instruction stringlengths 31 27k |
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Well, I am new to DSP so I apologize for the basic question. I am trying to transform audio 16bit PCM (which I store in normalized 0.0-1.0 range) samples into frequency domain using fft.
I understand that FFT transforms N points of time domain into two N sized frequency domain array. My problem is determining the size... |
How can I periodically estimate the states of a discrete linear time-invariant system in the form $$\dot{\vec{x}}=\textbf{A}\vec{x}+\textbf{B}\vec{u}$$
$$\vec{y}=\textbf{C}\vec{x}+\textbf{D}\vec{u} $$if the measurements of its output $y$ are performed in irregular intervals? (suppose the input can always be measured).... |
Say you are given a vector t =[...] for the time to be used to generate a sine wave of a certain frequency say FHz. I have all the data for t so I can conclude what dt is (my dt = 1d-5). Now I am to do an FFT analysis and plot magnitude vs frequency so I expect to get my original amplitude and also get back the origina... |
I have an assignment where I'm given the DFT of a sequence $x[n]$ as $X[k]=\{4,3,2,1,0,1,2,3\}$ and also
$$y[n] = \left\{ \begin{array}[cc]
xx[n/2] & \text{if n is even} \\
0 & \text{otherwise}
\end{array} \right. $$
and I'm supposed to find and sketch the DFT of $y[n]$.
So $y[n] = \{x[0], 0, x[1], 0 ... x[7], 0\}$ and... |
In their paper on Image Pyramids, Burt and Adelson state the following about pyramid generation.
An additional constraint is called equal contribution. This stipulates that all nodes at a particular level must contribute the same total weight($=\frac{1}{4}$) to nodes at the next higher level. Let $\hat{w}(0)=a$,$\hat... |
I need to get information about contrast of an image, and I compute its value in this way:
contrast=Pmax/Pmin
where Pmax is the pixel of the image with the highest value, an Pmin is the pixel of the image with the lower value.
Is this the correct way, or there is another approach?
|
I need to process an audio signal in an Android app. As shown in the two examples below, there is a 16kHz carrier signal and a lower frequency signal of varying frequency (0-4kHz). Does anyone know of an efficient algorithm to extract the lower frequecy signal in an Androd app?
http://mc.weatherflow.com.s3.amazonaws.co... |
I have a signal which contains sinusoidal components that oscillate at different frequencies. I think the phase of the sinusoids is changing with time. I could do a Fourier transform on small chunks of the signal and the phase of the sinusoids from that, but is there a better way to do this with a wavelet transform? I'... |
I performed an image downsampling with averaging operation(The image is divided into 2 by 2 blocks,then it is downsampled by 4 times.New pixel values od downsampled image is replaced with the mean of each block) as follows.
img = imread('cameraman.gif');
avgfilter = fspecial('average', [2 2]);
avgimg = filter2(avgfilte... |
Convolution of two sine waves (or tones as called in audio) is theoretically not defined as the integral is infinite. Taking finite duration windowed sine waves and doing there convolution computationally always contains a fundamental frequency equal to that of the lower frequency sine wave. I am not getting an intuiti... |
I am using real data of wind, which is a random one to get the power spectral density of these values.
How can I do to get this PSD in matlab, and Can I get an analytical expression for this psd.
Thank you.
|
I have a few questions about this assignment I am solving.
It is q3.33(a) from oppenheim's DSP book 2nd edition.
basically, the question gives you a pole zero plot, from which I was able to determine the correct x[n]. (Couple classmates confirmed my answer on that).
and it also gives you the y[n] of the form $(0.5)^n... |
I want to try to summarize some relationships between z transform, ROC and causality. But I am getting confused between left/right sided sequences, the ROC and causal.
If you need stability then the ROC must contain the unit circle. I can understand that.
