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Camera setup: We have setup some stereo cameras in a living apartment. That is, the indoor environment is monitored. With the stereo cameras, wide lens (3.5mm) are used to cover a big volume.
The height from the floor is around 2.8 meters. The objects (such as mug, bottle, telephone) are at least 3 meters away. For ins... |
I know there are methods to determine the degree of polarization of multi-dimensional signals, say [Bx(t), By(t)] -> DOP(f). Is there any way to extract a DOP for given vector time series in the time domain? Thanks!
|
In image processing, up to which value of mean and variance can a Wiener filter be used to denoise the image?
Wiener filtering can be used when variance of the noisy image is low, but I don't know the specific values of variance and mean.
What are limitations of the Wiener filter?
|
We know that the square window gives the lowest main lobe width possible, and that other windows after that trade main lobe width for side lobe height. I also understand that the main lobe width is inversely proportional to the length of the DFT aperature.
Given this, is there then a physical limit to how narrow a mai... |
I have signals captured from two channels of a measurement system. There is phase and magnitude differences between these two signals.
I want to apply nonlinear smoothing to the magnitude, but the phases should remain the same after the process.
This is what I have done both channels:
Perform FFT
calculate magnitude
... |
I am looking for a implementation of Hilbert-Huang Transformation. Ideally, in Java, but any language with similar syntax would prove useful. Could someone point me to a specific implementation, preferably in Java?
|
http://www.google.com/search?q=databending
It is, from what I understand, opening images as audio, applying effects, exporting back as image and voila! glitch-art
Example:
credit
But how does it really work? How/what change in frequencies(?) affect the pixels? It can't be totally random since some of them aren't comp... |
I have a dataset which contains a large number of time-series which have been filtered with acausal Butterworth filters. For a real-time application, I can only use causally filtered data.
Is it possible to turn the output of an acausal Butterworth filter into what would have been obtained if the original time-series h... |
How do I estimate PSD in matlab ? There are lots of methods to estimate, depending on application.
I am using an experimental data to identify faults ( e.g. interturn fault of electric motor), which gives periodic/regular waveforms (pre-fault) but erratically different (or could have some periodicity) waveforms (post-... |
I am trying to implement the STFT by MATLAB, and I read somewhere that we must do zeropadding to the subsequence of the signal before windowing it with a Gaussian window. What happens if we don't do any windowing?
|
Suppose $h(n)$ is a finite impulse response which is unknown. We can feed any input signal $x(n)$ into the system and observe the corresponding output signal $y(n)$.
From this, is it possible to estimate the input signal $x^*(n)$ which when passed through the filter gives the desired output signal $y^*(n)$ ?
I think a ... |
I am making my way through this paper here, and I understand everything up to equation 16, and I am stuck here.
Long story short, the vector $\mathbf{x}[i]$ here is an $N$ by 1 vector, consisting of the complex envelopes of signal + noise + interference across $N$ different sensors, taken at some time index $i$. We ha... |
What I'm looking for is the mathematical proof of why we can calculate derivatives as a simple convolution of a mask, and how do we get that mask?
I know it has something to do with how the derivative is approximated
Can anyone direct me to some book or anything?
|
I have a chunk of data (samples) that need to do autocorrelation using FFT method. All data are real and lies between -1 and +1. What I have done follows:
Zero pad the window with the length equals the samples' length.
FFT of result from 1
Result from 2 has Real and Imaginary parts. I used them to find Power Spectral... |
I posted this question at Math Stack Exchange a few weeks ago without any luck. It occurred to me I might have more luck by posting it here. The basic question is whether there exists a known relationship between the period of a signal (i.e. the inverse of the frequency) and its corresponding power, such as that depi... |
A sampled single dimensional signal is a vector.
A sampled image is a matrix.
Is video considered a rank-3 tensor since it is a set of images stacked on top of each other?
|
When we design a discrete Gabor filter for a 1D signal, how do we determine the variance of the filter, depending on the time step of the signal? Is there any empirical theorem/result that we should know about the Discrete Gabor Transform?
|
While reading Estimation of Symbol Timing Offset, i came across an equation like multiplying a signal with its conjugate.
$$
\hat{\delta} = \arg \max_{\delta} \sum_{i=\delta}^{N_g-1+\delta} \left|y^*(n+i) y(n+N+i)\right|^2
$$
I know multiplying
(5+4i) X (5-4i) = 25 - 20i + 20i + 16 = 25 + 16 = 41.
