instruction stringlengths 31 27k |
|---|
Doppler frequency is calculated by the formula $f_{d} = 2*v_{r}/\lambda$, where $v_{r}$ is the radial velocity and $\lambda$ is the radar wavelength.
Normally FFT is used to detect the doppler frequency(A bank of FFT filters is used for Doppler frequency estimation).
Assume that sampling frequency is fixed as $4*f_{ma... |
I am working with images and Fourier transforms. I am trying to understand what might be causing some artifacts in my output image.
I am starting with a 512x512, RGB image of Lenna. I FFT the image, and run a low-pass filter on it. Unchanged for everything inside the circle, 0 for everything outside the circle. Her... |
I have a device that connect with USB to my computer. This device have digital signal that I can sample it every few seconds. The signal that the device perform, is a digital signal with $X$ seconds of $1$ and $Y$ seconds of $0$ ($X$ and $Y$ are integers and consts).
What I want to do is to estimate if in $Z$ seconds f... |
I have some issues understanding the use of a window (it doesn't matter which one) to calculate the Power Spectral Density of a signal.
For example say I want to use the hann window, with $N = 1024$, knowing that my signal length is $X > N$, $X$ is arbitrary.
How should I calculate the PSD :
Take 1024 samples from th... |
I have designed my filter using firhalfband(N,Fo,A), which gives me the filter coefficients. Then I try to see the frequency response of it. I use these two function: fft, and freqz, but they are giving me different result in my figure, why is that? I wonder how these two MATLAB functions operate when taking the freque... |
The Spread Spectrum technologies such as DSSS and FHSS are successful against frequency-selective fading because the original signal is spread to a larger bandwidth and different frequencies are used a different times.
Do these technologies also mitigate the problems imposed by time-selective fading?
|
I'm reading through some material on wavelet and I'm finding it difficult to see why the following is true.
Let $\Psi(\omega)$ is the Fourier transform of the mother wavelet $\psi(t)$
$$\int_{-\infty}^{\infty}\frac{|\Psi(\omega)|^2}{|\omega|}d\omega<\infty$$
then many paper on wavelet would immediate jump to, is obviou... |
I am working in medical and nature images processing.
I want to find a popular method to segment texture image.
Could you suggest an approach to do so?
Both the feature extraction and classification.
|
I have 3 sensor inputs: $a(t)$, $b(t)$ and $c(t)$. I want to design a filter such that the weighted linear combination of the three is always a constant. Kind of like:
$$w_1(t)a(t) + w_2(t)b(t) + w_3(t)c(t) = k$$
So from my undergrad modules I think I need a adaptive filters. I can perform training to find $k$ and filt... |
I wrote an algorithm to do bicubic interpolation of an image.
I used the method desribed in the wikipedia page.
On simple images, the result looks good, but on more complex ones, I got strange artifacts in non-smooth zones.
What could be a probable source of those artifacts, and what can I try to get rid of them ?
My c... |
Following information is given about a signal $x[n]$
$x[n]$ is real and even signal
$x[n]$ has a period $N=10$ and Fourier coefficients $a_k$
$a_{11}=5$
$\frac1 {10}\sum_{n=0}^9 |x[n]|^2=50$
How can we obtain $x[n]$ from these information?
I know the first info makes the Fourier coefficients real and even, third info... |
I have a setup with an embedded DSPIC. This is used to link multiple audio inputs to the outputs and changed between them either when told to, or when one is active. This changing takes place on a hardware that is purchased and from what I can tell uses a relay. I have DSP software using a heterodyne filter to remove s... |
I have real-time audio application written in python, and I have question about detecting silence.
What's the most efficient way to distinguish silence from non-silence, in an input audio signal? By non-silence I mean any sound, even noise. I need to know when silence-period is in the process, as accurately as possible... |
In adaptive filters, the development of LMS algorithm typically starts from the Weiner-Hopf equation, while the development of RLS algorithm starts from the normal equation. As I understand, these two equations are the same, and both their solutions is the optimal coefficients that the adaptive filter has to find. Is ... |
I am trying to use generalized cross correlation with phase transform (GCC_PHAT) on some audio processing indoor, with the existence of low noise. I ve read here that by canceling the magnitude factor, which what pretty much PHAT does, the result of the correlation will be resistant to reverberation and low noise, but ... |
The original idea is to calculate distance from accelerometer input. However, accelerometer reading also contains the gravitational values, thus to remove gravity, I tried using Gyroscope. The idea was to calculate angle of rotation from gyroscope and then calculate gravity at this angle and subtract it from accelerome... |
I'm about to tackle implementing a Viterbi equalizer fo a GSM system in gnuradio.
