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I have two images – mannequin with and without garment.
Please refer sample images below. Ignore the jewels, footwear on the mannequin, imagine the second mannequin has only dress.
I want to extract only the garment from the two images for further processing.
The complexity is that there is slight displacement in the... |
I am calculating spectrogram of a audio file of 36 second using the following code snippet:
window = np.hamming(window_width)
length = signal.size
count = 0
for i in range(0, length-window_width, window_overlap):
cur_frame = signal[i:i+window_width]
window_frame = cur_frame*window
spec = sp.fft(window_frame... |
I got curious based on this question here, but basically, is there ever a real-life signal that exists where its Fourier transform does not exist? If a signal is not finite energy, then its Fourier Transform does not exist, so what might be an example, (if any), of such a signal in real-life?
|
I started learning audio dsp and the book I am working through is loaded with formulas and most of the time expects the reader to know what the relation is between these functions and why they are being used in the first place.
I'd reading a section on spectral shape, which includes sub sections on rolloff, flux, centr... |
I need for an algorithm which do steps, displayed on image below.
Are solutions for doing this exist? (Preferably on C# or JavaScript).
|
I designed a 2nd type Chebyshev bandpass filter with passband between 1 and 6 Hz and 30 dB attenuation in the stopband. When I looked at its Bode plot in Mathematica, I noticed "stability margins" (black lines) on the plot. I'm not familiar with this concept.
Does the stability margin have any impact on the stability o... |
hmm any one can help please? I have a signal that contains some modes in it. I've run a CWT transform but I haven't got my desired results. so I implement a test signal that contains freqs [0.1,0.3,0.6] Hz to see whats the problem with my codes. but the result was accurate. once I change the freqs to [0.1,0.4,0.6] or [... |
My data is sampled at $f_s$, and I'm interested in analyzing certain frequencies $0<f_1<\ldots<f_N<\frac{f_s}{4}$. I then take $T$ samples of data, $\{x_1,\ldots\,x_T\}$ where $T\gg \frac{1}{f_s}$, FFT it after padding to the next power of 2 after $T$ and then pick the FFT bins closest to my $f_i$ of interest to procee... |
How can I compare the way that word is pronounced with a sample? For example, most of dictionaries have sample word pronunciation recorded. The goal is to provide feedback to the user so that he or she can improve pronunciation.
I wonder if there are API available for such task or existing programs that are able to do ... |
I'm trying to do an FFT on live audio to see its spectrum.
So I have a sampling rate of 44100 Hz and from this take out 1024 bins on my FFT
So to test it I play a continuous tone at 15000 Hz and another one at 16000 Hz and look for it on my spectrum. I try to use higher ones to cut out noise from around. I have made th... |
Please excuse me for asking such a basic question. I was having a look at how AM radio works. So there is a carrier frequency and a higher frequency inside that that carries the data.
Does this mean that there would be two frequencies, one high and one low?
What about for stuff like 3G/GSM when stuff is sent over EM wo... |
I require an image processor which compares an image with a set of images and finds the equivalent image out of them.
|
What is meant by "spectral whitening" in DSP?
What effect does spectral whitening have when used in image processing? (visually or otherwise...)
Where might spectral whitening be useful in audio processing or analysis? What would a spectrally whitened audio signal sound like?
|
I'm having some trouble understanding FFT plots and measures.
According to this, this will give me the magnitude plot of FFT, but I want the area under from 1 to 10 Hz for example.
Is this next formula I made right?
My doubt is, what's the relation between Hz and rad/s, f and w?
|
I'm looking for wavelets that are both:
causal (a wavelet coefficient $W(j,k)$, where $j$ is scale and $k$ is translation, does not use data from $t > k$ to be estimated).
complex (such as the Morlet wavelet, to analyze phase differences between two time-series).
Do any exist?
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I am really sorry if it is wrong place to ask this question but i couldn't not find any stackexchange website appropriate.
i have some reflectance values of a material which are measured in 10.5 and 12.5 micrometers interval. This spectra is extracted from a spectral library. Then by using Kirchhoff's Radiation Law i c... |
Glottal Closure Instants (GCIs) are referred to the instances of significant excitation of the vocal tract.
I have seen in some papers that I can measure instant of glottal closure in a voiced segment directly from speech waveforms.
Can someone provide me information how detect GCIs ?
How Glottal Closure Instants can b... |
How is it possible to go from time domain amplitude values in floating point to frequencies?
I know the DFT is used, but what exactly is happening for it to be able to extract frequencies from amplitude values?
