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I have applied a complex wavelet transform on two signals :
a first one
a second one that is a concatenation of the first signal and another one
For the 4th coefficient (as an example, i've got the same problem for all of them), i get the following graph for its module (1- is blue, 2- is green) :
What could explain t... |
I want to separate out the components that I need to evaluate the pronunciation of a user, so I am dealing with the energy/stress/volume and intonation or pitch.
Focusing first on the pitch, is there any way to get the average pitch of the wave file? I would like to get the average of the user's input with respect to ... |
I`m confused on how to calculate a signal power. For example, suppose I generate a complex: signal $$x[n] = (1 + j3)e^{j2 \pi k n/N}$$
The DSP theory says the average power of a periodic signal with period $N$ is:
$$ \frac{1}{N} \sum \limits_{n=0}^{N-1} \left| x[n] \right|^2$$
Does that mean the signal power is indepen... |
I have just started my first proper course in DSP, and am a little confused about a statement in my textbook. The book states:
"Notice that when the Nyqust-frequency is defined as $\pi/\Delta t$, we will with $\omega_m = 2 \pi m/N \Delta t$ where $m = 0, \pm 1, \pm2, ...$ have $\omega_m \neq \pi/ \Delta t$, unless $N$... |
I need help with iris localization. I have applied the imfincircle function to locate it, but it only locates the inner boundary of the pupil and does not find outer boundary.
How do I locate the outer boundary?
|
I'm asked to design FIR filter, an ideal low pass filter.
I'm given with pass band frequency (wp) and stop band frequency (ws). Now, I have to choose Ns points from stop band and Np points from pass band. Taking the inverse will give me impulse response.
The problem is, Ns points are equally spaced and Np points are eq... |
Let's say we have 2 signals of frequencies 4 kHz and 8 kHz.
If we sample both signals with sampling frequency 10 kHz and pass the signal through a low pass filter with cut off frequency of 10 kHz, what frequencies will be present in the output signal?
|
I have two wave files each containing a sample of a note. I wish to compare, using a program, each sample and present how loud the first sample is with respect to the second if each sample is played back to the listener with the same physical conditions (volume, distance from speaker etc.). For example, if the second s... |
Assume a signal $x[n]$ with $N$ samples and its $N$-point FFT $X[k]$. Assume also the simplest power spectral density estimator:
$$ S_x[k] = \frac{\left| X[k] \right|^2}{N} $$
With this estimator, the average total power can be evaluated as:
\begin{align}
P_x = \sum \limits_{k=0}^{N-1} S_x[k] \Delta F
\end{align}
where... |
I want to extract Audio Features using RBM (Restricted Boltzmann Machine). For this, I am giving the spectrogram (PCA whitened) as an input to the RBM. For each audio file, The spectrogram is a matrix with no. of columns fixed but with different number of rows for each audio file. My question how can I train my RBM, or... |
The concept of frequency spectrum exists because it is possible to sum sines and cosines of various frequencies together to make any periodic signal. The frequencies of these individual sinusoids make up the spectrum of the final signal. Now when it comes to image processing we treat image as a signal similar to sound ... |
I have an image of size 160x160 of pixels. I am trying to convolve this image with two Gabor filters of different scales (7 and 9 respectively) but with the same orientation. I red in an article the following phrase :
The maps (the results of the two convolution operations) have the same dimensionality but they are the... |
This is just a question out of my curiosity, normally we do not do this
I have a x(n)=[1 2 3]
y(n) = fft[1 2 3 0] = 6 + 0i -2 - 2i 2 + 0i -2 + 2i %zero padding
y'(n) = fft([1 2 3]) = 6 + 0i -1.5 + 0.87i -1.5 - 0.87i
I used Matlab to generate these results.
My question is there way to obtain y'(n) from y(n)?
|
This is the spectrum of the signal:
And this is the signal:
On the FFT spectrum you can notice the high amplitude peaks and I think these frequencies are the main ones in the signal:
I think that the "low" harmonic must be clean, but I have the problem that low amplitude components are placed in every frequency inte... |
I saw a lot of use of tensor in signal processing. What is the intuition behind it? Is it simply a common representation for audio (1D), image (2D) and video (3D) signals?
|
A question asks for the energy and power of the following signal;
$$x[n]=\text{sinc}\left(\frac{n}{5}\right)$$ given that $$\Omega_c=\pi/5$$
Energy is $$E=\sum_{n=-\infty}^{\infty}{\left |\text{sinc}\left(\frac{n}{5}\right)\right|^2}\rightarrow \frac{1}{2\pi}\int_{-\pi}^{\pi}\left|X_c(\Omega)\right|^2d\Omega$$
My resul... |
I have got a signal in frequency domain. This is a frequency response function from software, so I can do nothing about it and have to leave it in frequency domain. I want to transfer the data to 1/3 octave spectra and I need some help on this.
