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I'm trying to use the goertzel algorithm to detect a pair of frequencies in a signal.
The sample rate I'm using is 8000 hz, and the size of the incoming sample buffer is 60 samples.
I have a large array of 600 samples in size, so each time a new buffer comes in I move the data 60 samples to the right in the large buf... |
I am working on an accelerometer on a mobile device. When an event occurs, I record around 30 acceleration values. The accelerometer, being extremely noisy is very unreliable. What makes it worse is that my motion during the acquisition time is also of the same order as the noise. Is there a good method to estimate the... |
With the following code, I get the "envelope" of a sound signal.
However in the "envelope" there are small oscillations (high frequency) : it is not smooth enough...
I imagine that it's possible to apply a low-pass to the "envelope signal" in order to remove the high frequencies oscillations, but I also imagine it's ... |
Suppose $x[n]$ and $y[n]$ are two nonzero signals(i.e., $x[n] \neq 0$ for at least one value of n and similarly for $y[n]$).Can the convolution between $x[n]$ and $y[n]$ result in an identically zero signal? In other words, is it possible that $\displaystyle\sum_{k = -\infty}^{k = +\infty}x[k]y[n-k] = 0$ for all n.
|
I have just started learning some DSP, I keep seeing things like memory in regard to filters. I am unsure exactly what this means and why it is useful. I understand that increasing the number of taps increases the memory etc.. and a tap is essentially just a coefficient for changing the signal.
|
What are the advantages / disadvantages of bilateral filter and anisotropic diffusion for noise reduction?
EDIT: To be more specific - what are the practical differences in terms of quality, speed or flexibility? How should I choose whether to use AD or BF for noise reduction?
|
I am running Wavelet packet analysis for a 1-D signal. I want to estimate dominate frequencies in the original signal.
Imagine I have following signal:
fn=[0.4 0.6 1.0];E=[0.03 0.05 0.1];A=[1.0 2.0 3.0];
wn=2*pi.*fn;a=E.*wn;wd=wn.*sqrt(1-E.^2);
y=A(1)*exp(-a(1)*t).*sin(wd(1)*t)+A(2)*exp(-a(2)*t).*...
sin(wd(2)*t... |
What is the difference between the Fourier transform, short-time Fourier transform and wavelets?
|
I know that for a given system, the Fourier transform of its impulse response gives its frequency response.
I want to find where this property comes from, but haven't been able to find if it's a definition or if there's a mathematical proof available, for both continuous and discrete-time systems.
|
I'm using a C# library that has the FIR filter implemented and I'd like to know what's the meaning of the ${z^{-1}}$ operation (delay) mathematically.
The triangle is multiplication, the circle is a sum, but what does the delay means in terms of mathematical operations?
|
I'd like to implement a digital notch filter, which has linear phase, in C++ or C#.
Example : a notch filter that removes 400hz frequency on a .wav file (16 bit, 44.1 Khz, stereo).
I'm a bit lost with many examples, I don't achieve to implement them.
Maybe some of you has a working example?
Thank you in advance.
|
I have a set of data in excel like this:
column A is time in ns and column B is the electric field strength.
now I want to convert the time-domain signal from these data to a frequency-domain signal with DSP toolbox in matlab.
to calculate Fs (sampling frequency) in order to use in signal analyse i use this formula:
t... |
I have a signal in matlab like this:
I have designed a band-pass filter in 16-32 MHz range:
and applied the filter to the signal,
the result is like this :
My Question:
what is the easiest way to convert this time-domain signal into a frequency based signal?
can I export the signal into workspace and use the data v... |
Why I get a different response from the same system (e.g. three phase inverter with LC filter) in state space form and in transfer function (Laplace) form when using the same PI controller values ($K_p$ and $T_i$)?
First I start with differential equations for each phase $a$, $b$ and $c$ to get state space model
\begi... |
So I just revised my pitch calculation algorithm using a harmonic product spectrum algorithm. I was just curious about why this explanation of Harmonic Product Spectrum states that you need to implement a Hanning Window to the data set. What would be the effect of implementing other Window functions on a data set (and ... |
I am programming a voice activity detection algorithm and I found a paper that recommends the use of HHT. I have been trying to program it and understand how to calculate $m1(x)$, but the shifting process confuses me.