If you need a causal system then the ROC must contain infinity... |
I am quit puzzled about a notation I found in Gardner's 1986 book titled Introduction to Random Processes. It is in the chapter on Cyclostationary Processes at section 12.4 which pertains to Linear Periodically Time-Variant Transformations:
(...) many modulation systems can be modelled as the scalar response
of a mu... |
I want to find out PSNR and SNR values of two images A and B of same dimension in Matlab.I used the following code
[peaksnr, snr] = psnr(A,B)
and getting an error
Undefined function 'psnr' for input arguments of type 'uint8'.
Then I converted both the images into double as follows
A = double(A);
B = double(B);
an... |
I am working on a position controller for a marine vessel. I have a measurement signal containing the y-position of the vessel that consists of both low frequency (<.1 rad/s) and high frequency (>.6 rad/s) motions. I am currently using a state observer that separates that combined motion in two separate states (a LF an... |
I need to implement an fft of size 2^20, but the hardware has imposed a limit of 2^10 fft size.Then how can i implement the FFT. please point me to some literature where i can go through the process
|
I have a bunch of piezo elements, that I want to use as "Mics", for that I will test their reaction in a Anechoic chamber. My question is what the are characteristics that I can measure that describe those elements and can help me analysing the generated sound later, I already have in mind the following :
Amplitude re... |
I understand this question may have been addressed however I have a major difference.
I started out with a sinusoid of form A*sin(2 pi freq t + phase) and I know all the components, this represents a force a system and produces a response that has a delay and that is periodic in a way, no fully sinusoidal though (I ca... |
How does one recover the complex IQ data from the output of a FM modulator. I have combined my signals correctly and performed FM modulation on them, now I'm looking for a way to find the IQ values that correspond to each sample output of the FM modulator?
I am writing a modulator for my signal to test the performance... |
I have a system that is forced by a sinusoidal function with specific amplitude, phase and frequency ( all these are known ahead of time). I get a deformation of sinusoidal form as well (think of a damped harmonic response), and I would like to find the phase lag. I was advised to use correlation but I don't have the f... |
Is there any known way of differentiating between frames of real footage and animated content? It seems like there's probably range of colors more likely to appear in a histogram for real footage, or maybe a level of color variation more likely to be in animation.
The case I'm looking at is converting GIF images to h2... |
I know what I am attempting is not easy but I have a spectrum with peaks at 10Hz 20Hz and 30Hz. I also have various amplitudes at these peaks. I want to recreate my original signal. I initially thought this could be done by adding up sine waves
y(t) = a1sin(2*pi*ff1*t)+a2sin(4*pi*ff1*t)+a3sin(6*pi*ff1*t)
with a1 , a2 ... |
Can anyone give some tips how to obtain Skeleton from distance map?
I already managed to obtain skeleton using erased method (e.g. zhang suen) but it's time consuming because it's an iterative method (the thicker the object the longer it takes).
OpenCV has well optimized distance transform function than can be calculat... |
this is a newbie question.
My goal is to create audio filter that cleans human speech from noise. I started with learning all about fft filters. At first I've created simple fft filter with rectangular window. But it was bad, I heard distortions in sound. So I decided to apply some window. I choose sinus window and now... |
I'm moderately skilled in programming classic microcontrollers. Embedded C, built-in peripherals, interrupts, the daily bread of an Embedded programmer.
I'm interested in expanding my skillset onto programming DSP. Of course the basic steps would be grabbing some development kit, some manuals and tutorials and get to w... |
I am trying to use cross correlation instead of FFT to find lags between 2 signals. I am trying to understand cross correlation by using it on sin and cos and expecting phase lag to be pi/2 but it is not:
clc
clear
frequency =1000;
t= linspace(-1000,0.1,1000);
sine_wave = sin(2*pi*frequency*t);
cos_wave = cos(2*pi*... |
I have been studying about Frequency Domain from many days but still i am not able to clear one of the confusion. We say that in frequency domain an image is represented as waves. What I am able to think is that if we take out an intensity profile column wise (i.e one row of intensity values), we have pixel location an... |
i am working recently on a project in which i want to implement a Kalman filter as being an observer, and i couple this observer with a state feedback controller that produces control actions necessary to control a mobile robot.
