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I'm looking for a DSP, around 500MHz, with an in-chip ADC (preferably 2 but 1 will suffice) that samples around 20MSPS. Does any such DSP exist or do I have to use external ADCs?
|
I am trying to further reduce the following expression (if there's a nicer way to write it):
$\sum_{k=-\infty}^{+\infty}\sum_{p=-\infty}^{+\infty} \sum_{l=-\infty}^{+\infty} w(k) \varphi_\mathrm{y}(l-p)\varphi_\mathrm{x}(k-n-l+p)$
where $\varphi_\mathrm{y}$ and $\varphi_\mathrm{x}$ are the autocorrelations of the signa... |
If my signal is
$$
f(t)=\exp[i\phi(t)],
$$
how do I show that
$$
\int_{-\infty}^{\infty} {\xi}\left|S_f(u,\xi)\right|^2 d\xi = 2\pi\int_{-\infty}^{\infty} {\phi'(t)}\left|g(t-u)\right|^2 dt,
$$
where
$$
S_f(u,\xi)=\int_{-\infty}^{\infty}f(t)g(t-u)\exp{(-i\xi t)} dt
$$
is the short-time Fourier transform of $f(t)$?
|
Say I have a function, $C=C\left(x\right)$, whose fourier transform is denoted by $c=c\left(k\right)$, i.e. $C\left(x\right)=\sum_{k=-\infty}^{\infty}c\left(k\right)\chi\left(x\right)$, where $\chi\left(x\right)$ is some complex valued basis function.
I know that the autocorrelation is given by $M\left(\Delta x\right)=... |
I'm looking for a way to implement a Speech Separation from a .wav file. Let's say I have a .wav signal containing the following words:
"The cat flew away"
Now I don't really want to do the speech separation after I've split the signal inside blocks of size NxM because I feel as though I could miss parts of the word.
... |
I am studying windowed Fourier ridges for my project. I know that it computes the instantaneous frequencies and amplitudes from the spectrogram. But I don't really know how it relates to signal processing, can anyone explain this concept to me?
|
Given a 2D image where each row of pixels is shifted by some amount horizontally in a sinusoid fashion, is it possible to recover the original image?
Imagine a camera that can capture one row at a time, and is moved vertically to capture the whole image. While it is moved, the camera vibrates, resulting in noise simil... |
I am trying to implement a Savitzky-Golay filter by following this Wikipedia page and in the first instance I have written the following Octave software code to create the convolution coefficients
clear all
m = input( 'Enter number of bars (must be an odd number): ' ) ;
% check no_bars is odd
if mod( m , 2 ) == 0 % i... |
What is the reason that a periodogram is not a consistent method for obtaining information about frequency response of a system?
To be more precise, why even after increasing number of samples doesn't the variance of the estimator decrease?
|
I have some data that was collected at irregular intervals, and I want to estimate the values on a regularly spaced time grid, so that I can take FFTs etc. I've not had to do this much before, and I'm wondering if there is an "industry standard" or best algorithm to use? I'm currently using the Matlab function pchip, b... |
Consider a static AWGN channel
$$
Y = hX+N
$$
where $h$ is a constant channel gain and the noise $N(0,N_1)$ is independent of the input $X$. The input is from a 2-PAM constellation, i.e., $X \in \ \left\{+√p,-√p \right\}$ with equal probability where $X= +√p$ is used to send a bit at 1 and $X=-√p$ is used to send a ... |
I have some misunderstandings with Q format. Please help me in understanding this.
If I say Q8 [unsigned] then there will be $8$ fractional bits. Is it ? How can I represent a $1$ and $1.5$ in this case?
If I say Q8 [ signed ] then there will be $1$ sign bit and $7$ fractional bits. Here how can I represent a $1$ and ... |
The idea is to get three classifications for crowd: High, low and medium based on an image captured inside a train compartment. Something like this:
This gets classified in the 'High' category
http://www.spiceflair.com/wp-content/uploads/2012/05/14a.jpg
This perhaps in the 'Medium' & so on.
|
I have a piece of code that I am translating and I am trying to understand the reasoning behind it. The application here is thresholding medical (cellular) images.