Could someone explain what exactly I need to do after I calculate the mean value between the two spli... |
The top answer to this question: Power spectral density vs Energy spectral density states:
However if random process is stationary, then it is for sure that it
has some finite power over some band of frequencies
Why is that? Also, is stationarity a necessary condition for a process to have finite power over a band ... |
On one hand, there is the statistical definition:
$R_x(\tau) = \frac{\operatorname{E}[(X_t - \mu)(X_{t+\tau} - \mu)]}{\sigma^2}$
and on the other hand, there is this other definition (e.g. see Dilip's comment here):
$R_x (\tau) = \int_{-\infty}^{\infty} x(t) x(t+\tau) {\rm d}t$
Why are these two definitions different? ... |
I am trying to write an algorithm that would automatically segment a piece of audio with bird calls recordings. My input data are 1 minute-long wave files and on the output I would like to get separate calls for further analysis. Problem is that signal-to-noise ratio is quite terrible due to environmental conditions an... |
The following picture is amplitude and phase of an image filtered using Gabor filter in the frequency domain. I'm trying to reproduce the results, I'm getting valid results for the amplitude I think, but I'm struggling with the the phase, I'm just getting a noisy image that does not resemble the picture. I'm referencin... |
I am programming a Voice Activity Detection algorithm. After researching several methods, I implemented a simple method that determines whether or not the person is speaking based off the RMS of the signal. However, I want to improve the algorithm to determine whether the sound coming into the microphone is actually a ... |
I apologize if this comes as a basic question, but I am struggling to understand why the PSD can only be estimated and not directly computed. For example, this thread discusses several such PSD estimation methods.
If I have a deterministic signal with a fixed number of samples, shouldn't I be able to directly determin... |
I understood that SNR is the ratio of signal power to the noise power. In terms of images, how the original image is affected by the added noise. In PSNR, we take the square of the peak value in the image (in case of an 8 bit image, the peak value is 255) and divide it by the mean square error. The SNR and PSNR are use... |
Sensor fusion algorithms can give a more precise 3D orientation (and possibly postion?) of a device by combining readings from an accelerometer, gyroscope, and magnetometer.
Can anyone explain, or provide links, explaining the details behind this? I want to understand the math and physics, such that if I have raw 9-DO... |
I am trying to understand how the Windowed Fourier transform works, so I used Matlab to create an artificial signal, and plotted the Windowed Fourier transform using a gaussian window.
$x(t) = \left\{\begin{matrix}
\sin(2\pi100t)& \text{for }0\leq{t}<\frac{1}{4}\\
\sin(2\pi50t)& \text{for }\frac{1}{4}\leq{t}<\frac{... |
I have a general task of discriminating images of shaded cylinders that significantly vary in 3d orientation, and almost insignificantly in size and shape.
I have treated it as a regression problem on a dataset of these oriented cylinders, but my problem is with feature computation for each sample.
Basically, what is a... |
I have a dozen source sound waves; each is about one second duration. I want to put each one through a digital filter (I'm truncating it to 128 taps).
Currently I'm doing this with a bog standard convolution. It is chewing up a lot of CPU.
I would like to switch to fast convolution.
Could someone outline the steps?
I'm... |
I'm struggling to implement a sampling scheme where arithmetic decoding is used to generate the samples.
This is based on excercise 6.3 from David MacKay's book, where you compare rejection sampling with arithmetic decoding sampling. For each symbol you want to generate with rejection sampling you have to generate an ... |
I'm trying to create a digital filter from a first order analog filter with transfer function
$$H(s)=\frac{1}{1+\tau s}$$
with time constant $\tau=.1\text{s}$, and sampling rate $f_s=1000\text{Hz}$.