The problem is that the governing equations for the mobile robot's motion contain errors, ... |
If I have a vibrating string, then there will be no motion at the endpoints. If I move away a bit from an endpoint then I'll get a signal, but particularly the fundamental will still be attenuated. Higher modes may suffer less or even benefit. This explains why the bridge pickup (closer to the edge) of an electric guit... |
I'm working with cascaded second-order allpass filters. Each filter is characterized by conjugate pole/zero pairs. In a long (>50) cascade of identical filters, the pole locations of the overall transfer function begin to diverge from their initial, theoretical location. They seem to fan out in a ring around where the ... |
I am having question about the HPF. In particular, it is having a high pass guassian filter which is applied to the original image below. High pass filter should only detect edge (lots of change in terms of intensity of the pixel). But how come the resulting image, still contains the two "white" things in the middle? S... |
I am spinning this question off from the question from johnny. Matt L. and I have had directly opposite conclusions to johnny's question.
I want to decouple the question from issues of causality and other goofy stuff.
So we have a simple first-order recursive system described with time-domain I/O equation:
$$ y[n] = p... |
In ~2006 the FAST corner detector was introduced. One of it characteristics is that you have to train a decision tree from some test image in order to learn the types of corners present and relevant in the test image. One of the consequences is that the images in which you will be applying the detector have to be somew... |
I ve been researching about harmonic distortion and THD. From what i ve seen on the internet, HD is caused by non-linear signal processing. I want to calculate it using an digital signal processor. For this, i suppose i need to compare the distorted signal with the original one.
My question is, is it possible to genera... |
If we have a 1-dimensional wave in time domain, it can be represented in frequency domain with x axis indicating the frequency of the wave and y axis indicating amplitude/magnitude of the wave. But what about a 2-dimensional wave in case of images in Frequency Domain? Because in 2-dimensional wave we have a cordinate (... |
I have some basic maths skill up to derivatives.
I ask you if you know any simple/concise book that can introduce me to DSP world. Thank you.
|
I want to apply the following steps in matlab
1) Read the audio data
2) Apply window function on that data
3) Apply fft on the result of window function
4) Make power spectrum of fft
5) divide power spectrum in Barks bands
[y,Fs,bits] = wavread('file1.wav');
Nsamps = length(y);
ham= hamming(Nsamps);
ham_fft = abs(fft(... |
How can poles and zeros exist at infinity?Can anybody explain using a system function?
|
I'm creating windowed sinc filters to apply them to certain signals that I'm dealing with. To design the filters, I'm using the approach described in the book "The Scientist and Engineer's Guide to DSP". Here's a brief resuming:
\[h[i] =
\left\{
\begin{matrix}
Kw(i)\frac{\sin(2\pi f_c(i - \frac{M}{2}))}{i - M/2} & ... |
I run into the following filter that is supposed to do some type of time averaging of $x$
$$y[n]=G\cdot y[n-1]+(1-G)\cdot\dfrac{x[n]-x[n-1]}{T} $$
where
$$G=e^{-2\pi c T} $$
$T$ is a time interval, and the samples of $x$ and $y$ come every $T$ seconds (we can assume $y[0] = 0$).
The definition includes a quantity $c$ t... |
I come from a background of computer science and I got interested in image processing but it requires a lot of signal processing, so could someone provide me a good beginner's book, that maybe first year college/uni students use?
I understand that dsp is crucial for image processing, so I focus on the first thing for n... |
Can anyone please clarify the difference between correlation/convolution and matrix multiplication? As I thought either convolution or correlation is similar to matrix multiplication. I read this question difference between convolution and multiplication, that is about simple multiplication.I am asking about matrix mul... |
This has to do with creating the synchronization circuit for MSK in a DSP as described in "Minimum Shift Keying" by Subbarayan Pasupathy published in IEEE Communications Magazine.
