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I am working on a video motion feature implementation. I am implementing a work in a research paper. What I want to implement is this:
The frames contained in a video clip can be thought of as a volume obtained by considering all the frames in time. This volume can be decomposed into a set of two 2-D temporal slices... |
i have two audio signals .wav and i want to convolve them, the trouble is that the frequency do not fit, they aren´t just the same frecuency-lenght in the wavread arguments. i am supposed to use the lesser frequency to resample the other one using interp1 and splines. I really appreciate your help.
Thanks
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I used MATLAB to produce a simple sine wave over time:
frequency=7
sampling_frequency=1000;
total_time=1-1/sampling_frequency;
step=1/1000;
theta=0;
time=0:step:total_time;
y=Amplitude*sin(2*pi*frequency*time+theta)
I then used the following code to fast fourier transform:
Y=fft(y);
ploty=Y/length(y);
stem(abs(ploty)... |
It is perhaps important to start with the fact that I am a complete beginner in DSP.
I have got a number of audio recordings (of sampling rate 22 kHz) - of bird songs - which I have been trying to analyse using FFT (in Matlab/Octave). Particularly, I am trying to show using Machine Learning classification algorithms th... |
I have an image convoluted by a rectangular PSF of a blur kernel. The fourier transform of the PSF has pattern of a sinc function: $\frac{\tau sin(\pi \tau f)}{\pi f}$. The image size is 256x256.
Would my following assumptions about the PSF on its period in frequency domain and width of its pulse in the PSF be correct:... |
I am not sure if this question will be a little off-topic on this forum, that I will give it a try anyway, since it implies signal process arguments.
By using MATLAB, I'm on my way to generate a turbulent wind field. This wind field is generally a Gaussian process with all the time series having 0 mean value and unitar... |
How exactly we should describe the process of Entropy Coding in wavelets ? can't it be simply called Coding ?
Here is the block Diagram of Image compression using wavelet,
I am confused about the "Entropy Coding" part. Can i simply use Huffman coding to code bits ?
|
Suppose a random signal $x(t)=\sum\limits_{n=-\infty}^\infty Z_n \delta(t-n\tau)$, where $ z_n = Z$ and $Z$ is a random variable with equal probability to be $+-1$, is passing through a low pass filter $h(t) = sinc(\frac t \tau)$ and the output is $y(t)$. and I want to prove the output signal is cyclostationary proces... |
I am looking for a method to calculate the instantaneous pitch variation within a speech signal for time-warping application. Sample by sample frequency variation knowledge within a frame will give more accurate output in time-warping.
Till now, I read about a method based on Teager's energy operator. Please suggest me... |
I need help to develop an object recognition system. It needs to identify an object in an image by comparing it with an image in an existing database. For example my database may consist of images of cars, buses, cups, etc. If i give a certain image as an input i want the code to check and tell me whether a car(as in t... |
I know this is probably a simple question, but I haven't been able to find a satisfactory answer anywhere...
Say you have a time series signal of finite length N. Call it $y[n]$. It looks like a sine-gaussian perhaps but with some random effects. The mean of zero, and no trend is present.
Now, say you receive this s... |
When we say we set a threshold to the edge finding, what values are we exactly thresholding on? I know a threshold value to limit only certain edges to show up, but what value of the edges is compared with the threshold?
In both the sobel and canny edge detection, they are comparing with a threshold value but I am neve... |
I know this question may not be very relevant to programming, but if I don't understand the theory behind image processing I'll never be able to implement something in practice.
If I got it right Gaussian filters are convolved with an image for noise reduction since they compute a weighed average of a pixel's neighborh... |
For audio classification we are using cepstral coefficients, but how should I decide the number of coefficients?
|
I'm trying to gain a better understanding of windowing functions, but so far I can't tell why they're used. As an example, I'm familiar with the Hamming Window:
$ w(k) = 0.54-0.46cos( \frac{2{\pi}k}{N-1})$ if $0<k<N$
But I don't understand how this can be done without distorting the signal. Are these specially chosen f... |
I am sampling a 5mhz blocks at an IF of 40 MHZ. I am sampling at 11MHz per bandpass sampling theory, but when I run the FFT I get all the aliases.