Applying the bilinear transform in Matlab however appears to yield a filter with the a different 3dB point than expecte... |
I am going to be straight forward and say this is one of the problem sheets and I am stuck on one of the questions. I already know the answer as I have been given them but I am trying to work backwards to see how to derive from the answer. so the question is:
Find another cosine y(t), whose frequency is as close as ... |
Are the results of applying Kalman filter and recursive linear MMSE estimation process the same ?
|
While studying an OFDM system, I have encountered statements like coherence bandwidth is the reciprocal of delay spread and coherence time is the reciprocal of the Doppler spectrum.
Can somebody help me to understand this mathematically?
How is coherence bandwidth related to delay spread, and how is coherence time rela... |
I'm trying to use Savitzky-Golay Filter to smooth experimental data. Savitzky-Golay computes $c_{n}$ in the following equation
$$ g_{i} = \displaystyle\sum\limits_{n=-n_{L}}^{n_{R}} c_{n}f_{i}+n$$
According to Numerical Recipes book by William H. Press. I've bought the library. The next step to calculate $g_{0}$ which... |
I have a problem that is similar to the state space model in Kalman filter but the observation matrix $G_t$ of $$y_t=G_tx_t +w_t,$$ is random. The elements of $G_t$ are i.i.d. random variables with a given distribution. Is it possible to use the Kalman filter to this problem? If it has already been used please provide ... |
I'm just beginning to study Fourier transforms and I'm having trouble getting this concept down.
I know what a low/high pass filter does to a signal.
What is the relationship between Fourier magnitudes (dB) and frequency (Hz)?
How would Fourier magnitudes get affected when a low/high pass filter is passed on a signal... |
I am confused between symbol rate offset, symbol timing offset and sampling clock offset. Can anyone tell be the difference between these terms.
|
Professor asked me what filter is better -1/3 2/3 -1/3 or -1/4 1/2 -1/4, the answer was second. But what those coefficients means? It was during discussion of linear filter but i'm not sure was it about it.
|
What are filter banks and what's their application? What's the meaning of $H(z)$ in descriptions of filter banks?
|
I have implemented a QPSK modulator and demodulator using classic text-book examples: Square Root Raised Cosine Impulse shaping and Matched Filtering.
The problem is that I've been told that I could only used square waves for the modulation instead of sine waves on the emitter side (only +/-Vmod) and this really degrad... |
What happens if I ignore Nyquist? I have a 16.368MHz digital signal coming out of a GPS front-end chip. My microcontroller (which is reading that digital stream of data) operates at a maximum of 32MHz so sampling at ~16MHz seems unlikely. If I undersample my signal with say M = 3, then I have a 5.456MHz signal which... |
I'm trying to create a spectrogram of a complex signal 'y', but I'm running into a problem where the spectrogram that I create appears to have its spectral features "displaced" in frequency space, and I think that it has something to do with not having pre-processed the signal correctly.
I know the signal's:
sampling r... |
I went through the answer Kalman filter in practice and it seems we must know all the first and second order properties of random variables to apply the Kalman filter.
But when I only have a set of data $y_1,\dots,y_n$ and a given model $G$ and $F$ with assumed noise processes, could someone please explain to me how to... |
In most examples and FFT code that I've seen, the output (frequency magnitudes) of the forward DFT operation is scaled by N -- i.e. instead of giving you the magnitude of each frequency bin, it gives you N times the magnitude.
Operationally, this is simply because the DFT is calculated by taking the inner product of th... |
I have a vector of zeros and ones with length ~2000. I need to detect and filter out the once, that appear periodically. There may be none to 3 such periodic signal's components. Typical periods are in range 10-90, but due to measuring noise, some spikes may be shifted by 1-3 steps in any direction.
I assume there is... |
I'm trying to simulate the damping of the hand on a guitarstring on an already recorded/sampled open guitarstring sound. I've been trying to use low pass filter and had a moving frequency range but that didn't make it sound like a palm muted string, just the loss of higher frequencies.