Is amplitude between -1 and 1 mapped from pressure levels? How are different frequencies detected in a block ... |
I am using Cmor-Fb-FC wavelet transform in matlab environment. Fb is bandwidth and Fc is center frequency parameters. with trail and error procedure I select Fb-Fc as 5-1, and I've got rational output. But I want to know whats the rule of selecting proper value for Fb-Fc parameters? Imagine I am using 300 samples over ... |
With Python/Scipy, if I have a simply BPSK signal sequence [1, -1, -1, 1, ...], how can I generate a real-valued baseband waveforms with oversampling rate m, and common pulse shapes such as root raised cosine, Nyquist, etc.?
In Matlab I found this fdesign.pulseshaping function, is there an equivalent function in SciPy?... |
I have a DS-CDMA system and the chip width is fixed but every user have a different spreading code length. I want to have a simulation of signal to interference noise ratio (SINR) VS. bit error rate (BER). If I assumed that SINR is given like -25:0 dB. How can I calculate noise power (it is thermal noise) and the inte... |
I read that an "improper system" "has more zeros than poles; it is not causal, cannot be implemented, has a strictly proper inverse and has infinite high-frequency gain."
Does causality fail due to instant signal change? I try to make up a recurrence, where $x_n$ depends on both $x_{n-1}$ as well as $x_{n+1}$ but they ... |
I have data from a LIDAR unit that I would like to get the spectral density of. Unfortunately, the only thing I remember from my Fourier analysis class are the methods that I know will not work.
The data comes from a 1D LIDAR scan of a (mostly flat) surface, which returns radial distance at evenly spaced $d\theta$'s. I... |
I know that to downsample by a factor of 2.5x one can upsample a signal by a factor of 2 and downsample by a factor of 5. I am wondering what is the best and more efficient way of downsampling by a factor of 2.5x?
|
I have a few questions regarding the implementation of Bark scale filter banks for the calculation of bark frequency cepstral coefficients.
1) Which filter shape is most commonly used and why?
Triangular band pass similar to the mfcc's as seen here:- A Comparative Study of Filter Bank Spacing for
Speech Recognition
Tr... |
I know that the Complementary Filter has the functions of both LPF and HPF. But I think my understanding on the principal behind it is still unclear.
I am quite new on digital signal processing, and maybe some very fundamental explanations will help a lot.
Say I have a Complementary Filter as follows:
$$y =a\times y+(1... |
This is my first question at this site, please excuse me if I'm not asking the question properly.
I have an acoustic signal of a test and I want to remove the "thin" peaks, because they don't really affect the structure that I'm studying. For that I've thought of a Low pass filter in order to retain only frequencies b... |
I'm trying to decode a Hamming Block Code transmitted using a QPSK modulation.
For the moment I can sucessfully decode it using hard decision but in order to improve performance I'd like to use soft decision. I've read some online courses about the Viterbi decoder and I think I understand how it works.
The thing is tha... |
From a book, I learned that image interpolation (or upsampling) can be written as a convolution like this:
$$g(i,j) = \sum_{(x,y)}f(x,y)h(i-rx,j-ry)$$
while $r$ is the upsampling rate.
But I have a time understanding this formula, since AFAIK, the basic idea of image interpolation is to bring near-by pixels into consid... |
I am attempting a project and I have been given bat calls (in .wav format) which are not ultra-sound navigation calls. I'm looking to isolate the different species of bats and monitor their activity.
The problem I am having is that I do not have a prior training set to be able to feature matching on the samples to try ... |
I read several papers which proposed a method to identify modal parameters. such as natural frequency and damping ratio.I've tried to implement those methods by Matlab.but I couldn't get the result.It seems I missed something.anyway it must be some problem with my codes.Please check it and see what's the problem. here ... |
Remember that Christopher Walken skit on Saturday Night live where he told the band “it needs more cowbell” https://en.wikipedia.org/wiki/More_cowbell ? This site automated it http://www.morecowbell.dj/ using echonest http://developer.echonest.com/ . That’s all fine. Question is: “What is the actual algorithm behind... |
I have two electric signals from different sources and I think that are similar.
I have the FFT of the signals and I have their spectrums of frequency in separate arrays.
How should I proceed to find the percentual of similarity the two signals?
|
I'm having some problems understanding the FFT. Is the frequency resolution in the spectrum calculated as
$\frac{\textrm{sampling rate}}{\textrm{number of FFT points}}$ or $\frac{\textrm{sampling rate}}{\textrm{0.5 * number of FFT points}}$?