I went through the internet and ieeexplore and I think that this document explains it best:
Application of Viterbi Equalizer in GSM
System
My question concerns the Matched Filter in the block diagram.
Normal Burst in GSM has 8 possib... |
if we create a continuous signal from epoched data (say, EEG data with 100 epoches), what kind of problems it may cause( i am only aware of edge artifacts ) and how i can avoid them in analysis (say, power analysis)? Thanks in advance,
|
Suppose you have the ability to inject any arbitrary waveform into a piece of analog rf hardware and collect and digitize the output for analysis.
If you wanted to characterize/estimate the transfer function of the hardware, what waveforms would you use?
|
Hey I want to remove DC offset from tetra signal captured as inphase and quadrature phase samples(I and Q samples). I and and Q samples are 16 bit signed values. I tried averaging 2000 I and Q samples out of 2040 sample burst and subtracting corresponding averages from I and Q sample values recieved.But it doesn't seem... |
I have a signal that is composed of a sinusoid. I will send it over the air through radios. Part of the signal will go through processing on a computer, which means that it will be converted from analog to digital, and back again.
I have two questions. How can I model any frequency shift that will occur in this syst... |
I have a signal which is a matched filter output.
whose values are given here:
Amplitude values
Input to this matched filter is radar return signals mixed with additive gaussian random noise.
Its impulse like peaks are at indices 2201,3005,4443,4444,6524(by manual inspection).
How can I detect these peaks programmatic... |
If I have a white noise fed into a filter what would be the output of the filter - what would you expect to see?
I know that white noise has a unit power spectral density (PSD), how these thing relate to each other?
|
Problem statement
In the dataset above, you can notice a continuous path and then a noise-band at around 320-400 on the x-axis. I want to be able to apply a filter that amplifies the continuous path while suppressing the noise-band. The data inside the noise-band is useless.
Attempt 1
I am new to signal processing, bu... |
I was reading a DSP book where they had a problem on Z Transform.
Determine the z-transform and ROC for signal $\displaystyle x(n) = a^nu(n)$?
The solution states that
$$
\displaystyle X(z) = \displaystyle\sum_{n=0}^\infty (az{^{-1}})^n
$$
So
$$
\displaystyle X(z) = \frac{ \displaystyle {z} }{z-a}
$$
Hence ROC $|z| ... |
Relation of Entropy and SNR : Based on this question and answer, I had another question that struck me and I am curious to know, if somebody can shed some light, on the following situation: $y= desired_{signal} + noise$ is received by the receiver and the $error = y - \hat{y}$ where $\hat{y}$ is simulated at the receiv... |
Given a signal $$x(t)=16\cos(20\pi t+\frac\pi 4)+6\cos(30\pi t+\frac\pi 6)+4\cos(40\pi t+\frac\pi 3)$$how can I calculate the power contained in a frequency interval, say 12Hz to 22Hz.
The total power is $\frac{16^2}{2}+\frac{6^2}{2}+\frac{4^2}{2}=154$
|
There are several state-of-the-art algorithms in image denoising including Non-Local Means, BM3D and so on.
I am wondering whether there is some well-known state-of-the-art algorithms in image edge detection (with reproducible code) ?
Thanks a lot for your answering.
|
How is FT of $ \delta $(t) equal to 1 ? Normal FT gives the result $\infty$. Can someone please explain? I did the normal integration and substituted the limits.
