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I have just a few questions regarding the rise time of filters.
A rise time seems to be measured by using a step input, with an input typically from 0 to 1. However this is what I cannot seem to find out...
-Does the rise time stay the same if input is from say 0 to 0.5 or some other arbitrary input?
-If not, how can I... |
Basically, I'm running a server and two clients of the Unreal Engine and trying to figure out if there is lag in player movement on one of the clients (the clients are different in way that's not important). For each client I am sampling the (X,Y,Z) location in world space every frame. Here's a sample from one player ... |
As stated by the convolution theorem, a convolution in spatial domain is equivalent to a
multiplication in the frequency domain.Nevertheless, when should the first be preferred over the latter?
|
Convolution is used to find the output when the input and the impulse response is known. I am curious if there is a similar operator, like convolution, or a method that can determine the output of a system. I am asking this question, right of the top of my head, so I don't have any concrete example to apply it to.
If ... |
I have a time-series signal (voltage, specifically) that was divided by 1000 before doing an FFT-based Power Spectral Density (PSD) (using LabView), giving the output in dB.
Question: What is the effect of this de-amplification on the validity of the PSD? And how can I amplify it back?
|
Given a length-n vector (the elements of vector describe samples of any time changes).
There are $m$ elements, and other values $n-m$ (coming right after the first n elements)
I need to predict what their value of other $n-m$ elements .
I would like to know which prediction method is recommended to use for solving ... |
I am designing a schematic in which a Counter is to be increased whenever a particular band of frequencies is received at the Audio input.
We are receiving the audio input as electrical Signal and I have already designed the bandpass filter with the required passband. How can I design the Counter for the same purpose?
|
my field of engineering is condition monitoring. I use vibration analysis to determine the current state of health of a machine and then recommend any course of action required to ensure it does not fail during a production period. See www.efftek-uk.com for further description.
The most important tool I have is ESP whi... |
I am currently replacing a hand-made image processing library by OpenCV. The programmer who worked on this library had implemented the sobel gradient function. This implementation is right except that he first converts his image to grayscale with the formula thereafter (the arithmetic mean):
Y = 0.333 * R + 0.333 * G +... |
I have a bunch of data that went through a hardware filter before being digitized. The data has been taken this way for quite some time, and everyone is fine with that. Now, there is a new application in development that requires unfiltered data. Rather than wait a year to collect new data, we would like to "unfilter" ... |
To reduce out of band energy, it is common to apply a Raised Cosine filter to the front and back of an OFDM symbol. It is also common to overlap the windowed portions of adjacent symbols to eliminate amplitude dips.
For an OFDM symbol consisting of a Cyclic Prefix followed by NFFT points from an IFFT, is it common to i... |
For linear time-frequency representation, the resolution is limited by Gabor limit. However, if interference is allowed, the resolution may be higher. What is the possible resolution in such a case? In "Electroencephalography Feature Extraction Using High Time-Frequency Resolution Analysis ", they stated
We all kno... |
I would like to know, why does the periodic signal in time always give a discrete frequency spectrum in FT?
I know the equations, but I simply dont understand why is it so.
Thanks!
|
I'm working on a project on which we use WebRTC and the RecordRTC library to record audio files.
I would like to be able to process the audio files server-side to ensure a good synchronisation.
To do so, I'm thinking about marking the start of each record with an inaudible sound marker in order to analyse new records a... |
I have a matrix [66 x 100,000] (it's some EEG data, across 66 electrodes) and want to find the mean power in the 8-13hz range. How can I do this?
I have so far: x = fft(data) What do I have to do next?
Furthermore, when I plot this I get the following::
I was hoping for something more like this:
|
The 2D discrete image signal's quaternionic fourier transform can be described as F(u,v), and its 2D analytic signal often be described as:
$$
F_q(u,v) = (1 + \mbox{sgn}(u))(1 + \mbox{sgn}(v)) F(u,v)
$$
$$
\mbox{sgn}(u) \triangleq
\begin{cases}
1&, u>0\\
0&, u=0\\
-1&, u<0
\end{cases}
$$
$F_q(u,v)$: quaternionic anal... |
I am working on a system processing ultrasonic echo reflection. The transmitter creates a short ultrasonic pulse and the receiver analyses the reflection.
