so excuse my simplified words. I'm testing the performance of a digital filter on a (partly) noncontinuous signal: As you can see, the signal is not continuous at some points (like it is stopping and starting over again) When I apply the digital filter to it, I get this: The filter shows ...
What is the purpose of DFT & IDFT blocks which are placed after the Analysis filter bank & before the Synthesis filter bank. The reason i am asking this question is because I tried to design Polyphase DFT filter bank & got the same output with & without DFT-IDFT block.
In the above image I have created a time series (in cyan) that is a simple addition of a major cycle (in green), a minor cycle (in cyan) and a linear trend (not shown separately). What I would like to do is filter this time series to remove the minor cycle so what is left would be the trend plus major cycle (in red). ...
So I have the following data: [[-344 13771 4600 ] [-275.2 12478 6410 ] [-206.4 19443 830 ] [-137.6 69392 3830 ] [ -68.8 143737 3780 ] [ 0 189278 16870 ] [ 68.8 184486 5090 ] [ 137.6 188466 9380 ] [ 206.4 185023 21680 ] [ 275.2 128133 1460 ] [ 344 51288 1950 ] [ 412.8 10854...
I have an x-axis which is in logarithmic scale representing frequency, I would like to have NumPoints on this axis that are equidistant on the log scale, I have the following variables: Fmin = 100 Fmax = 4000 or 8000 NumPoints = 10 What formula should i use to calculate where the NumPoints-2 points between Fmin and Fma...
Let $x$ be a $N \times 1$ vector in $\mathbb{R}^{N}$ where $M$ components are zero and the remaining $N-M$ components are standard normal random variables. $x$ may not be sparse e.g. $M$ may be small. I am interested in bounding $||Ax||_{l_{2}}$ where $A$ is a $K \times N$ matrix $(K<N)$. This made me think to look f...
I have an FDM signal from 0 to 70 MHz. The total signal contains 4 FDM signals from (0-fs/4), (fs/4-fs/2), (fs/2-3fs/4) & (3fs/4-fs). I used fvtool command in matlab to plot these signals. Then i downsampled the total signal by 4. Downsampling is time is expansion in freq spectrum. Thus on paper my new signal should o...
I have a set of signals consisting of impulses at different instant of time i.e. its not a continuous signal. What I want is a mathematically intuitive way to distinguish between any two such signals? I understand that the time of occurrence of impulses can be used to differentiate between the impulses. But what I need...
I have several analog signals with constant amplitude in time. What I want is a way to convert these analog signals to an impulse signal i.e. a signal that consists of impulses occurring at different instants of time. The conversion should be such that it should be possible to differentiate between different analog sig...
I would like to estimate the peak-to-peak amplitude of a periodic signal whose frequency components are known. This is, I have the frequency spectrum (a peak in the fundamental frequency and other peaks in its harmonics) and I would like to compute the peak-to-peak amplitude. Is that possible without reconstructing the...
Let's say you have an original image and a version of the same image that may have been convoluted with a Gaussian blur. How could you demonstrate that the Gaussian blur has been applied and calculate the blur radius?
Suppose a complex $(N \times N)$ matrix $\mathbf{T}$, multiplying a complex vector $\mathbf{x}$ with length $N$. On a given application that I'm studying, which is irrelevant for this discussion, I can get the same results of $\mathbf{Tx}$ by the following operation: \begin{align} \mathbf{W_1} \Re\{ \mathbf{y} \} + \ma...
I've always thought that image convolution is a weighted sum of the nearby pixels using the convolution kernel as weights. But from the description on Wikipedia, this is what cross correlation does. Convolution seems to be equivalent to 1) reversing the convolution kernel 2) doing cross correlation with this reversed k...
I have derived the Yule-Walker equation as shown below. \begin{align} \hat s(n) &= -\sum_{k=1} ^ p a_k s(n-k)\\ e(n) &= s(n) - \hat s(n) = s(n) + \sum_{k=1}^pa_k s(n-k) \end{align} In order to minimize the power of the error: $\displaystyle \frac{\partial E\left[\lvert e(n)\rvert^2\right]}{\partial{a_k}} = 0 \quad \te...
