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I want to design/implement a simple feedforward/feedback all-pass filter, and I am having some troubles and questions with it.
From Shroeder design (based on J. Smith article here):
I came up with the following pseudocode (for first-order all-pass filter, meaning the delay was set to 1 sample, and b0 = aM):
float a... |
I'm wondering what basic characteristics of shadows are. I'm looking from a camera image/video processing point of view. I'm considering the case of an outside scene in direct sunlight.
These are the characteristics I could quickly come up with, but I guess there are more.
There are strong edges on the shadow borders
... |
I am working on an arabic OCR for printed scanned documents ... Some of the scanned documents may written with font size of height 8 which is quite small ... I want to resize the height to 60 pixels but some artifacts may occur due to nature of arabic characters .. some characters may overlap .I had used local threshol... |
I'm a complete beginner to object recognition.
I need to identify and count how many of a certain type of Lego, in a picture full of different pieces of Lego, using MATLAB.
The pieces of Lego I need to identify, are of a certain shape, colour and size.
I need suggestions for algorithms that can perform this task, and ... |
Does anybody know a simple way to recognize vowels from MFCCs? (not words, just phonemes: a,e,i,o,u).
I heard this is a vastly known problem in DSP with several solutions like SVM, GMM and cosine distance. I tried this last method using a good audio processing library and a representative training set, but seems that ... |
I'd like to simulate a certain $E_s/N_0$ in a baseband OFDM MATLAB modem to generate SER curves.
I randomize 50 QAM symbols with $N$ subcarriers each in frequency domain, add null frequencies (guard bands) to each symbol, perform IFFT and then add CP. Then I convolve the "time domain" symbols with a certain channel wit... |
I'm hoping someone - anyone can answer a question on the Siglib library call to perform an FFT on data.
There's no examples anywhere I can find that provide a real application using data from an ADC, so that's where my wheels are spinning.
The examples from Numerix have only canned data, generated by the function SDA_... |
how do you compute the channel gain from path loss index in wireless communication?
|
As far as I know, bandwidth is measured in herz or 'radian/second' in digital signal processing or in analog communications. But in computer networking it is measured in 'bits/second'. I don't get how these two are related. Can anybody please help?
|
I made this question https://dsp.stackexchange.com/questions/19731/newbieplotting-fft-and-impulse-response-coefficients-using-java
I have the IRC: Impulse Response Coefficients:
$h(k), k = 0, 1,.., N-1$
Low Pass: $h_D(k)=\sin(k\omega_c)/{k\pi} , k\neq0; h_D(0)=2f_c$
High Pass: $h_D(k)=-\sin(k\omega_c)/{k\pi}, k\neq0; h... |
As known, we can get a 'phase spectrum' from the DFT of a input signal. Assume, we do a DFT on a given equal interval sampled 50Hz AC signal, the signal is a complete whole cycle, but the start sample point may at any point. And I get the amplitude and phase from the DFT, but what's the reference point of the phase?
|
I would like to measure the time delay between two signals. By applying the cross correlation I get the time delay or number of samples by the location of the cross correlation peak, however I am interested to know what peak value itself identifies ?
I get a large value for the cross correlation peak.
Thank you!
|
I'm trying to implement the VAD algorithm in this paper:
http://ms12.voip.edu.tw/~paul/Papper/Steganography/iLBC/(VAD)Real-Time_VAD_Algorithm.pdf
I'm polling a microphone on an Android phone, and getting short[]'s which I want to perform operations on. The audio data is 16bit PCM data.
I've gotten the basics of the al... |
I'm doing vehicles tracking project. The video is a one-way, no vehicles occlusion.
After doing background subtraction, erosion and dilation, I get bounding box objects for each frame, and I can find $centroid(c_x,c_y)$ from that.