So I am trying to figure out how to constrain the fft interpretation to the appropriate bands.
IE i may be sampling the band from 80-85MHz by mixing it down to 37.5-42.5 M... |
I am a newbie in speech processing and experimenting to get a feel. I have extracted some speech segments using a window function and I want to find distance between a pair of segments. First, I took DFT of a segment and downsampled from the resulting frequencies (like 16 out of 256) and used them as features.
Then, I... |
Can someone explain sharpen in a non-technical way. So, I don't mean that you need to multiply every pixel with 5 and subtract it with the pixel left, under, above and right.
|
To find the complex Fourier coefficients of this sequence, then The $k^{th}$
coeff. $c[k]$ is given by the DTFS formula
$$
c[k]=%
%TCIMACRO{\dsum \limits_{n=0}^{N-1}}%
%BeginExpansion
{\displaystyle\sum\limits_{n=0}^{N-1}}
%EndExpansion
x[n]\exp(-in\frac{2\pi}{N}k)
$$
(Lets forget about the $1/N$ scaling for now). My q... |
I am trying to better understand discrete-time PLLs. I drew a picture of a simple feedback system here:
I want to be able to describe and understand the system at v(t) and y(t). I found a fairly good reference with the following diagram:
I am still having a hard time and I think if I could see a signal go through thi... |
I want some data to fit the corresponding Gaussian distribution.
The data is meant to be Gaussian already, but for some filtering reasons, they will not perfectly match the prescribed and expected Gaussian distribution. I therefore aim to reduce the existing scatter between data and desired distribution.
For example, m... |
I have a little problem with the subject Theory of signal. We were told to make amplitude and phase spectrum of cosinus in MATLAB, but we don't have results nor materials from our teachers, so we have to study from many different materials and you can imagine what this is like. So I'm a little confused.
However, I did ... |
I am working on lane detection system and I would like some suggestions from you as to how to detect lanes from a filtered image which I got after performing certain operations. Firstly, I should mention a few things:
The lane detection would be in a real road environment unlike here.
As of now, I am running it in rea... |
This question is moved from Stack Overflow.
Sorry about this noob question, because I never work with matlab and signal processing before. Here is what I want to do: I have a fixed length of byte array X, now I want to encode it to a sound file, I also want this process to be reversible, which means the sound can be co... |
I am using Naudio open source library and I am trying to do some simple filtering. The problem is that I hear some "clicks", not too loud. The library offers me the possibility to use at least two buffers, so the computing time doesn't introduce a delay between them. Because in the most of the time I am dealing with a... |
I have a output signal $Y(t)= \sum_{n=-\infty}^{\infty} Z_n \delta (t-n\tau) h(t)$ and $Z_n$ is a random variable with equal probability to be $+-1$ and h(t) is $sinc(\frac t {\tau})$
and the Expected value of $Y(t)$ is
$E(Y(t)) = E[\sum_{n=-\infty}^{\infty} Z_n \delta (t-n\tau) h(t)]$
$=\sum_{n=-\infty}^{\infty}h(n\ta... |
I want to estimate a channel based on LTE 3GPP EVA with given power delay profile (set of average power and delay of channel taps).
tau = [0, 30e-9, 150e-9, 310e-9, 370e-9, 710e-9, 1090e-9, 1730e-9, 2510e-9]; % relative delay (s)
pdb = [0, -1.5, -1.4, -3.6, -0.6, -9.1, -7.0, -12.0, -16.9]; % avg. ... |
The transfer function of a Low pass filter is H(w). From this I want to develop a high Pass filter. I read in DSP by Proakis, Sec 4.5, that the high pass filter can be obtained by translating H(w) by PI radians. I did not get how it will form a High Pass Filter? I think tt should form a Bandpass
|
Does anyone know how does the "Songify" app algorithm work?