Is it because $ \delta $(t) is a unit impulse function so as it's height is large it's width is very small so no matter what the FT will always be equal to ... |
Can someone please confirm if this is indeed the correct way to compute DWT:
load image
load Haar wavelet
for i = 1: Level
periodize edges of image
convolve image with Low pass Haar wavelet filter and store result in G
convolve image with Low pass Haar wavelet filter and store result in H
convolve G with low ... |
I read that in A/D converter we usually use rounding or two's complement truncation and not the sign magnitude or 1's complement truncation as in this case the quantization error is correlated to the input signal.
Can some one with A/D quatization knowledge alabaret this for me.
Is the quantization error in two's compl... |
Input: Microphone recordings of digits from 0 to 9 from different speakers.
Output: The digit from 0 to 9.
I am doing this for fun. So first I will train my neural network using some samples and then use it to classify digits.
Problem is every person takes different amount of time to say the digit also every person ... |
Why is frequency modulation (FM) not used on the current amplitude modulation (AM) radio stations? The Police have FM 2 way radios that use only 5 kilocycles of modulation. Instead of the current AM radio stations that use the same bandwidth of 5 kilocycles. The limiter & discriminator or ratio detector filter out the ... |
Peoples' ears can hear sound whose frequencies range from 20 Hz to 20 kHz. Based on the Nyquist theorem, the recording rate should be at least 40 kHz. Is it the reason for choosing 44.1 kHz?
|
Let's say sampling rate is $Fs = 44\mathtt{kHz}$, now I have $N = 2048$ samples, then I can get $N/2 + 1 = 1025$ frequencies.
I'm confused by Matlab's FFT documentation that says the frequencies are present at:
(Fs/2) * linspace(0, 1, N/2+1)
It is very probable that the real frequencies are not among the frequencies g... |
I am doing the project on speech recognition.I have implemented both the mfcc features and neural network.I am a bit confused in joining the two blocks.For example i have three different words with their '.wav' files.I extract 13 mfcc features from these three words respectively.There will be a matrix of mfcc e.g (13x1... |
I was reading the book "Spectral Analysis of Time Series" By Herman Koopmans.
On Page 55, he explains that a specific type of non-stationary signal which is the result of adding weakly stationary ergodic noise to a deterministic signal can be decomposed to Wiener spectra. I wonder why there is a need for signal to be e... |
I'm trying to estimate the power spectrum and time delay for random signals. the signals were taken from two sensors, 10 cm apart. My aim is to estimate the PSD first then estimate the time delay in order to obtain the structure velocity.
I've tried the traditional way by using matlab built-in functions like fft, xcor... |
I'm trying to teach myself about the basics of feature extraction from digital signals. Specifically, I want to try monophonic then polyphonic pitch recognition. I don't have any university level math or computing science experience (I'm self-taught), so I find the formulas in many texts and websites to be a road-block... |
Audio is a sequence of waveforms which have peaks, troughs, 0, just like below.
How to define the x axis (y = 0) in the picture, how to choose the 0 point in the waveform since there are so many points on the waveform?
Why not move the waveform above or below the x axis(y=0)?
This question seems a little stupid. But I... |
I'm trying to estimate the average coherence of discontinuous snippets of a signal in MATLAB. For Welch's spectrum estimates, the following works:
w = blackmanharris(50); % window
n = numel(w); % fft length
o = 0; % overlap
x = rand(200,1); x1 = x(1:100); x2 = x(101:200);
s = pwelch(x, ... |
So, I'm completely new to digital signal processing, but while reading a piece this morning about quantization it got me daydreaming: could a machine ever be fast enough to sample the position and amplitude of each particle that makes up a wave?
|
In the process of trying to learn dialup, I've managed to learn/figure out everything except how to convert a demodulated/filtered signal back into the original data.
Here I have a random two-bit signal (red) sampled at 16000Hz at 2400 symbols/s (gray lines separate symbols), as well as the filtered signal (blue). The ... |
I have two related questions on this topic. First, about how wide should the filter be around the carrier frequency? What is used to determine this?
Second, in Python, I must use 'numtaps' when implementing a filter. What exactly are numtaps? How can I determine what to set this value to? Setting it to about the c... |
I was reading in a book that below stated are the Properties of Discrete-time Sinusoidal Signal:
Property 1. The discrete time sinusoids whose frequency are separated by an integer multiple of 2$\pi$ are identical.