The function in question takes a histogram and returns the lower threshold value. The bit of code that I am translating mutates the histogram in a way tha... |
In a direct sampling RF front-end, we can get real samples.
But when we sample the signal with two ADC with a sampling delay $ \frac{4n+1}{4fc}$ where $n$ can be $0,1,2 \dots$
Then we have In phase(I) and Quadrature phase(Q) samples outs of two ADC.
But there is a problem when we sample "I" before symbol transition and... |
The LP-PSOLAR algorithm uses a the Linear Prediciton Coding (LPC) to calculate an error vector from a signal. This error vector is used to calculate the pitch markers. What do these pitch markers represent in the original signal in generall? Do they represent a change in fundamental frequency or a change in phoneme (I ... |
Consider the measurements of two modulated AC signals (voltage and current) which may have noise, harmonics and a DC bias. Each measured signal consists of a list for the time and a list for the corresponding values.
My goal is to find out the phase shift in degrees between these two signals in a robust way.
I would li... |
I have floating point data signal. I apply lossy control to it and I need some kind of metric to determine its quality.
In image processing, there is PSNR (or MSE). I tried to use the same for it, It gives me some kind of result. But is there anything better (or any standard used for this kind or problem) ?
|
What I am attempting to do is take a signal (.wav audio file) which contains calls from different Bat species. Here is a graph representation of the signal:
What I have done so far is:
1) Emphasise the signal
2) Compress the signal
This is the output I got plotted:
I need to remove the signals that do not warrent con... |
I know that in many edge detection algorithm the first step is choosing a scale to work on.
We actually operate on the chosen scale by applying a gaussian blur on the original image.
I wanted to know , what is the relation between the detected edge's width to the sigma.
e.g. If I want to detect lines 5 pixel wide , w... |
I just studied the Algorithm of ANPR (Automatic Number Plate Recognition) from the chp 5 of book Packtpub.Mastering.OpenCV.with.Practical.Computer.Vision.Projects in which they detect the vehicle licence plate , but their are some restriction in that algorithm , i call it restriction because its specific for one countr... |
I am trying to find out how can I count the number of people in a room using the feed coming in from a CCTV camera fixed in one of the top corners of the room. The number of cameras can be 2 at max but 1 for now. The objective is to only be able to tell how many persons are present in the room, in real time. The precis... |
I am reading a nice book on DSP
The Scientist and Engineer's Guide to Digital Signal Processing
It mentions
In electronics, the mean is commonly called the DC (direct current) value.
How can a signal have no DC component if the mean is the DC component? Surely every signal's values can be averaged?
|
I'm trying to do something similar to Shazam (paper here) in Python. Basically the idea is to use FFT with sliding window to transform both WAV files and recorded audio to the same spectral representations for recognition purposes.
The only problem is the spectrograms I extract from certain songs just seem to generate... |
In image processing, scale-space is a technique to represent an image at different scales. But this really confuses me, since I always think that the word scale just means the size of an object, so the scale-space representation of an image shall have different sizes of the same image.
However, it turns out that the s... |
We can describe a BPSK signal, x, as:
$$x=cos(2\pi f_c t+n\pi),\;n\in \{ 0,1 \}$$
The BPSK signal is then transmitted through a narrow band frequency flat Rayleigh channel, described by a single complex exponential $h=Ae^{i\theta}$. $A$ has Rayleigh distribution and $\theta\sim\mathrm{Unif}(0,2\pi)$. The received signa... |
Section 13.2 of the HD Radio standard for AM defines a pulse shaping function that appears to be some strange variant on a raised cosine filter:
More specifically, H(ξ) is a raised cosine filter with total width (1+α)T, and W(ξ) appears to be a "stretched" version. Why do they do this?
|
I was reading section 4.3.3 from chapter 4 of "A Wavelet Tour of Signal Processing",
"...Let $\bar{f}(t)$ be a continuous time signal defined over $[0,1]$. Let $f[n]$ be the discrete signal obtained by a low-pass filtering of $\bar{f}$ and uniform sampling at intervals $N^{-1}$.