There are 2 inputs:
speech (or amateur singing), and
underlying music.
Somehow the speech is altered and mixed into the music. Which DSP algorithms are needed to achieve this effect?
Pitch-shifter?
Time-stretcher?
Fundamental (F0) frequency detection?
A... |
Consider this system:
y[n]=y[n-1] + x[n]
It can be verified easily that this system has a finite impulse response (by putting x[n]= delta[n] i.e impulse, and evaluating impulse response h[n]=y[n]). Yet for a unit step input U[n], this system is an unstable system. On wikipedia it is mentioned that FIR filters are... |
Before today I had no idea there is any transform an such; I was only aware with DCT and wavelet transforms but today one of my teacher who is doing research on image processing project had told me that contourlet transform is way better than wavelet transform.
It is better because wavelet transform extract frequencies... |
I'm trying to use and understand SciPy's deconvolve for a project I'm working on. I'm having some trouble understanding how to use it.
What I would like to do is to take two PMFs from discrete gaussian distributions and recover an unknown distribution using deconvolution. My understanding is that if I deconvolve the PM... |
I'm reading up on DSP implementations and Q number format for representing decimal and decimal fractional numbers. Why would you use the Q format and not a standard like IEEE 754? When would you use one instead of the other? They both encode numbers into a sequence of bits, right?
|
Here you can see that the transfer function applied to a cosine input will give you a sinusoid and a transient term:
$$
x(t) = \underbrace{(x(0) + x'(0))(2 e^{-t} - e^{-2t}) + \frac{2}{5} e^{-t} - \frac{1}{2}e^{-t}}_{{\rm goes\ to\ } 0 {\rm\ as\ } t \rightarrow \infty }\ \ \ + \frac{1}{10} \cos(t) + \frac{3}{10} \sin... |
I'm starting to brush up on the Kalman Filtering I learned a couple decades ago. From what I remember, if you have a measurement vector
$$
z=H x + v
$$
and the $n$ components of the measurement noise $v$ are uncorrelated with each other, then you can process the measurements sequentially in the KF equations. Is there... |
I have been posting all over SE looking for a good method of finding the maximum of a trigonometric polynomial. A trigonometric polynomial is the sum of a set of sinusoids,
$$a_0 + \sum_{k = 1}^{N} a_k \cos(k \theta + \phi_k)$$
and can also be written in the form
$$ \sum_{k=-N}^{N} c_k \, e^{i k \theta} $$
where $c_k$ ... |
I have been trying to implement the CLAHE algorithm and came across this page which states step by step procedure for the algorithm.
I understand the initial steps to perform HE of tiles in the image. But can't figure out how to perform the interpolation in the last step to remove the artificial boundaries between the ... |
Am reading Rick Lyons - Understanding DSP book about Sampling. I have few doubts related to this chapter.
what is the difference between Bandpass Signal and a Passband Signal ?
Can somebody give an example for a Bandpass system which produces Bandpass signal as output !
When it comes to Baseband Sampling ( Low pass Sa... |
I've got a real 2D image of propagating waves in both directions. How can I separate this image into one with the upward-propagating waves and one with the downward-propagating ones?
This example has one wave in each direction. The x-scale shows time, the y-scale shows position, and intensity shows the amplitude of the... |
I need to approximate a function f, but I cannot do so with frequencies that exceed 1kHz
What is the best approximation I can get? Is taking the Fourier transform then zeroing any term above 1kHz the best approximation? Or can I fiddle with the lower order terms and get a better fit?
This is a made up scenario, but I h... |
I know this is very basic question and i am new to this site. But can anybody help me with my question. How to convert a sound file to other data format say using some type of transform (FFT or DFT) with which i can manipulate the data of the sound. Lets say i get frequency values from the sound i should be able to man... |
I would like to know the basic differneces between Gaussian filter and Bilateral filter in digital image processing...