Asking this because the spectrum is symmetric for real-valued inputs. So, say... |
I have some biomedical signals. If I calculate their magnitude spectrums, I see power-line interference at 50 Hz. I was excepting a harmonic at 100 Hz, but it turns out there is none. However, there is a harmonic at 150 Hz. Is this common or should I worry that I have done something wrong?
|
As the title suggests , I want to know how noise cancellation is achieved by adding multiple images together . I know the theory behind it , but simple adding the pixel values of 100 noisy images of the same sample would simply produce a white image . Can anybody elaborate , how it is normally done.
|
What exactly is one looking for in audio data to find different instruments or energy sources?
So the FFT can extract frequencies, but how can e.g. percussion be detected alongside e.g. a bass line?
Would the frequency bins be checked into a frequency mapping of instruments? This would lead to collisions in possible in... |
As a beginner in signal processing, i'll try to explain my problem most thoroughly.
I'll firstly explain the idea of the transformation the real valued data has undergone.
This includes the ideas and reasons why it does work and the way to revert the procedure. Then i'll explain my difficulties in reverting it.
First o... |
Is there any approximate relation in FFT coefficients?
For example, consider time domain signal $x_n$. Its $N$ point (assume $N$ to be even) DFT is given as $X_0,X_1,...,X_{N-1}$
Can I say something about the nature of $abs(X_1)/abs(X_0)$? I am trying to see the variation in DFT coeffcients and for that I thught one ca... |
The shift theorem says:
Multiplying $x_n$ by a linear phase $e^{\frac{2\pi i}{N}n m}$ for some integer m corresponds to a circular shift of the output $X_k$: $X_k$ is replaced by $X_{k-m}$, where the subscript is interpreted modulo N (i.e., periodically).
Ok, that works fine:
plot a
N = 9
k = [0, 1, 2, 3, 4, 5, 6, ... |
Why exactly is X(0) the DC component of a signal?
How is it equal to N times x(n)'s average value and why it is at X(0)?
|
I'm fairly new to digital signal processing. I've been trying with limited success to apply a time-varying bandpass filter to a signal (question at stackoverflow). Instead I thought I'd try a different approach. Say I have a function which approximates the data I'm trying to filter from my signal. Is there a technique ... |
I have a sinusoidal signal embedded in noise, and I would like to estimate it's amplitude and frequency, so I can subtract it out.
I realize that an FFT can do the job for me, and that's what I've been doing - I took an FFT of the input signal, and looked at the frequency bin with the highest amplitude, used that as an... |
I'm new to image processing, and I'm trying to get my feet wet. I have taken a picture of some jigsaw puzzle pieces and I want to isolate the pieces.
I am using the Python SimpleCV library to do this, and so far I've managed to get pretty decent results using findBlobs(), hueDistance(), and drawMinRect(). Here's what ... |
I am reading a book on filter design and the author mentions to evaluate a transfer function, let 2 * PI * f (w) vary from 0 to PI to get the first half of the response.
How exactly is this done in code and what is meant by vary in terms of actual values?
Should f not change as well to get the response of different fr... |
what would be the most efficient ("cheap" in terms of system resources etc.) way to compare 2 sound files?
Here is the scenario. Sound A comes as an input. There is sound B stored in memory. We need to see if sound A is identical to sound B. We know that both sounds can be copies of the same sound file, but there is a... |
I am new to the topic of compressed sensing. I read a few papers about it by R.Baranuik, Y.Eldar, Terence Tao etc. All these papers basically provide the mathematical details behind it, i.e., Sparsity, RIP, L1 norm minimisation etc. However, can anyone provide the MATLAB codes that achieve compressive sensing?
Thanks i... |
I am in need of an open source library for computing Fast wavelet transforms (FWT) and Inverse fast wavelet transforms (IFWT) - this is to be part of a bigger code I am currently writing.
The things I am looking for in the library:
1) Contains a good variety of wavelet families (Daub,Haar, Coif etc.)
2) Ability to run... |
I have a signal which consists of an onset and a release period.
The onset, sounding like a thud, excites frequencies across the entire frequency range (0-22 kHz). It lasts for about 0.1 s.
The release, sounding like a ring, excites 2-5 specific narrow frequency bands. Their amplitudes decay exponentially, albeit at s... |
Wireless networks operate on a high frequency, say some several gigahertz. Are the signals really sampled at that high frequencies? If I got it correctly there's some modulating/demodulating in signal transmit, does that effect the situation somehow?
|
I'm trying to recreate a gassuain pyramid using the following scales:
There seems to be two concepts used here:
1) When an image is halved, applying a gaussian kernel of σ will apply as 2σ.