Its discrete wavelet transform can only ... |
So, in my class I have to learn how to derive the power relation in a Double Side Band, Full Carrier Amplitude Modulated Wave However, I have hit a wall somewhere.
The basic equation is this :
$P_{total} = P_{carrier} + P_{lower-side-band} + P_{upper-side-band}$
Then,
$P_{carrier} = \frac{E_c^{2}}{R}$ where $E_c$ i... |
I asked this question here: Audio Separation of .wav signal but it wasn't clear, so, here is my second attempt:
First off, assume that I have a .wav file containing a sentence as follows:
"My name is Michael" I would like to extract, from this, the following:
"My" -> Phoneme (1)
"Name" -> Phoneme (2)
"Is" -> Phoneme... |
This might be more of a statistical question, but let us assume that we take a spectrogram, or STFT, where time is on the x-axis, and frequency is on the y-axis in this matrix. What information might be lost - or 'changed', if we decide to normalize each column by its own standard deviation, so that the variance of eac... |
Given this problem
I know that the autcorrelation and power spectrum are fourier pairs, so when taking taking fourier transform of Rxx, one should end up with Sxx
However, when I take the transform of $R_{xx}(m)$, I end up with $S_{xx}(\omega)=\frac{2\alpha }{\alpha ^{2} + \omega ^{2}}$, which also matches up with fo... |
For a real signal, the power spectrum will have Hermitian symmetry. If I consider the one-sided spectrum (as negative frequencies, or anything above the Nyquist frequency are redundant) must I then multiply my power spectrum by a factor of 2? I have seen this done but I can't understand why the "redundant", symmetrical... |
Consider the $m$-dimensional VAR process
$${\bf x}_t = \sum_{l=1}^{P} A_l{\bf x}_{t-l} + {\bf e}_t$$
where the componenets of ${\bf e}_t$ are spatially and temporally independent and follow a generalised Gaussian distribution i.e. $e_{i,t} \sim GGD(\alpha_i, \beta_i)$.
Given my data $X$, I am trying to estimate the VAR... |
Say I have a signal that is not x-indexed. That is, the x-axis of the signal is the distance traversed by car and the y-axis is the heading direction of the car at the corresponding distance.
Can I apply the Fourier Transform to this signal?
If so, what is the physical meaning of this transformation? I believe that the... |
I have gone through the below related posts of filtering noise from the signals
MATLAB, How to filter a discrete signal?
MATLAB: Filter a noisy EKG signal
But the signal I want to filter contains noise which is not periodic as shown in the images below:
Can anyone suggest a filtering technique for the noise labeled w... |
I am in a intention of Porting Voice Activity Detection algorithm in real time hardware,so i was successful in classifying speech and silence/noise but now the problem is replacement of silence frames with comfort noise(preferebly any stationary noise)
so can any one suggest any opensource CNG algorithms or how a comfo... |
I am new to signal processing and I am trying to get the basic concepts right.
I have a signal and I performed the FFT of the signal in MATLAB. I also did the PSD of the signal in MATLAB using the psd command.
So I am confused about:
1) When we get the FFT of the signal, It's basically showing what frequency components... |
There are various types of ADC in used, like,
ramp-compare ADC
Wilkinson ADC
pipeline ADC
Flash ADCs
Sub-Ranging ADCs
Folding ADCs
Pipelined ADCs
Successive Approximation (Algorithmic) ADCs
Integrating (serial) ADCs
but Successive Approximation (Algorithmic) ADCs (ADC0808) is most commonly used.
I do not understand ... |
I have the following system:
Where $ {H}_{1} \left( f \right) = {H}_{2} \left( f \right) $ and $ \theta \sim U[0, 2\pi]$ independent of any other factor in the system.