If I send two pulses (back to back or with some small defined delay) I receive two superimposed reflections.
I can process these superimposed reflection with the IIR... |
I have the following vector:
%create vector of sampling
t=0:pi/100:2*pi;
x=sin(t);
What is the sampling frequency (of this vector) ?
Thanks
|
I am trying to estimate the fundamental frequency in a signal to corroborate the result which I have got by FPGA implementation of an FFT algorithm. Hence, could someone give me some idea how to do so?
|
How to create a vector of random numbers(in Matlab), when each value is [0,1] .
And the sampling rate should be fixed ?
thanks
|
I have a question.
Suppose we have a signal $x(n)$, length (samples) $N=400$ which have been sampled with $f_s=8000 \mathtt{Hz}$. Also suppose $X(k)$ - the DFT transform of this signal.
How many zeros we must add at the end of $x(n)$ in order to change the frequency resolution of the DFT to $1 \mathtt{Hz}$?
My question... |
Is it possible to compute the output from one linear time invariant system which unit impulse sequence is:
$$h(n) = \left(\frac{1}{3}\right)^nu(n)$$
and the entry in the system is:
$$x(n)=u(n)-u(n-2)$$
where $u(n)$ is the unit step sequence?
Or I need more information (so something missing)
|
I am reading the Wiki for DFT. There is a part for circular convolution theorem which sounds a bit odd saying:
$$ \mathcal{F} \left \{ \mathbf{x\cdot y} \right \}_k \ \stackrel{\mathrm{def}}{=} \sum_{n=0}^{N-1} x_n \cdot y_n \cdot e^{-\frac{2\pi i}{N} k n}
=\frac{1}{N} (\mathbf{X * Y_N})_k, \,$$ which is the circular... |
I am currently writing my bachelors thesis about on-line handwriting recognition.
In this context, there are different methods to make the input data more useful to classifiers / to make learning more effective:
Scaling the drawn symbol
Shifting the drawn symbol
Rotating
Re-sampling
Wild point detection
I think all o... |
From my understanding, the Fourier series is a way to describe an arbitrary continuous signal in terms of sinusoids of varying frequencies and amplitudes, as shown here in Equation 1.
Why are the phases of the sinusoids not considered?
|
I would like to know is there any possible approach to recognise digits of the first (and second as well, as you can see in the example figure) order of magnitude. I've thought of neural network with full, half and possibly quarter training samples of the digits I want to recognise, but I don't think this is the best a... |
I have three questions regarding the correlation meters used in audio.
How are they calculated? I have heard correlation is performed by convolving one signal with the time flipped version of a other, but this seems impossible in real time.
How does this differ from the pearson's correlation coefficient used in statis... |
I have recorded samples of lungs sound (person breath) signals from electronic stethoscope.
How can I measure (In percentage) the similarity between two audio signals (lung sound), the general idea is to take a clean recorded signal(without environmental noises, Laboratory conditions) and compare it with the regular re... |
I am doing a baseband simulation for IEEE 802.11 g. The standard uses 64 IFFT length ($3.2 \mu s$) and 20 Msamples per second. This result in one sample per symbol. I am wondering what the structure will be if I use 40 Msample per second, i.e will the IFFT size change, or interpolation will be used?
Thanks
|
Literature says that a stationary signal is ergodic, if its ensemble average = time averages. Should it be the statistics computed by time averaging = statistics computed by ensemble averaging?The way I understood from the book in the link is as follows :
Case1: There are N samples in a time series I compute the mean1... |
I have made a stft_func in MATLAB which takes window length i.e t_short as a parameter for window length. Now, this function is giving me a very blurred image with $f_s=1000\mathtt{Hz}$, radar target $10\mathtt{GHz}$ (fm) and $T_{short}=0.5 \mathtt{s}$
How to find frequency resolution from spectrogram.?
|
I have the following (experimental) image, with its correponding raw data in .csv format here.