Property 2. The frequency of oscillation of discrete time sinusoids sequence increases as $\omega$ inc... |
I was trying to solve the below system is a Time variant or Time Invariant System:
$y(n)=nx^{2}(n)$
Now for a system to be time invariant the system property should be :
y(n,k) = T[x(n-k)]
OR
y(n,k) = y(n-k)
for all possible values of k.
Coming back to system equation:
$y(n,k) = T[x(n-k)] = nx^{2}(n-k)$ delayed by ... |
I am working with Gabor filter with level set method as in A Level-Set and Gabor-based Active Contour Algorithm for Segmenting Textured Images. As you known, Gabor filter gave many filter output. Some of them is not necessary. Could you suggest to me some method that can choose best filter output that satisfy maximum t... |
This is my first time working with Mel-frequency cepstral coefficients and I have two questions....
Are some of the 13 coefficients strictly related to some sound properties like frequencies, amplitude, pitch or something ? For example if I'm interested only in sound pitch, are there some coefficients that carry that ... |
Suppose I have impulse response like [1/25,2/25,3/25,4/25,5/25,4/25,3/25,2/25,1/25].
I did convolution with 600 samples of test signal (it seems that I did some filtering).
Then I calculated the same thing in frequency domain:
Found frequency spectrum of test signal and frequency response.
Multiplied them and via I... |
Assuming we have a discrete signal $ { \left\{ x \left[ n \right] \right\}}_{n = 1}^{N} $.
Which has a Discrete Fourier Transform / Series.
Now, assume I'd like to estimate its Discrete Fourier Series coefficient yet some samples of $ x \left[ n \right] $ are missing (The indices are known).
How could that be done ef... |
i have a measured signal (e.g. voltage sine wave) and i would like to generate a sine wave which matches this meausred signal (with the generated sine wave i remove the harmonics from my measured signal), just like a PLL block in simulink.
for that i have to do the 2 steps:
1- Calculate the frequency of the measured s... |
In Welch periodogram Power Spectral Density estimate, we devide the N-long signal into K segments, each of length L with overlapping D, such that N=L+(K-1)*D.
In this paper, just above eqn (7), the authors mention that the step of performing the DFT is to be done to each (windowed) segment (of length L) such that the ... |
This question is related to my other recent questions. In the last question, I learned that I should filter about 4 * WPM (words per minute) around the carrier frequency if I want to recover Morse code. One problem is, learning the words per minute. I filter because I want to differentiate times when the system is k... |
I'm implementing a BCI-Wheel Chair Control System where signal are extracted from the arm into MATLAB -> feature extraction -> classification -> Control signal
The delay should be minimal between muscle movement and when control signal is sent
I've read that windows is not optimal for this purpose (not even 8.1?) Can a... |
I am trying to estimate the signal-to-noise ratio of a wave file (The wave file is a recording of a conversation in a room). I found examples where the Matlab scripts compare the wave file data to a noisy wave file. This noisy wave file is generated by either mixing Gaussian noise into the actual recording or they have... |
Here is a bunch of code from MATLAB documentation: Designing a Basic Monostatic Pulse Radar:
wav = phased.RectangularWaveform(...
'PulseWidth',1/pulseBandwidth,...
'PRF',PRF,...
'SampleRate',Fs);
matchingcoeff = getMatchedFilter(wav);
hmf = phased.MatchedFilter(...
'Coefficients',matchingcoeff,...
... |
I understand what books states about cross-correlation of signals and their mathematical representation:
For continuous functions f and g, the cross-correlation is defined as:
$(f \star g)(\tau)\ \stackrel{\mathrm{def}}{=} \int_{-\infty}^{\infty} f^*(t)\ g(t+\tau)\,dt$,
where $f*$ denotes the complex conjugate of $f$ a... |
I am working on a tuning app. I have so far tried 3 different libraries with a number of different algorithms. However, when I compare to other tuner apps, I seem to be getting frequency doubling on certain notes with certain instruments. Sometimes frequency halving depending on other factors. I have tried adjusting th... |
I have an aerial image, like this
I want to detect the outline of the ship without shadow. How can I do it? I tried threshold in both graylevel, RGB and HSV. It doesn't work perfectly.
|
Brief background
There is a known problem that high order modulation schemes suffers from phase noise of reference generator. It's very important issue for e.g. line-of-sight (LOS) modems exploiting QAM-256 or even higher. Some compensation schemes are possible to solve this task. The main challenge in these schemes is... |
I read across places that cross-correlation of two signal (a radar sending a signal and receiving it back) one can detect the location (distance) of a target from a point and that cross-correlation can be used for comparing signals.