So, the next thing I want to do is blobs tracking. Here is what I'm thinking on how to do... |
I was reading through the Ambiguity function, and I was wondering if it's possible to calculate analytically (or numerically) the ambiguity function of a sum of sine waves. In case of a single sine wave the envelope is a rect(), hence the ambiguity function is a $sinc()$ with variables the delay time and the Doppler fr... |
Hi there I'm trying to measure time delay of one percussive event recorded by two contact mikes on a glass surface. I decided to check the cross correlation (via FFT) and detect the index of the max in the function. The method is really jittery as the medium affects the transfer function of the signal. I wonder if ther... |
I would like to implement the example code from SigLib's example "fsk.c".
The defines are below
#define SAMPLE_LENGTH ((SLArrayIndex_t)512)
#define NUMBER_OF_LOOPS 4L
#define SAMPLE_RATE 9600.0
#define BAUD_RATE 1200.0
/* Number of samples ... |
What would a straight-forward way to convert an audio payload (PCM 16bit) from 6 kHz to 8kHz? I understand that this is an interpolation problem.
I fear that a linear interpolation would introduce too much noise. If so, what would be a better interpolation? I was looking at Lanczos resampling but I've seen it used in ... |
I'm studying DSP, but I have weakness about it.
I was reading this: http://www.eetimes.com/document.asp?doc_id=1275556
But, I don't know how select the Low Pass Filter.
I am assuming that this depends on the sampling frequency, But I need explanation "as with clay".
I know what is a FIR Low Pass Filter, but I don't kno... |
Can someone find or make a visual illustration of how time domain aliasing cancellation works in lapped transforms such as the MDCT? I think a graphic would make it a lot easier to understand, and I can't find any. For instance, this image clearly shows how the overlap-add process combines windowed chunks of a signal... |
Dear Signal Processing readers,
I want to introduce 'noise' into parts of images.
Until now, I worked with rectangluar images for a similar purpose and did the following, using (inverse) Fourier transform (2D) with MATLAB:
fft = fft2(image); % run fourier transform on image
amplitude = abs(fft); % extract power sp... |
I'm trying to get rid of some periodic flicker noise through post-processing of the recorded images. The reason for these artifacts is that the electronic rolling shutter of the camera reads each line sequentially, and this mechanism results in brightness variations in the image, when the picture is taken under fluores... |
A linear phase filter will cause a delay in the output signal of the filter.
If I'm not mistaken, a filter with phase equal 0 would cause no delays in the output signal.
Can someone explain me why isn't it possibly to generate a 0 phase response filter? I mean, theoretically...
I'm studying for a test and I can't find ... |
Currently, I have two signals, the main components of both signals are 60Hz, but both also have weaker response at 180Hz + small amount of noise. As shown in the photo below, I want to find the phase difference between the 60Hz component of first signal, and the 180Hz component of the second signal.
As you can see the... |
I have a theoretical question about optimal similarity metrics for comparing data sets.
In reading this linked paper, pp. 488-489 [1], I read the following 2 interesting statements. On page 488:
In the ... scenario in which $A$ and $B$ differ only by Gaussian noise, then it can be shown that $\rm SSD$ [Sum of Squares ... |
Can someone help me:
I have a signal which consist of three frequencies; for example, $f_1=800\,\text{Hz}$, $f_2=1050\,\text{Hz}$, $f_3=1600\,\text{Hz}$. How can i compute the fundamental frequency?
Also, is there a method called "frequency histogram method" which is used to compute fundamental frequency? Can someone e... |
I am reading a book on Laplace transform, and in the section on the convergence of Laplace transform for various signals the following theorem is stated, without any proof :
If a signal's Fourier transform be zero in some frequencies, or have discontinuities, it will not have Laplace transform (like sinc function, or p... |
I tried the following code in MATLAB R2013b
clc;
clear all;
VideoFreader=vision.VideoFileReader('video7.mp4','AudioOutputPort',true,'AudioOutputDataType','int16');
i=1;
while ~isDone(VideoFreader)
[I,Audio]=step(VideoFreader);
a(i)=Audio;
i=i+1;
end
There are no errors, but warning as follows:
Warning: The... |
Apologies if the question title is confusing.