But am not getting t... |
Musical residual noise is created when a noisy signal passes through a noise reduction algorithm. For instance, please see the paper: Esch, T; Vary, P: "Efficient Musical Noise Suppression for Speech Enhancement Systems", IEEE Intl Conf Acoustics, Speech and Signal Proc, (ICASSP): 4409-4412, 2009.
Noise reduction tec... |
While I was going through the short time processing of speech signals, I got stuck at the relationship between the frame rate and the sampling frequency.
Is there any relationship between them for an audio signal so that the change in sampling frequency will have an effect in the frame rate? If yes, what is mathematic... |
I have a question regarding the wavdecand dwt functions in MATLAB.
I am using wavdec(y,3,db4) on a size 1024 signal, which means I am trying to compute the third level db4 transform of the size 1024 signal y. If I am thinking about this correctly, theoretically the returned matrix after the transform should also be of... |
I am using CMOR(complex morlet) wavelet in Fourier space in order to reconstruct my signal and also estimate damping and frequencies of the embedded modes. there are two main parameters in cmor. the Fb and Fc. bandwidth and center frequency, with increasing the Fb i get better result for frequency estimation, but i can... |
I trained the data using BOG with SVM , i trained it , now i just want to open it through xml , i don't want to train it again and again on every run ,
I trained the svm and now loading it using its loading function
CvSVM svm;
svm.load( "trainsvm.xml" );
But how to do the same with vocabulary file which i use for clus... |
Direct Linear Transform (DLT for short) is a method of homography estimation, it solves the overdetermined linear system via SVD $$Ah=b$$to find a solution $h$ under constraint $\|h\|=1$. Actually it finds the least square solution which minimize $\|Ah - b\|$.
I understand the basic idea of this algorithm, but it is re... |
The question is:
"Weight and height measurements of a child taken every month"
Classify this signal as:
one/multidimensional
single/multichannel
I'm not understanding how this can be described as a signal and hence how can it be classified.
|
I was wondering whether we can recover a signal $x(t)$ from its power spectral density using signal processing techniques.
|
I am facing a trouble implementing the fft and appreciate your advice. My data structured as follow:
The data are retrieved from the share market for one day change in the share price. The time range from (11:00:00 to 15:00:00) this means 16200 Sec. In matlab its represented as serial so it will be from 0.4584 to 0.64... |
I am Fourier analyzing at a bunch of signals of equal length and find that $$\frac{d\ln({E[P]})}{d\ln({PT'})}$$ is constant across the dataset, where $T$ is a vector of the periods (inverse of frequencies) of each signal, and $P$ is the vector of the power corresponding to each of those periods. $E[P]$ is the mean pow... |
First, I apologize for the vague title, to be honest, I'm not even sure how to describe my question in a title. If someone would like to edit it so its more reflective of what I'm asking, then please do.
I have simulink model running in real time gathering data and doing stuff in real time (1ms).
One of the things its ... |
I have a before/after image pair with pixel transformations only (for example, grayscale, or a hue shift. No blur, border, sharpen, etc.). I'm trying to find a function such that for every pixel (r, g, b) in the original image, f(r, g, b) = (r', g', b'), (r', g', b') being the pixel in the "after" image.
I was thinking... |
Let $g[n] = f[-n]$(n is the time index). Then what will be expression for $g[n+k]$.
Is $g[n+k] = f[-n + k]$ or $g[n+k] = f[-n - k]$. Which one is true and Why?
|
Here i will first try to put my application agenda
Typically i am sampling at 8Khz rate and filtering speech components using a BPF(digital)
now i have got the 0-4Khz components in which essentially <250Hz are not speech so i will eliminate them also
now i am trying to find out entropy of frame and trying to mark ... |
I've been doing FFT's (on sound) for a while and usually use the highest intensity or the ones above average to get the spikes in an FFT result.
What would be a good method to get all the 'spikes' in an FFT. I can't use averages or peaks because the frequency response drops off. Visually its quite obvious there is a sp... |
I used the measured position to estimate the velocity using a Kalman filter.
