instruction stringlengths 31 27k |
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Let's say we have the following matrix:
0 0 0 0 0 0
0 1 0 0 0 0
1 1 0 0 0 0
1 1 0 0 0 0
0 0 0 0 0 0
0 0 0 1 1 1
0 0 0 0 1 1
0 0 0 0 0 0
I want to calculate the size of every connected region of 1s and also get the corresponding matrix elements of each region. The size should be computed as (width + heigth) / 2.
In the... |
I want to use Robust PCA for image inpainting, i.e. to estimate anomalous (bad) pixels. I assume that the bad pixels are known (mask).
I know how to calculate the Robust PCA using Alternating Direction Method of Multipliers but if somebody has already an implementation in Matlab I would not say no to sharing it. ;) But... |
I would like to know whether this can be done and if then how would this feat be acomplished in Matlab given a filter calculated with Remez?
h = remez(...);
If this is not possible then I would appreciate an explanation whether this can be done approximately (approximations of the zeros and poles) or whether this is si... |
Required to find symbol error rate vs $E_b/N_0$ for 4QAM, 16QAM & 32QAM. Thought that SER & BER are the same but did my research to find that BER is $1/\log_2(M)$ of SER. Could you please confirm this?
Also found SER for:
4QAM to be: $\displaystyle \text{erfc}\sqrt{\dfrac{E_b}{2N_0}}$
and that of 16QAM to be: $\disp... |
I will be using one central node/latop to play a tone and will have a second node/laptop to record and read the sound and be able to say this node is on the left or right. So trying to figure out the best way to distinguish between the two tones. I want to have a tone set for left and one set for the right. I want the ... |
I have designed few shelving and peak filter. And I have calculated the a, b coeficients for several gains for these filters. Now I would like to create some kind of characteristic in Matlab so I could present my design of these filters?
Something like this for shelving filter
|
I saw in some codes people always taper the ends of a signal using a cosine taper.
Is that necessary and why?
|
What does End Effect in signal processing basically means? I would like to see some reference book or papers if you suggest me.
Thanks
|
I need to implement people counting(head count) into one of the product.
I am able to detect faces and count them but only when the persons are actually looking into the camera, if they are not looking into it they will not be detected.
So, what I want is to an idea about detecting human rather than detecting only face... |
I apply an FIR filter passing the band around 50/1000 and attenuating the other three frequency components 30/1000, 70/1000 and 110/1000.
Both filters are generated using the Remez algorithm. The first filter is of length 300 and the second is of length 100.
Why is the beginning of the filtered signal attenuated and w... |
I'm new to matlab sound processing and I am trying to create a 2D array for the sound code below. I have to make the sound have 400 samples(i.e 10ms) at 20hz, but having trouble getting the code to work. Any help would be greatly appreciated.
The steps are:
(1) Construct a two-dimensional array (n x 2 array, i.e.,... |
I need to identify certain features of the audio signal recorded from microphone in stethoscope. These sounds are only samples i've found, but the final signal will be probably a bit noisier (maybe not, i don't know yet). Below are examples of three signals. Normal one and one of the person sick with atrial spatial def... |
TV is L1 norm of gradient of an image. So we've to find gradient of the image (which is still matrix, right?). Then take the sum of absolute values of the gradient matrix (So now it must be a scalar, right?).
Can I simply use imgradient function in matlab for first step? It gives two values magnitude and direction, so... |
I have a play function that is called twice, each plays a channel (a column of the two dimensional array) in both speakers. What I am trying to do is to play both channel at the same time. How do I silence a channel and set the the elements in the corresponding column to 0. I also need an output array that im trying ... |
Problem: The signal $cos(2\pi14100t)$ is sampled at $F_s = 400 Hz$. It is then upsampled with a factor 3 and then reconstructed ideally with a new frequency $F = 500 Hz$.
I now want to find the new signal that is created.