In my opinion, I just need a couple of band pass filters which in the frequ... |
I am having difficulty in calculating the SNR for the following scenario:
In a cellular network
Transmit power=36dBm
Noise spectral density=-174dBm/Hz (AWGN)
Bandwidth=10MHz
I have some other parameters such as pathloss and Rayleigh fading, but I think my biggest problem is the conversion of dB to power.
|
I am generating a sine wave from a DAC at a constant frequency. If each full cycle is specified by N data points what is the relationship between the accuracy of the rendered sine wave and N?
N=1 will obviously be pulses. N=4 (peaks and zero crossings) will yield a triangular wave of sorts. Is there some equation that ... |
I have an unusual low-pass filter problem that I would like to realise with a passive RC network (Note 1).
It's easy enough to find prototype filters for passive RLC networks, or for the various active RC filters.
I am having difficulty finding prototype filters for the classic filter types (e.g. Butterworth, Chebychev... |
I have a dataset which records the arrival time for a series of photons.
I am using lomb-scargle method of scipy.signal to try to find some kind of period.
Since the arrival time is random,how to express the intensity of the signal?
If I bin the dataset and let the photon counts in every bins be the intensity , the da... |
The Auto-Correlation function of Walsh-Hadamard codewords does not have a good characteristics.
It can have more than one peak and thus, the Walsh-Hadamard codes do not have the best spreading behavior or correlation property.
The cross-correlation function of the Walsh-Hadamard codewords can also be non-zero for a num... |
There seems to be a number of different ways for telecommunication. What is the most recent (and maybe standard) technique used today? It seems the simple AM/FM modulations are definitely somewhere in the past, then there are these spread spectrum systems... All hints welcome :)
|
I read the DAFX book by Udo Zölzer about the distortion effect at pages 124-125
and it says that suitable simulation of the distortion is given by the function:
$$f(x)=\frac{x}{|x|}\left(1-e^{x^2/|x|}\right)$$
Can someone explain this formula and what kind of signal we get?
From what I understand 'x' is the sampled si... |
I'm trying to suppress the sidelobe of OFDM but having troubles showing its spectrum in Matlab. After IFFT, I get a time-domain signal, but in which way I can get its spectrum(or PSD)? I want to analyze the out band power emission, with rectangle window in the time domain the out of band power curve decrease slowly, so... |
My understanding if a frequency in my signal doesn't line up exactly with a bin, it is smeared over a few bins to the right and left.
How can the instantaneous magnitude be determined? (do I have to worry about the windowing function?)
|
Based on a paper in here, they have proposed a method to optimize morlet function parameter. I have tried to implement their technique, but I can't get rational results. any idea?
In here you can see what they have done briefly:
Imagine you have a signal with following frequency and damping pairs :
$(f_i ,\zeta_i), (f_... |
To get a panning effect I know that there is a need for a Circular Matrix.
The angle is can be set as a default or I can let the user to select the angle.
So if the angle is $\theta$, the circular matrix will be:
$$
\begin{pmatrix}\cos \theta & \sin \theta \\ -\sin \theta & \cos \theta\end{pmatrix}
$$
and the gain i... |
I want to know if there is a generalized way to select a signal from a set of signals. Let me clarify it: I have a number of signals $S = \{S1, S2, S3,...\} $.
Some of the signals are not important to me like, all having all zeros for data, or all same values, or only one datum comes in at the first occurrence then re... |
http://astarte.csustan.edu/~tom/SFI-CSSS/info-theory/info-lec.pdf compares a bacteria DNA vs a random string:
My first step was to generate for myself a "random genome" of
comparable size to compare things with. In this case, I simply used
the Unix `random' function to generate a file containing a random
sequen... |
When you plucked on a string, the string is vibrate and except the fundamental frequency,
there are an integer multiple of the fundamental frequency, which are the harmonics.
