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Python number crunching faster? Part I

Christopher FeltonChristopher Felton ● September 17, 2011●4 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.


More free Ebooks

Sami AldalahmehSami Aldalahmeh ● September 13, 2011●2 comments

A handy roundup of free engineering ebooks and journals, with a focus on where DSPRelated readers might actually find useful material. The post points to InTechOpen, highlights several MATLAB books, and even calls out a numerical methods title plus some sensor fusion content. If you like browsing for practical references, this is a good place to start.


Bank-switched Farrow resampler

Markus NentwigMarkus Nentwig ● August 13, 2011●2 comments

Markus Nentwig proposes a bank-switched variant of the Farrow resampler that breaks each impulse-response segment into multiple sub-segments, enabling accurate interpolation with lower-order polynomials and fewer multiplications per output. This trades increased total coefficient storage for computational savings. The post explains the concept, connects it to polyphase FIR interpolation, and provides Matlab/Octave and C example code for practical evaluation.


Impulse Response Approximation

Christopher FeltonChristopher Felton ● August 5, 2011●3 comments

A stepped-triangular impulse approximation represents an FIR low-pass using a cascade of recursive running-sum filters, offering big savings in computation. Christopher Felton outlines the quantization step that maps a true impulse into three stepped-triangular types and shows how the approximation is built from recursive running-sum and sparse-sum blocks. Inspect the frequency tradeoffs and decide if the efficiency gain is worth the approximation error.


Orfanidis Textbooks are Available Online

Rick LyonsRick Lyons ● July 12, 2011

Two classic signal processing textbooks by Sophocles J. Orfanidis are now available for download from his Rutgers webpages. The first, Introduction to Signal Processing, includes errata and a homework solutions manual. The second, Optimum Signal Processing, includes a solutions manual plus MATLAB, C and Fortran code. Note that Prof. Orfanidis retains copyright on both books, All Rights Reserved.


ICASSP 2011 conference lectures online (for free)

Sami AldalahmehSami Aldalahmeh ● July 5, 2011

For the first time, the oral sessions of ICASSP 2011 were recorded and posted online for free, giving engineers worldwide easy access to the conference. The talks span speech and communication signal processing, plus eclectic topics like bio-inspired methods, where Prof. Sayed uses a distributed LMS model to reproduce group predator and prey behavior. Expect some theoretical material, but many presentations are practical and inspiring for DSP practitioners.


Do Multirate Systems Have Transfer Functions?

Rick LyonsRick Lyons ● May 30, 2011●3 comments

Multirate systems can fool you into thinking standard z-domain analysis always applies. Rick Lyons shows why CIC decimation and Hogenauer implementations do not have a single z-domain transfer function from the input to the downsampled output, because downsampling breaks the one-to-one frequency mapping of LTI systems. Use the cascaded-subfilter H(z) up to the decimation point, then explicitly account for aliasing when predicting the decimated spectrum.


FREE Peer-reviewed IEEE signal processing courses

Sami AldalahmehSami Aldalahmeh ● April 26, 2011●1 comment

IEEE Signal Processing Society is offering a small set of free, peer-reviewed courses covering topics like wavelets, speech analysis, and statistical detection. The post points to these endorsed materials as a useful way to browse vetted DSP learning resources without paying for formal coursework.


A Fixed-Point Introduction by Example

Christopher FeltonChristopher Felton ● April 25, 2011●22 comments

Christopher Felton walks through binary fixed-point representation with clear examples and a simple W=(wl,iwl,fwl) notation. He argues for designing to range and resolution rather than bit counts, then shows how multiplication and addition affect bit growth and alignment. These concrete examples make it easy to see why rounding, resizing, and radix-point bookkeeping are essential in DSP implementations.


DSP Algorithm Implementation: A Comprehensive Approach

Sami AldalahmehSami Aldalahmeh ● April 13, 2011●6 comments

This post lays out a practical pathway for taking DSP algorithms from high level simulation to production hardware, comparing GPP, DSP, FPGA and ASIC platforms. It presents a stepwise methodology starting with nested loop programs, then exposing parallelism with data flow graphs, using SystemC transaction level modeling to bridge to Verilog or VHDL, and explains why that flow speeds design and simulation.