The pattern represents the scattering of a laser field off a point defect located in the center. One can clearly see a V-like shape of the scattered waves (here drawn in green). Unfortunately, there is also quite a lot of ba... |
I have some confusions regarding reverberation time (RT60). I need to calculate reverberation time from a given power envelope.
This is what I get. As you might see the line is curved close to zero (at top left corner) and then becomes straight. Is t-intercept equal to RT60, or do I need to extend the straight line at ... |
Despite,reading up on this I'm unable to find crystal clear clarity on the differences b/w phase and grpdelay functions and interpretation of grpdelay's output. I have designed a low pass butterworth filter having a cut off frequency of 0.04Hz. There is a delay in my filtered output compared to my original signal. I wi... |
Does modified Morlet wavelet function satisfy the admissibility condition ?
Do you have any reference for the answer please.
Thanks.
|
I know that we can get the phasor representation(frequency domain) of a signal through Fourier transform, but the picture above gives a method that can get the phasor representation of a signal in a different way, in this method according to the writer, it numerically "demodulate" the signal to create a phasor represe... |
Today, a news blurb appeared that claimed that attackers could identify the location in which an audio record was taken by matching the mains hum to known data. Sadly, I fear that this might be feasible. The mains hum has quite a tight tolerance, but any aberrations (that cannot be done away with) could be spotted easi... |
I'm looking for implementation of MPEG-7 descriptors which are compatible with OpenCV's recent versions and containing "region-based shape descriptor ART(Angular Radial Transformation)".
I encountered an stackoverflow question which had very informative answers:
Link in one answer was containing only the following des... |
I try to design a FIR All-pass filter with random phase in the frequency domain. I am a bit confused by my result and am not sure if the reason is a programming error or a misconception about convolution and impulse response of myself. I use Python and Numpy:
import numpy as np
mag = np.ones(129) # magnitude is unitiy... |
I am acquiring data through an oscilloscope, which I am then filtering. I am doing all of this in Python.
I need a bandpass filter with stop bands from 0 to .4MHz, and 4MHz and up. My pass band needs to be in the remaining region and would ideally be as large as possible.
I checked out a text from the library to help d... |
How can I analyze a basic sine wave for its frequency phase and amplitude at a particular point in time in MATLAB? Are there any tools to do that? I am trying to do additive synthesis of inharmonic sounds in MATLAB, but having a great difficulty to extract the needed phase amplitude and frequency data of the partials I... |
I was searching on the Internet hopelessly about some materials regarding detecting a sound in another sound. Say, I've got a recorded short sound (which may be anything, fragment of speech, fragment of music, noise, their combination or something else) and a longer sample, which contains this sound somewhere. The prob... |
We all know the famous Wiener Filter,
$$G(\omega_1 , \omega_2)=\dfrac{H^*(\omega_1 , \omega_2)}{|H(\omega_1 , \omega_2)|^2 + \dfrac{N(\omega_1 , \omega_2)}{S(\omega_1 , \omega_2)}}$$
How to we use it practically? Meaning, given $y = s * h + n$ where $s$ is the original image, $h$ is a LSI operator, and $n$ is some nois... |
I am having trouble understand what I think should be a pretty simple concept. Conceptually I can understand that a signal that has been down-converted, and then delayed in time is different than a signal that has been delayed in time and then down-converted.
I am trying to compensate for this difference on an FPGA (I... |
I know that frequency is not the same as the rate of change of amplitude, but why?
Surely a high frequency is when your amplitude changes rapidly. Does an FFT perform something similar to differentiation of a function?