I've got some code below in MATLAB, which identifies squares/rectangles in an image.
img = imread('image.jpg');
imgbw = im2bw(image);
filter = bwareafilt(imagebw, [800 10000]);
fill = imfill(imgbw,'holes');
area = bwareaopen(fill,15);
final = bwlabel(area);
dets = regionpro... |
I try to understand what following algorithm from this gimp plugin does to denoise an image:
/* Wavelet denoise GIMP plugin
*
* wavelet.c
* Copyright 2008 by Marco Rossini
*
* Implements the wavelet denoise code of UFRaw by Udi Fuchs
* which itself bases on the code by Dave Coffin
*
* This program is free s... |
I am working on a project. I want to segment a logo from a car and the picture's size is 3648*2432. The logo is selected by the red rectangle.
I extracted this area and turn into gray level. The area's size is 249*173. Then, I use histogram equalize to enhance the edge. Finally, I use adaptive canny to find edge.
H... |
I'm trying to understand how FM demodulation works. I read a lot on the internet and as far as I understand, the main process is to extract the phase difference (that is also the output) between a quadrature signal and the input signal, and that use this difference to keep those two signals locked in phase.
However the... |
If an All Pass system has poles at $0.8\exp\left(-j \dfrac{\pi}{4}\right)$ and $0.8\exp\left(j \dfrac{\pi}{4}\right)$, There would be $2\pi$ jumps in the phase (principal value) at angular frequency $\omega= \dfrac{\pi}{4}$ and $\omega= \dfrac{7\pi}{4}$, but why would there be $2\pi$ jumps also at $\omega= \dfrac{3\pi}... |
The game Rocksmith and some polyphonic guitar tuners (like the PolyTune) are able to recognize, at least to some extent, multiple fundamental frequencies in single-source audio.
I know there are some methods for multiple fundamental frequency detection, like MUSIC and ESPRIT, but is it likely that these products implem... |
Does anyone know what algorithm is implemented in the Matlab function resample for timeseries?
I had a look on the manual but it just gives the instruction to use it: http://it.mathworks.com/help/matlab/ref/resampletscollection.html
Since the syntax is different from the one used for the usual resample (http://it.mathw... |
I need to create a sound source locator using three microphones. I plan to start off by connecting three microphones to my windows PC, and then taking input from them using matlab to monitor their individual microphone outputs in real-time. How do I go about this? Can I just take a usb hub and connect three usb-microph... |
In the context of a microcontroller, I have much less memory and far less processing power. This means, I can only buffer a reasonable window of samples (e.g. 16 or 32), and may not be able to use more advanced curve-fitting options, such as this one.
This plot shows the raw data along with filtered output. The filte... |
Given $A$ is an input signal and $B$ is an output signal:
I know that the frequency response function (FRF) is defined as the Cross Power of $A$ and $B$ divided by the Auto Power of $A$.
Logic tells me I should be able to get the same results using FFT of both signals by taking $\cfrac{\texttt{FFT}(B)}{\texttt{FFT}(A)}... |
Can someone help me with this problem:
Suppose we filter signals with the below pre-emphasis filter:
$$y[n]=x[n]-0.8x[n-1]$$
I have to calculate the impulse response of the filter (ok easy) and then I have to find the magnitude of Frequency Response (dB) at $F = 2000 Hz$ with $Fs = 16000 Hz$
Can someone tell me the for... |
I am very new to the concept of computer vision (and image processing), and trying to understand the algorithm used for edge detection.
One thing I'm currently struggling to figure out is the Laplacian of Gaussian
In this case, how to you determine the value of x and y? I thought the centre of the kernel was the orig... |
This is probably as basic a question as it gets, but still the answer eludes me. I have a function from a library that I can use to both modulate, and demodulate an FSK encoded piece of data. For the sake of simplicity, let's just say it's one byte.
From the example provided, it exists in the perfect world of PC memory... |
I'm trying to compute FFT of size (N/2) for sequence of real values of length N. I'm using this recipe.