I just know that both are used to smooth out an image but I would really appreciate if somebody let me know what is the purpose and scenario of each of these filters.
Thanking in advance!
|
Can I use the word amplitude instead of magnitude when I describe FFT bins? I dont see any similar word in my language.
|
I have designed a low pass filter that smooths the output coming from an accelerometer attached to a vibrating machine. I designed this assuming cut-off frequency $f_{c1}$, and sampling frequency $f_s$. Now the issue is when I change the machine , I have to change the cut-off frequency and $f_s$ manually, else the outp... |
My task is simple; I want to simulate analog low-pass filtering of an input signal, using Python. Note that the input signal is an array of values, not an analytical function.
My first question is if it simply is possible? If so, where is the line drawn between making a digital (FIR/IIR) filter and simulating an analog... |
i have 15 set of data which is the s-parameter, consist of 2 column (1001x2) each set. The first column is frequency and the second column is magnitude in dB. I have transform the magnitude from (dB) into value and frequency into time. But i have no idea how to transform these 15 sets of data into an image by using Mat... |
I am currently working on a gesture recognition system (for an Android Application). I think that I have completed the Image processing stage, where I am able to extract the contour of the hand (I am wearing a glove to avoid background subtraction for now).
I am also getting the bounding ellipse/rectangle, the centroid... |
Does the length of a Fourier Transform affect the ability to discern a small event in the data?
This is a tricky question to ask, let me clarify. I don't care about when the event happened and I don't care about the resolution of the event (it doesn't have to be some precise measurement) but I want to SEE the event. I... |
I have a signal $X(t)=\sum_{n=-\infty}^{\infty} Z_n \delta(t-n\tau)$, $Z_n$ is a random variable with equal possibility of $\pm 1$ and I know the power spectrum of this signal is $\frac {1}{\tau}$ from the solution.
I have try to calculate the power spectrum by the auto-correlation and then calulated by its Fourier tra... |
I am attempting to implement observer based state feedback in C and can't figure it out.
Here is pseudocode of the algorithm:
//Initialize Stuff
loop:
//read process variable
y = read(ch1)
//compute control variable ... |
I'm trying to understand the maths behind this Fast fixed-point $\tt atan2$ calculation with self-normalization. In particular, equation $(2)$ for theta1 appears to provide a first-degree expansion of some series. Similarly, equation $(2a)$ under "QUICK NOTE" seems to expand this series.
What is the series that's bei... |
I have 2 arrays of 800000 input and output data samples. The system is a kind of oven that works from 0 to 10 volts. The sample time is 0.001s.
I have to identify the model of this system, but first of all, given that the data are clearly dirty, I would like to filter the noise.
How can I do it with the System Identif... |
I would like to emulate a signal with bandwidth (like BPSK) for some testing, but the two signal generators I have access to do not have any BPSK options installed. If I know the frequency (2462MHz) and the bandwidth (16MHz) I want, is there a way to mock this up easily with two signal generators? I feel like this is... |
I am trying to record a sound two times and compare the two recordings. I am playing exactly the same sound both times (for example, play a small .wav file twice and record it each time), so I hoped that my attempt would not rise to the complexity of Speech Recognition...
I have tried computing the DFT (using FFT) and ... |
I read a paper about directional filter bank (Improved structures of maximally decimated directional filter Banks for spatial image analysis) in image processing and I don't understand clearly about the process. It says they use '2-band structure' like this image:
Could someone explain it to me how this process works? ... |
I have an fft function in matlab/octave that works great on an imported mono wav/audio signal that is 1 second long, but when the signal imported is more than 1 second the frequency values become incorrect. To check this I created a 2 second signal using audacity the first half of the 2 second signal (1 second of it) ... |
According to JOS on stiff-string synthesis, stiff strings (like on a piano) introduce inharmonicity (i.e. the harmonics of the tone are not all in tune) due to different frequency components of the wave travelling at different speeds. This inharmonicity is called "dispersion".