But in simulations, the Kalman gain changes quickly and then remains constant when position and velocity continue to change. For example, position and velocity change at times 0 to 0.5(s) and 3 to 4(s), respectively. But the Kalman gain only... |
I am trying to understand the motivation for the above question for signal processing and image processing? Can anyone answer this question, or provide me with resources for this?
|
I am trying to Frequency modulate a sine signal using Matlab. I have written the following code for the same:
clc;
Fc = 5000; %Carrier Signal
fs = 20000; %Sampling Rate
t = 0:0.00001:0.002;
freqdev = 50; %Frequency Deviation
x = sin(2*pi*1000*t);
subplot(3,1,1);
plot(t,x);
y = fmmod(x,Fc,fs,freqdev,0);
subplot(3,1,2);
... |
I have a question concerning methods to find different aspects of "similarity" of signals. So, first, I'd like to characterize the signals to be dealt with:
We have a number of close to rectangular signals between values 0 and 1, starting and ending with some distinct flanks; furthermore, the signals are subject to som... |
I noticed that spline interpolation with a degree higher than 3 (everything beyond cubic splines) have a very high interpolation error, hence the prediction is mostly awful. I've come across various lecture notes, slides and Youtube videos that simply indicate that cubic splines (3rd degree) are optimal and that anythi... |
I am working in digital image processing field, recently I am studying about Image Noise, I just want to know that whether this noise removal is part of image enhancement or image restoration.
I have read some papers on it, at some points authors are saying that image enhancement from a given noisy image, and on the o... |
In the following example I have used various examples found on the web to understand fft and detecting signals with matlab.
fs = 1000;
t = 0:1/fs:1.5-1/fs;
x = 3*cos(2*pi*20*t + 0.2) + 1*cos(2*pi*30*t + 0.3) + 2*cos(2*pi*40*t + 2.4);
N = length(x);
% define the frequency components of the series
freq = 0:N-1;
freq... |
Could someone explain to me why the computation speed for the fast Fourier transform increases by padding the series with zeros to the point that its length is close to a power of 2? This is common in the matlab environment, for example:
http://www.mathworks.co.uk/help/matlab/ref/fft.html
|
Usually the filters for detecting edges have the sum of all the values in the filter matrix equal to 0.
For example: [-1 -2 -1, 0 0 0, 1 2 1].
Why is it so ?
|
I'm not sure how to approach this problem. I will describe what I am trying to do, and what some of my matlab outputs look like. I'm not trained in signal processing or anything, so please be patient reading.
I have a fourier transform of some signal, and its frequency domain covariance matrix. For example, say my s... |
I'm working on a device that produces a signal with sampling rate of 100 kHz. Every 40 ms, I receive 16000 points, from which I have to extract a frequency close to 3.55 kHz. I have developed a method that extracts the frequency of the signal very close to the Cramér-Rao lower bound, but I need to extract the amplitude... |
i'm trying to plot 1)the magnitude and 2)the phase response of an IIR eq with only its coefficients.
i've found somewhere that one can find the magnitude response starting with the transfer function :
ex: y(n) = a1*y(n-1) + b0*x(n) + b1*x(n-1) + b2*x(n-2)
... has the transfer function
Y(Z) b0 + b1 Z^-1 + b2 ... |
Consider a set $U$. My signal is a piece-wise constant "function"
$Sig: t \mapsto s$, i.e. the signal at time $t$ equals to some subset
$s \subset U$. One can see $Sig(t)$ as a stochastic process.
For a given sequence of time points $\{t_i\}$, we have sequence of
partial observations $\{ Ob(t_i) \}$. Each observation $... |
What I'll get if I take FFT of a frequency domain data.
I have two questions related to this.
For ex, x is a time domain data where X = fft(x), now fft(X)= what i'll get ?
While reading SCFDMA, data is first DFT'd and a subcarrier mapping happens then we are taking an IFFT which will transmitted using a radio. Accord... |
Price/volume data from financial markets are like signals. Can these information be applied to signal processing to generate profits successfully?
|
I am developing C++ code to do some modeling that will take as input a parametric model of the complex frequency response (amplitude, phase) and produce as output an impulse response, e.g. using the iFFT. I would like to compare this to a "windowed" impulse response that I have measured for a loudspeaker in a room. The... |
If one has a 2-D array and would like to take the 1-D Fourier transform along a direction $\theta$ degrees off the horizontal, is there a better/faster way to do this rather than rotating the image by $-\theta$ degrees, taking the 1-D FT across the columns, and then rotating the result back by $\theta$ degrees?
|
Recently I read about image noise. I read about noise characteristics, but I am unable to understand the following two points:
The sources of noise in digital images arise during image acquisition
(digitization) and transmission.