Incomplete solution: With the frequency before sampling $\pm 141000 Hz$. After sampling the normal... |
I have 2 sounds that I created and trying to make them beeps. Maybe 4 or 5 repeated beeps but have a pause in between each beep but make it still part of the signal. I think I have it so that the sound is a single frequency but just need make it a repeated with pauses and still part of the same signal. Any help would b... |
Could some one tell reasons to use the MATLAB imreconstruct function when processing images? I already have studied the topic and what the function actually does to the image. However, I cannot figure what we would use it for. What output should we expect from calling doing this function?
|
I have a .pgm format 480 * 640 gray scale image named 'columns.pgm'. Using PCA (principal component analysis), by preserving 40 principal components I compressed original image with 1218 KB to 301 KB using a Matlab function which I wrote. Now about the entropy. The entropy of the original file was 6.93 and of the compr... |
I was going through some of the basics of fourier series and fourier transform. And I came across one topic "Fourier transform of periodic signal". But I am not able to understnad why we need to go for fourier transform because we already have fourier series to take care of periodic signal.
Any help is appreciated.
|
I normalized the spectrum of a time series and windowed the spectrum, then something strange happened. The code below can run in MATLAB or Octave.
figure;
n = 2e3;
t = 1:n;
m = n/2+1;
f = linspace(0,1,m);
x = rand(1,n) - 0.5;
y = fft(x);
subplot(2,2,1); plot(t,x,'k');
axis tight; xlabel('t/s'); title('original signal ... |
assume that I have a set of image and segmented image of a object in image (training set). Now, I want to measure features of object in image based on the training data. But I don't know how many features in image. And which is best feature for image segmentation? Gradient feature or entropy feature or ....Could you ex... |
I am trying to store the signal I have created in a single variable and graph the signal in two subplots in one single graph, one for left channel, one for right channel.
Fs = 44100;
N = 400
t = (1:N)*(1/Fs);
freq = 500;
sound_array = zeros(N,2);
%starts N x 2 matrix
sound_array(:,1) = sin(2*pi*freq*t);
sound_array(:,... |
I'm trying to write a tempo detection algorithm for analysis of audio samples. I'm roughly following the approach described in Scheirer, the first stages of which give me some very nice beat onset plots, which I'm happy are correct. The next stages involve the actual tempo detection process, which go something like thi... |
What fading model should I use for simulating rain loss in my network simulations? I am using NS-3 for this and I'm quite lost on how to go about this.
My simulation is just simple packet streaming and I want to test the effects of various interference on throughput.
|
I've been looking at 2 different methods to acquire a known prn code in a CDMA signal. The first trys Xor prn code chip to I sign data. Then average the result over a time period to determine the data bit. I have not had reliable success with this method. The second which I am going to try performs a real multiply of a... |
I was wondering what would be the statistics of the output of the filtering of an extremely wideband signal (e.g. 10 Gbps random data stream) through a narrowband low-pass filter (e.g. 10 MHz LP). I have the feeling that the statistics of the output would be somehow Gaussian due to the central limit theorem, but I cann... |
I am using FFTW3 to transform the autocorrelation function of a particular function. I expect the transform to be real since the autocorrelation function is symmetric with respect to negative and positive lags and its Fourier transform give the power spectrum. However, I get nonzero complex part and negative real parts... |
I'm applying at each 2048 samples the pitch synchronous algorithm to pitch scale some audio files, everything seems to occur well, but I've noticed some glitches, artifacts, clicks between every boundary frame!
I'm testing off-line in large áudio files(I'm processing frame-by-frame), what I tried to do was keep the old... |
When computing wavelet transforms with PyWavelets (Discrete Wavelet Transform in Python), e.g.
import pywt
cA, cD = pywt.dwt([1, 2, 3, 4], 'db1')
>> print 'cA', cA
cA [ 2.12132034 4.94974747]
>> print 'cD', cD
cD [-0.70710678 -0.70710678]
how can I specify the scale parameter of the wavelet?