Given the input is identical, Is this an LTI system?
Could you prove it?
Otherwise I don't have explanation why if the input is White Gaussian Noise wi... |
I am dealing with a complicated optimization problem with two parameters $N, K$. Setting $K =$ constant and finding the minimum for different $N$'s lead to a set of function values. I can do this for $K = 2,3,\cdots,8$ and $N = 5,6,\cdots,40$. Larger $K$'s or $N$'s take more time to calculate (e.g. 1 hour).
I've plotte... |
i've read that it is possible to interpolate between two sets of filter coefficients (if they are close of course), but how to interpolate between two sets of coefficients which are of different orders? Is it possible ? If not what is the possible way to fade between two IIR filters ?
thanks
Jeff
|
Are there any known methods to use existing keypoint detectors (which generally work on grayscale images) on color images? of course, one could always add the results of the detector on all three channels to get $3N$ keypoints, but I was wondering if there exist other established methods.
Specifically, I am looking to ... |
I am working in image cryptography using an ICA algorithm like JADE. To evaluate my work, I am used the SDR and PSNR.
I would like to know what the acceptable values of the SDR and PSNR in image blind separation are.
|
Which one is better for frequency detection? In what circumstances? Advantages and disadvantages of both? Which would be better for a Voice Activation Detection algorithm? Which would be better for an instrument tuner? How would one detect pitch user an HHT transform? Is it faster than FFT? How would one implement HHT?... |
I work with images in gradient domain because I sometimes need to manipulate image gradients (i.e. Laplacian filter responses) rather than intensity values.
However, I need to apply a geometric transform (distortion) to an image already in gradient domain.
Is is then meaningful to reintegrate the distorted image back t... |
Sorry if this is elementary. If I add a small sinusoidal signal to my data and then take FFT of both my original and the modified one and then inverse FFT the division of the two, I should then be left with the sinusoid. But when I do the above, I get a shift in my output, as if the phase is lost. What am I doing wrong... |
I am trying to understand one simple thing in wavelets and I searched a lot, but did not get any idea.
I need to perform Multilevel 1-D wavelet decomposition on my signal. I have arrays
signal[] and time[]
( time[x] corresponds to the time at which signal[x] occurs ) . After I perform
wavedec(signal,level=1,'... |
I am doing my project on speaker recognition. One of my friends is doing his project on speech recognition. Both of us need to calculate the MFCC for feature extraction.
Is the MFCC going to be the same one for both of us? Is there any difference between the algorithm of MFCC for speech and that for speaker recognitio... |
Let the function be
$$
f(t) = e^{-3t} u(t -1)
$$
The criterion is set at 70% of the total signal energy.
I know how to find total energy, but not how to find the the $W$ (or frequency) at which the signal is a certain amount.
|
I have worked with state space models in relation to Kalman estimation. Here I have always seen state space models with fixed state size over time, i.e. the state transition matrix is square.
Let us for example define:
$$x_{t+1} = A_t x_t + B_t u_t\\
y_t = C_t x_t + D_t v_t$$
Then $A_t$ is $n \times n$.
I suppose I cou... |
Camera setup: We have setup some stereo cameras in a living apartment. That is, the indoor environment is monitored. With the stereo cameras, wide lens (3.5mm) are used to cover a big volume.
The height from the floor is around 2.8 meters. The objects (such as mug, bottle, telephone) are at least 3 meters away. For ins... |
I know there are methods to determine the degree of polarization of multi-dimensional signals, say [Bx(t), By(t)] -> DOP(f). Is there any way to extract a DOP for given vector time series in the time domain? Thanks!
|
In image processing, up to which value of mean and variance can a Wiener filter be used to denoise the image?
Wiener filtering can be used when variance of the noisy image is low, but I don't know the specific values of variance and mean.
What are limitations of the Wiener filter?
|
We know that the square window gives the lowest main lobe width possible, and that other windows after that trade main lobe width for side lobe height. I also understand that the main lobe width is inversely proportional to the length of the DFT aperature.