According to another page on the same site... |
Let $x[n] = \cos(wn)\ \ $ be our signal. Let the other signal be $x_2[n] = \cos(2wn)\ \ $ . The spectrum of $x[n]$ and $x_2[n]$ are well defined and consist of impulses at the relevant frequencies with weight $\pi$. However, if we interpret $x_2[n]$ as the decimated version of $x[n]$, the spectrum of $x_2[n]$ must have... |
Let's start with a simple temporal law, that I want to analyze in the frequency domain :
$x(t)=\frac{1}{2}(A+Bcos(2\pi \nu_0 t+\phi_0))$, with :
$A=4$
$B=10$
$\nu_0=10^8$
$\phi_0=10^8$
I made a simple DFT (in Ruby) with $N=256$ samples at $f_s=6.10^8$ and got the following results, that are self explanatory, hopefu... |
This question is a little theoretical. Say I have 3 points that are bound within a certain 3D volume of some shape (i.e. box, cylinder, cone, etc). By using PCA, I find the most two dominant directions and reduce the data dimension to 2D.
All of this is done by the following:
Compute 3x3 covariance matrix of the 3D d... |
I was looking over some DSP and came across the following signal: $x[n] = 1/n$. So I wondered whether it had a z-transform but I soon realized that it fails to meet two conditions described here:
...(1) the signal starts at a finite time and (2) it is asymptotically
exponentially bounded...
So I modified it to meet... |
I have been struggling to find a well-cited approach to this. Essentially, I am experimenting with different background subtraction algorithms, but none seem to perform well when there are significant lighting changes (say, from a window in the scene getting significant glare at specific times).
The question is simple... |
I have a periodical function $f(x)$ that is sampled in time, at a rate of 1 millisecond.
Together with this, I also have a second function $t(x)$.
I want to find the correlations between periods of the first function and the second function $t(x)$.
How can I approach this issue?
|
I am trying to make sense of FFTs and binary data.
Say I have a series of random binary data, which is measured with a repetition rate of 400Hz (interval time of 0.0025s). I have a total of 12489 points, which corresponds to a total measurement time of about 31 seconds.
I would like to be able to learn more about what ... |
I have two audio signals, both with supposedly matching frequencies at each time window. What I would like to do is to verify that the frequencies are identical. However, one was created at a higher power than the other, so simply checking for a match at each sample won't work. I've already cross-correlated the sign... |
Is there any good documentation about training your own classifiers(with examples included) different from this one provided by MathWorks.
Thanks in Advance
|
For the differential equation
$$\frac{d^2y(t)}{dt^2}+\frac{3dy(t)}{dt} + 2y = x(t),$$
I was able to find the frequency response as $$H(j\omega)= \frac{1}{-\omega^2+3j\omega+2}.$$ However, I am not sure how to find the impulse response, which is the next part. Please help, and thank you in advance.
|
I am working on camera motion detection.How do I decouple translation and rotation? I have obtained rotation by using sensors. I understand we can do something as simple as X'=RX+T , where R-rotation, T-translation and X,X' are the points involved. These points are 3D points(either world or camera coordinate system). H... |
I've got an array of different values representing the y-values in a plot:
y-values = [0.07, 0.09, 0.04, 0.15, 0.14, 0.05, 0.02, 0.2, 0.18,... ...0.06, 0.2, 0.17, 0.19, 0.21, 0.3]
I want to split it in 3 intervals:
the first one should return the values to the first big amplitude (here: [0.07, 0.09, 0.04, 0.15])
the... |
I'm trying to do some simple filtering for an audio signal using a window-sinc low pass filter. Supposing that my buffer has M values and the filter kernel size is N, after convolving these two arrays I would obtain an output of m+n-1 values. Which part should I take into consideration? I'm asking this because, if I re... |
For audio classification we are using the cepstral coefficients and their derivatives. Which I calculate using the Matlab libary voicebox.