In Matlab for example ... |
I want to find the SER and BER of the 4QAM,16QAM,32QAM. I know that the formulas for SER and BER are different but I cant find them in any text book.
The formulas for the BER that I found:
4QAM = Q(sqrt(2 ebno))
16QAM = 3/4 Q(sqrt(4/5 ebno))
32QAM = (8-sqrt2)/10 Q(sqrt(15/31 ebno)
can anyone confirm and give me the fo... |
By which value can the constellation of a 32QAM be normalized? for example 16QAM is normalized by squareroot of 10.
|
My aim is to determine the power spectrum of a signal of finite duration using
Its autocorrelation function
Performing FFT directly on the time-domain data
The function is defined as follows in the finite time interval from 0 to 10:
$$ \cos(10t) + \sin(20t) $$
I am using the library FFTW 3.3.4 in C source code and ... |
Why do we need different modulation techniques? Is it because we need to convert from analogue to digital data sometimes? Also what are typical applications where modulation schemes are used apart from wireless scenarios?
|
I was wondering which is the best language for image processing? I know MATLAB has a good library and user community for it. I havent checked it out, its there for python as well.
What is the most efficient language for this?
|
I am looking for methods to enhance noisy images, where:
some pixels in the image are very noise,
some other pixels do not contain so much noise.
My first thought is to build an adaptive Gaussian filter.
This means that the Gaussian kernel will depend on the (estimated) noise status of the current pixel (large radi... |
I have been working on Digital signal processing since Feb-2013. I have good command over it. Now I want to expand work in field of IMAGE and Video processing. Please suggest me best practical guide books, blogs, online links and other supporting matters related to IMAGE and Video processing .
|
I've got a FIR filter, which I developed in IC. Now, when I got my results, I ended up with two signals:
Theoretical signal calculated through MATLAB as a response to my input.
Real response given by the hardware.
What is the best method of calculating the error of the real response with respect to the theroetical on... |
When transforming a noisy signal via Fast Fourier Transform from time to frequency domain there is a "Processing gain" of the FFT which increases as number of bins increases. I.e. the more bins I have the more the noise floor in the freqeuency domain is reduced.
1.
Actually I don't understand completely, from where thi... |
If $\omega$ is the frequency of a continuous time sinusoidal signal, $-\infty < \omega < \infty $.
But when it comes to discrete time sinusoidal signals, the range becomes limited : $-\pi \le \omega_{discrete} \le \pi $.
Does that mean discrete sinusoidal signals cannot represent all frequencies possible ? What is the ... |
From this derivation to get the differential of a band pass filter: Id like to find out how we set the upper frequency cut-off and the lower frequency cut-off.
$$i(t)=\frac{v_i(t)-v_o(t)}{R}=C\frac{dv_o(t)}{dt}$$
So we get
$$v_i(t)=v_o(t)+RC\frac{dv_o(t)}{dt}\tag{1}$$
For the high-pass filter we have
$$i(t)=\frac{v_o(... |
I want to convolve two discrete functions $f$ and $g$ using convolution stride size $a$ to get the result as $s_{a, i}$:
$$s_{i,a} = \sum_i g_k f_{ai-k}$$
I know that simple convolution with $a=1$ can be calculated via FFT. I can do it even now, but I have to drop out all the points except $ai$. I think this way is not... |
I am trying to follow a paper where they say they apply the Kalman filter, but don't give the forumulation for the Kalman filter! Moreover I have looked at two references Wikipeadia and Durbin and Koopman 2001, but they seem to give slightly different formulations, I would like to know which is either is correct!
The s... |
From the textbook Signals and Systems by Oppenheimer ...