Here is my code in matlab:
clear;
clc;
x = [1, 2, 3, 4, 5, 6, 7, 8];
n = length(x);
z = [1 + 2j, 3 + 4j, 5 + 6j, 7 + 8j];
Z = fft(z);
Fe = (Z + (conj(Z)))/2;
Fo = (-1j*(Z - (conj(Z))))/2;
for k = 1 : n/2-1
F(k) ... |
Can the inverse system of a stable system be unstable?
For the class of LTI systems, the criteria for stability of a system with impulse response $h(t)$ and systems function $H(s)$ are:
$h(t)$ be absolutely integrable.
or in s-domain:
$H(s)$ have the $j\omega$ axis (or $s=0$) in its convergence region, and do not ... |
In my understanding, unlike shifting and scaling, image rotation actually rotates the coordinates whereby interpolating the pixels at discrete positions. It is like a projection that discrete values are projected to another plane which are along with the pixel value.
Thus, I am wondering whether we could do 2-D image r... |
Suppose I have two signals $f,g$ which are orthogonal in the sense of the usual inner product on $\mathbb R^d$. Then if I'm not mistaken these signals have disjoint spectral support, i.e their fourier transforms have disjoint support. Suppose further that I know the spectral supports aswell. How can I make use of this ... |
Hello I'm new on image processing.
I know there are a lot of threshold and region growing algorithms to use.
But I would like to know what is the trick for choosing suitable method for an image ?
Currently I'm trying to extract characters from this image
and most of the methods I've tried didn't really satisfy me. S... |
Currently I hope to use scale space representation to filter one image. Features in one image can be filtered using an Gaussian smooth filter with one optimal sigma. It means different features in one image can be expressed best in different scale under scale space representation.
For example, I have one image with on... |
Theory
Suppose an OFDM transmitter with four subcarriers is implemented using the structure shown below.
If the serial data input rate is 40 bps, each parallel stream will run at 10 bps. Theoretically, the spectrum (assuming BPSK modulation for each subcarrier) of the IFFT output $x$ should look like shown below (the ... |
What type of filters can be used to remove Poisson noise in an image? I tried applying Gaussian filter to few points in the spectrum.
|
I tried to write my own circular convolution function in python using the fact that for two signals $f$ and $g$ we have
$$ \widehat{(f * g)} = f \cdot g $$
So I tried this
from scipy import array, zeros, signal
from scipy.fftpack import fft, ifft, convolve
def conv(f, g):
# transform f and g to frequency domain
F =... |
I'm quite new to this; I'm try to classify textures as defective or non-defective. I've used a Gabor filter bank with Matlab which outputs a column vector of the Gabor features of an image. I have a data set of non-defective images and defective images.
My question is, what can I now do with this (or these) feature vec... |
I have the following image:
and I would like to fill in its contours (i.e. I would like to gap fill the lines in this image).
I have tried a morphological closing, but using a rectangular kernel of size 3x3 with 10 iterations does not fill in the entire border.
|
I have time domain history data. Then, I convert it into FFT to get the frequency domain history data. The problem is, how to determine the bandwidth of the FFT data? Is it automatically synchronize with my time domain history data? The bandwidth of time history data is $50 Hz$-$500 Hz$.
|
Problem.
There is a discrete signal $f[i]$ (example below).
It is known, that $f[i]$ have a form of rectangular pulse with additive white Gaussian noise.
$f[i] = s[i] + n[i]$,
$s[i] = \alpha(\theta[i - i_{1}] - \theta[i - i_{2}]) + c$,
$i_{2} > i_{1}$,
$i_{1} > N$
Where
$\theta[i]$ is a Heaviside step function,
$n[i... |
I need to extract (estimate) a square-shaped signal from the main signal. This square-shaped signal should be located within a specific range, in fact all we have to do is to estimate its height. A square-shaped signal is not periodical, it's length can vary from 1/12 to 2/3 of overall length.
