20
votes
Accepted
What are your tips for DSP self-study?
Tips for DSP self-study huh. Well, ...studying 'signals and systems' is a great idea and having Matlab software means you have the tools to learn an awful lot. I think Dr. Steven Smith's book "The ...
9
votes
Accepted
Entropy : do we prefer higher or lower entropy?
You seem to have a number of misunderstandings, which I'll try to clarify while also trying to help with your questions.
The entropy of a source $H(S)$ gives the average codeword length to encode a ...
7
votes
Accepted
Educational purpose - What is the correct way to simulate a multipath fading channel which has ISI
Lets say we want to transmit a sequence of discrete data $\left\lbrace x[n] \right\rbrace$. But because we are living in analog world, the sequence must be modulated.
Call $T_s$ is symbol duration ...
6
votes
Accepted
Why are equalizers moving average model (FIR)
The reason why almost all linear adaptive equalizers are implemented as FIR filters is that FIR filters are always stable and that there exist relatively simple and effective adaptation algorithms. ...
6
votes
Accepted
Why use parametric based estimation methods - confusion regarding terms
Hi: I'll try to answer as briefly as possible and only with respect to statistics. not dsp.
In statistics, if you have a nice pdf such as the normal distribution, then maximizing the likelihood is ...
6
votes
Accepted
Doubts on LMS derivation
Here I expected $y(n)$ is to be computed by convolving $x(n)$ with $h(n)$, but in the equation given by Wikipedia it is shown as a matrix multiplication
$y(n) = h^H(n).x(n)$. Are these two ...
6
votes
Why Zero Padding in the Center of the DFT Interpolates / Upsamples the Signal (Sinc Interpolation / DFT Interpolation / Periodic Interpolation)
What you are experiencing is technically called interpolation by DFT; i.e., interpolating a time-domain sequence $x[n]$ by properly zero filling the middle portion of it's DFT $X[k]$ (and taking the ...
5
votes
Accepted
Convolution in frequency domain
If you have two DFTs $A[k]$ and $B[k]$ (note the correct representation of a sinusoid at DFT bin number $1$)
A = [0,-j,0,0,j]; B = [1,1,1,1,1];
with the ...
5
votes
Accepted
Relationship between information retrieval and source separation in signal processing
There are, a few discrepancies that might be making a difference here. My suggestion would be to edit the question for clarity. There are quite a few assumptions that lead to non-straightforward ...
5
votes
Accepted
Proper method for generating Channel coefficients in MATLAB
Complex channel coefficient is just a way to represent the independent real coefficients. You just need to generate ...
4
votes
What Resources Are Recommended for an Introduction to Signal Processing (DSP)?
The DSP neophyte who has some mathematical maturity may want to start with
Martin Vetterli, Jelena Kovačević, Vivek Goyal, Foundations of Signal Processing, 2014.
which is freely available online. ...
Community wiki
4
votes
What are your tips for DSP self-study?
I can recommend online course - Coursera DSP. There are very good introduction in mathematical basis of DSP and review of main DSP themes.
Online courses are symbiose of self-study (study time ...
4
votes
Accepted
Design of equalizer for wireless communication
Ok, there is some misconceptions in your question. I strongly recommend you to read a little more about the topics, but I will try to help you a little. My answers and some comments:
...linear ...
4
votes
Accepted
Matlab: Difference between flat fading and Rayleigh multipath fading, terminologies and concept
The fundamental idea to keep in mind is that in a wireless channels with reflections, if you transmit $s(t),$ you'll receive $$r(t)=\sum_{i=1}^Na_is(t-\tau_i).$$
Another important idea is that ...
4
votes
Accepted
Concepts: Ideal and Non Ideal Channel Resulting in Inter Symbol Interference
The book doesn't say that the impulse response must be zero for an ideal channel. It says that an ideal channel has exactly one, and not more than one, non-zero component, i.e. the ideal channel's ...
4
votes
"Don't window a transient signal"
To build on Laurent's answer, here is an example. Top frame shows an example transient signal: a damped sine wave. As the signal decays very quickly, the first 0.1s of the signal is the most ...