I am not sure if this is the right place to ask this question. I am asking it anyway.. In Aircraft Airbus 320, control inputs from the pilot's sidestick are processed by two computers for different purposes. in case of failure of one computer, other computer can handle the total processing job perfectly. My question i...
So it recently dawned on me that Bessel filters, despite being listed along with the other common types, are really an oddball that belongs in a different "class", and I'm trying to learn more about it. The rectangular magnitude response represents the ideal frequency domain response, for the transition band is zero a...
I am working in digital image restoration field. I have read all things about convolution, that, for an LTI system, if we know its impulse response, then we can find its output by just using convolution between input and impulse response. Can anyone tell me that what is the main mathematical philosophy behind it? Your...
What I have: A digital signal sampled at Fs composed of a QPSK modulation at Fmod plus some additive white gaussian noise. What I want: $$ modPower/signalPower $$ The idea behind is that I want to detect if there is a communication at Fmod on the channel or if there's just noise. What I do: Over the observation w...
I am using an accelerometer to record the motion of a shake table (a rocking table) which moves forward and backward in one direction. The signal is very noisy and decided to use Extended Kalman Filter. What equation should i use for the model?
I have a list of collected data points that I need to take a DFT of, however with the problem that about one quarter of the data points in the middle are missing, and so even though the existing data points are evenly spaced their is still a large time gap that needs to be accounted for. I looked up some things on a va...
I need to calculate BER (bit error rate) for a system using spreading spectrum DS over ultra wideband (UWB). Modulation is bpsk. The spreading sequence lengths are not equal for all users. Receiver is matched filter. I know the equations of DS-UWB transmission signal and template signal which I should use at receiver t...
Before I start my question, I should probably mention that this question might not be the right question to ask here, but I tried academiabeta, and stackoverflow, but without getting any to-the-point responses. I also noticed that there were some questions of similar interest asked here, but mine is little different an...
My idea is to find a word in wav/mp3 files. It is just one word. I have a database with a few hundred songs. I am looking for pattern matching (this is not an exact pattern) it can be an word i choose but only one. Can you recommend a pre-processing algorithm. you run the algorithm on the files? Can you recommend a re...
I have a wave file whose frequency spectrum looks like this. It contains audio, which I want to keep while removing the rest. The problem is that the surround noise changes, just those distinct voice patterns remain. I marked the voice patterns for clarity: What could an algorithm look like / a workflow in adobe aud...
For a multivariate time series $$\underline{y}(t)=[y_1(t) \dots y_n(t)]^T$$ What is the difference between the components $y_i(t)$ being spatially independent and instantaneously independent?
The Rician factor is the ratio between the mean power of the LOS path to the mean power of all other (non-LOS) paths. Increasing/decreasing this factor means increasing/decreasing the mean power of the LOS path. Does increasing the Rician factor improve or degrade transmission quality?
I am currently working on an audio watermarking project in MATLAB. I currently have a code I am using to construct a nxn 3 Band Wavelet Transform matrix. However, when I try to construct a matrix that is larger in size, I get the error "Maximum variable size allowed by the program is exceeded" or "Out of Memory." There...
Question I've been attempting to resample a GPS signal in MATLAB. I've built a few FIR filters using fvatool and from handmade transfer functions (punched out with a HP-35s). Most are kaiser-windowed LPFs, but some are least-squared LPFs. All of the filters I've built have nominal responses that should adequately filte...
I'm trying to demodulate a QPSK communication. For the moment I do: Signal detection by computing the ratio between the power of the carrier wave compared to the signal power. Initial clock synchronization with a preamble. Initial phase recovery with the same preamble. This works great with my signal but now I want...
I've been driving myself absolutely crazy trying to do autocorrelation analysis on some data that I've got. I've read from multiple sources that the ACF of a random telegraph signal / random telegraph noise is given by an exponentially decaying function where the decay constant is the average rate of switching between...