Furthermore, what else is needed other than my amplitude values to plug into an FFT to get frequencie... |
Why do people calculate impulse response? Generally systems are designed after modelling the possible input signals and calculating the appropriate filters for the desired outputs. Where does impulse response calculation become important?
|
The reason why Hu moments were implemented in OpenCV and why Zernike moments were not implemented is looking like their performance similar as stated in this paper.
As stated in the paper Zernike moments' advantage is their reconstruction facility.
If there is a reliable implementation of Zernike moments compatible w... |
I am working on building detection from aerial images using Image Processing in MATLAB.
I am doing an image segmentation based on the local homogeneity of image. I want to apply K Means Clustering in CIE LAB Color Space with some clusters on the anisotropic diffused image for color quantization and generate the image f... |
I'm currently stuck on a problem. I have a DT LTI system $\mathcal{T}$ that depicts the first backward difference:
$$y[n] = \mathcal{T} \{ {x[n] \} } = x[n] - x[n-1]$$
Its impulse response is therefore: $h[n] = \delta[n] - \delta[n-1]$, where $\delta[n]$ is the DT unit impulse.
I want to compute the impulse response $... |
I'm working this summer on a very interesting, albeit difficult to understand at 100 %, project and it turns around reducing the glare effect than could be applied on a receiver when he gets signals coming from two emitters sending $X(t)$ (more energy for this signal compared to the other) and $s(t)$ (much weaker).
Let... |
I'm testing a background subtraction algorithm with sample image sequences. I am wondering how one can evaluate the accuracy of a background subtraction algorithm results given ground truths?
The only idea I have in mind right now is to take the difference between a result image and its corresponding ground truth imag... |
I am a beginner in DSP and is familiar with $\sin(\theta)$ declaration (as studied in trigonometry) - where $\theta$ is the angle it makes with X - axis. Please tell me what does $\sin(2\pi f t + \phi)$ equation means and how $\theta = 2\pi ft + \phi$.
Thanks in advance
|
I am a beginner in DSP but get confused when some formulas like below are referred:
cos(φt) · { A·cos(φt+Δ) } = ½·A·{ cos(Δ) + cos(2φt+Δ) }
x(n) = Acos(ω0n+φ)
T= 2kπ/ω
...........
The problem is that I do not know what terms like 'φ' or How T is equal to 2kπ/ω.
I know that I am lacking some elementary knowledge - but... |
Let us consider following code:
fs=100;
N=1000;
t=0:1/fs:N-1;
y=10*sin(2*pi*100*t)+0.5*randn(size(t));
plot(y)
plot(t,y)
I have applied following threshold methods:
thr = 0.4;
% Perform hard thresholding.
ythard = wthresh(y,'h',thr);
But I can't see any difference, also soft threshold method,codes are below:
yt... |
Here is a small MATLAB function that computes B in B/A given A and the impulse response h of B/A
function b = numf(h,a,nb)
%NUMF Find numerator B given impulse-response h of B/A and denominator A
% NB is the numerator order.
nh = max(size(h));
impr = filter(1,a,[1 zeros(1,nh-1)]);
egqs = toeplitz(impr,[1 zeros(... |
The following text is from Digital Image Processing(Gonzalez and Woods)[3ed, Ch 2.2,p46]
If a sensor can detect energy radiated by a band of electromagnetic spectrum, we can image events in that band. But the wavelength of electromagnetic wave required to “see” an object must be same size or smaller than the object.Fo... |
I've a fundamental problem with understanding terminology used in DSP. What does it mean to tune a radio (for example, a software defined radio) to a certain carrier frequency $f$? The received signal is an arbitrary waveform and not a sine wave of frequency $f$. I understand AM modulation: you take a sine and manipula... |
I want to match a small template to a larger image, that the distance between the template and the subimage with the same size of the template is minimized. It can be solved directly or by applying 2D cross correlation, and both methods have an O(n^4) time complexity. Are there any method to simplify the algorithm by e... |
I was reading my book where the z-transform of a signal is derived to be ${1-e^{-2bT}z^{-1}}$ . Then it goes on to say that by applying the bilinear transform we can get
$$\frac{2(1+bT+(bT-1)z^{-1})}{T(1+z^{-1})}$$
I know a little bit about bilinear transform but this example is using it in a different context. Can so... |
FIR Networks are a really great kind of Neural Networks. It allows very precise predictions of missing samples for example when ones wants to reconstruct missing signals or adaptive noise cancellations, etc
I'd like to understand how they work and a great paper about it is the one by WAN in 1993 : http://www.cslu.ogi.e... |
I need to model the dispersion which is introduced by an EML in Matlab. The parameters are:
Lambda: 1310nm
PAM4 power:
Pmax = 1.09 mW
Pmin = 0.13 mW
Bandwidth = 56Gbps
|
As I understand it, FIR filtering is a linear process. That mean for me that the whole filtering process will have a fully predictable behavior.