Given this, is there then a physical limit to how narrow a mai... |
I have signals captured from two channels of a measurement system. There is phase and magnitude differences between these two signals.
I want to apply nonlinear smoothing to the magnitude, but the phases should remain the same after the process.
This is what I have done both channels:
Perform FFT
calculate magnitude
... |
I am looking for a implementation of Hilbert-Huang Transformation. Ideally, in Java, but any language with similar syntax would prove useful. Could someone point me to a specific implementation, preferably in Java?
|
http://www.google.com/search?q=databending
It is, from what I understand, opening images as audio, applying effects, exporting back as image and voila! glitch-art
Example:
credit
But how does it really work? How/what change in frequencies(?) affect the pixels? It can't be totally random since some of them aren't comp... |
I have a dataset which contains a large number of time-series which have been filtered with acausal Butterworth filters. For a real-time application, I can only use causally filtered data.
Is it possible to turn the output of an acausal Butterworth filter into what would have been obtained if the original time-series h... |
How do I estimate PSD in matlab ? There are lots of methods to estimate, depending on application.
I am using an experimental data to identify faults ( e.g. interturn fault of electric motor), which gives periodic/regular waveforms (pre-fault) but erratically different (or could have some periodicity) waveforms (post-... |
I am trying to implement the STFT by MATLAB, and I read somewhere that we must do zeropadding to the subsequence of the signal before windowing it with a Gaussian window. What happens if we don't do any windowing?
|
Suppose $h(n)$ is a finite impulse response which is unknown. We can feed any input signal $x(n)$ into the system and observe the corresponding output signal $y(n)$.
From this, is it possible to estimate the input signal $x^*(n)$ which when passed through the filter gives the desired output signal $y^*(n)$ ?
I think a ... |
I am making my way through this paper here, and I understand everything up to equation 16, and I am stuck here.
Long story short, the vector $\mathbf{x}[i]$ here is an $N$ by 1 vector, consisting of the complex envelopes of signal + noise + interference across $N$ different sensors, taken at some time index $i$. We ha... |
What I'm looking for is the mathematical proof of why we can calculate derivatives as a simple convolution of a mask, and how do we get that mask?
I know it has something to do with how the derivative is approximated
Can anyone direct me to some book or anything?
|
I have a chunk of data (samples) that need to do autocorrelation using FFT method. All data are real and lies between -1 and +1. What I have done follows:
Zero pad the window with the length equals the samples' length.
FFT of result from 1
Result from 2 has Real and Imaginary parts. I used them to find Power Spectral... |
I posted this question at Math Stack Exchange a few weeks ago without any luck. It occurred to me I might have more luck by posting it here. The basic question is whether there exists a known relationship between the period of a signal (i.e. the inverse of the frequency) and its corresponding power, such as that depi... |
A sampled single dimensional signal is a vector.
A sampled image is a matrix.
Is video considered a rank-3 tensor since it is a set of images stacked on top of each other?
|
When we design a discrete Gabor filter for a 1D signal, how do we determine the variance of the filter, depending on the time step of the signal? Is there any empirical theorem/result that we should know about the Discrete Gabor Transform?
|
While reading Estimation of Symbol Timing Offset, i came across an equation like multiplying a signal with its conjugate.
$$
\hat{\delta} = \arg \max_{\delta} \sum_{i=\delta}^{N_g-1+\delta} \left|y^*(n+i) y(n+N+i)\right|^2
$$
I know multiplying
(5+4i) X (5-4i) = 25 - 20i + 20i + 16 = 25 + 16 = 41.
But am not getting t... |
Musical residual noise is created when a noisy signal passes through a noise reduction algorithm. For instance, please see the paper: Esch, T; Vary, P: "Efficient Musical Noise Suppression for Speech Enhancement Systems", IEEE Intl Conf Acoustics, Speech and Signal Proc, (ICASSP): 4409-4412, 2009.