I am looking for ANY advice anyone can give me regarding on this problem. I'm looking to start a project on subtitles, basically, if we take a Youtube video of a breaking news story, subtitles are generated automatically without the need of physically inputting what the speaker is saying. Now, I understand the complexi...
I'm trying to filter some motion noise from an ECG signal. To do that I'll try to implement an adaptive filter using the LMS algorithm. I think I have to calculate the MSE of this: E[e^2 ] = E[(s + n )^2 ]+ 2E[(s + n)X ]W^T + WE[X^T X ]W^T in which s+n is the noised ECG, X (n) = [ 1, Accx (n), Accy (n), Accz (n)] are t...
I'm going to have to give a bit of context for this question to make sense. I am working on a project which includes audio source localisation in 3-D space through TDoA (Time Difference of Arrival - from here called $\tau_{ij}$, between sensor $i$ and $j$) measurements. In other words I have $m$ sensors that each have...
Just starting research in audio watermarking and have read many papers regarding audio watermarking but I am still a little confused on what exactly gets embedded into the signal. For example, one paper I read embedded an image or written signature into the audio. My question is what typically gets embedded in the sig...
I have 2 audio .wav files recorded by the the same person with the same word. Each audio file has a format of 16bit, 44.1kHz and is a mono channel. I would like to know how to compare these 2 audio files if it is by the same person. What I know is to process the files to produce a floating point result of values betwee...
How get cross-correlation peak and based on it calculate correlation score for similarity of two audio samples. SO far I've FFT two samples complex conjugate second multiply results IFFT cross-correlate with itself(autocorrelate) Thanks for any advice
I have to find a convolution of two signals $h[n] = 0.5^nu[n]$ $x[n] = u[n]-u[n-3]$ the final sum, which is correct is: $$\sum_{m=n-2}^n 0.5^mu[m] $$ note that i replaced n-k with m, that is $ m = n-k $ So, in regards to parameter n, i have to decypher the formula. The result is: $$0, n<0$$ $$2(1-0.5^n*0.5), 0\le n \lt...
I currently have a 16bit mono file. I would like to read this data to retrieve only the data chunk part and not the entire file with the headers so that I can apply fft on it. I tried finding solutions to read only the data chunk but the codes I managed to find were either not complete / seems to have errors. I am doin...
I am using wavread in MATLAB, which outputs the data in a .wav file in double precision. I am wondering how this relates to binary, such as 0's and 1's. Thank you!
It's my first attempt at 3D analysis. I am trying to segment a 3D CT data. My case study is very simple: trying to segment circular objects in my data. Do you have any suggestions on how to achieve the 3 steps: 1. First of all I would like to visualize my data in 3D 2. Then segment by value, e.g. all voxels whose value...
I have a task from my boss on calculating Ballistocardiogram (BCG) with Photoplethysmogram (PPG) using MATLAB. I would like to know how it can be done. Any kind soul can guide me? As for what I know about how the test will be like is that I will need to take the BCG data and compare it against the PPG (I know Matlab ha...
Using the figure below as an example, I wonder if it is possible that the small oscillations in the signal at the time of each step shift could be caused solely by the application of a low-pass filter to a pure square wave?
I am wondering if there is a formula for the period of the sum of $N$ cisoids of the form $e^{j\frac{2\pi}{N} i\, \omega_k}$ where the frequencies $\omega_k \in \left[-N/2,N/2\right]$, $k=0,\ldots,N-1$ and $\forall\, p,q,\, \omega_p \neq \omega_q$. This kernel arises, for instance, in the expression of the Non Uniform ...
I am trying to find a specific block of data that should hold valid audio data that I want to process. For instance, I record at 44.1kHz, but I only transmit a .5s of content. Since the buffer I store the recordings in contains both the recording as well as background noise, is there a way to isolate the sample I desir...
I'm trying to write a correlation function on an embedded device. Because its embedded, I need to make sure that the memory/CPU usage is minimum. Here's the deal: - I have a saved waveform sampled at 1 MHz - I have timing info for the waveform - I am going to be taking a new waveform at 200 kHz with new timing info -...