I know that the HPS algorithm is a tool to find the fundamental frequency, but
I don't understand the downsample part. How can I know how many... |
Is there any speech enhancement algorithm library?
I need to process a sound to voice recognition, and I want to remove noise from audio source.
|
I don't know if this question is related to DSP, but I'll give a chance.
Introduction:
These days there are a Guitar Synthesizers which allow the guitarist
to play another instruments like: brass, drums, strings, pianos and other guitar models
from his guitar.
It possible to turn any guitar with the MIDI Pickup, that ... |
I am studying a deconvolution problem for which observations include noise. I am considering using one of several common regularization methods, including Tikhonov's solution, truncated Singular Value Decomposition (SVD), or Wiener filter.
A manuscript I am reading, Riel and Berkhout, "Resolution in seismic trace inver... |
I want to build a navigation system with gyroscope/odemeter and gps. I have konwn there is a famous method called kalman filter for this application. And now, I have done some experiment in matlab with kalman filter to get hide state vector which included vehicle's current postion/velocity/heading etc. But out of my ex... |
I have the following matlab code
N = 500;
M = 32;
K = N + M - 1; % Length of full convolution
x = randn(N);
y = ones(M);
% first one: convolution
r = conv(x, y, 'same');
% second one: multiplication
z = ifft(fft(x, K, K).*fft(y, K, K));
idx = round((K-N)/2+1):round((K+N)/2); % Indices of the central portion
z = z(id... |
Are there any differences if you use Maximum Likelihood or Maximum a Posteriori to estimate the Point Spread Function for image deconvolution?
|
I have an image with size $256*256$ then i get the Fourier transform of the image, If i only take the central region of the Fourier transform whose size is $64*64$ and get the inverse FT ,What is the effect on the output image$(64*64)$?
Down-sampling, Blurring, or both.
|
I'm trying to implement a QPSK Mod/DeMod system in C++. So far i have a bit generator that generates N bits and stores it into an array:
srand(time(0));
for (int i = 0; i < N; i++)
{
bitstream[i] = rand() % 2;
}
Now I parse through the array to group bits of two and send those two bits to get modulated.
for (int ... |
Norbert Wiener came up with the Wiener Deconvolution in 1949 or something like that. But what happened after that? When was Richardson-Lucy deconvolution developed for example? And what happened between that and Adobe's deblur demo in Creative Cloud?
|
I am having some trouble understanding how a digital signal gets converted back to an analog signal.
Why is the "sample and hold" effect a problem?
|
Why does image deconvolution still work with image without sharp edges? Take this image for example:
|
I capture signal (sound) in 44100hz and I need get response if there is peaks at amplitude, actually. Is this possible? How I can do this in real-time?
|
Is there an accurate way to create an aliased image from the Fourier transform of the original image?
in other words, i have the Fourier coefficients of an image and i want to make down-sample in frequency domain.
|
I am relatively new to signal processing techniques and Matlab and need a bunch of test data in the form of white noise, as defined on the Wikipedia page, with constant, flat power spectrum.
I tried the matlab built-ins wgn(m,n,p) and randn(m,n) to get vectors of white noise, and then used different combinations of th... |
My computer vision / image processing professor uses the term "Label", to represent part of an image, but cannot explain in a way that makes any sense to me what it is. She says this is a somewhat common term, but it is not clear what it is. Its somewhat similar to a pixel, but different. I can't find a definition anyw... |
I have doubt, please clear.
A transform is said to be unitary transform when the product of the transform matrix and its conjugate transpose (Hermitian) matrix is equal to Identity matrix.
DFT for N= 4 gives
c =
4 0 0 0
0 4 0 0
0 0 4 0
0 0 0 4
Thus, it satisfies the... |
I am trying to implement a Chebyshev type I low-pass IIR digital filter in C. I have got the SOS Matrix and scale values from Matlab.
What is the direct equation or algorithm to implement such a filter?
|
I am having a hard time understanding why my textbook takes the squared magnitude of the calculated DFT coefficients.
I know in DFT, you would normally take the magnitude to get the spectrum, how come it is the squared magnitude which gives me the spectrum when iam using goertzel?
|
If $x[n]$ is a discrete random sequence with uniform distribution having mean $\mu = 2$ and variance $\sigma^2 = 3$, if passed through a moving average filter to get an output of
$$\frac{x[n] + x[n-1] + x[n-2]}{3}$$
what would be the the variance of the new signal?