If we define $h_k[n]$ to be the impulse response to $\delta[n-k]$, then for the system to be time-invariant, we must have
$$h_k[n] = h_0[n-k]$$
I don't understand how the last equation follows. By definition of time-invariance, if the input is shifted by $k$ un... |
I want to simulate data from the following model:
$\textbf{z}_k=\textbf{H}\textbf{x}_k+\textbf{v}_k$ $\textbf{v}_k \sim N(\textbf{0},\textbf{R})$
$\textbf{H}$ does not change over time
$\textbf{x}$ is a vector of loadings
$\textbf{R}$ is a diagonal of constants
$\textbf{x}_k=\textbf{F}\textbf{x}_{k-1}+(\textbf{I}-\tex... |
From the MATLAB code below where do these theoretical equations for Rayleigh fading and white Gaussian noise come from? Or how are they derived?
h = 1/sqrt(2)*[randn(nrx,ntx,N/ntx) + 1i*randn(nrx,ntx,N/ntx)]; % Rayleigh channel
n = 1/sqrt(2)*[randn(nrx,N/ntx) + 1i*randn(nrx,N/ntx)]; % white gaussian noise, 0dB variance... |
I am using Matlab R2012b 64-bit on Windows 7 in order to estimate the power spectrum of a simple signal that is:
$$\cos(10t) + \sin(20t) $$defined in the time interval from 0.0 to 10.0
Here is what I did:
1) There was no predefined value of points so I took 1001 points by forming a time array and sampled the function a... |
Is phase and amplitude information necessarily lost when undersampling if you have a constant periodic single frequency sinusoidal?
My second question is: How can one determine the undersampling frequency for such a case? And how to retrieve the amplitude and phase.
I want to know if this is possible for a sampling fre... |
I have the Fourier Transform of a signal. How can I determine the Nyquist Rate of that signal? What is the formula for that?
|
Apologies if I'm using incorrect terms, and if this question's already been answered here. I have a data set which consists of a high DC signal modulated by AC signal (taken from custom-built biomedical instrumentation). Now the problem is that my signal has some low-frequency features that I'm having a hard time getti... |
I'm learning how to construct a discrete signal as a sum of impulse responses of the system. The final formula is then:
$$
y[n]= \sum_{k= -\infty}^\infty x[n] h_{k}[n]
$$
My question is, what is the amplitude of the impulse response, because i thought an impulse response was already a scaled response to the unit impu... |
I am currently working on functional connectivity analysis of EEG, and need to bandpass filter my data into different frequency bands (Delta, Theta, Alpha and Beta). An important thing is that the filter doesn't cause phase distortion. I know a butterworth filter and processing the data in both the forward and reverse ... |
In these days we can able to easily calculate the bandwidths with the help of technology, but in olden days where people used the analog phones, how did they calculated the bandwidths?
|
as you can see on page 314 the derivation of the ln[f etc.] equation is done via the matched filter that i have also provided below. i would like to know how this equation is used to derive the ln[f etc.] from its previous probability functions, and how it then finally becomes in the form of ||a-b||<||a-b||
any help wo... |
I've been reading a little bit about how DCT works in image compression using the 2D transform, however I don't know how this applies to audio using a 1D transform.
Suppose I have an audio segment, I mean, some samples of a mono audio vector.
Ai = [-0.0288,-0.0304,-0.0354,-0.0406,-0.0469,-0.0518,-0.0647,-0.0831 ...]
A... |
I am study linear classification(Y=aX+b) But it is still weakness that is overlapping problem. Could you suggest to me how to reduce overlapping problem in linear classification or how to find these parameters that helpful for overlapping reduction. In my question, I don't considier SVM method because it is more diffic... |
I'd like to validate changes to our app music player in our Continuous Integration unit tests (more like integration tests, really). My thoughts were to start with an audio file that contains only a pure tone, or a series of pure tones. Our automated test would stream this file over the internet and play it through our... |
I am trying to verify for myself that aliasing actually makes it impossible to distinguish the "real" and the aliased frequency. What I have done it set the sampling rate to 20 Hz and created two sine wave with 2 Hz and 18 Hz. As far as I understand the 18 Hz signal should alias to a 2 Hz signal when the sampling rate ... |
I keep reading that a constant group delay over a frequency band is desirable for a filter. But a constant group delay means a linear decrease in the phase delay, which would distort the signal.