Noise reduction tec... |
While I was going through the short time processing of speech signals, I got stuck at the relationship between the frame rate and the sampling frequency.
Is there any relationship between them for an audio signal so that the change in sampling frequency will have an effect in the frame rate? If yes, what is mathematic... |
I have a question regarding the wavdecand dwt functions in MATLAB.
I am using wavdec(y,3,db4) on a size 1024 signal, which means I am trying to compute the third level db4 transform of the size 1024 signal y. If I am thinking about this correctly, theoretically the returned matrix after the transform should also be of... |
I am using CMOR(complex morlet) wavelet in Fourier space in order to reconstruct my signal and also estimate damping and frequencies of the embedded modes. there are two main parameters in cmor. the Fb and Fc. bandwidth and center frequency, with increasing the Fb i get better result for frequency estimation, but i can... |
I trained the data using BOG with SVM , i trained it , now i just want to open it through xml , i don't want to train it again and again on every run ,
I trained the svm and now loading it using its loading function
CvSVM svm;
svm.load( "trainsvm.xml" );
But how to do the same with vocabulary file which i use for clus... |
Direct Linear Transform (DLT for short) is a method of homography estimation, it solves the overdetermined linear system via SVD $$Ah=b$$to find a solution $h$ under constraint $\|h\|=1$. Actually it finds the least square solution which minimize $\|Ah - b\|$.
I understand the basic idea of this algorithm, but it is re... |
The question is:
"Weight and height measurements of a child taken every month"
Classify this signal as:
one/multidimensional
single/multichannel
I'm not understanding how this can be described as a signal and hence how can it be classified.
|
I was wondering whether we can recover a signal $x(t)$ from its power spectral density using signal processing techniques.
|
I am facing a trouble implementing the fft and appreciate your advice. My data structured as follow:
The data are retrieved from the share market for one day change in the share price. The time range from (11:00:00 to 15:00:00) this means 16200 Sec. In matlab its represented as serial so it will be from 0.4584 to 0.64... |
I am Fourier analyzing at a bunch of signals of equal length and find that $$\frac{d\ln({E[P]})}{d\ln({PT'})}$$ is constant across the dataset, where $T$ is a vector of the periods (inverse of frequencies) of each signal, and $P$ is the vector of the power corresponding to each of those periods. $E[P]$ is the mean pow... |
First, I apologize for the vague title, to be honest, I'm not even sure how to describe my question in a title. If someone would like to edit it so its more reflective of what I'm asking, then please do.
I have simulink model running in real time gathering data and doing stuff in real time (1ms).
One of the things its ... |
I have a before/after image pair with pixel transformations only (for example, grayscale, or a hue shift. No blur, border, sharpen, etc.). I'm trying to find a function such that for every pixel (r, g, b) in the original image, f(r, g, b) = (r', g', b'), (r', g', b') being the pixel in the "after" image.
I was thinking... |
Let $g[n] = f[-n]$(n is the time index). Then what will be expression for $g[n+k]$.
Is $g[n+k] = f[-n + k]$ or $g[n+k] = f[-n - k]$. Which one is true and Why?
|
Here i will first try to put my application agenda
Typically i am sampling at 8Khz rate and filtering speech components using a BPF(digital)
now i have got the 0-4Khz components in which essentially <250Hz are not speech so i will eliminate them also
now i am trying to find out entropy of frame and trying to mark ... |
I've been doing FFT's (on sound) for a while and usually use the highest intensity or the ones above average to get the spikes in an FFT result.
What would be a good method to get all the 'spikes' in an FFT. I can't use averages or peaks because the frequency response drops off. Visually its quite obvious there is a sp... |
I used the measured position to estimate the velocity using a Kalman filter.