I solved a question to find cross-correlation of two finite length sequences where:
x(l... |
Given some frequency-domain representation of an impulse response.
How does the number of frequency-domain sampling points influence the outcome of an inverse FFT, if I keep the sampling frequency constant?
Example 1
Take x(t) = e^(-t)*sin(t) for t in (0,10). Sample the signal, take its fft. Omit every second point of ... |
Suppose you have k samples from each of the N elements of a uniform linear array (ULA) of sensors:
What is the physical meaning of a covariance matrix?
How do you form a covariance matrix with the samples?
How do you decide how many samples you need to use to form the covariance matrix?
|
I am planning to use Slepian or DPSS window in my application where I want central lobe to be concentrated and also have low bandwidth:
http://en.wikipedia.org/wiki/Window_function#DPSS_or_Slepian_window
However, since the generating function is missing and looking at some online resources was not very helpful. So, I a... |
Suppose we have the following signal model
$y(t) = A_1\sin(\omega_1 t+\phi_1) + A_2\sin(\omega_2 t+\phi_2) + ... + A_p\sin(\omega_p t+\phi_p) + z(t)$
When we are sampling at sampling frequency $f_\mathrm s$ we know that the sampling frequency must be at least 2 times the maximum frequency, but what about a condition fo... |
How is the periodogram defined? I have been searching the internet a lot, and I find two different definitions:
periodogram is the squared magnitude of the DFT
periodogram is the squared average of the DFT
which one is correct?
|
I am trying to implement what's called the "second spectrum". Basically, you do this:
Take a time series of length $N$.
Divide it into $m$ segment, each of length $N'=N/m$.
For each segment $m$, do a Fourier Transform. The result is the 'first
spectrum',$S_i^{(1)}(f_1), i=1:m$
Divide each spectrum into $n$ octaves, an... |
Having just read the rather long article on Wikipedia about window functions, I still do not see the mathematical point in using window functions when calculating spectral estimates (using e.g. Welch's method).
For example,
The Fourier transform of the function cos ωt is zero, except at frequency ±ω. However, many ot... |
I cant understand this when I have a system thar compress the input as follows
$$y(t)=x(2t)$$
Now to check for time invariance now I should input $x(t)$ to get
$$y_1(t) = x(2t)$$
Then I should input $x(t-t_\circ)$ to get
$$y_2(t) =x(2(t - t_\circ)) = x( 2t - 2t_\circ)$$
Now checking the output shifted we get
$$y_1(t-t_... |
In MATLAB I generated a signal and its autocorrealtion function using:
signal = ones(1,500).*5; % a vector of the number 5
acf = autocorr(signal,numel(signal)-1));
To my suprise the resulting ACF vector is all NaN.
But If i generate the signal with numbers between zero and one (0.3 in this case),
signal = ones(1,50... |
I read LTE PHY profile and do not understand the relationship between number of FFT and bandwidth. For example LTE downlink channel bandwidth 1.25MHz and 5MHz have 128 FFTs and 512 FFTs accordingly. Another example is 802.11 a bandwidth 20MHz, FFT size 52.
I design an OFDM system system at carrier frequency 910MHz, wit... |
I'm reading up on fourier theory, especially the transforms. I implement the math as spectrograms in C++ to get a better understanding of what is going on.
I've made an implementation of the short time discrete fourier transform which works well, however as for transforms derived from the fourier transform, they seem t... |
When windowing non-overlapped segments of data stream, part of the data is being "ignored" at the boundaries of the segments due to the tapering of window functions (non-rectangular). Some information is lost in further processing.
My question is: for a given window function, does overlapping the segments by $50 \%$ p... |
I am trying to code audio signal processing software in java.