Let's assume that I have a group delay of 5 [time units] over a frequency band from 100 Hz to 110 Hz.
That means that the pha... |
In continuation to the previous question Conceptual questions from signal processing I have a doubt which is: Consider an Autoregressive model (AR(2)):
$$
y(t) = ay(t-1) + by(t-2)
$$
and a FIR (Moving Average, MA(2)) model
$$
x(t) = a\epsilon(t-1) + b\epsilon(t-2).
$$
According to the reply in the prev question, in... |
I've a circular array of eight FM antennas. I'm doing digital beamforming to form eight beams with the following beam patterns:
The objective is to reduce my side lobes even further, and to get narrower main beam. What I'm interested in to know if it is possible to take two adjacent beams, and do another step of beamf... |
Which filters response is maximally flat at origin?
It seems to me that both High Pass Filter and Band Pass Filter have maximum flat response at origin, although it is almost zero in case of Bandpass Filter. Am I right?
This is a question that I came across in my friends exam yesterday.
The solution is given is that it... |
sinusoid x(t)=1+sin(2*pi *500)
sampling rate :1000 samples/s
Can I have 4 points DFT output with the above data.
|
I am trying to design a high pass filter to remove baseline drift from an ECG signal.
the baseline drift is of very low frequency like 0.3Hz or so with an amplitude of 25% of the ECG signal. But the bandwidth of ECG signal itself is 0.5Hz to 150Hz. I have been trying fdatoolbox in matlab to design the HPF but I was not... |
I was wondering if there was a simple approach to smooth images, e.g using a Gaussian Kernel without introducing boundary effects which are also present when using Discrete Fourier Transforms?
For both approaches, the missing pixels at the left-hand side will be taken from the right-hand side and vice versa.
Right now ... |
i'm working on image matching mainly SIFT and SURF for my thesis. i read Moravec's and harris corner detectors and understand them quiet well in my own opinion.A COMBINED CORNER AND EDGE DETECTOR[1]
I even understoond harris-lapace detector but had no luck with understanding it's descriptor.Indexing based on scale inva... |
I am working in linear classification such as $$y=ax+b.$$ where $x=[x_1...x_n]$. But the method has weakness that is overfitting problem. Could you suggest to me one method or one way to reduce overfitting problem in linear classification (don't consisder other method ex SVM).
|
In the image, I connects to universe, but II and III not.
I need to detect both II and III, also I if possible.
Is it possible with current computer vision libraries?
|
The equation to approximate an input signal with a unit impulse in Continuous Time, is shown below, before we take the limit $\hat{x}(t)=\frac{lim}{\Delta\rightarrow0}\sum^{\infty}_{-\infty}x(k\Delta)\delta_\Delta(t-k\Delta)\Delta$ <-- why is there a final $\Delta$ multiplying the $\delta_\Delta(t-k\Delta)$?
|
PROBLEM
Two real, causal time series
$$f_k = a^k \quad \text{and} \quad g_k = b^k$$
where $a \neq b$, $|a| < 1$, and $|b| < 1$, are given for $k = 0,1,2,...$
Find the linear convolution $h_n$ of the time series $f_k$ and $g_k$.
ATTEMPTED SOLUTION
I think perhaps my attempted solution here can be improved upon. We have ... |
I'm working on a modem system using a 16-QAM scheme. The transmitter runs an IFFT to modulate the signal, and has a 12-bit DAC. The receiver runs an FFT to demodulate the signal, and has a 12-bit ADC.
Here are the general parameters:
Sample rate is 512,000 SPS
FFT size is 64 samples
4 bits per bin/frequency with a t... |
My situation is as follows: i am trying to generate a waveform the hard way, by constructing the samples one by one and then saving the result to a .wav file using Python.
When the frequency is constant, everything is fine: i use $y(t) = \sin(2 \pi \cdot f \cdot t)$.