Could someone help me find good m... |
I want to implement a Time-Freqency analysis by wavelet packets. I don't know what level and sampling rate should I use. I've tried different value for level(3,4,5,6,). but I couldn't get an accurate frequency identification. here is my little scrip and results, hope one can say whats the problem, Thanks in advance.
... |
Is there a method in order to compute some sort of edge detection / convex hull of a signal x[n] (in orange here). Here of course the orange signal is not convex nor concave, but only piecewise...
(I would like to know if this could be an alternative way of computing the envelope of a signal, alternative to the traditi... |
I’m trying to figure out how to go about differentiating between a single-voice speech and a multiple-voice conversation in which voices overlap. In other words – the moment of overlap is what I’m interested in. I want to be able to listen to a vocal conversation between two people, and detect when voices collide/overl... |
I have found several sources describing how to use Doppler velocimetry/anemometry to measure the Doppler shift with analog signals, but i cannot find much about using this method for digital signals such as Bluetooth. I was wondering if it were possible to use Doppler velocimetry/anemometry with digital signals? If so... |
I am dealing with a problem similar to principal component analysis. Aka, I have a matrix and i want to recover the 'most efficient basis' to exaplin the matrix variability. With a square matrix these are the eigenvectors, weighted by the eigenvalues.
Originally, I was dealing with square matrices, and I used eigendec... |
I'm an absolute beginner.
I'm doing a project that involves estimating the time delays between the sound signal arriving at different microphones in an array (the project is a bit complicated but I'll find my way through). I'm planning to use a panda board and connect the microphone array to it. But before going about ... |
I'm trying to calculate symbol error probability of variant of pulse position modulation considering only AWGN channel. The decoder is maximum likelihood so the implementation of the detector is by using vector correlators.
$$
\arg \max_{x} <y,x>.
$$
Attached is the waveforms vector representation of the transceiver & ... |
I have data that is sampled at 12 kHz. The downsample function in software such as MATLAB only allow you to downsample by an integer factor, i.e. from $12\to{6,4,3,2,1,\ldots}$ kHz. How would I downsample by a non-integer factor, say, from $12\to{10,9,8,7,5,...}$ kHz (pseudocode/description is fine)? You can assume tha... |
I found a very interesting piece of code for MDFT polyphase filter bank here.
Unfortunately, there doesn't seem to be a paper describing the theory. Does anyone know some reference for the code?
I'm especially interested in these 3 topics:
What are the data in the channels exactly? Are they supposed to be real, imagin... |
Hey guys so I have an input sequence that are real values that represent intensity of a column of an image. I also have the transfer function in the Z domain of a 2nd order high pass filter. I have the following transfer function using the bi-linear transform estimate.
Say my input signal is defined as x[n] = 27, 588,... |
Question
Considering that $1\over{2\pi}$ is the frequency of $\cos x$ and also of $\cos x - 2\cos 2x$, what is the frequency of $\cos x - \sqrt 2\cos\sqrt 2 x$?
Thoughts
Perhaps "frequency" isn't the interesting quantity. I think the period is infinite. That would make the frequency zero. Maybe what I want to know is b... |
What are the formulas for signal and noise power gain of digital filters (FIR and IIR)?
For a FIR, I've seen in Harris' windowing paper that the DC gain is the sum of the filter weights.
$$
G = \sum_{i=0}^{N-1} w_i
$$
For a FIR, I've seen that the noise gain is the square root of the sum of the square of the weights.
... |
How does one choose the proper windowing function to apply in an FFT? What are the benefits of larger sizes and smaller ones? Which would be the best for pitch estimation? Also which window function would be best for pitch estimation?
|
Does anybody know if natural waterfalls generate pink noise or brownian noise? I have tried to search about this but the matches are non-scientific and contradictory.
For example
Livescience.com mention Brownian noise to resemble big waterfall.
Soundonsound.com magazine says Pink noise to occur in nature, e.g. waterfa... |
I've been reading about Super Resolution Image reconstruction (Reconstruction of high resolution image from multiple low resolution aliased images contain sub pixel shifts), and i want to know why SR is possible? this document explains why SR is possible, but i didn't understand completely!
please can you explain more ... |
I want to extrapolate a signal X of length 11, using the weiner filter coefficients W of length 7. The procedure I am using is as follows:
Compute the autocorrelation matrix upto lag 8 .