Imaging sensors can be affected by ambient conditions
Interference can be added to an ima... |
I have a time series of data with columns labeled Months and Rows labeled Years. So, my data looks as:
1900 -5 8 8 6 15 ......
1901 -3 8 8 6 15 ......
1902 -12 4 8 6 15 ......
1903 -34 1 8 6 15 ......
1904 -5 8 8 6 15 ......
The ab... |
what kind of filtering is it? How signal/noise ratio in the output mix of the duration of accumulation would look like, knowing
Signal/noise ratio=0.1
,The number of cycles of accumulation of up to 200
,Limits of change in the signal/noise ratio of 0.1-3
|
Since Fourier transforms are applicable to only periodic functions, and unit impulse function{..0,0,1,0,0..} doesn't seem periodic, is this possible?
Is it like we are considering unit impulse function to be periodic in the range $(-\infty,+\infty)$?
|
I'm surveying for feasibility techniques for a potential future project that's rather out of my past areas of expertise, and I have a fairly basic question. Of course if it looks feasible I will have to learn a whole lot more on the topic.
What is the equation to compute the number of FFTs/IFFTs per sample required, gi... |
I'm currently working on a project that involves in Speech and recognising Phonemes (in kind of real-time).
The current algorithm works by always listening for input from the Mic, and input that is above a specific threshold then this is recorded to a .wav file and sent to the server for processing. E.g.
listening...
... |
I am trying to understand the MACE correlation filter as per the paper
Mahalanobis, A., Kumar, B.V.K.V., Casasent, D.: Minimum average correlation
energy filters. Appl. Opt. 26, 3630–3633 (1987)
The equation to train the MACE filter is this
I however am abit confused about what values you use for u. From my readin... |
Investigation of the method of suppression of random noise by coherent signal accumulation
Purpose - to identify opportunities for the coherent accumulation for cases of stationary and quasi-stationary signal.
Suppose that the input is the mixture of observed signal and a random white noise (i.e. noise with uniform spe... |
Goal: I have a video stream from a webcam. I need get frequency of the green color changes on the video.
Video color space is RGB
The signal looks sinusoidal with a frequency range 0.5 - 2.4 Hz
I performed the following steps:
Capture Video frames into collection.
Processing frames by Gaussian pyramid down level 4. (f... |
Hi everybody. I'm very new in Kalman filter. Today I try to design Kalman filter to get estimated postion, velocity and acceleration from measurement position (by linear encoder). I don't use optimal input $u_k$ and my Kalman equations are:
$\displaystyle \quad\
\hat x_k^- = A\hat x_{k-1}
$
$\displaystyle \quad\
P_k^... |
I have to compare two .wav files.
I am converting the wav file to an MFCC file. Why am I getting different MFCC matrices when using different Matlab toolboxes (one of them is giving a matrix having 20 rows while the other two are giving matrices having 13 rows, and even those two matrices are not same)?
Why are the t... |
I need to append (not add) a bandlimited signal F(b) to a bandlimited signal F(a) and keep the result bandlimited without modifying the part that corresponds to F(a).
Both are bandlimited to the same frequency (for example to 1/4 the Nyquist frequency). If F(a) is 100 samples long and F(b) is 100 samples long, the resu... |
I have a current signal represented by an array in python, which I would like to add shot noise to. Shot noise is defined as
$$
σ_i = \sqrt{2qI \Delta f}
$$
where $q$ is the elementary charge, $I$ the current and $\Delta f$ is the bandwidth of interest. I guess the simplest way to add the noise would be to to add it ... |
So I am getting to grasps with Fourier transforms. Intuitively now I definately understand what it does and will soon follow some classes on the mathematics (so the actual subject). But then I go on reading about the laplace transform and there I kind of lose it. What is the moment of a signal? Why is the fourier trans... |
How can we determine whether the receive filter is optimal or not in a digital receiver?
