20 votes
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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 ...
Richard Lyons's user avatar
9 votes
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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 ...
MBaz's user avatar
  • 14.9k
7 votes
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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 ...
AlexTP's user avatar
  • 6,080
6 votes
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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. ...
Matt L.'s user avatar
  • 88.9k
6 votes
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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 ...
mark leeds's user avatar
  • 1,107
6 votes
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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 ...
Peter K.'s user avatar
  • 25.2k
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 ...
Fat32's user avatar
  • 28k
5 votes
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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 ...
Matt L.'s user avatar
  • 88.9k
5 votes
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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 ...
A_A's user avatar
  • 10.6k
5 votes
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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 ...
AlexTP's user avatar
  • 6,080
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. ...
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 ...
SergV's user avatar
  • 920
4 votes
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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 ...
JohnMarvin's user avatar
4 votes
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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 ...
MBaz's user avatar
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4 votes
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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 ...
Matt L.'s user avatar
  • 88.9k
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 ...
Daniel K's user avatar
  • 309
4 votes
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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}{\...
Maximilian Matthé's user avatar
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,...
MimSaad's user avatar
  • 1,976
4 votes
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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 ...
robert bristow-johnson's user avatar
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 ...
TimWescott's user avatar
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4 votes
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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\...
Cedron Dawg's user avatar
  • 7,520
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 ...
TimWescott's user avatar
  • 11.8k
3 votes
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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 ...
JohnMarvin's user avatar
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}\\ $...
Peter K.'s user avatar
  • 25.2k
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 ...
Disciple's user avatar
3 votes
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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}(...
Nir Regev's user avatar
  • 620
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.
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 ...
Matt L.'s user avatar
  • 88.9k
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 ...
msm's user avatar
  • 4,225
3 votes
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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 ...
Maximilian Matthé's user avatar

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