Using Levinson recursion, invert the autocorrelation matrix A.
Multiply the matrix of cross-correlations with the inv(A) to get W
Ob... |
So I have discrete pixel values in the image and each row or column would make a signal. I would like to filter some of the rows or columns to remove high frequency components from them. Should I use the Z transform (difference equations) or work with the FFT of the data to perform the filtering?
|
Wigner transform and continuous wavelet transform are both some kind of time-frequency representation of a signal.
What are the similarities and differences between them? Could you give some comparison between them? Let's restrict ourselves in 1D signal at the moment.
|
if we pad any signal with enough zeros we can get the same result as linear
convolution whilst using the fft function we compute circular convolution.
Why?
|
I'm currently trying to develop an algorithm to control the camera characteristic of a camera (the camera characteristic can be controlled to adopt it to the different lightning situation which can occur)
In this case I need a good measurement of the exposure/lightning on the current image taken (i.e., if it's under/ov... |
I have been working on a Harmonic Product Spectrum Algorithm. Now, all the literature I've read about the subject tells me to downsample an N amount of times. How does one determine what this N value should be? Here is my implementation of Harmonic Product Spectrum so far. Feel free to tell me if I have gone wrong some... |
I build application witch will be transform piano sound to musical notes. I found some pitch detection algorithms with implementations (mainly based on autocorrelation or fft), but chord recognition, of course is not working.
What is best method for instrument as piano(both hands + chords)?
Is there any implementati... |
When I was studying spatial frequency and entropy, it occured to me that "is there a relationship between entropy and frequency?".
As you can see, if we consider only horizontal line, the frequency of (a) is lower than (b).
Intuitively, I think that magnitude of frequency is proportional to entropy. But Google said not... |
I would like to do research on automatic music composition. At the lowest level , I would like to start from discrete frequency levels as states. Here I would like to know the mathematical tools I can use for this? I have heard of markov chains and chaos theory etc used in this purpose. But one professor recently told ... |
I want to remove the background from image. Let's say I have this background:
.
Then I put pen drive and tippex on it.
.
Is it possible to remove the background image from second image? If so, then how can I do it?
I'm using the AForge.NET Framework.
UPDATE
I want to remove background like in this research paper. Is ... |
I have a signal that I want to make it sound like a whisper. This is probably better than using a voice conversion system, but I was just want to do some experimenting before.
I was thinking maybe try to reduce the frequency of the signal? Maybe downsampling? Does anyone have a idea how to get closer to a whisper/unvoi... |
In continuous time it was possible;
$$
u(t){\longrightarrow} \boxed{\quad\textrm{system}\quad} {\longrightarrow} y(t)\implies
\delta(t)=\frac{du(t)}{dt}{\longrightarrow}\boxed{\quad\textrm{system}\quad}{\longrightarrow} \frac{dy(t)}{dt}=h(t)
$$
Does the same apply for discrete time system i.e.
$$
\delta[t]=\frac{du[t]}... |
Let $\Omega > 0$ and denote by $\mathcal{B}_\Omega$ the subspace of $L^2(\Bbb R)$ consisting of signals that are bandlimited to $(-\Omega, \Omega)$. Denote $\mathcal{L}_{\Omega} : L^2(\Bbb R) \rightarrow L^2(\Bbb R)$ the ideal lowpass filter that projects signals $x \in L^2(\Bbb R)$ into $\mathcal{B}_\Omega$: i.e., $\... |
I'm trying to port some MATLAB code to Python and am running into some strange behaviour. I am implementing a 5th order Butterworth bandpass filter. The sampling rate is 30 Hz.
Running MATLAB R2012b, Spyder 2.2.0 with Python 2.7, SciPy 0.12.0 on Windows 7 x64.