For a decision region, how can we choose a noise value zk?
|
Could it bring any meaning? Is this kind of convolution be useful is solving anything?
|
I have an ADC that will have a guitar track as an input. I want to find which notes are hitting via the frequency domain. Given that guitar frequency range is from 82Hz to 1046Hz which sample rate (window) should I choose? I had read on an article about some kind of rule but I can't find it..
|
I am trying to implement an EKF to estimate my position and velocity states by using accelerometer measurements as well as periodic GPS (position) measurements. Basically I want to use the constantly use the accelerometer and re-calibrate it for every position measurement that is available. When there is GPS dropouts... |
I delved into the state of the art of algorithms for Poisson noise estimation in order to estimate the variance, I found that the Expectation Maximization algorithm is very used and it is very effective and easy to use,
But I have not found the matlab code, I found only one that is was used for classification,
can you ... |
I'm a new guy in image processing and computer vision, so this question might be stupid to you.
I just learned some feature detection and description algorithms, such as Harris, Hessian, SIFT, SURF, they process images to find out those keypoints and then compute a descriptor for each, the descriptor will be used for ... |
First off: I'm really new in the signal processing department and MatLab as well, so please bear with me here.
What I want to achieve is to split my audio signal into smaller 25ms frames, where the frames overlap with 50%, apply a window function, thus the overlapping (probably Hamming or Hann), and then add it togethe... |
i am looking for discrimination of speech from animal sound if you take a 150ms frame of speech and bird as shown below, what feature can i extract from them so that i can effectively state the difference
zero crossings ?
cross correlation of successive frames seems high in bird sound ? isn't it
any other feature fr... |
Is there a function in MatLab to scale a given signal to a specific range, like how soundsc does it with the input?
I have to signals, one is a wave file, using wavreader (I know one should use audioreader now) to load the file, and another file that is being read using fopen and fread.
Now I want to compare those to s... |
Image registration algorithms can be classified into intensity-based and feature-based algorithms, I want to know When and Why should i choose intensity or feature based algorithm?
another question Is there a difference between image registration and optical flow?
|
Suppose the input signal is represented as $$y=x+w$$ where $x$ is the signal to be estimated and $w$ is AWGN. I design a high pass filter $H$. What does $$H^{T}Hw$$ represent? Will it be just AWGN but attenuated or will it AWGN completely distorted. We can also assume that $\sigma_w = 0.2$
|
I am looking for -or trying to create- a filter with a piece-wise monotonic second derivative in such a way that when placed on a non-periodic input signal, changes in sign of the second derivative happen as soon as possible and piece-wise monotonicity of the second derivative (so also of the first derivative and the f... |
In various software defined radios, there are three important parameters to set when receiving: frequency, bandwidth and MSPS (million samples per second?). What does it mean to receive with a triple of parameters $(freq, b, s)$ (freq-frequency, b-bandwidth, s-MSPS)? Radio listens to frequencies in $[f-b/2,f+b/2]$ rang... |
I want to program a microcontroller in C but I am asking about the theoretical aspect of it. I think I've got an idea what I am going to do but there is one thing I can't quite get. To begin with, I will use a guitar, so there is a 82Hz-987Hz range. So roughly, I'm going to low-pass the data on 1KHz, "window" them and ... |
I am making preparations to implement FFT on a wav File.wav Only Java.
I'm reluctant to use matlab
I have a File wav stereo, 16 bits.
I need to calculate the FFT, but I have confusing.
My File has a Frequency of 44100, the samples in bytes are 10 minuts ->
(10 minuts)*(60 seconds)*(44100 sample rate)*(2 bytes because ... |
I am looking at working broadly in the area of Speech Signal Processing, and would be grateful if someone can clarify these doubts!
1) Are there any differences between Processing and Analysis of Speech Signals, in terms of the scope they provide for research? Arent both of them concerned in detecting the spoken langua... |
I am reading about different integral transforms. Fourier transform being a subject I will soon take a class on. Looking at the wavelet transform there is one term that I do not understand, that is the noiselet. What would a noiselet look like and how is it defined? Is it some basis of the wavelet transform? It is hard... |
What is the minimum period P (in samples) of the signal $e^{j(\frac{M}{N} )*2πn}$ for the following values of M and N?
M=1,N=3
M=5,N=7
M=35,N=15
I have got the answer for the first pair of values, which is 3. But I could not get the other two correct. Please help.
|
In a soundfile, I'd like to remove a frequency and all its harmonics. What is the best solution in your opinion ?
Do a loop with 20 notch filters, one after another.
// f is the fundamental frequency
for (i=0; i++; i<21)
{
file.notchfilter(f*i);
}
Question : does each notch filter create some unwanted side-effects, ... |
I was reading this paper for a project work.