I am getting 2 answers like 1 and 3. Which is correct... |
As I understand it, in image processing, when it is desired to apply a filter of some sort to an 2D image, a kernel is applied to all the pixels of the 2D image in a process called convolution. Median, Gaussian, Bilateral, all are examples of kernel. These kernels are applied in a similar fashion to that shown in this ... |
Currently I am working on chord recognition, and get confused on several aspect. One of them is choosing windowing for reduce the spectral leakage.
from what I read, hanning/hamming function is a good choice for windowing, but currently I read about welch windowing function, and its quite good either.
So, can somebod... |
Hi guys I'm currently studying for finals and I came across a square pulse problem.
I ended up getting this:
I'm wondering if I can simplify this to a sin function. If so how? And please correct me if I did the problem wrong in the first place.
|
I am using MIT-BIH arrhythmical database where I have a digital signal of 1200 Hz so 1200 samples per second.
This means that analog filters have already applied to remove the frequencies over Nyquist frequency, so no aliasing.
However, I want to take equispaced sampling, every two sample, simply by the following in Ma... |
I am new to seismic data processing and I really have no understanding of the term 'phase'. So, could anybody give me a simple explaination of the term?
Thank you
|
I am doing some post-hoc analysis of a dataset consisting of a series of movie frames that are contaminated by a strongly periodic artifact. I would like to remove this artifact from my frames.
For ease of plotting I've just reshaped my array M of pixel values to [nframes, npixels], then averaged over all pixel values ... |
I am having some troubles understanding the algorithm.
My algorithm looks like this.
how come am i able to compute the DFT coefficient using this algoritm.
As far as i know
$$
X[k] = \sum_{i=0}^{N-1}x(n)W_N^k
$$
which using Goertzel is the same as $$X(k) = y_k(N) $$ and since $$y_k(n) = v_k(n) -W_n^kv_k(n-1)$$ a... |
I am studying digital signal processing and our professor gave us a task of modulation and demodulation and designing an appropriate FIR low-pass filter in Matlab.
I have done the modulation and demodulation but I am stuck on the filter designing. I want to know how the coefficients of filer are determined.
|
When you perform a discrete Fourier transform (DFT), and compute the DFT coefficients, $c_k$ and $X(K)$, what do these values tell us about the signal it came from?
|
In radix-2 FFT, why is the bit growth equal to $1+\sqrt{2}$ per stage?
I want to know the logic as to how that value has been obtained. Is it just empirical or has it been derived?
|
How is iterative refinement is applied to the estimate obtained by Lucas-Kanade algorithm? I am familiar with the following steps:
Estimate velocity at each pixel using one iteration of Lucas and Kanade estimation
Warp one image toward the other using the estimated flow field
Refine estimate by repeating the process
... |
Actually I want to calculate the cross correlation coefficient of any signal on comparing with all mother wavelets like Morlet, Daubechies using MATLAB.
|
I want to calculate delay between an input and an output audio signal of my audio processing system. The input and output signals are available as signed 16 bit integers. To try out, I tried the autocorrelation of the input signal with the following numpy commands:
import numpy as np
import wave
wfp = wave.open('test.w... |
First off, I apologize for possibly incorrect use of jargon, as I am new in this area.
I have a type of signal that I am assuming can be described by an arbitrary number (let's say $n$) of basis vectors, which are linearly independent. I am measuring a signal through one sensor, which gives me a vector of amplitudes. ... |
One way to measure the frequency response of a filter is to input separate cosine/sine wave at the desired frequency, measure the steady state output, stack the inputs/outputs into complex arrays, and divide. A more efficient way is to use the DFT, and somehow we get away with just inputting one sinusoid.
Input $x_n =... |
Thanks porten!
Thanks Bitstream!
But our professor has said that design your own filter!(not used built in function of matlab).