I've read some literatur... |
I'm working on a computer vision application (using openCV) and the goal is to measure the width of an object before/after it falls on a container. So after the fall its position will be random, something like this to be clear:
The only object I want to measure will be clear filtering the image and will be always on ... |
I'm looking for some advice, I am hoping one of the many forum experts could help me with some advice on the sampling and processing of signals. I am using an STM32F4 processor with 12 bit ADC and DSP core. Presently, I can use either CMSIS DSP or Siglib libraries. Here's a point description of the problem space:
I can... |
I'm trying to plot my FFT of File. This i my result.
I made my FFT application:
I want to obtain something very similar to plot of WaveSpectra, please tell me my mistakes..
My FFT
The result from WaveSpectra
Other result from WaveSpectra
Here my code:
double[] Samples = null;
try {
RandomAccessFile rafr = new Ran... |
I used the Matlab function fft but I am not sure which function is the correct inverse to go back to the time domain.
|
I appreciate anyone that takes a moment to help me with this problem. I've been banging my head against the keyboard for a while, searching forums and DSP tutorials and I can't figure this problem out.
Here's a couple links I found very helpful from this forum on the subject:
http://www.dspguide.com/
http://wiki.scipy.... |
Hello I'm trying to ocr on a special image.
I've done adaptive threshold , polartocartesian transformation.
this is my original image
this is adaptive threshold
this is polartocartesian transformation
what do you think is this image good enough for OCR or do I have to optimize my preprocessing steps ?
|
I have developed a function which down samples an input signal.Say I have an input signal with a sampling rate of 512 samples/sec and would like to down sample it 128 samples/sec, then the down sampling factor is discrete and it is 4. In this case, I believe that the input signal will successfully be down sampled to 12... |
I am new in DSP.I have the flowing code
t = linspace(0,1,1000);
x = cos(2*pi*60*t);
CWTcoeffs = cwt(x,1:64,'sym8','plot');
colormap jet;
and my output is:
their have scale in y axis but i represent in frequency
what is the solution in matlab
|
I have a rectangular image which need to transform so the image fits an arbitrary convex quad. Easy? Except I have an important constraint, the features on the original image must maintain their relative area after the transformation.
I've tried a perspective transformation, but of course feature's relative area's chan... |
I have a small team project: use Matlab implement a Gui 5-band digital equalizer using FIR and IIR filters (2 cases) - The sample rate of all audio files is standard audio 44.1kHz. Coding is not my concern, however, I have 2 questions
(1) What are 5 center frequencies that I should choose ?
I saw a homework problems se... |
I already have a low-pass filter and want to remove the DC offset of a signal with a high-pass one. I guess you can subtract a low-pass filtered signal from the original signal and the result is a high-pass filtered signal?
My low-pass filter:
x = input, raw signal
y = output, low-pass filtered signal
n = current samp... |
This is probably a simple question, but I have a hard time understanding how c. converts to d. in the following picture.
Further reading as an answer is acceptable. After all, I am only asking this because the PDF I’m reading doesn't really explain why c. results in d. The more I look at it, the more I am confused.
|
I would like to design a very simple MATLAB Gui program to adjust the magnitude of an audio while it is being played via a slider (slider1). However, the program any adjustment will not response until the audio is finished playing. How could I fix this?
The audio is played after a push button is pressed
function PlayF... |
What is the difference between single channel separation and single microphone separation? Are they equal?
|
I will train my own cascade, using python and opencv. I want it to be as precise as possible, so as I read it is common that training takes one or two weeks to compile. I want to use cloud computing to not wait that long. I research but cannot find a single guide that explains what to do. I have pozitive and negative ... |
In BCI competition 3 dataset 5, It has been stated that we have to take 8-30 Hz frequency and break it with resolution of 2 Hz then it will result in 12 components but I am finding out only 11 components.Why??
The description can be found out here...
http://www.bbci.de/competition/iii/desc_V.html
|
I want to create a program in matlab that could act like karaoke software. At this moment i have following lines of code:
clear all;
[y1, fs1] = audioread('miszmasz.wav');
[y2, fs2] = audioread('voice.wav');
%checking if there is an audio file
if y1 == 0
return
elseif y2 == 0
return
end
%spliting into left a... |
I have just started studying Digital Signal Processing. Can someone explain what is the difference between a Discrete Signal and a Digital Signal in simple words?