We train on 1.6 million 32*32 color images that have been preprocessed
by subtracting from each pixel its mean value over all images and then
dividing by the standard deviation of all pixels over all images.
I've trouble distinguishing between "from each pixel its mean... |
I have got this question: let's suppose that I have three signals, representative of the same variable, but taken on three different component of a same device; in my case, three load signals, representative of the out-of-plane bending moment Mx of each of the blade of a wind turbine.
Is it any possible to effectively ... |
I have the following inverse system
$$G(s)= s^2 + 2s + 3$$
How do I implement it in Simulink? Note that the transfer function is only accepted if and only if the order of the numerator is less than the order of denominator.
|
I do not seem to know how to extract from a Matlab plot. For example in the plot given below, I have a simulated $P_b$ for Rayleigh fading and theoretical AWGN channel Pb plot against SNR ($\large \frac{E_b}{N_o}$).
My question is how do I get the value of $\large \frac{E_b}{N_o}$ for a particular value of $P_b$. For ... |
I am trying to recreate the following results:
...from the following data: Dropbox Link (FS40.dat)
The time is in milliseconds (frequency is 40000 Hz)
The article state that they used Complex Morlet wavelet to create the spectrogram:
"Power estimates from the averaged LFPs were calculated from
time–frequency spectrogr... |
I am learning Fourier analysis and without any teacher, just trying to read books on my own. I think I have made some decent progress but they are a couple of points which are still very unclear to me and that I can't find explained in any of the books I have.
One of the sources that I found in this document, which is ... |
Is it possible to detect if a sample of human voice by smartphone/computer(44Khz) is sourced from another electrical device or originated from a human being
I want to build a model in matlab such that, samples a vocal signal and decides whether it is a real human being talks a live or recorded, is it even possible?
|
I need some information about Stockwell transform (also known as the S-transform):
How can I implement it in MATLAB?
*Does it give you the damping ratio $\zeta$ of a signal like the complex Morlet continuous wavelet? $$x(t)= e^{-\omega_{n}\zeta}\sin(\omega_{d}t+\theta )$$
What are its advantages compared to CWT?
The ... |
How to find bandwidth of a composite signal that can be decomposed into five sine waves with given frequencies and having all peak amplitudes same ?
|
I am involved in designing a 10th order high pass filter with cutoff frequency 300Hz.
My hardware AIC3204 has a 5 biquads of direct form I, 24bit coefficient
so I have chosen the FDA (filter design and anylysis) tool of MATLAB to design a 10th order filter and convert it to SOS sections 5 such and realize coefficients ... |
I have a signal. I applied Gaussian smoothing to it and then for baseline reduction I appied Tophat filter to the smoothed version.
This is the original signal:
This is the final signal:
I read that KL Divergence helps in finding the information loss between two signals but then again a condition was that the element... |
Reading up on causality, I understand the mathematical definition, in so far as that a causal system, is one where the output depends only on the current time, and possibly the past time, but never the future times.
I have also seen that we can 'make a (transfer) system causal' by 'shifting it in time by an appropriat... |
I need to explore the algorithm of the digital guitar tuner or the digital guitar effects algorithms.
Is there any Book or articles that you recommend of?
All I know is that relate to DSP.
thanks
|
Black rectangles on these charts show same signal, which is actually color sequence from a image, taken from different positions.
It is easy to see that all images contain same pattern. Is here some algorithm which can detect these similarities, ideally implemented in Python ?
What about cut signals in multiple win... |
Let's say I have an IIR filter that is a notch type with a peak at -20db. How do I modify the coefficients of the filter so that the peak is at -10 db for example?
I know I can scale the denominator coefficients so that the global peak amplitude can be set, but it affects the whole filter, which I don't want.
Here is ... |
The Wikipedia page on sound localisation states:
Localization accuracy is 1 degree for sources in front of the listener
and 15 degrees for sources to the sides. Humans can discern interaural
time differences of 10 microseconds or less.
However, I can't find any more detailed information on localisation accuracy.
... |
I'm an undergraduate student and I am currently trying to find suitable topic for my bachelor thesis. The broader topic I've selected is "Image processing on GPU", so basically I'll be developing image processing algorithms specifically for GPU.
The problem is, I need to do a novel approach, not reinvent a wheel. Ther... |
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