At the beginning of the circuit he shows signal passing through a "squarer". The explanation reads:
"Although the MSK signal $s(t)$ has no... |
I have come know of Huffman and Arithmatic encoding that try to curtail redundant data and thus compress data. Why are they called Entropy encoding? What does that mean? Why not just call them compression encoding?
|
I am not sure if the question is off topic but I need some input before I can finalize anything.
Actually to do my higher studies (in research) I wanted to make sure that I land my career in Digital Image / Signal Processing only. I have a strong background in Computer Application development. Now the Signal Processing... |
I'm implementing (in software) FIR filters. I have done two implementation: convolutional one and based on multiplication in frequency domain with FFT/IFFT. I'm using standard FFT implementation (not my own) and overlap-add processing.
FFT parameters are automatically selected to be like this: $fftSize = 2 * 2^{\lceil\... |
I am learning Fourier Transform from many days but till now I am not able to understand what does phase angle image show us or tell us? They say that MAGNITUDE tells "how much" of a certain frequency component is present and the PHASE tells "where" the frequency component is in the image. I understood about the MAGNITU... |
I’ve got what it looks like a periodic signal but the periods seem not equally repeated, I want to estimate the PSD, particularly I need to obtain the dominant frequency of this data. I’ve estimated the autocorrelation and then applied FFT, the results as shown in the figures below.
It seems that FFT algorithm is not t... |
How would one implement a frequency divider? I know I can resample up and then treat it as if I'm at the same frequency, however I want to divide the frequency of my signal by 2 and keep the same number of samples.
I've tried multiplying by a cosine wave at half the frequency then low pass filtering. However the prob... |
I'm currently in the process of writing DSP code, based on the amazing RTLSDR project, and using this as a vehicle to teach myself the basics of DSP (I have an extensive background in software development, but none whatsoever in physics, EE or DSP.)
Specifically, I am trying to implement FSK from scratch using only the... |
I am designing an algorithm for image processing. I am using MATLAB for the processing of the images(video). How I can detect the position of the corners in the image?
|
I have a continuous-time signal that is of the form
$$x(t)=A \sin(2 \pi f t+\phi_1)+B \sin(2 \pi (5f) t+\phi_2)+C \sin(2 \pi (7f) t+\phi_3)+n(t)$$
sampled at 50us where $f$ is the frequency I want to find out A,B and C are constants where $$0.15 \cdot A>B+C+|n(t)|$$. I don't need to know the value of any of the phases ... |
I need to do a DFE for QPSK simulation in MATLAB.
The system consists of a QPSK signal transmitted with power = 1 which is then pulse shaped with a square raised-cosine filter. Add AWGN and a filter receptor with ISI = [0.25 1 0.25].
Summarizing and clarifying, the sequence is:
QPSK signal -> Upsample -> Rcosine -> AW... |
For my course I need to implement a 30 point Cooley-Tukey DFT by transforming it into a 5x6 matrix.
I have tried to implement using the following Matlab code:
clc; clf; close all; clear all;
N = 30;
rows = 5;
columns=6;
data = linspace(0, 2*pi, N);
data = sin(data)+0.1*sin(pi*data);
plot(data);
count=1;
matrix=(reshape... |
The Fourier spectrum is in the Figure, how to find the essential bandwidth?
|
I seek a scaled audio file format for my dsp output.
I'd like to produce "scaled wav files", when I've done my pre processing. Eg. if the amplitude is 15.5 (of whatever unit), you should be able to open a standard wav/whatever file and see "15.5".
If you produce a normal 16 bit wav file, you have to tell the user, to ... |
I am designing an FIR filter.my specs are fs=300MHz, Fc=45MHz, Fs=75MHz, passband gain=3dB , stopband attn=>40dB. what parameters I values of a have to provide in firpm for the minimum order filter design.