I am looking for some literature for how to choose the frame length and overlap of frames. Most literature that I know of are considering speech recognition, but for this applicatio... |
I am fairly new to both Matlab and FFT. I would like someone to explain to me the relationship between the frequency of the wave being processed, the number of point (usually called NFFT), the Nyquist Fs and frequency vector. Basically how do you find the peak frequencies? how does FFT represent the frequencies? Any an... |
I am trying to implement Wavelet transform in an embedded environment. The problem I face goes like this:
Two vectors, A and B. A is input samples. B is vector co-efficients.
I take convolution of A and B, call it C.
Now I go back to B and up-sample the co-efficients.
Multiple C with new up sampled signal of B.
Prob... |
I'm trying to implement some watermarking algorithm found in a paper*. This is a line of the paper:
Do the H-level DWT for all the renumbered segments.
Then in simulation section the author explains the wavelet used for experiments.
DWT transform adopted the common wavelet "Daubechies-1" and level H =
3.
I just d... |
I've a scatter plot with x- and y-axis.
Now I want to calculate a rectangle which displays where 80% of the points in this plot are.
I'm not very familiar with statistics, so I don't know the method I have to use. Read something about percentiles. Is that correct?
I know I have to calculate where 80% are on the x-axis ... |
I have a source that emits $M$ equiprobable messages, which are assigned signals $s_1, \dots,s_M,$ that are equidistant by $a$. That is, if we plot the $s_k$ signals in a horizontal axis they are dots separate by the same distance $a$. And so I would like to determine the optimum receiver and the corresponding error pr... |
The question is revised to make it more specific and constructive.
I am doing imtransform, yet the xdata and ydata are very much different from that of size(output).
[ output, xdata, ydata ] = imtransform( imgDouble{1}, tform );
Real data test:
K>> size(output)
ans =
854 1230
K>> xdata(2) - xdata(1... |
I'm looking for a kind of error correction code or solution that can correct my codeword in this case:
My message holds $k$ bits, and $2k$ bits codeword (rate is $1/2$) is produced by the generator matrix, e.g. $k=5$, so it yields $10$ bits codeword $\rm 1001100111$. If the codeword is erased for some reason and it bec... |
I am using IIR filter for smoothing
$$y[n] = ax[n]+(1-a)y[n-1]$$
My question is, if I add another IIR filter, will it be the second order of IIR filter? If not, what it can be called?
My second filter is
$$y_2[n] = ay[n] + (1-a)y_2[n-1] $$
|
I have computed the MFCCs for two audio files and applied the DTW to both sets of coeficients for both files. Now each such pair the two files is described by one single value (computed by the DTW). Let's call this value Diff.
If the audio files differ more, then Diff is greater, which is great. (so Diff is proportiona... |
I have created a Low Pass filter (I am trying to understand the topic of DSP, and it's hard.. But moving along with samples, books and a nice post on Stack Overflow).
I am saving the 'output' to a file, and I hear a constant clicking noise (very rapid, medium-pitch). The volume (intensity) of the 'clicking' gets louder... |
The choice of frame length for calculating the Mel Frequency Cepstral Coefficients of a signal, depend on the how much the signal change over a small time scale.
I guess that human speech change rapidity while some animal sound change more slowly, but what is the measure of small time scale of an audio signal?
|
I would like to create white noise in the frequency domain, and then transform it to the time domain using python. To understand the problem, I simply generated white noise in the time domain, and transformed it to freq domain:
import scipy.signal as sg
import numpy as np
import matplotlib.pyplot as plt
e = np.random... |
Consider filtering square n*n images by square,separable m*m filters. What are the general computation time for the following approaches 1) FFT 2) by 2-D convolution 3) by two 1-D convolutions. (Capital letters to indicate implementation-dependent constants)
p.s what does it mean for a 2-D filter to be separable?
... |
I have four .wav files and each contain bird chirps. Given an input .wav file, I need to find out which bird it is.
I have some basic understanding of signal processing. I have used spectrogram to analyze my audio files and now I don't know how to make the frequency spectrum comparison to find out the solution.
