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10 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
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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
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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
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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
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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
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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
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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
  • 6,605
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
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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
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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
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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
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3 votes
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Lloyd Max Quantization and Clustering : Part 1

Different. Lloyd-Max is a special type of scalar quantizer design which is optimized (in terms of MSE) to source pdf. Hence the quantizer is generally non-uniform. Lloyd's algorithm (and the more ...
msm's user avatar
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3 votes
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Help in understanding from book expression of variance of an estimator : PRBS vs real valued input

A PRN sequence is a Pseudo-Random Noise sequence, often generated by using an Linear Feedback Shift Register (LFSR) with the feedback taps done by using a primitive irreducible polynomial in GF{2}, ...
Dan Boschen's user avatar
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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
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
  • 4,295
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

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
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Why do digital modulation when the signal entering the channel is analog?

I think part of your confusion is actually between discrete-time and digital signals. Some textbooks do not clearly distinguish between them. All signals that exist in the physical world are real. So,...
MBaz's user avatar
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3 votes
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Is there a rule of thumb for selecting the variance of the input?

You can define your model as $$\hat{x}[n] = \frac{1}{||\mathbf{x}_{n}||}\mathbf{A}^T\mathbf{x}_{n} +\frac{1}{\sqrt{\mathsf{SNR}}} w[n]$$ In this way, you define a range for $\mathsf{SNR}$ in dB (e.g. ...
msm's user avatar
  • 4,295
3 votes

Digital Signal Processing DSP Overview study documents

There are some books (some free) that cover most DSP topics with a claim of easy approach, probably they cover the most important fundamentals: 1- Understanding Digital Signal Processing_R.Lyons 2- ...
Fat32's user avatar
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3 votes
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How is the number of amplitude levels decided in modulation techniques?

There are $M$ symbols in an $M$-QAM constellation where two carriers I and Q are amplitude modulated, and there are $N=\sqrt M$ amplitudes for each I and Q carriers (for $M\ge4$). In $16$-QAM there ...
msm's user avatar
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3 votes
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Doubts and some confusion on variance for complex rv

If the imaginary and real components each has a variance of 0.5, then what would be the total variance? The variance of the complex RV in that case would be $0.5 + 0.5 = 1$. If on the other hand, ...
MBaz's user avatar
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3 votes
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Performance of adaptive filters

Q1: Dependence of estimation performance on number of data points Since LMS and RLS are adaptive filters, their estimation performance improves as the number of their iterations increase. Hence more ...
Fat32's user avatar
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3 votes
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which filters are these?

When only a rough pole-zero plot (without exact locations of them) is given for a filter, then the only procedure that remains to evaluate the frequency response magnitude will rely on the geometrical ...
Fat32's user avatar
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3 votes
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Find the minimum sampling rate

I would not use a formula for the understanding of the so called bandpass sampling (or undersampling) operation. Instead try to analyse the situation by yourself considering the signal spectrum, ...
Fat32's user avatar
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