I am putting a=[0.01 0.01]. is it correct?
|
I have some two tone origami paper. This particular paper comes in six types, each with a different pair of colors. Here's an image that shows all six:
Given a picture of one of these sheets, I want to figure out which type it is, and do so in a way that's resilient to common changes in lighting.
Extracting the actual... |
I have some data $(X,Y,Z)$, which is a set of measurements $Z$ over a $2D$ space $X$,$Y$. The $Z$ data on this space is continuous except for some discontinuous jumps in certain
domains of $X$,$Y$ (due to calibration errors).
For example, let's say $X$ and $Y$ are both between $[0,1)$, and let's say my underlying tru... |
I have been struggling with this issue for months now, trying to understand which one to pick: FFT or cross correlation.
I am given a sine wave (with phase, amplitude, frequency and time vector known), and I have to extract the phase shift between that sine wave and another signal. The new signal has the same frequenc... |
In recent years, I have started doing work in image analysis and have used the Barbara and Lena images. Early on, I became aware of the interesting history surrounding the Lena image and that got me wondering:
Who is Barbara?
Where was the photograph shot?
What was the original intent of the photographer? Presumab... |
To create a scale space, I applied a Laplacian of Gaussian filter on the following image:
After the scale space was created, I plotted circles around local maxima in scale space. However, instead of $9$ circles, I got many more as shown below(image below not to scale).
The following video shows the LoG filter respons... |
I have a sine tone, doing some math, and finally gets an harmonically accumulated sine tone in complex numbers. I do want to preserve the phase information of imaginary part on the other end, which will continue processing the complex signal. So,
Is there any way i can safely pack the complex number into any of standa... |
I want a image. Firstly I want to do fourier transform and get the magnitude and phase information. And then make a new random phase, finally use the changed random phase and original magnitude to make the inverse fourier transform. But I can not achieve it?
The code is shown here.
img = imread('flag.jpg');
imafft... |
The result of calculating the fourier transform using numerical integration is:
the result of using Matlab's FFT is:
So where did I go wrong here? I know the FT of a Gaussian should be another Gaussian. Shouldn't they be the same? Furthermore I would have thought the
Edit:
Following a suggestion using the fftshift f... |
I'm confused by the SMA. I'm reading a paper which states that "SMA is the square root of the summation of the filtered acceleration signal". So I understand that it's $sma = \sqrt{\sum{samples}}$
I check the referenced paper, which says that "SMA is calculated after dividing area under the acceleration curve with the ... |
I need to find the similarity between two signals, i.e., measured signal and the reference signal(e.g. a simple sinusoidal signal). What I mean by similarity is, if there is any decentralized glitch or any change in freq for one particular cycle, the similarity measure should give 'False' or a value in 0 - 1 scale for ... |
I'm a newbie and I have a few questions regarding ECG signal. Here is my plot of raw ECG signal with a sampling frequency 200Hz in Matlab.
Does it look good to you? Why is it flipped?
|
I'm currently working on a project that involves using an electron gun and it would be really nice to know the spot shape of the electrons coming out of the gun (the frequency of electrons at some x,y away from the center of the beam). Our current assumption is that the beam's spot shape is somewhat gaussian.
We have ... |
I am currently working on a frequency domain real-time application on a digital signal processor. Currently for one time frame of my algorithm I read in time domain data into a buffer, perform a Fourier transform with a Hanning window (win), which I generate using MATLAB with the following code:
win = 2*hanning(N,'peri... |
I'm trying to process an audio stream through a series of peak filters at static frequencies that range from 0 to nyquist. If I start the range from anything below about 25Hz, the audio gets corrupted in some way (I haven't looked at the values yet, so I don't know what the corruption looks like) that results in no sou... |
Hello does any one know or at least can lead me to the right way?
What does not change pronunciation of words when someone is speaking. The idea is to create a app that checks if word/sentence is pronounced correctly. To do so I need something in pronunciation that does not change at all or changes in small range.