Thanks in advance !
|
I try to combine my 7 band FIR filters into 1 FIR in order to process an audio signal. I use convolution to combine their coefficients but the effective is not the same as I apply 7 different FIR filter. I also have the gain problem
H = conv(conv(conv(conv(conv(conv(h1,h2),h3),h4),h5),h6),h7)
My questions are
Is co... |
I am trying to understand why auto and cross correlation helps find the best fit line in least squares. I have an equation as stated here:
$Ax=b$ -- I have not exact solution, so I use the least squares equation.
$$A^TAx=A^Tb$$
I can solve and use the equation but I am trying to figure out why cross and auto correlatio... |
I compute the output of a LTI system, can someone tell me if my answer is right..? and help me with my others questions?
The impulse response is: $h(n) = \left(\frac{1}{2}\right)^nu(n)$ , entry is $x(n)=u(n)-u(n-1)$ in which $u(n)$ is unit sequence.
(1) We know that the outpout of this LTI system is $y(n)=x(n)*h(n)$
(2... |
I failed to do this question on the exam and finding it very difficult, I would be glad if you can help me solve it. How shall I start?
|
I am reading about audio processing for my university project and I found out that DCT basicly filters out some frequencies. I wanted to use for example high-pass filter and then DCT for compression, but isn't dct a filter?
What's the difference?
|
I know the answers of the below question but I dont know why, except linear, could you explain the rest? Small tick: correct, small tick with dash incorrect answer.
|
I do like to generate the FFT from time domain history data (experimental lab) using MATLAB. I am not so sure if my coding is right? Can someone verify the coding.
My code is as follow:
clc;
signalx = Ch1;
N=6678;
t=Time;
%Plot FFT Vehicle signal
zx = fft(signalx);
plot(abs(zx));
title('FFT Veh... |
Some of you may know the infamous Audio EQ Cookbook. You can take a look here.
We tried to implement the functions there to achieve a implementation for a Lowpassfilter.
However, when we try to lower our variable cutoff-frequency, we have some issues. With decreasing cutoff-frequency the signal is increasing in volume... |
I've been following the recent lectures on Coursera for image processing. In the lecture on the use of the discrete cosine transform (DCT) for image compression, he says that for images that are "Markovian", the DCT has the same properties as the Karhunen-Loeve Transform. He uses this argument to support the use of th... |
I am recently acquiring data through ADC on signals ranging from $0.1\textrm{ Hz}$ to $50\textrm{ kHz}$ ($F_s = 128\textrm{ kHz}$). I noticed that apart from the white noise, I have something similar to $1/f$ noise with a cutoff frequency near $1\textrm{ Hz}$. There isn't much that I found on equalizing this $1/f$ nois... |
I met with a problem: if I sample audio data at 11kHz (yes, that is slow!) for 10 seconds and apply a Fourier Transform to the data, at what frequency will a 15 kHz tone appear?
So the original signal is 15kHz? which is greater than half of sampling rate 5.5kHz, right? and thus alias appears. And then, in this case, ho... |
I'm dealing with a light sensor tracking a bursting LED. Unfortunately signal is weak and there are other light sources in environment. I've used optical filters on sensor but there is still some noise.
The LED burst with a fixed duration but the distance and time vary. So I'm interested in finding those 5 pixels wide ... |
I have a 2D image which contains R G and B values. I think this should be purely based on averaging of pixels and selecting the threshold to decided whether this is day picture or night one.
Can you provide any kind of different ideas to solve this problem?
I am shifting this question from the https://stackoverflow.com... |
To obtain the gradient from an image, one can use a linear filter that for example is derrived from the derrivative of a gaussian. Then convolving the image with this linear filter gives you the gradient image.
