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Compressive Sensing - Recovery of Sparse Signals (Part 1)

Mamoon Mamoon November 28, 2015

The amount of data that is generated has been increasing at a substantial rate since the beginning of the digital revolution. The constraints on the sampling and reconstruction of digital signals are derived from the well-known Nyquist-Shannon sampling theorem...


Analytic Signal

Mehdi Mehdi November 26, 20155 comments

In communication theory and modulation theory we always deal with two phases: In-phase (I) and Quadrature-phase (Q). The question that I will discuss in this blog is that why we use two phases and not more.


Multilayer Perceptrons and Event Classification with data from CODEC using Scilab and Weka

David NorwoodDavid Norwood November 25, 2015

For my first blog, I thought I would introduce the reader to Scilab [1] and Weka [2]. In order to illustrate how they work, I will put together a script in Scilab that will sample using the microphone and CODEC on your PC and save the waveform as a CSV file.


Maximum Likelihood Estimation

Mehdi Mehdi November 24, 2015

Any observation has some degree of noise content that makes our observations uncertain. When we try to make conclusions based on noisy observations, we have to separate the dynamics of a signal from noise.


Approximating the area of a chirp by fitting a polynomial

Alexandre de SiqueiraAlexandre de Siqueira November 15, 20158 comments

Once in a while we need to estimate the area of a dataset in which we are interested. This area could give us, for example, force (mass vs acceleration) or electric power (electric current vs charge).


Deconvolution by least squares (Using the power of linear algebra in signal processing).

Agustin BonelliAgustin Bonelli November 12, 20152 comments

When we deal with our normal discrete signal processing operations, like FIR/IIR filtering, convolution, filter design, etc. we normally think of the signals as a constant stream of numbers that we put in a sequence


The Most Interesting FIR Filter Equation in the World: Why FIR Filters Can Be Linear Phase

Rick LyonsRick Lyons August 18, 201517 comments

Rick Lyons pulls back the curtain on a little-known coefficient constraint that makes complex-coefficient FIR filters exhibit linear phase. Rather than simple symmetry of real coefficients, the key is a conjugate-reflection relation involving the filter phase at DC, which collapses to ordinary symmetry for real taps. The post includes derivations, intuition using the inverse DTFT, and a Matlab example to verify the result.


Phase and Amplitude Calculation for a Pure Real Tone in a DFT: Method 1

Cedron DawgCedron Dawg May 21, 20151 comment

Cedron Dawg shows how to get exact amplitude and phase for a real sinusoid whose frequency does not land on an integer DFT bin. The method treats a small neighborhood of DFT bins as a complex vector, builds two basis vectors from the cosine and sine transforms, and solves a 2x2 system using conjugate dot products to recover real coefficients that give amplitude and phase. A C++ example and sample output verify the formulas.


Exact Frequency Formula for a Pure Real Tone in a DFT

Cedron DawgCedron Dawg April 20, 20152 comments

Cedron Dawg derives an exact closed form formula to recover the frequency of a pure real sinusoid from three DFT bins, challenging the usual teaching that it is impossible. The derivation solves for cos(alpha) in a bilinear form and gives a computationally efficient implementation (eq.19), with practical notes on implicit Hann-like weighting and choosing the peak bin for robustness.


DFT Bin Value Formulas for Pure Real Tones

Cedron DawgCedron Dawg April 17, 20151 comment

Cedron Dawg derives a closed-form expression for the DFT bin values produced by a pure real sinusoid, then uses that formula to explain well known DFT behaviors. The post walks through the algebra from Euler identities to a compact computational form, highlights the integer versus non-integer frequency cases, and verifies the result with C code and printed numeric output.


Deconvolution by least squares (Using the power of linear algebra in signal processing).