4
votes
Accepted
Confusion regarding pdf of circularly symmetric complex gaussian rv
Let me try to establish the relation between the univariate PDF for a real Gaussian and the univariate PDF for a complex proper (i.e. circular symmetric) Gaussian.
You know that
$p_x(x)=\frac{1}{\...
4
votes
Computer exercises and solutions in signal processing
"Digital Signal Processing: A Computer-Based Approach" by Sanjit Mitra is what you need I guess, especially the exercises at the end of each chapter. There is a booklet on the Internet again by Mitra,...
4
votes
Accepted
Unable to understand the derivation of the update equation for LMS
so with an LMS filter, we have a time-variant $N$-tap FIR filter:
$$ y[n] = \sum\limits_{k=0}^{N-1} h_n[k] \, x[n-k] $$
$x[n]$ is the input signal, $y[n]$ is the FIR output, and $h_n[k]$ are the FIR ...
4
votes
Finding the time domain equation corresponding to a given DFT
(1) I'm not sure my intuition is right
Almost, but you're missing the fact that if you're sampling at 4000kHz, the signals alias; after sampling at 4000kHz, a sine wave at 1500Hz is ...
4
votes
Accepted
Finding the time domain equation corresponding to a given DFT
Here's the pure math:
$$ x[n] = \frac{1}{N} \sum_{k=0}^{N-1} X[k] e^{i \frac{2\pi}{N} nk } $$
$$ x[n] = \frac{1}{8} \left[ (5.657 + i 5.657) e^{i \frac{2\pi}{8} n 3 } +(5.657 - i 5.657) e^{i \frac{2\...
4
votes
Recommendation for courses / studies on digital signal processing
I've kind of grouped your subjects into larger overall subjects.
Note that there's a lot of overlap here, with the possible exception of actually making it work in a microprocessor (except -- in my ...
3
votes
Accepted
Matlab: Correct method of generating Rayleigh fading channel
Question 1:
Yes, it is a multipath Rayleigh Channel with $ 1000 $ taps. You need to know that when you generate complex coefficients, and the real part and the imaginary part of this coefficients ...
3
votes
Accepted
properties of System
For question 1: apply the definition of time invariant: find the output as normal; find the output with the same input but delayed by $T$
$$
y_1(t) = \frac{dx(t)}{dt}\\
y_2(t) = \frac{dx(t-T)}{dt}\\
$...
3
votes
What are your tips for DSP self-study?
DSP would be a quite broad subject and where to start would depend on your ability to pick up information, learn, understand and implement it. A basic course for supplying us with the basics to ...
3
votes
Accepted
What is the Technique to Find Variance of Estimation Error
I don't understand the subscript $n$ notation, however, in the least squares problem that is given by:
\begin{equation}
{\bf{y}}={\bf{H}}{\theta}+\bf{n},
\end{equation}
where ${\bf{n}}\sim\mathcal{N}(...
3
votes
What Resources Are Recommended for an Introduction to Signal Processing (DSP)?
I would add to the list the book "Digital Filters", by Richard Hamming. A short classic, rather than a heavy tome.
Community wiki
3
votes
Accepted
Linear Prediction of AR Process
From the definition of the process you know that
$$x_{n+1}=x_n-0.2x_{n-1}+w_{n+1}\tag{1}$$
Since $w_n$ is white you can't predict it, so the best linear predictor for the given process is the filter
...
3
votes
LTI systems inverse of each other
You may also look at the transfer function ($s$-domain) of the two systems:
$$H_1(s)=\frac{1}{2}e^{-s}$$
$$H_2(s)=2e^{s}$$
And the two systems in series will result in $$H_1(s)H_2(s)=1$$
which means ...
3
votes
Accepted
Beginner level confusion regarding symbols used in modulation
An M-QAM modulation contains M different constellation symbols. Hence, in one symbol you can encode $\mu=\log_2(M)$ bits. (i.e. 6 bits for 64-QAM for example). Now, the bits are equally distributed ...
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