However, if i change the frequency to be a function ... |
For an efficient and optimized digital FIR filter design, there are two methods available broadly, Equiripple filter design & Least Squares filter design. A general method for designing a filter is also Frequency Sampled FIR design but it is not an optimized design
The basic knowledge I have is that Equiripple filter... |
I found the following comment here
The DTFT can be used when the samples are not equally spaced in time, the DFT cannot
My initial thought was that this had to do with periodicity of the basis complex exponential in DFT, but then this issue is unrelated to the uniformity of sampling itself.
What is the correct explan... |
The title might be unclear, but the problem is this. I have a signal sampled 1500 times with a rate of 60/s, and a sensor array 512 units large. There is a lot of noise, echo and other frequencies being picked up, but I am interested in only one. First I do a spike removal, then a bandpassfilter (butterworth) around th... |
In terms of signal analysis (especially frequency decomposition) would it be bad to use cyclic autocorrelation (AC) instead of acyclic/linear AC?
My use case:
I'm trying to characterize events in a sensor stream by periodicity/frequency. At the moment, I'm using DFT spectrum (spectral density) and AC of sliding windows... |
I wonder if there is a method works as follows:
1) The algorithm is trained with some images of a particular object (I train my Image with 1000 banana images)
2) I use the training data to segment a test image (semantic segmentation)
The critical point here is, the training images DO NOT contain labeling information (f... |
Let say I have a time-varying signal which is a mixture of two signals with different dc offsets. If I apply the fft on that signal, I will only get the dc value (at f=0hz) with stronger energy. How can I obtain another dc value which has less energy?
Is there any other methods that I can use to extract both dc values?... |
I want to determine the power of the noise component of a time dependend signal x(t) (similar to picture below)
With constant sampling rate I use Parceval's Therorem and calculate the standard deviation with
$$
\sigma^2 = \frac{\text{abs}(\text{FFT}(x))^2}{N}
$$
where N is number of data points and fft matlab dft impl... |
I don't understand why FFT return different maximum amplitude as the signal length increase. I would except that with a large signal length according to the frequency, detected amplitude will be very accurate.
From where this periodic signal is coming ?
Here is the related python code I used to generate the plot:
%mat... |
I'm implementing navigation system for my robot. There are two ways to get data from it: odometry(encoders from motors) and camera. Both of information sources can give me estimate of robot's position. But camera has very big noise. I've found that it is useful to use Kalman filter. After some literature looking I have... |
I wonder to know what is the best way to handle not equally spaced in time signal when performing FFT ?
I guess it depends on the signal itself. I work with signal with about 1000 - 5000 samples and they can contain between 0 to 20% missing samples. And I am interested in both main frequencies AND amplitudes.
Currently... |
In their paper Scalable Recognition with a Vocabulary Tree, Nister and Stewenius mention that after an offline training stage, new images can be inserted
on-the-fly into the database. However, at the end of the training stage the IDF is computed and normalized. My question is, how would you incrementally update such a d... |
Is there a 'Nyquist Criteria' for the minimum samples per chip that are needed for a correlator? I've heard 2, but will 1 work? What are the implications of having less than the minimum (in the context of radar and communications)?
|
i understand that there are linear equalizers such as MMSE and ZF, however there are also non-linear equalizers such as MLD. i would like to know what is the difference between a non-linear equalizer and linear equalizer, and how one can be distinguished from the other?
|
I want to calculate averaged power of a time-domain signal by means of its spectrum. I guess Parseval is the right tool.
So I sample a sinus of 100 Hz 10000x within one second.
Unfortunately the sum of the squared samples euqals not the sum of the FFT amplitudes (weighted by the number of FFT bins). Where is the mistak... |
I need help solving the following blurring function question.Assume an image $f(x,y)$ is moving in front of a camera so that $_0()$ and $_0()$ are the time-varying components of motion in the x- and y- directions respectively. The camera’s shutter opens at = 0 and closes at = . Assume that the shutter opening and clo... |
Hello everyone i want to do dynamic texture video sementation using the Fourier transform in MATLAB. I am applying 3-D fft on dynamic texture video frames (using matlab function 'fftn') and reconstructing video frames back from the phase spectrum only (using matlab function 'ifftn').