I am designing my own filter.The problem is still there.First, i have to know filter's order(N). One way is to find from its transition width. The pass-band edge frequency is 500Hz and stop b... |
I currently have a .wav signal that was recorded at a 48 kHz sample rate, with a central frequency of 5.260 MHz and bandwidth of 4 kHz. I'm trying to get some information from this signal, specifically the location of the frequency peaks using matlab, which I would expect to be around 5.260 MHz. The code I'm using to g... |
This might sound silly, but I have a language barrier type of question, and I was hoping someone could clear up this misunderstanding:
I've seen the term "early vision" several times in papers I've read, but the term often doesn't accompany any later publications to compare with when using the term. At first I assumed ... |
I understand FIR & IIR digital filters are often implemented in fixed point DSP hardware. I have also seen them implemented in software. Do companies make special purpose digital filtering chips that use floating point precision (float) and (double)? When fixed point hardware is used, what are the choices for the bits ... |
When using Butterworth filter in MATLAB, if I chose an order too high, sometimes the filtered signal would grow without bound to infinity, and if I decrease the order, the problem disappears, has anyone encountered the same problem, and why?
|
I have a mic having sensitivity chart as below which has relative dB on y-axis showing
1V/0.1Pa
what does this convey how to read my mic's sensitivity
a document on microphone sensitivity from analog says the microphone sensitivity is measure with 1Pa pressure as reference
confused in half way after seeing my datas... |
The ripples in the amplitude spectrum of a linear spectrum is undesirable. I desire to obtain a flat spectrum response. Hence, how can I remove these ripples? Also, which factors (like time bandwidth product, chirp rate) are affecting the ripples, and how?
I designed a chirp with a frequency from 10 to 90 Hz in MATLAB.... |
To be correct I want to determine variance of white noise before integration (brown noise is white noise after integration). For example, I have a following model (in MATLAB) of noises acting on my sensor:
How can I determine variance (power) of 'white noise 2' from samples (total noise)?
Of course noises are band-li... |
What does the squared magnitude of the DFT coeffiecient tell me,
where do I use it?
what does it define?
$|X(k)^2| = ???$
I am curious to know because the Amplitude is given by the formula $A_k = \sqrt{|X(k)|}$
So it cannot be the amplitude.
It would help me alot if anyone could help me.
|
I have solved the problem of finding the filter's order. The order, $N$ is 144 (N= 4/Normalized of BW).
Since the $F_s$(sampling frequency) is 7200Hz, $f_p$(pass-band edge frequency) is 500Hz and $f_s$(stop-band edge frequency) is 700Hz.
Now, I want to:
Create an ideal impulse response i.e, sinc function and then
App... |
I am very new to Fourier analysis, but I understand that through the use of the Fourier transform a signal in the time domain is displayed in the frequency domain, where frequency values are normally displayed along the x-axis, and amplitude is displayed along the y-axis. However, at one point in the textbook I am usi... |
I have ordered a USRP kit to do radio signal processing in software. Is there a way to use the same device to process sonar data? What would I do upconvert the sonar to 50 MHz?
|
my guide has told me to get familiarized with state estimates and kalman filter ....the problem is that:
1) I am totally new to this topic and finding it really difficult to understand due to lack of materials. so, can someone give some suggestions on how to proceed.
2) I am also suppose to incorporate delayed data int... |
I'm creating a program for localizing types of barcodes.
(I know plenty exist, that's not the question :) )
I'm focusing for the time being on 1D barcodes.
The difficulty I have is that the barcodes I need to be able to locate could be situated anywhere within an image with other information (text/images/...), inequal... |
I'm trying to implement some code for watermarking on audio based on a scientific paper. I'm stuck in the part of the pseudo code where they calculate the fourth order cumulant of the approximation coefficients of DWT. I have no idea what a cumulant is even when I tried to understand the theory.
I found the following ... |
I have a corpus of several thousand sounds. Now I pick a sound sample S, and I want to find in my corpus the sound S' that is the most similar to S.
The application is for music, for a composition that I am working on :) I found a lot of questions which are all related to speech recognition. The problem here is differe... |
Can anybody tell me the advantages of the lifting scheme and why it is better then a normal filter bank?
|
I have a sin wave of 16Hz, sampled at 2048 samples persecond. I performed the FFT of the sequence and did a spectrum plot. My plot gave me two spikes at both ends of the graph. But i was told the spike is meant to be at 16. how exactly am i supposed to do the plotting in order to get this exact spike?
|
Dear all,
Please find attached image,
I am not able to understand why is the underlined term is to be included.