TI DSP Predictions

Jeff BrowerJeff Brower ● September 19, 2007●5 comments

Jeff Brower lays out two bold predictions for Texas Instruments that could reshape the DSP developer ecosystem. He argues TI will offer a supported real-time Linux on their C6x DSPs now that legal obstacles have eased, and that TI may acquire an FPGA company to own the board space around its chips. Read to weigh the technical and strategic impact.


'z' as in 'Zorro': Frequency Masking FIR

Markus NentwigMarkus Nentwig ● October 2, 2012

Markus Nentwig shows an efficient way to build steep wideband FIR filters by combining upsampled and complementary stages, then masking their spectra. He provides a Matlab and Octave design program that uses a generic least-squares optimizer to place coefficients, letting you explore filter sizes and oversampling while cutting computational cost significantly compared to a conventional symmetric FIR.


Fitting Filters to Measured Amplitude Response Data Using invfreqz in Matlab

Julius Orion Smith IIIJulius Orion Smith III ● October 11, 2010●2 comments

This post is a redirect notice for a code snippet now hosted elsewhere. If you were looking for the invfreqz example on fitting a filter to measured amplitude response data, this page simply points you to the new location and asks you to update your bookmark.


Why is Fourier transform broken

Sami AldalahmehSami Aldalahmeh ● October 4, 2011●2 comments

Many engineers know the Gibbs phenomenon without grasping its root cause. This post shows that the problem comes from using the incomplete metric space of continuous functions, C[a,b], for Fourier series, and explains how switching to Lp spaces resolves convergence in the mean but allows functions to differ on sets of measure zero. It also reminds readers that Fourier analysis gives no time localization, so be mindful of its limits.


Constrained Integer Behavior

Christopher FeltonChristopher Felton ● May 26, 2014

Overflow and underflow are not always bugs, they can be useful in DSP when fixed-width integers wrap during processing. Christopher Felton demonstrates with moving-average (recursive-windowed-averager) and CIC filter examples how 2's complement wraparound in MyHDL's modbv cancels between an integrator and a comb via pole-zero cancellation. He also covers fixed-point resizing choices, saturation versus wrap, and how rounding error can accumulate.


Free DSP Books on the Internet - Part Deux

Rick LyonsRick Lyons ● December 4, 2008●1 comment

Rick Lyons updates his curated list of freely downloadable DSP textbooks, adding titles across communications, implementation, spectral analysis, audio restoration, mathematics and music theory. The post highlights readable introductions like Prandoni and Vetterli's Signal Processing for Communications and Vetterli and Kovacevic's Wavelets and Subband Coding, while reminding readers that these copyrighted books are free only for individual download and not for redistribution.


Evaluate Noise Performance of Discrete-Time Differentiators

Neil RobertsonNeil Robertson ● March 28, 2022●8 comments

Differentiators can be wildly different at rejecting noise, even when they share the same usable bandwidth. Neil Robertson introduces the Differentiator Noise Power Ratio, a practical Gaussian-noise metric and a compact formula that uses the filter coefficients to quantify output noise and SNR loss. The post also gives MATLAB guidance for designing and comparing FIR differentiators so you can pick or build filters with much better noise performance.


The Sampling Theorem - An Intuitive Approach

Stephane BoucherStephane Boucher ● January 26, 2015●1 comment

Scott Kurtz from DSPSoundWare.com has put together a video presentation that aims to give DSP engineers an intuitive grasp of the Sampling Theorem. The short, approachable video focuses on conceptual understanding of sampling and aliasing rather than mathematical formality. Watch the presentation on DSPRelated and share your reactions in the post comments to join the discussion.


SDR: Does it makes sense? Part 1/2

Praveen RaghavanPraveen Raghavan ● January 7, 2008●1 comment

Software-defined radio is still a question mark for deployment, but Praveen Raghavan argues the economics are starting to line up. He points to rising process costs, especially as devices move toward FinFET technologies, and notes that consumer products usually need only a couple of wireless standards at once. That makes multi-standard silicon look more practical than ever, and he tees up the next question, what should actually be software definable?


New Discussion Group: DSP & FPGA

Stephane BoucherStephane Boucher ● September 11, 2007●8 comments

Stephane Boucher has launched a new discussion group for engineers implementing DSP functions on FPGAs. It is meant to become a focused place for sharing ideas, but he notes it may take a few weeks before enough members join for the discussion to really get going. If FPGA-based DSP is your thing, this is an open invitation to get involved early.