In MATLAB:
[b,a] = butter(5, [0.75*2/30, 5.0*2/30], 'bandpa... |
I tried to apply a Gabor filter with a specific scale (according to my values of lambda and sigma, so it is (7x7) and for 4 orientations (0, $\frac{\pi}{4}$, $\frac{\pi}{2}$ and $\frac{3\pi}{4}$) to a input gray scale image.
In my code, three steps are achieved:
Create a Gabor Filter
Read an RGB image, then convert i... |
Suppose you have N samples of a sinusoid of frequency $f_0$.
$$
x[n] = A\sin[2\pi (f_0/f_s)n ]
$$
Suppose you take an M-point FFT (where M may be greater than N) of your N samples of x. What will the amplitude of the peak of the spectrum be?
Reference: http://www.designnews.com/author.asp?section_id=1419&doc_id=236273&... |
I've recently come across this YouTube video that shows a PCM stream being played back via the 1-bit internal "PC Speaker", and after hours of googling I've come to think this is done via Sigma-Delta modulation.
So far, what I gather from the process is that it takes the original digital signal, oversamples and approxi... |
I want to create a microphone array that can be connected to my laptop. Previous discussions in stack exchange and other forums suggested me to get a USB audio interface. I would want to do a lot of real time signal processing on the sound arriving at the microphones (To be precise, I want to calculate the time delay o... |
Suppose,I have an input of length of lenght 100. data=[x0...x99];
I take a window of lenght 11 from x0 to x10.
windowed data=[x0 x1 x2 x3 x4 x5 x6 x7 x8 x9 x10],
now, I compute 7 wiener coefficients which are
[w0 w1 w2 w3 w4 w5 w6];
next I will move by window one sample forward,
new windowed data= [x1 x2 x3 x4 x5 x6... |
We can capture frequency response, phase response and reverb in recording of impulse response. Then we can use it to simulate audio system and/or space by convolving it with dry audio signal.
Is there similar technique for measuring and simulating non-linear distortion?
I thought of measuring how the amplitude of outpu... |
I am new to signal processing, and was reading about filters. I came upon the Sobel filter which has the following forms,
$\textbf{G}_x = \begin{bmatrix}+1 & 0 & -1\\+2 & 0 & -2\\+1 & 0 & -1
\end{bmatrix} * \textbf{A}$
and
$\textbf{G}_y = \begin{bmatrix}+1 & +2 & +1\\0 & 0 & 0\\-1 & -2 & -1
\end{bmatrix} * \textbf{A}$
... |
I have a program which takes in data from an oscilloscope, and due to reflections in the medium, we will get random signals at random intervals. The two grey windows at the right bottom show the original data (below red line), and after FFT forward and backward (above red line).
The data set is originally 10,000 point... |
I don't quite understand why the textbooks say it is impossible to implement an ideal low pass filter.
If I was to take the FFT of a discrete signal x[n], with Matlab's fft function I'd be returned with a complex sequence X[n] which represents the magnitude of each frequency from 0 to n/2 and is repeated. If I was to t... |
Suppose you have N samples of a sine wave of frequency $f_0$ sampled at $f_s$.
$$
x[n] = A\sin[2\pi(f_0 /f_s)n]
$$
How do you get a delta function in the frequency domain (i.e. all of the energy is in the center one bin of the FFT)?
Does the number of samples (N) need to be a prime number of cycles of $f_0$? If so, wh... |
I have designed a simple 3-state $[x\ \dot{x}\ \ddot{x}]$ Kalman filter which is updated by measurements for $x$ and $\dot{x}$. The filter is tracking a peak on a surface in the $x/\dot{x}$ plane. At each iteration, I detect a number of peaks on the surface, each with an $x/\dot{x}$ measurement pair.
I need to select t... |
In Harris Corner detection, is there any particular reason why this function is chosen:
$R=det(M)-k*trace(M)^2$
We want to a function to evaluate how fast E(u,v) fluctuate accross different direction, but can't we use other function such as:
$R=1-(\lambda_1+\lambda_2)^2$
|
MATLAB has a spectrogram function for the time-frequency analysis of a single signal. It also has a cpsd function for estimating the cross-frequency spectrum for two signals. However, cpsd averages across windows, collapsing the time axis into a single estimate.