The gradient magnitude can also be approximated by use of (grayscale) morphological gradients, such as: (dila... |
What would be the key features of an audio track or set of tracks (like 5.1) for identifying film audio tracks that have a clear unbalance between speech levels and sound effects levels (apart from the obvious rms diference between tracks)? And how could they be extracted or calculated?
It's for a machine learning proj... |
I can't seem to understand how to derive the "twiddle sum" property:
$$\sum_{n=0}^{N-1}W_{N}^{kn}=N \ \delta[k\bmod N] $$
where $$ W_{N} \triangleq e^{\frac{j 2 \pi }{N}} $$
and $$ \delta[n] \triangleq \begin{cases}
1 & \text{if } n=0 \\
0 & \text{otherwise}
\end{cases}$$
I've tried doing the following... |
I was just wondering... The formula I learned to calculate the energy of the signal is expressed in the time domain:
$$E_x^{\text{time}} = \sum_{n=-\infty}^{\infty} |x[n]|^2$$
Then, what does the amount of energy gotten from the magnitude spectrum mean?
$$E_x^{\text{frequency}} = \sum_{f=0}^{fs/2} |X[f]|^2$$
Supposing ... |
Can someone tell me if the way I solved this problem is right?
Suppose I have a signal $X_a(t)= 0.5cos(700πt)+0.6cos(720πt)+0.1cos(780πt)$ where $F_s=8000Hz$
I want to determine the minimum samples which must be available in order the frequency resolution of DFT be ok.
I did this:
Firstly calculate each frequency of ... |
I have position and acceleration measurements and I would like to apply a Kalman filter to estimate the velocity of the system.
I am not sure yet about how to procede, but I will check the already answered questions on this website (like Estimating velocity from known position and acceleration , Kalman filter with acce... |
I can use the following type of barcodes in my project:
Code 39
Code 128
QR Code
Data Matrix
PDF-417
They will be printed with standard office printers and with boca printers. It should also be possible to show them on smartphone screens.
They will be read with webcams or Windows tablets. It would also be helpful I c... |
$$r_x(t)=\begin{cases}
2-2t ,& 0 \le t \le 1\\
2+2t ,& -1 \le t \le 0
\end{cases}$$
So how do I get original $x(t)$ from this?
I tried integrating the equations, and the value I got was $1$.
|
I am having a hard time in trying to use a Kalman filter to obtain velocity
from acceleration and position measurements. I think the main reason is that
I am not familiar with Kalman filters (I had only a couple of lectures about it
and I implemented it only once).
To start I had a look to some questions here and also ... |
I have been acquiring data through ADC and into memory, the input signal are square waves that step down in frequency as time goes by. Due to memory, I have a fixed maximum number of samples per FFT. But I would like to compare their magnitudes, and phase along side each other. The max length of FFT is good enough for ... |
I am new in this field (junior year) and I have a question may a little lazy for the majority of the forum.
I have a speech signal $ x(t)=0.4\cos(100πt)+0.3\cos(1800πt)+0.2\cos(980πt)$ and I know that the sampling frequency is $F_s=1000$
If I want to find the signal form after sampling the only thing I have to do is re... |
I am trying to wrap my head around the original wavelet shrinkage papers of Donoho and others from the mid-90s. There are a couple of curious things that confuse me, and I'd appreciate some input:
The universal threshold is often (and also in secondary literature) given as $\sigma\sqrt{2\ln n}$, but sometimes I see $\... |
I cannot find anything truly helpful in my book, and I find mostly theoretical answers and not practical.
So I am posting the exact exercise I want to solve, and through it I hope I'll work some things out.
Our data is that we filter voice/speech signals with this filter:
$y(n) = x(n) - 0.8 x(n-1)$
a. What's the impul... |
Can someone verify this statement?
The hold operator is assumed to be a zero order hold
Then the laplace transform of this hold operator has a well known form $(1-e^{-sT})/s$
Let w approach 0, we see that our resulting function is $(1-e^{\alpha T})/\alpha$ where $s = \alpha + j\omega$
What does it mean for the time hol... |
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