So, could someone explain why a universal deterministic and optimal filter design method to obtain the desired response doesn't exist and why recursive approach seems to be ... |
Since Fourier transform of a random stationary process in time (in the case of existence) is not necessarily real, my question is what is the relation between the covariance of real and imaginary parts of the transformed random variable.
More specifically, according to Wiener-Khinchin theorem for the time function $x(t... |
Under the simulation of a communication system suffering from noise (AWGN) and co-channel interference, what is the appropriate way of calculating SINR?
Could one calculate it at a packet level as:
$$
SINR_{packet} = \frac{\sum{\left \| x \right \|^{2}}}{\sum \left \| n \right \|^{2}+\sum \left \| z \right \|^{2}}
$$
... |
Fourier transform of a continuous signal is just the projection of the signal on the sinusoidal family for imaginary part and the same family with phase offseted by a quarter of period for the real part.
Beside the magical Euler's formula $e^{ix} = \cos x + i\sin x$ that ease the analysis of the transform and the fact... |
I am trying to implement speech recognition but I'm struck with a few questions:
There are 12 Coefficients of MFCC. What are their names?
What is the range of values of MFCC coefficients?
If I want to relate MFCC coefficients graphically, what will be unit of weights on their edges?
|
The way I understand it, computational science covers a wide range of fields, all of which use computers and mathematical models to solve a problem involving one of the sciences (physical, life etc.).
But is DSP, which does use maths to manipulate a signal, a form of computational science?
|
I'm trying to do a program that can judge whether 2 audio files have the same pronunciation or not by comparing the frequency of them with each other.
The first step, finding the magnitude and phase data from the audio file by using Discrete Fourier Transform, is done. I now have a list of the DFT data from both files ... |
Suppose two continuous complex domain signals are convolved, how is the magnitude and phase of the resultant signal related to the magnitudes and phase of the original signals?
|
I am not sure why zig zag manner scan is used in DCT. I can guess that might because it moves from low frequency component to high frequency component and as most of the energy is stored in low frequency component that's why this way of scan is favored but not quite sure.
|
I'd like to estimate the power spectral density of the signal attached here, the sampling rate is 100 Hz.
If I don't do windowing (i.e., use boxcar window), the result seems to be a simple $1/f^2$ noise:
import json
from scipy import signal
from pylab import *
x = json.load(open('x.json'))
f, p = signal.welch(x, fs=100... |
-----------------deleted by author to merge with valid account
|
I am working on a system that is implementing a digital filter that was designed for another system that had an FPU and a 12bit ADC. The system that I am working on is using the same filter without an FPU using a 12 ADC and fixed point arithmetic.