For ... |
I am confused with 2d magnitude plot of frequency spectra. So we have 2 images, the first one is shown at the top, and the dilated or enlarged version of the white box is shown at the second row.
For the dilated version, in the spatial domain when travelling in the $x$-direction, you see a fast change in intensity, a... |
I have to compress an image using SVD. I did a few things, but all of them produced a black & white image. However, the output must be RGB. How can I compress an image without changing its original format?
|
I'm asking as I'm looking at Wavelet transforms, and I see that it is possible to have a TF decomposition or a representation of the signal.
Looking for example at the code here we can see that the TF representation
(tfr_morlet function)of a signal returns the averaged power and optionally the intertrial coherence.
The... |
Can someone please explain how to determine if a system is invertible or not. The particular systems that I am looking at are:
$y(t) = \cos \left( x(t) \right)$ - I believe this is non-invertable because of the periodic function
$y[n] = nx[n]$ - I think this is invertable and the inverse is $x[n]/n$
$y[n] = x[4n + 1]$... |
We know that applying a filter on an image is called correlation or convolution depending on the filter angle. In Gonzalez I have read that we can apply linear convolution on an image. Here I have a doubt.
Suppose we have a vector image of 4 elements and a filter/kernel also of 4 elements then we get the output of li... |
I am making an application where I record sound on an IOS device using the built-in microphone. I want to process the sound in such a way that only frequencies in the 16-18 kHz range remain; the other frequencies should be eliminated. I read in some posting that I can use a band pass filter to get this result. Any help... |
I'll start off by saying, yes this is a homework problem, and no I do not want the answer. I simply need to know where to start.
The input to a system with a frequency response of $H(\omega)$ is $x(n)=A\sin(\omega_0 n+\theta)$
What is the expression for the systems output, $y(n)$.
As I said, I just am unsure of where... |
In my filter the fs=300MHz. and no. of coefficients is 31. I have following queries.
(1) what will be the number of multiplications per second?
(2) can multiplication be done in one clock cycle?
(3) why normally multiplication per second is calculated and not addition/substraction per second in the filter implementatio... |
Now that we can operate in the frequency domain by way of FFT. We can implement ideal lowpass, highpass and bandpass digital filters. then why there many textbooks talk about designing digital Butterworth filter, digital Chebyshev filter etc, by the so called impluse invariance method, bilinear transform method?
I cann... |
In my filter design, FIR gives order 14 and IIR gives order 8 or 4.
so what are the disadvantages of IIR design?
|
In reading a paper on demodulation there was this sentence:
"Using complex-envelope notation, the noise corrupted waveform at the
IF filter output can be written as...".
I've not heard this term, what does the complex-envelope notation mean?
The equation below that statement is:
$$z(t) = \mathrm{Re} \left\{ \widet... |
This is a general question on the laplacian operator, which has two different versions. The first version is :
\begin{matrix}
0 & 1 & 0 \\
1 & -4 & 1 \\
0 & 1 & 0
\end{matrix}
The second version includes the diagonal:
\begin{matrix}
1 & 1 & 1 \\
1 & -8 &1 \\
1 & 1 & 1
\end{matrix}
Including the diagonal d... |
I have a signal that I am filtering on an MCU (STM32F405) that I get from a codec (CS42L52). It is being sampled at a sampling frequency of 48kHz (actually between 48400 and 48600). I have a ping pong buffer setup with DMA and am using I2S to send the signal between the MCU and codec.
I seem to have a glitch somewher... |
I'm computing discrete short time Fourier transform. Data is split into overlapping chunks and gaussian window is used for each chunk. However, I'm not sure how much overlap there should be between chunks.
If they are too sparce I will loose information between them (gaussian window for each chunks is shown here):
On ... |
I have computed 12d-MFCC features on 3 consecutive audio frames, say:
For 1st frame: a1 a2 a3 ... a12
For 2nd frame: b1 b2 b3 ... b12
For 3rd frame: c1 c2 c3 ... c12
How to organize these MFCC features, in a1 a2 ... a12 b1 b2 ... b12 c1 c2 ... c12 or a1 b1 c1 ... a12 b12 c12? Is there any difference between these two ... |
Currently I am trying to use MFCC feature for audio training in order to do human voice classification (can later distinguish the person who speaks).