How to compute Image gradients? In X Direction - Kernel : -1 0 1 In Y Direction - Kernel : -1 0 1 Let's assume my input image is: unsigned char* imgImage; Could any one please share code. I am newbie to image processing. Is it possible to do in OpenCV with this same kernel? Thanks a lot in advance!
I'm trying to figure out how to implement a continuous EQ filter. I have in the past built a parametric EQ and then built a 10 channel EQ using 10 of these parametric filters. This gave nice results. However I'm now trying to create a continuous EQ filter. Is this possible with a parametric EQ? By selecting the Q f...
Okay, this might be a very basic question, but I'm completely new to the field. I have a $40$ Hz sine wave ($1000$ samples, $fs = 1$ kHz), which I have amplitude modulated to a $4$ Hz sine wave ($1000$ samples, $fs = 1$ kHz). In the time-domain, the result looks like I expected. However, when I calculate the FFT, there...
Everybody explains that white noise has all frequencies equally strong. But, this immediately means that Ultraviolet catastrofe inevitably happens if power > 0 stays constant at any frequency and, what is similarly unacceptable, White noise is identical to single Dirac impulse since delta pulse is a constant in the F...
I see the following expression from http://en.wikipedia.org/wiki/Scale-invariant_feature_transform The quadratic Taylor expansion of the Difference-of-Gaussian scale-space function, with the candidate keypoint as the origin is $$D(\textbf{x}) = D + \frac{\partial D^T}{\partial \textbf{x}}\textbf{x} + \frac{1}{2}\textb...
A bit of background information: I plan to start experimenting with digital signal processing on microcontrollers. The particular microcontroller I plan to use (TI Stellaris LM4F230H5QR, from the Stellaris Launchpad) has a 12 bit ADC integrated on it, while on the other hand, all DACs I that look like possible candidat...
I'm doing research on how we can do matching between 2 audio files using Android to see if they have the same content or not. After doing lots of research, I have concluded that I must do 2 main things: FFT Correlation Am I on the right track?
I have a file which is a recorded voice of a single word. The format of this audio file for me is unknown. Could you help me to decode or figure out how it is compressed? I was trying to open it as a PCM (8/16 bits), VOX, uLaw, aLaw, but it does not work.. all I can hear is a loud noise. Audio file example is at: http...
Is it possible to compare two audio samples(meaning find word from one in another) realtime with cross-correlation?
I am reading several papers applying anisotropic diffusion in the transform domain like Shearlet, curvelet. All of them have got excellent results. The physical meaning and principle of applying anisotropic diffusion in space domain is quiet straightforward -- the gradient. I am confused as to the physical meaning and...
Let's say I have two images that are parts of the same bigger image, cut using two masks (for simplicity, let's assume that the masks are just rectangles of the same size). The images are presumed to overlap. The concrete examples are: image stabilization (two different views of the same background) and object tracking...
Update This question is purely theoretical and should not account for any psychoacoustic effects, or practical implementations. Perhaps the question should be rephrased to "How long will it take to apply the smoothest possible fade, in a single precision system"? A sine wave is generated within a 32-bit float system...
I have a device with an accelerometer and I want to make it as precise as possible (using algorithms). Currently no DSP algorithm is running - I am reading values as they are. I've made 3 tests - in each I rotated the device around one of the axes and recorded the values coming out from accelerometer. It would be perfe...
i have a problem in ofdm modulation .... there is a stream of bits that is converted from serial to parallel and is applied to an IFFT block. based on the definition of IFFT the output should be an complex valued number. how does it represent the real and imaginary parts in the digital domain( bits 0,1) ?
I have a set of songs for which I extracted the magnitude spectrogram using a Hamming Window with 50% overlap. After extracting the spectrogram, I did some dimensionality reduction using Principal Components Analysis (PCA). After reducing it to lower dimensionality, I reconstructed the spectrograms from lower dimension...