The ADC resolution is 3.3v / 2^12 = 0.000805664062v
With an FPU each AD... |
I am trying to filter EEG signals using butterworth filter and filtfilt. I have gone through a lot of documentation and these 2 commands seem sufficient for filtering. However, the results are bizarre.
from scipy.signal import butter, filtfilt
import sys, pickle
from numpy import *
import matplotlib.pyplot as plt
def ... |
There are two real signals in the form of $A_i sin(wt+p_i), i=1,2$. Suppose frequency $w$ of both the signals is the same and amplitude $A_i$ and phase $p_i$ are different. The first signal has unknown $A$ and $p$, but they are constant in the time of observation (for simplicity only, actually they aren't). The second ... |
I have a real signal that is being sampled via ADC and passed to a Xilinx V6 FPGA. I then do a Hilbert transform on it using the FIR compiler core and get the complex signal (I and Q) to tweak the signal some. Now I am struggling with how to do the inverse Hilbert to get back to a real signal again.
I understand in t... |
I have a UWB radar that Im playing with. The problem is that my algorithms at the moment can clearly, reliably and validly pick up 1-2mm movements of a pendulum i've set up. However, when I use it on my chest to find my heart, things doesnt work. The radar should be set up so that the waves penetrate skin and bone, but... |
For a project I am working on, I need to incorporate GPS data into an audio file. As things currently are, the easiest way for me is to record the audio in a multi-channel .wav file, and add the GPS data as a square wave in a separate channel.
GPS data will be binary, i.e. either an arbitrary high or low signal for an ... |
I'm trying to make some simply autofocus algorithm for my device and pretty confused with 2D FFT results.
As I know for now - number of high-frequency's components in 2D FFT will say me that image is more focused, than other one where 2D FFT got more low-frequency's components. And i do not know how to calculate this p... |
A reviewer has asked me to re-filter my data to remove baseline drift.
Each sweep is 170ms sampled at 1000Hz and the reviewer wants it high-pass filtered at 0.5Hz. The original bandpass settings on the hardware were 0.15 to 1000Hz
Although this is easy to code in matlab, the epoch is very short compared with the stop ... |
Autocorrelation of white noise should have a strong peak at "0" and absolutely zero for all other $\tau$ according to this. Then why is output of this code a cone shape (with the expected of strong peak at "0" instead of being flat elsewhere except at "0")
y = ones(1,1000);
z = awgn(y,1);
[m l] = xcorr(z);
plot(l,m);
|
I am getting a signal from a device. This signal contains a non-random noise signal in it. I want to remove that noise from the signal. This noise is correlated to the motion the device is having. So now I've attached a accelerometer to the device to measure the motion produced by the device. I'm getting the data of X,... |
As a part of my research i have to use the following z-transform in matlab 'filter' function so as to derive the convoluted signal from the original one.
$$\frac{1}{{\rm Li}_{k}(z^{-1}e^{-b})}$$
But since i read that the polylogarithm can be expressed as a function only for specific values of k (k can take many value... |
This question is a follow-up for my previous one: Reduce the "glare" effect of a receiver within a system.
I have broken the problem into pieces and I humbly think that I've got a way to solve it it with Duet BSS. However, there is still a problem: I have got
$$Y(t)=h_1(t)\otimes X(t) + h_2(t)\otimes s(t)$$
Is there a... |
I have a set of data over which I would like to interpolate, with a sampling rate say about $f_0$ Hz with a significant uniform random sampling jitter, such that it's more or less:
$$f_0 - \frac{1}{4}f_0 \le f_s \le f_0 + \frac{1}{4}f_0.$$
Normally, with uniform sampling, I can perform sinc interpolation:
$$x(t) = \sum... |
I'm reading the book
Vinay Ingle, John Proakis - Digital Signal Processing Using MATLAB® (3rd Edition) and I came across a function to shift sequences. Mathematically,
$$
y(n) = \{x(n-k) \}
$$
if $m = n-k$, then
$$
y(m+k) = \{ x(m) \}
$$
function [y,n] = sigshift(x,m,k)
n = m + k;
y = x;
end
This function shifts no... |
I'm reading a book about dsp and doing some exercises.
Here is question that confused me:
A peak appears at index number 19 when a 256 point DFT is taken of a signal
1) What is the frequency of the peak expressed as a fraction of the sampling rate? Do we need to know the actual sampling rate to answer this question?