I have problems to determine the proper time length of audio training samples? To be clear, say I have computed 12d-MFCC features over 20ms window (10ms overlapping), if ... |
I'm trying some wavelet analysis of EEG signals, using the phase lock measures
from [1], specifically the S-PLV measure.
In order to calculate that we perform a wavelet transform on the signals and
one of the parameters we have to set is the number of cycles for the wavelet.
The authors suggest that the number of cycle... |
I am looking a way to make outputs match. For len 160 wouldnt the error be significant? For 40 ? I have included some sample output showing the error.
Direct Form 2 Original
void highpass(const short int *x, short int *y, int len, int filtID, long int *mem)
{
int i;
const short int Pcoef[5][3] = { { 16384, -3... |
I'm implementing an Speech Recognition System over a Cortex-M4 processor (STM32F4-Discovery) and I'm facing a problem when a compute the RIFFT.
I believe that I'm passing to the function the vector in the right form, but when I compare the output to the one obtain by MATLAB, it's quite different.
I've also tried to per... |
Given the input to the IIR filter
$$
x[n]=\delta [n]+0.25\delta[n-2205], \quad \text{where}\quad \delta[n]= \begin{cases}1, &n=0\\0, & n\neq 0\end{cases}
$$
What I want to do is to design an IIR-filter that cancels the "echo", such that I'm only left with the $\delta[n]$. To do so I of course make an IIR-filter, where ... |
What is the important advantage of adopting a wavelet based denosing scheme such as SURE, SUREshrink, etc. than using approaches involving low pass, high pass, band pass, band reject filter for signal denosing?
More precisely, what is the advantage of using wavelet denosing on EEG signals than other methods.
Thank you!... |
I have amplitude modulated sound file signal and after demodulation using FFT I got fundamental frequency and its harmonics and of course noise.
How to remove that fundamental modulation frequency and its harmonics from spectrum using Matlab… any suggestion… m-file…?
|
What is maximum data rate a wireless link can support using BPSK for 10^-6 BER, Bandwidth of channel is 200 kHz and SNR 21.335dB ?
As per my understanding , Eb/No can be calculated for BPSK for given BER using curves and value found is 11.29dB, and after that we can use relation
Eb/No = S/N * W/R
R comes out to be aro... |
Let's say that I have a sampled signal x[n], it is being, in this exact order, up sampled by 2, down sampled by4, up sampled by 4 and down sampled by 2 to produce y[n].
It seems to me that it should be pretty self evident that since we up sampled the signal by 2 and down sampled it by 2, then up sampled it by 4 and dow... |
I thought this was supposed to be an obvious question, until I finally set up my real time system.
So basically I have a transmitter that sends 128 samples/second to a receiver. The transmitted information is stored as an object in MATLAB and continuously updated.
When people talk about real time signal processing, I'm... |
The "textbook" OFDM block diagram is as the below figure. But I wonder if it satisfies the sampling theorem. Here, we have N points $X[k]$ before IFFT, and
$$x[n]=\frac{1}{\sqrt{N}}\sum_{k=0}^{N-1} X[k] \mathrm{e}^{\displaystyle -j\frac{2\pi k n}{N}}$$
This derives from the analog signal
$$x(t)=\sum_{k=0}^{N-1}\mathrm... |
I want to compute the autocorrelation matrix by using the covarience method.
X(n) is the harmonic process $X(n) = \sum\limits_{i=1}^3 A_i e^{jn\omega_i} + V(n)$
$V(n)$ is unit variance noise.
$ |A_i| = 2$ for $n = 1,2,3$
The phases of $A_i$ are uncorrelated random variables that are uniformly distributed between $–... |
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