2) When an image has a scale of σ, and a gaussian kernel of σ is applied, then the resulting image will have a scale of sqrt(2)*σ... |
I need to analyze some vibration data from accelerometers (g-Hz) but I don't quite understand the meaning of each component that the registered signal contains. Basically it is Magnitude and Phase. I'm able to get the Real/Imaginary parts from those two as well.
Today I plotted Real vs Imaginary parts (Nyquist plot/dia... |
I am making a software model of an existing digital synthesiser and I have made audio recordings of single notes with particular parameter settings. I made a program in C++ which sends a MIDI sysex message to change the parameter in question, then triggers a MIDI noteon and noteoff message. The sound is recorded to a w... |
I have two power plots. I am trying to normalize it for comparing them . So , do we normalize it after converting to db scale? or do we normalize the power as such?
|
Is there a formula to calculate the frequency of a phantom fundamental based on the harmonics, and would this formula be applied directly to the energy bins of an FFT?
|
Firstly I should state that my maths knowledge is limited.
I'm currently designing an acoustics modelling application which uses a rectilinear FDTD grid for modelling pressure variations - but I've become stuck upon trying to implement a fibrous layer boundary. From Kowalcyzk 2008, the fibrous boundary model is define... |
I have got a txt file which contains voltage and time data. When I did fourier transform using fft function in matlab I got complex numbers. I understand that the real part is the magnitude and the imiginary part is phase. How can I determine whether there is any aliasing by looking at that complex numbers?
Another qu... |
whats is the effect of pading zeroes to a sequence in FFT?
for eg: x[n]=[2 3 4 5] corresponds to X[K]=[14 -2+2i -2 -2-2i]
while x[n]=[2 3 4 5 0 0 0 0] corresponds to X[K]=[14 0.58-9.65i -2+2i -3.414+1.65i -2 3.414-1.65i -2-2i 0.58+9.65i]
so what is the relation between X[0] , X[1] and X[2] in the new sequence?
please h... |
I have a signal with 262144 samples and "sampled" with 200Msps.
I want to use the cwt (Continuous Wavelet Transform), for making a time-frequency scalogram.
Also, I worked in some scripts in python and Matlab, but it's very hard for the machine for make the process and empty the memory of the computer with 262144 sampl... |
I'm working on improving the modem communication speed that can be achieved over a 10km cable. The current modem achieves a speed of 9600 bps via hardware (phase locked loop) and I'm looking for ways to at least double this, and do it all in firmware. Here are some specs for the project.
Cable characteristics
Length: ... |
I know there are lots of papers proposing algorithms for the problem of reconstructing a signal from modulus of its Fourier Transform (so-called Phase Retrieval Problem).
Also, recently it is studied for the case of sparse signals.
I want to know whether there is any paper or survey working on "reconstruction of signal... |
This is probably off-topic since it isn't really a question, but I thought that this GIF of the fourier series of a square wave was too cool not to share.
|
I hope this isn't a dublicate . I checked but maybe didn't see everything.
Here goes :
I've been scratching my head over some EQ related questions. I read a lot of papers but my limited math and DSP knowledge prevents me from fully understanding them, so I thought I'd ask the experts.
I hghave a good library (http://... |
I'm reading an article about multi-scale representation of image, and it is said that, convolving the image with a Gaussian kernel at different $\sigma$, then different scale representation is created.
This really confuses me, because I think the image is just blurred by Gaussian with different $\sigma$, each blurred v... |
I'm new to signal processing and I'm learning on my own. Therefore I might ask questions not worthwhile your time.
At the moment I'm trying to figure out the concept of a lowpass filter.
Using matlab I created a complex signal x:
x = rand(1,1000)+rand(1,1000) .* 1i
And i created a filter Num with the fdatool, the spec... |
i have my output as vector of 60000 complex number representing 10 sequence in frequency domain , each of length 6000. i need to plot the phase and amplitude of intermodulation distortion from this output. can anyone suggest me how it can be done. i tried with magnitude and angle unwrap in matlab selecting particular b... |
I recently read that with compressive Sensing it possible to sample the signal at a rate lesser than that suggested by Nyquist Criterion. However, I am still not getting how this is possible. Can someone suggest an intuitive way of understanding this concept of sampling at lower rates? And if this is possible, do moder... |
I'm wondering if someone can clear the following up for me as I'm having a difficult time understanding the concept.