But in simulations, the Kalman gain changes quickly and then remains constant when position and velocity continue to change. For example, position and velocity change at times 0 to 0.5(s) and 3 to 4(s), respectively. But the Kalman gain only... |
I am trying to understand the motivation for the above question for signal processing and image processing? Can anyone answer this question, or provide me with resources for this?
|
I am trying to Frequency modulate a sine signal using Matlab. I have written the following code for the same:
clc;
Fc = 5000; %Carrier Signal
fs = 20000; %Sampling Rate
t = 0:0.00001:0.002;
freqdev = 50; %Frequency Deviation
x = sin(2*pi*1000*t);
subplot(3,1,1);
plot(t,x);
y = fmmod(x,Fc,fs,freqdev,0);
subplot(3,1,2);
... |
I have a question concerning methods to find different aspects of "similarity" of signals. So, first, I'd like to characterize the signals to be dealt with:
We have a number of close to rectangular signals between values 0 and 1, starting and ending with some distinct flanks; furthermore, the signals are subject to som... |
I noticed that spline interpolation with a degree higher than 3 (everything beyond cubic splines) have a very high interpolation error, hence the prediction is mostly awful. I've come across various lecture notes, slides and Youtube videos that simply indicate that cubic splines (3rd degree) are optimal and that anythi... |
I am working in digital image processing field, recently I am studying about Image Noise, I just want to know that whether this noise removal is part of image enhancement or image restoration.
I have read some papers on it, at some points authors are saying that image enhancement from a given noisy image, and on the o... |
In the following example I have used various examples found on the web to understand fft and detecting signals with matlab.
fs = 1000;
t = 0:1/fs:1.5-1/fs;
x = 3*cos(2*pi*20*t + 0.2) + 1*cos(2*pi*30*t + 0.3) + 2*cos(2*pi*40*t + 2.4);
N = length(x);
% define the frequency components of the series
freq = 0:N-1;
freq... |
Could someone explain to me why the computation speed for the fast Fourier transform increases by padding the series with zeros to the point that its length is close to a power of 2? This is common in the matlab environment, for example:
http://www.mathworks.co.uk/help/matlab/ref/fft.html
|
Usually the filters for detecting edges have the sum of all the values in the filter matrix equal to 0.
For example: [-1 -2 -1, 0 0 0, 1 2 1].
Why is it so ?
|
I'm not sure how to approach this problem. I will describe what I am trying to do, and what some of my matlab outputs look like. I'm not trained in signal processing or anything, so please be patient reading.
I have a fourier transform of some signal, and its frequency domain covariance matrix. For example, say my s... |
I'm working on a device that produces a signal with sampling rate of 100 kHz. Every 40 ms, I receive 16000 points, from which I have to extract a frequency close to 3.55 kHz. I have developed a method that extracts the frequency of the signal very close to the Cramér-Rao lower bound, but I need to extract the amplitude... |
i'm trying to plot 1)the magnitude and 2)the phase response of an IIR eq with only its coefficients.
i've found somewhere that one can find the magnitude response starting with the transfer function :
ex: y(n) = a1*y(n-1) + b0*x(n) + b1*x(n-1) + b2*x(n-2)
... has the transfer function
Y(Z) b0 + b1 Z^-1 + b2 ... |
Consider a set $U$. My signal is a piece-wise constant "function"
$Sig: t \mapsto s$, i.e. the signal at time $t$ equals to some subset
$s \subset U$. One can see $Sig(t)$ as a stochastic process.
For a given sequence of time points $\{t_i\}$, we have sequence of
partial observations $\{ Ob(t_i) \}$. Each observation $... |
What I'll get if I take FFT of a frequency domain data.
I have two questions related to this.
For ex, x is a time domain data where X = fft(x), now fft(X)= what i'll get ?
While reading SCFDMA, data is first DFT'd and a subcarrier mapping happens then we are taking an IFFT which will transmitted using a radio. Accord... |
Price/volume data from financial markets are like signals. Can these information be applied to signal processing to generate profits successfully?
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I am developing C++ code to do some modeling that will take as input a parametric model of the complex frequency response (amplitude, phase) and produce as output an impulse response, e.g. using the iFFT. I would like to compare this to a "windowed" impulse response that I have measured for a loudspeaker in a room. The... |
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