Shouldn't it be simply the exponential term?
|
Following this link:
http://software.intel.com/sites/products/documentation/doclib/ipp_sa/80/ipp_manual/GUID-5457048E-3C59-43E4-8A99-FB3873DCB769.htm
I have produced the following code:
const int len = 512;
Ipp64f pInputSignal[len];
int tapslen = 27;
int numIters = len;
IppsFIR... |
Some animals are specially trained to repeat human phrases, others just make sounds that people interpret as speech.
In this video a rooster seems to say "Cristo Viene" (Christ is coming).
Taking the spectrogram of that rooster to analyze the formant position of the vowels I can't find any matches (I was using a forman... |
I am an experimental research engineer working on a Laser experiment. The details do not matter but due to practical limitations my data is influenced by the following transformation during data acquisition, and I was wondering if there is a mathematical method or an algorithm to "undo" its effect on the captured signa... |
If I have a measured signal $y$, true signal $x$, and a convolution matrix $A$ that is a Toeplitz but not circulant matrix, I can write the convolution as
\begin{equation}
y = A x \ .
\end{equation}
However, I would like to analyze the system using diagonalized coefficients of the Fourier transform. A re-formulation su... |
When I searched for octave, I listened to the fact that if you take an octave band the lower cutoff is double the upper cutoff.
Where is this name octave coming from, that is, where does the figure eight show its significance in the above statement?
In the same way, if the above is clarified, I have some doubts on 1:2 ... |
My doubt is in regards to the usage of the function cceps in Matlab.
The function is giving me the cepstrum for the entire speech signal of 3 or 4sec. This works well when I play back the signal after taking the inverse (using theicceps function). However, in order to analyze the given speech signal, we are supposed t... |
I am having some problems understanding how the Fourier transform method is used to determine the FIR filter.
As far as i have understood, you start by using the ideal impulse reponse for the specific filter design you want to have. Those are given in this table
As the the ideal impulse reponse contains infinite impu... |
If we filter out ideal white noise using an ideal LPF of cutoff frequency 10 KHz and then sample it at 30 KHz , is the resulting discrete signal statistically independent? I would like to know the statistical behaviour of the output signal.
I was attending an on-line test. My answer was told wrong.Below is the snapshot... |
I am detecting vehicles from the video (taken with a camera), and the vehicle detection part works fine.
My problem is that I should always extract only one representative frame per vehicle, and not the whole sequence.
As an example, if I have a few seconds of video (and possibly multiple vehicles), my approach current... |
I have a wav file recorded from a hydrophone. I am not quite familiar with signal processing so I really need some advice from you.
I have gathered the information which looks useful and related.
sample rate = 250 000 Hz
Receive sensitivity of hydrophone = -180 db re 1V/μ Pa
Peak-to-peak voltage range = 5.64 V
Pre-ampl... |
I have a simple problem I would like to figure out for fourier analysis of seismic data.
Let us say that we have a signal $z[n]$, of length $N$. If I take its (same size) FFT, I will get $Z[k]$, also of length $N$.
I know that zero-padding will not improve my resolution, however, what I am trying to figure out is: If... |
I've been doing some work with creating visual output on oscilloscopes using audio; here are some examples:
http://www.youtube.com/watch?v=UJ8-vySqEAI
This is using Max/MSP to read from a buffer of 3d geometry as audio output to an oscilloscope screen. I'm interested in doing transformations on full frames of video as ... |
I would like to create a very simple system that allows a user to train (once or multiple times) a spoken phrase (from 1 word to a whole sentence.)
Then the (same) user would speak the phrase back, and the system would score how close to the original the phrase was matched. If certain words were missed or incorrect, th... |
I would like to know how to compute the statistics of the discrete Fourier transform of a noise signal. To illustrate what I mean, I will first explain in detail a computation I have managed to do myself.
Suppose we have a discrete time series of values $x_n$ with $n$ from 0 to $N-1$. Each $x_n$ is a random variable, u... |
Here Above are the frequency responses of different size microphone(condenser)
what actually is the relation between the microphone capsule size and frequency response ?
The conclusion is that a large diaphragm microphone will have a more limited frequency range than a small diaphragm. This is illustrated in the gra... |
BM3D is considered state of the art in Image Denoising. I am trying to implement the algorithm in Matlab. Right now, I am done with the 3D wavelet transform in the first step, and to be precise, the equation 3 in the aforementioned paper. What I want to do now is to carry out the equation 5 on my de-noised patches in ... |
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