I'm looking to implement and learn about the processes and uses using short-time Fourier transforms. I have had experience using DFT's and wanted to expand my knowledge.
From research that I have read ... |
First of all I apologize for the long question. I am working with a 3-axis accelerometer and a 3-axis gyroscope and I need to calculate the correlation between the different signals (acceleration in x,y,z and the quaternion x,y,z,w). In the end, I need to get 21 correlations, one for each pair. I was thinking of using ... |
I am stuck on a research problem and do not know what kind of methods to use. I hope people in this forum can give me some good ideas, I've always had good brainstorming sessions here.
My problem is that I have a laser and a stack of thin films made up of three different materials. If I shoot the laser at these thin f... |
I want to do sample rate conversion by subsequently upsampling with factor I=5, and then downsampling with factor D=9.
I have designed a nyquist sample rate conversion filter h() of length M, with matlab's filterbuilder tool for interpolation factor I=5, and decimation factor D=9. Since D > I, I have chosen the normali... |
I've found good literature for offline proccesses, and I've gotten good results. However, when it comes to online applications, I've found nothing good. Applying any filter without a linear phase response will give me too much distortion once I get to displacement. However, any filter I've used with linear phase respon... |
I understand that sampling rate should be twice the signal bandwidth for successful reconstruction. But what I don't get is how this translates in the time domain.
For example, I have a sine wave pulse with frequency 100MHz and duration of say 50ns. This signal has a bandwidth of 20MHz (1/50ns = 20MHz). Theoretically, ... |
I'm studing about Kalman Filter and Particle Filter in multiple target tracking in computer vision (tracking pedestrians). Reading sientific papers I'm colliding with a lot of sentences like:
The assumptions of linearity and Gaussianity of the Kalman Filter
cannote manage complex scenario.
Now.. I'd like to find so... |
We call the conversion from a continuous time signal $f(t)$ to a discrete time signal $f_s[k]$ "sampling". Is there a name for the reverse operation, i.e. creating a continuous time signal from a discrete time signal?
|
I'm looking for your opinions.
I have obtained a Speech Signal and computed a STFT on this signal. I have split the signal into blocks of (100 x 256) and computed the STFT based on different Windowing functions. But, they all bring back similar results (outputted on the same block) I know this might make sense, but, I... |
I would like to improve my knowledge about Fourier Transform and related subjects.
What are the recommended resources in order to self study about it?
|
To start with, I'm pretty new to DSP, so forgive me failing to understand any basic concepts.
I'm working with an iOS app that depends on the input from the analog audio jack whilst monitoring the signal through output (headphones, etc). Unfortunately, iOS devices suffer from crosstalk given a high enough volume due to... |
I am trying to form an approximation of the wavelet transform from a partially sampled image.
Reconstruction in the 1D case is easy. We have $w = h x$, with $w$ as the wavelet coefficients, $h$ as the inverse of the Haar matrix and $x$ our signal. $w$ and $x$ are $n \times 1$ and $h$ is $n \times n$. Where we don't sa... |
I would like to recognize the boundaries of a hexagonal tiling in a photograph, like in image below:
It seems to me, that a standard approach at a square grid is to first detect corners (e.g. canny) and then extract the longest lines via a Hough transform or something similar.
This does not look as optimal solution ... |
I am trying to create a program that can count the number of nuclei in such an image:
What I've already done is the following, step by step:
Apply an Alternating Sequential Filter (closing and opening the image with gradually bigger structuring elements)
Apply a distance transform
Apply watershed segmentation using t... |
Matlab says
x = hilbert(xr) returns a complex helical sequence, sometimes called the analytic signal, from a real data sequence.
How did the name complex helical come about? How is it related to analytic signal?
|
I've seen some other questions on here about zero padding (like this one) but I'm still a little confused about my situation.
I'm attempting to zero pad my input time series data, so I get an interpolated frequency spectrum. I had initially tried this with NumPy's FFT package, and I checked my algorithm on generated d... |
I'm wondering if anyone knows of any open source solutions to note recognition. I'm planning on building a program for online sheet music, which recognizes your notes, and tries to tell you what you can improve, and how.
However, for piano for example, I don't even know if it's possible to have software that detects va... |
I am referring to this report on pitch estimation.
The author does envelope extraction using Hilbert transform before applying AMDF.
What is the advantage of doing that?