I have found moving part pixel int... |
I'm looking for some advice on where and what to start reading for learning to solve this.
I've the time series of the position coordinates (x,y) of an animal in an open field (just a cage). I want to detect the time instants when the animal stops or starts "walking" (i.e.: moving from one place to another, not in the ... |
I have to process building floor plan images to extract walls from the structure. It is trivial in case of binary images; but the images in our case are colored, and have different colored walls. This restricts us from using any thresholding operation as we cannot assume the walls will always be the darkest. How can I ... |
This might be a silly question, but I'm pretty rusty on my signals. The problem is the following:
I'm using a program which does not have a random signal generator, but I can sum up sines; and I need to have a "white noise" signal.
What would be a good way to "approximate" a white noise signal with sines?
I know that I... |
It is possible to do IQ demodulation by using a zero crossing detector and a timer to take 4 samples of a sine wave if a reference signal is present. When there is no phase difference between the signals the 4 samples are taken at exactly 0, 90, 180 and 270 degrees by using the timer.
Does this subset of lock-in/synchr... |
From the definition I know:
$$|\sum_{n=-\infty}^{+\infty}x[n] \cdot z^{-n}| < \infty\space\space\space(1)$$
Is there any sequence of x[n] which can not be written as
$$x[n] = y[n] \cdot u[an-b]\space,\space\space\space a,b \in \mathbb{Z}$$
(where the dot means a multiplication and $u[k]$ means the discrete step functi... |
In OFDM system, after IFFT a CP is added at the beginning of the OFDM symbol by copying it from the end of the symbol.My question is that with CP why we use cyclic convolution instead of linear convolution. I would appreciate if someone please could explain the reason of cyclic convolution here.
Many thanks!
|
I'm studying image warping and I'm unsure how to warp a starting rectangular image into another image where the box corner points have been changed as I want:
(image from http://www.inf.ethz.ch/personal/pomarc/courses/CompPhoto/cpv03.pdf)
As far as I've understood, I need to:
1) Decide the four points (e.g. by using d... |
I will first describe the Independent Component Analysis (ICA) steps, so that the question becomes clearer, and more relevant to similar questions.
Assume we have $N$ sensors distributed in space. From each sensor we obtain a time series of length $T$. We collect these time series as rows in an $N \times T$ matrix $X$... |
I have a question which is closely related to this one:
FFT Processing Gain
^that discussion is a bit general so I want to ask a very objective question to see if I can apply that knowledge:
If I collect one million time samples and perform a one million point DFT, roughly what DFT processing gain (in dB) improvement c... |
I have a problem of the following form. There are two signals, x(t) and y(t). The model for the system is such that:
$$x(t) = x'(t) + f(y(t))$$
where $f(y(t))$ is a variable offset introduced by interaction of $x$ with $y$. The function $f$ is unknown. As an additional constraint, $x'$ is expected to vary slowly an... |
Studying for an exam I'll have I ran into a question about computing the localization of a Gaussian Filter in 1D, but sadly I have no idea of what such localization is. Can you give me a hint?
Thanks
|
In the diagram below there are three multiplication operations required to
compute each filter output sample. I'd like to draw a block diagram of an equivalent filter but requiring fewer than three multiplies per filter output sample.
I need to transform the filter structure to Transformed Direct Form II, that it ca... |
These two filters have very similar frequency responses, but those responses are not identical except at a single frequency $\omega_0$. I want to find $\omega_0$.
One way I've seen that done is like this:
$$\begin{align}
Y_1(z)&=X(z)+2X(z)z^{-1}+X(z)z^{-2}\\
Y_2(z)&=X(z)+X(z)z^{-1}
\end{align}$$
if $Y_1(z)=Y_2(z)$
$... |
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