Purpose: It will use audio signal processing library to filter out noise from the raw PCM based input (.wav) of heartbeat recording.
Steps involved
Select input .wav file.
Apply Low Pass filter at 33 Hz.
Apply High Pass Filter at 2 Hz.
Apply Amplificatio... |
lets assume following signal
$$x=a_1 \sin(2\pi f_1 t) + a_2 \sin(2\pi f_2 t).$$
This is passed through a simple single-pole filter with cut-off frequency $f_c$. How is $x$ delayed by the filter (peak of the cross-correlation)?
In the case $a_2=0$, the time delay between the original and the filtered frequency is
$$T_1... |
I am trying to learn how to solve a bunch of digital signal problems and I have trouble understanding the solutions provided by this book I'm using. Basically, this problem asks me to determine spectrum amplitudes for half-wave rectified sine, as the title mentions. The signal looks like this:
The development in compl... |
The human eye has 3 types of color cones: red, green and blue. Their spectral sensitivity peaks in short (S, 420–440 nm), middle (M, 530–540 nm), and long (L, 560–580 nm) wavelengths. See diagram below.
Then we have the CIE standard observer. Wikipedia says it can be thought as the spectral sensitivity curves of thre... |
In many signal processing books, it is claimed that the DFT assumes the transformed signal to be periodic (and that this is the reason why spectral leakage for example may occur).
Now, if you look at the definition of the DFT, there is simply no that kind of assumption. However, in the Wikipedia article about the discr... |
I have the following signal:
and this as the solution to the problem:
$$\begin{align}
X(j\omega) &= \int_{-\infty}^{+\infty}x(t)e^{-j\omega t}dt \\
&= \int_0^\tau \frac{E}{\tau}t\; e^{-j\omega t}dt + \int_\tau^T E \; e^{-j\omega t}dt \\
&= \frac{jE}{\omega}\left[e^{-j\omega T}-e^{\dfrac{-j\omega \tau}{2}} \mathrm{sinc... |
Excuse my silly question, but i really want to know does changing the sampling rate affects the energy (bandwidth) of a signal? therefore improves cross correlation output?
|
I am studying the mel-frequency cepstral coefficient
(MFCC) and have downloaded a pdf which is related to MFCC from this link. It is about the "ETSI ES 201 108"
ETSI ES 201 108 - Speech Processing, Transmission and Quality
aspects (STQ); Distributed speech recognition; Front-end feature
extraction algorithm; Com... |
I have a digitally sampled signal(sampling frequency = 4000 Hz) which contains analog frequencies from 0 to 1000 Hz.
I wish to have a frequency resolution of 100 Hz(like 0,100,200,300 Hz .. 1000 Hz) and I have to improve the frequency resolution at the 0 to 100 Hz range by 10 Hz. That is at the frequencies 0,10,20,30,.... |
Can anybody explain the meaning of Doppler tolerance for the waveforms(Linear frequency modulated, Barker code, Polyphase codes) used for Pulse compression.
The Doppler frequency shift $f_{d}$ due to a moving target of velocity $v_{r}$ is given by
$f_{d} = 2*v_{r}/\lambda$ where $\lambda$ is the wavelength of the tran... |
I am quite new to the idea of equalization. I have a few queries regarding the same.
In my application I require to equalize a channel whose impulse response is an IIR response. Is it possible to design an adaptive equalizer based on LMS algorithm to equalize it? I tried some MATLAB simulations and it worked well (the... |
I have a stereo sound for Inter aural time difference consisting of sine wave(16 samples as 16 bit PCM Data) for left and right audio. From time to time I would like to advance either the left or right audio so that sound in one ear comes earlier than the other. If I just put another sine wave of different frequency i... |
in mixed (digital and analogue application) where you have to convert a signal from continuous to digital time domain and then back again to continuous time an anti-aliasing LPF is needed and in general also a LP reconstruction filter.