2... |
I was pointed here by a chain of stackoverflow posts. I'm a software developer looking to make a module that recognizes noises like snorts and coughs. Here was my original question on stackoverflow and here is where I found this SE board.
In short, I'm trying to use a Raspberry Pi and a microphone to pick up noises li... |
I have very limited resources as I'm working with a microcontroller. Is there a taylor-series expansion, common lookup table, or recursive approach?
I'd prefer to do something without using math.h's sqrt()
http://www.cplusplus.com/reference/cmath/sqrt/
|
Is there a type of filter for processing spectrum data that will subtract spikes in intensity? I was thinking of something like a low-pass filter but that takes away data if the intensity is too high. I am trying to do this since there is a lot of bright intensity measurements in a certain frequency range that are maki... |
I want to do some simulations of speech recorded on microphones. In order to simulate the channel response, are there any standard (FIR) filter parameters available that I can use?
|
What is the proper way to calculate the frequency content or spectrum of a signal with a variable (but measured) sample rate? I.e. get similar results as you would from an FFT, but for data not evenly spaced in time.
I know I can resample the data by interpolating on an evenly-spaced time vector then apply a FFT, but ... |
I want to implement ETSI/BRAN Indoor Model A channel, i have the delays for the paths tau=[0 10 20 30 40 50 60 70 80 90 110 140 170 200 240 290 340 390]*10^(-9); and the Average Relative Power (dB) pdb=[0 -0.9 -1.7 -2.6 -3.5 -4.3 -5.2 -6.1 -6.9 -7.8 -4.7 -7.3 -9.9 -12.5 -13.7 -18 -22.4 -26.7]; and i want to create time... |
I want to filter sensor signals for a data acquisition application on an ATmega microcontroller. The interesting parts of the signal are in the range 0-5Hz, and I'd like to leave these mostly unchanged, but strongly attenuate 50Hz power line hum and anything above that.
The signal will later be evaluated in the time do... |
I read a paper that said about correlation kernel that defined:
$$W(x-y)=(α/1+d(|y − x|))$$
where $α = (\int(1+d(y − x)dy)^{-1}$, $(d(|y − x|))$ is spatial Euclidean distance from the central pixel.
My question is what is correlation kernel and benifit of that kernel comparison with gaussian kernel? Thank you so much... |
I am searching for the algorithm that takes (audio) signal as an input and outputs weighted intensity of the frequencies of some narrow band (axis X) in time (axis Y). Frequency in the middle of that band would have highest weight.
In other words: It is like spectrograph but I am interested only for the weighted avera... |
If $x[n]$ is input signal, $y[m]$ is output signal, $h[m]$ is impulse response of the channel, then the following relation holds
$\lvert Y(\omega)\rvert = \lvert H(\omega)\rvert^2 . \lvert X(\omega)\rvert$
where $F(\omega)$ is the DFT of $f[n]$.
To verify this, I wrote following MATLAB code
x = randn(1,1000);
h = [1 2 ... |
I want to compare the performance of SIFT and SURF against some standard image set.
After a bit of research, I found that ROC curves are one of the best ways to do this task.
In order to plot ROC curve TP rate and FP rate are needed.
Output of SIFT and SURF are detected keypoints. How exactly can I generate ROC curves ... |
I am trying to make a panadapter for a software defined radio and I am a little bit stuck and hope someone can help me.
I have an I Q signal from my radio. (Actually I am using an IQ wav file from the internet recorded at 44800 2 channels 16bit).
If I use HDSDR the spectrum looks like this:
The display from my app lo... |
How to recover the carrier of a PSK signal that has high modulation index (eg. 750 symbols per second at carrier frequency 1500 Hz)? I guess that a PLL will not be able to lock onto it with such frequent phase transitions. Maybe upconverting or frequency multiplication of the signal before the PLL is the solution?
|
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