Agustin BonelliAgustin Bonelli November 12, 20152 comments

When we deal with our normal discrete signal processing operations, like FIR/IIR filtering, convolution, filter design, etc. we normally think of the signals as a constant stream of numbers that we put in a sequence


Multilayer Perceptrons and Event Classification with data from CODEC using Scilab and Weka

David NorwoodDavid Norwood November 25, 2015

For my first blog, I thought I would introduce the reader to Scilab [1] and Weka [2]. In order to illustrate how they work, I will put together a script in Scilab that will sample using the microphone and CODEC on your PC and save the waveform as a CSV file.


Feedback Controllers - Making Hardware with Firmware. Part 8. Control Loop Test-bed

Steve MaslenSteve Maslen March 21, 2018

Built around modest FPGA hardware, this post presents a practical test-bed for evaluating high-speed, low-latency feedback controllers. It covers ADC/DAC specifications, basic and arbitrary test signals, and an IFFT-based generator that can produce thousands of simultaneous tones for rapid Bode, phase, and latency measurements. The article also compares two IFFT strategies, explains turbo sampling, and shows open- and closed-loop test configurations.


Hidden Linear Algebra in DSP

Sami AldalahmehSami Aldalahmeh June 17, 20105 comments

Linear algebra is hiding in plain sight inside many DSP techniques, not just abstract theory. By treating linear systems as matrix operators y = A x you reveal Toeplitz structure in LTI systems, connect to covariance matrices, and gain geometric intuition via eigenvalues and eigenvectors. This matrix viewpoint complements convolution-based thinking and offers practical tools for filter and channel analysis.


Implementing Impractical Digital Filters

Rick LyonsRick Lyons July 19, 20162 comments

Some published IIR block diagrams are impossible to implement because they contain delay-less feedback paths, and Rick Lyons shows how simple algebra fixes that. He works through two concrete examples—a bandpass built from a FIR notch and a narrowband notch using a feedback loop—and derives equivalent, implementable second-order IIR transfer functions. The post emphasizes spotting problematic loops and replacing them with practical block diagrams.


A brief look at multipath radio channels

Markus NentwigMarkus Nentwig October 31, 20078 comments

Markus Nentwig walks through a hands-on RF experiment that makes multipath and fading visible using a network analyzer and simple dipole antennas. He shows how reflections produce frequency-domain notches when path differences equal half wavelengths, and how doubling distance increases free-space path loss by roughly 6 dB. The post explains why narrowband signals often see flat fading while wideband links become frequency-selective, motivating OFDM and multi-tap channel models.


Python number crunching faster? Part I

Christopher FeltonChristopher Felton September 17, 20114 comments

Christopher Felton walks through simple benchmarks comparing raw Python, numpy, and PyPy for numeric workloads, and shares what surprised him about performance. He shows that idiomatic Python optimizations such as list comprehensions and built-ins plus the PyPy JIT can sometimes beat a numpy approach for small tests, and explains why native PyPy numpy progress matters for scientific users.


Instant CIC

Markus NentwigMarkus Nentwig May 8, 20124 comments

Modeling CIC decimators in floating point is simpler than you might think, Markus Nentwig shows, if you treat the filter as a finite FIR by sampling its impulse response. The post compares a naive float time-domain implementation, an FFT-based frequency-domain approach, and the recommended method of computing the impulse response and using an off-the-shelf FIR filter, with code and plots.


Simulink-Simulation of SSB demodulation

Josef HoffmannJosef Hoffmann June 13, 20211 comment

This post walks through Simulink models that implement SSB demodulation and modulation, using Richard Lyons' phasing method as a foundation. It shows practical models for simple carrier multiplication and for the phasing method with cosine and -sin paths plus Hilbert filtering, and it highlights sampling, decimation, filter choices, and delay alignment to make the techniques work in simulation.


Modelling a Noisy Communication Signal in MATLAB for the Analog to Digital Conversion Process

Parth VakilParth Vakil October 30, 200713 comments

Practical signal modeling treats receiver noise as a fixed power source, not something tied to the transmitted waveform. Parth demonstrates why using MATLAB's awgn(sig,SNR,'measured') can misrepresent an analog front end and provides a short function that scales your signal so the added AWGN produces the desired receiver noise variance. This prepares realistic inputs for upcoming ADC simulations.