Are there some applications (if so what are) in which the reconstruction filter is n... |
Problem: I have a photo of an object (a manufactured part like the attached photo below), using my Andoird phone camera I want to verify if the object in camera preview matches to the template or not. (in other words, is it the same part as the template or not)
I can make the user to move the camera in order to have s... |
I am trying to combine two sets of data together, each of them having different sampling rates. For example if I sampled a 100 Hz sin wave at 1KSPS, and another 150Hz sin wave at 1.5KSPS, would it be possible to combine them for FFT without using non-uniform FFT techniques? Or would my best case be interpolating the sl... |
If I have a signal with frequencies up to 2000 Hz but I'm only interested in the frequency range up to 500 Hz, is it correct to sample at 1000 Hz $(f_\mathrm s \geq 2 \cdot f_\mathrm{max})$? Or do I have to apply an analog lowpass filter with cut-off frequency 500 Hz first?
I read this similar post, but didn't understa... |
The frequency resolution of DFT is
$\Delta f = \frac{1}{T_{0}} = \frac{1}{NT} = \frac{f_{s}}{N}$
where $f_{s}$ is sampling frequency, $T_{0}$ is sampling time, $N$ is number samples, and $T$ is the sampling interval.
Where does this definition come from? Espescially the $1/T_{0}$, why this equality?
|
Let
$$
A=
\begin{bmatrix}
a_{1,1} & a_{1,2} & a_{1,3}\\
a_{2,1} & a_{2,2} & a_{2,3}\\
a_{3,1} & a_{3,2} & a_{3,3}\\
\end{bmatrix}
$$
and
$$
B=
\begin{bmatrix}
b_{1,1} & b_{1,2} & b_{1,3}\\
b_{2,1} & b_{2,2} & b_{2,3}\\
b_{3,1} & b_{3,2} & b_{3,3}\\
\end{bmatrix}
$$
and $C=A\circ B$, i... |
$X$ represents sample in frequency domain and $x$ represents samples in time domain.
NOTATION 1
$ X[k] = \sum\limits_{n=0}^{N-1} x[n] \ e^{-j \frac{2\pi}{N} n k} $
$ x[n] = \frac{1}{N} \sum\limits_{k=0}^{N-1} X[k] \ e^{j \frac{2\pi}{N} n k} $
NOTATION 2
$ X[k] = \frac{1}{N} \sum\limits_{n=0}^{N-1} x[n] \ e^{-j \frac{2... |
I am asked to extract three curves from palm images.
original image
expected result.
The curves has lower intensity than neighboring pixels. So I try to morphology to extract the pixels. The matlab code is followed.
RGBImage = imread('E:\ImageProcessing\Palm\Matlab\IMG_20140817_144... |
I am trying to detect if a car did accelerate or did brake by using the accelerometer of the iPhone.
In the figure below I plotted the collected data. (To collect the data the phone was laying flat in the car facing the top of the phone to the front).
The issue I am having now is that visually I can clearly see where... |
I am digging into Haar like and LBP features. More precisely its implementation in OpenCV. Every article or forum entry I found states, that LBP is faster than Haar. My local tests also confirm this expectation. Yet I don´t understand why are LBP faster. As far as I know, OpenCV uses MB LBP features. I looked into some... |
The task is to detect arrow from medical images. After segmenting, arrow-like objects can be connected with other connected components (noise). In this context, detecting a part of the object that characterizes an arrowhead would be a solution since template based shape descriptor does not perform well.
Please help!
... |
I am doing some analysis in python working with a photodiode. The signal is more or less periodic with some phase shifting in certain areas that give rise to faster or slower periods but overall still very periodic.
I am trying to see how "good" our signal is by computing the Signal-to-Noise ratio (SNR), but so far I ... |
i recorded a tone between two devices and applied cross correlation on the recorded and the sent signals.
I want to know if there is a multipath effect on my result (on MATLAB). what can i use for that?
|
I am a self-learner of DSP by reading related books.
While reading Time domain Analysis of DT (Discrete Time) system, I suddenly got introduced to difference equation representing a system. It then further stated -
The response of any DT system can be decomposed as :
Total Response = Zero State Response + Zero Input r... |
I'm working on image project using Matlab & I heard about 2D-otsu algorithm as effective segmentation with low SNR and low contrast images, but I can't find enough information about it or how to implement it. I'll be thankful if anyone can help me in such case.
|
Subsets and Splits
No community queries yet
The top public SQL queries from the community will appear here once available.