Is there a function in this same family that does not av... |
fdesign.arbmagnphase is a MATLAB built-in function that will create a filter that best fits the frequency-response that it is given.
I'd like to implement a similar function in Python. What algorithms are used to synthesize filters based on an arbitrary frequency response?
|
Goal: For a particle moving uniaxially, to estimate position ($d$) and velocity ($v$) from noisy acceleration ($a$) and very noisy position (GPS) measurements using a Kalman filter.
Catch: The accelerometer has a DC offset, i.e. its zero is not zero but some number $\Omega$ such that the "true" acceleration $a^{true}$ ... |
Please explain me the concept of Orthogonality and Orthonormality. I understand that the mathematical definition of orthogonality between two vectors is that their dot product is zero. It means that there is no correlation between the two vectors. And in case of Orthonormality the vector length would be 1.
I am unable ... |
How can I find out frequency and peak value of a signal's fundamental?
Please do consider the following:
The peak value of some harmonics might be higher than that one of the fundamental
The frequency of the harmonic is not an integer (e.g. 3.45 Hz)
I have information about where to search for the frequency (3.45 +- ... |
I am asked to find a pair of sequences, each one of which contains three distinct values and where the convolution is an all-zero sequence.
I've came to the conclusion that one of the sequences must be infinite but just can't think of any examples...
|
If I have a time domain array of N samples, and swap the final half with initial half, then perform an FFT, what will be the effects on the frequency domain? Will the frequencies' magnitudes remain the same as the ones in the frequency domain of the original array?
What if instead of swapping halves, I just circle 1/10... |
I have a signal with length of $N$.
The sampling frequency is $f_s$.
Window size is $w$
The noverlap size is set to $m$.
And $nfft$ is selected for frequency resolutions.
I am running a STFT with matlab spectrogram function.
[S,F,T,P]=spectrogram(x,window,noverlap,nfft,fs)
what is the effect of nfft , $f_s$ and... |
What is computational complexity of continuous wavelet transform
cwt ?
What is computational complexity of wavelet packet transform wpt ?
What is computational complexity of Stockwell transform transform wpt ?
for the cwt : imagine I have a signal with length of $N$ and I
scale vector is 0:1:s
what will happen if I di... |
When we implement the 1D-short time fourier transform, the formula is as such:
$$
Sf[m,l]=\sum_{n=0}^{N-1}f[n]g[n-m]\exp\left(\frac{-i2\pi l n}{N}\right).
$$
the window $g$ used is normally symmetric, why is this so? (Hint: Use the phase of Short time fourier transform.
|
I have to implement a complete receiver in digital communications. The transmitter gives me an output vector which has real and imaginary values. It uses 16psk for modulation.
How would I demodulate that signal?
I was assuming the output with constant amplitude with different phases, but the magnitude of the whole ve... |
I have a half second audio data sampled at $44.1\textrm{ kHz}$. If I FFT it I get frequency components in $2\textrm{ Hz}$ increments. Now suppose I want the component of an arbitrary frequency, like $32.45\textrm{ Hz}$, how can I get it?
Looking at the DFT formula from Wikipedia:
$$
X_k = \sum_{n=0}^{N-1} x_n e^{-i 2 \... |
A band limited signal of $5$ kHz is observed at the output node whose sampling frequency is $600$ kHz. What is the minimum order of the filter?
Does it mean:
a signal that is from $0$ to $5$ kHz on the frequency axis when drawn
a signal from $-2.5$ kHz to $2.5$ kHz
a signal with bandwidth of $5$ kHz centered anywher... |
I am trying to code a simple algorithm in MATLAB that would be used to detect a single word. The goal would be to have a user record the word one time to act as a template, then make the user repeat the same word to try and detect it. Searching on this website has answered many of my questions, but I still have some i... |
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