Is speech a narrowband or wideband signal? I read somewhere that hilbert envelope extraction works better for narrowband signals.
When I extracted e... |
I asked a question earlier but I didn't get any answer for it. So now I am simplifying it: what are Cross-Spectral Density (CSD) and Power-Spectral Sensity (PSD)? What is their application? How can I get them in MATLAB?
$$S_{kl}(\omega)=\lim_{T\to\infty}\frac{1}{T}E\{Y_k^*(\omega)Y_l(\omega)\} $$
$$S_{kk}(\omega)=\l... |
I'm learning feature-detectors from this lecture notes, and I don't quite understand the Normalized Laplacian of Gaussian filtered image.
Here is the original image:
this is the output presented in the lecture notes, filtered by Normalized Laplacian of Gaussian with $\sigma=2.502$:
and here is mine, using scipy.ndima... |
I believe that I read, for digital modulation with multiple levels, that the AGC makes use of the fact that the symbol sequence, before noise etc., can be modeled as a uniform distribution.
$$\text{Source $\rightarrow$source-coding $\rightarrow$FEC$\rightarrow$ interleaving $\rightarrow$ bit-scrambling $\rightarrow$ mo... |
I am using complex morlet wavelet to estimate frequencies and damping ratios in a 1-D signal. I have done it so far, and the results are accurate enough. Now my question is how can I modify or optimize my method to get better result. like mixing it with other method such as Neural networks or anything else. I know this... |
Say I am trying to measure a exponentially decaying signal ($S_y$), that constitutes a portion $\gamma$ (about 5% so $\gamma=\frac{1}{20}$) of total signal measured($S=S_y+S_b$ and $S_b=S_0 \beta e^{\frac{-t}{C_b}}$ and $S_y=S_0 \gamma e^{\frac{-t}{C_y}}$). Where $\gamma+\beta=1$ and $C_Y$ is about $\frac{1}{4}$ of $C_... |
I am trying to speed up some of my code and for physical reasons I need to do an FFT shift to get the FFT data in the standard DFT fourier representation. In Matlab this is performed in the frequency domain, with a data swap.
I had believed that I found a numerical trick, by performing the shift in the time domain, be... |
A bit of noob question, but can't find a definite answer anywhere:
What is the difference between Gray-scale resolution of an image and the Image contrast.
Are they different terminology for the same thing?
|
I have Digital Image processing (DIP) in my academics and I am looking for some reliable and daily updating sources for learning DIP Filters for Enhancements,color conversions, Filtering purposes through MATLAB , please suggest me some websites or tutorials which can help me. Thank you.
|
Background:
I am a newbie in DSP. I am implementing a simple Kalman Filter that estimates the heading direction of a robot. The robot is equipped with a compass and a gyroscope.
My Understanding:
I am thinking about representing my state as a 2D vector $(x, \dot{x})$, where $x$ is the current heading direction and $\d... |
This is primarily a question about how to name my filter. Suppose that I have a photon detector, and that I want to measure the rate at which photons arrive, with a simple gated counter (because this is the only hardware I have). The photon rate is just the counter value divided by the gate interval.
It seems to me t... |
I am using Q1.15 fixed-point values to represent a 12-bit signed integer value. I was wondering if it matters if I use the 11 MSB or LSB of the fractional part to represent my 11 magnitude bits? Only thing I can think of is using LSB might in some cases reduces the chance of overflow errors and maybe allow me to use l... |
As I understand aliasing is because you sample a high frequency with a low sample rate, and the sample points don't have enough information about the high frequency component. Here is an example of aliasing:
w = 5.0;(*signal frequency*)
dt = 0.66125;(*sampling rate*)
f[x_] := Sin[w x]
ls = Table[f[x], {x, dt, 200 dt, ... |
I'm trying to get the average frequency response of an entire piece of audio by splitting it into segments, doing an FFT on each one, and averaging the magnitudes together. Its working great until I get to the last segment, which nearly always is shorter than my FFT size, so I pad it with zeros until its the right size... |
suppose $x_n$ is an $N$-point sequence with periodic auto-correlation
$$
c_n=\sum_m x_m x_{(n+m) mod N}
$$
How to find $N$-point sequence $y_n$ with periodic auto-correlation $d_n = c_n - A$, where $A$ is a constant?
|
I'm developing an application which one of its capabilities is peak detection.
I expect to handle three types of the similar data (Acceleration vs Frequency) but measured in different laboratories and with different sampling and noise. The data is not equally spaced in frequency.
Here is the plot of each of them:
De... |
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