Can so... |
What algorithms or methods can be used for the tracking of audio pitch contours? (Such as in musical vibrato and glissando, or for tonal vowels in languages such as Chinese?). Will a PLL work even for pitched tones that have a weak or missing fundamental? Or is there a more suitable tracking method?
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I am training a classifier of my own, specifically for detecting body arms. According to the training process I have to provide positive samples and negative samples, wich will be used in the different stages of the cascade classifier. As Mathworks says here, there is a tradeoff between having a higher false positive r... |
I am trying to get the fundamental frequency of a signal that only has a single pitch. I coded out the autocorrelation function using FFT and already got the autocorrelation result. Unfortunately, I don't know how to get the fundamental frequency from the autocorrelation result. Can someone help me? My JAVA code for ge... |
I have been trying to implement image classification by extracting facial features. I tried starting with some already established methods like Multi Kernel Approach, using Independent Component Analysis etc. Is there a very basic algorithm for image classification with which I can proceed the implementation ?
The dat... |
I need to divide an audio signal into frames in which the signal can be assumed to be stationary. How can I measure whether an audio signal is stationary in a frame?
|
I already have an algorithm for making panoramic image mosaics by image registration and then blending images in the overlap regions.
The following image from Wikipedia illustrates several images under translational movement stitched together by finding optimal dividing lines in the overlap regions:
Now I would like t... |
Is there anyone with Image Processing background who can suggests which one is better for Image Processing algorithms among CUDA and FPGA. I'm specifically looking to write code for Wavelet Transform and Discrete Cosine Transform and eventually writing code for Quantization and Arithmatic coding, but i am confused whic... |
What's an extremely efficient way of detecting when a user has stopped talking into a microphone? When I use systems like Siri it can detect when I've stopped talking almost immediately, even when there's background noise. My initial guess was to get the average volume of the background noise first but it seems there i... |
I'm trying to segment a specific part in a gray-scale image and I want to achieve this by applying the OSTU algorithm, but after using this function:
im = imread('ready.jpg');
oim =otsu(im,10);
figure,imshow(oim, [1 10]);
I could show the output image, but I do not know how can I save the result to be able to use it l... |
Take:
$$
(u*v)(k) = \sum_{i=-\infty}^\infty u(i)v(k-i).
$$
The $k$ is there, it's because you want to define
$$
\ldots\ldots, (u*v)(-3), (u*v)(-2), (u*v)(-1), (u*v)(0), (u*v)(1), (u*v)(2), (u*v)(3), \ldots\ldots
>$$
etc. The number in the parentheses is $k$. Thus, for example, when $k=4$, we have
\begin{align}... |
I hope this ultra-noob question is not entirely out of place here.
I'm trying to make a super simple ringtone: 0.5s silence + 1s "pure tone" + 0.5s silence + 1s "pure tone" + 0.5s silence. By "pure tone" I mean something like a sinusoidal or square wave with a fixed frequency (let's say 350Hz for concreteness). This ... |
I'm having trouble implementing OFDM with a raised cosine (RC) window in Matlab. I know how to generate an OFDM signal and how to show its spectrum, I just don't know how to generate the window extensions. Each OFDM symbol is extended by $TW$ samples at both ends to smooth the transitions between successive symbols, th... |
What are the options for recovering the phase of a BPSK signal in software? The only resources I can find online contain circuit diagrams – there seems to be no one willing to explain this to people unfamiliar with analog circuits.
I'd like any explanation of how carrier recovery works in theory, and would also love ps... |
I am trying to understand the periodicity of the DFT. How can this property (both in time and frequency domain) be used and can be helpful while developing on a DSP?
It would be good to see some source code or pseudo code, in MATLAB preferably, where this property is exploited/demonstrated.
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Consider filtering square $n\times n$ images by square, separable $m\times m$ filters.
Give general equations for the computation time for the following approaches to this:
Filtering using FFT
2-D convolution
Two 1-D convolutions
Use capital letters to indicate implementation-dependent constants.
For particular ... |
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