2004 Comp.dsp Conference Presentations

Below is a synopsis of the Presentations made by comp.dsp Usenet members at the 2004 Comp.dsp Conference held at Danville Signal. The PowerPoint and PDF files that we have on those presentations are available here as well. We hope you find the information useful.

Signal Processing for OFDM Systems by Eric Jacobsen (Intel Corp.)

This presentation explores the requirements imposed by typical multi-carrier modulation, like that used in Orthogonal Frequency Division Multiplexed (OFDM) systems, on the signal processing in a typical transceiver. The differences with respect to single-carrier systems are investigated, and the flexibility and opportunities created by multi-carrier modulation are also discussed.

Powerpoint Presentation PowerPoint File
Powerpoint Presentation (Print Ready) Print


Interpolated and Frequency Sampling FIR Filters by Rick Lyons (Besser Associates)

This presentation introduces two computationally-efficient, linear-phase, finite impulse response (FIR) filter structures. The Interpolated FIR structure is nonrecursive, while the Frequency Sampling structure is recursive. (It's true, you can build linear-phase recursive filters!) While these FIR filter designs have not received their deserved attention in the literature, they definitely belong in your filter design toolbag. Design examples will be discussed showing how these filters can be more efficient than standard Parks-McClellan-designed FIR filters.

Presentation - IFIRs PDF

Presentation - FSFs PDF


One-Bit Delta Sigma D/A Conversion - Theory & Implementation by Randy Yates (Sony Ericsson Mobile Communications)

This presentation covers oversampling and noise-shaping in one-bit D/A conversion. Classic architectures for interpolation and delta sigma modulation are presented, along with the use of dither in a delta sigma converters. A real-world implementation of a one-bit delta sigma D/A converter on DSP hardware within a Sony Ericsson handset will be discussed including psycho-acoustic noise-shaping within the modulator. Anomalies encountered in the prototype development of the converter are revealed, their solutions presented, and their sources theorized.

Presentation (part 1) PDF
Presentation (part 2) PDF


Q-Math Refresher - Understanding Q Math Basics by Shawn Steenhagen (Applied Signal Processing)

This presentation answers questions such as "where does that extra sign bit come from?" and "how do I know how many bits to shift when doing mixed Q-math arithmetic?". Examples are presented to help show the math behind the mixed Q-math rules.It provides some insights into making fixed point software development easier and more intuitive, describing a base set of library functions and #define macros that make C statements like: q15 q15_my_var = Q15(0.33); possible.

Presentation PDF
Powerpoint Presentation PowerPoint File


Survey of Forward Error Correction and Its Use in Modern Wireless Applications by Eric Jacobsen (Intel Corp.)

The ability to correct channel induced errors is critical to the reliability of most wireless communications systems, and recent advances in both theory and practice has led to a continually moving state-of-the-art in Forward Error Correction for these applications. A brief history of forward error correction is provided to set the context of the discussion. Fundamental theories are covered and how they apply to different approaches in detecting and correcting transmission errors is explored, as well as how channel impairments and system requirements may favor one type of system over another.

Powerpoint Presentation PowerPoint File
Powerpoint Presentation (Print Ready) Print


The Swiss Army Knife of Digital Networks by Rick Lyons (Besser Associates)

This presentation reviews the long list of useful signal processing functions that can be performed with a single digital network. That simple network is an Nth-order comb structure in cascade with a general 2nd-order recursive structure. The process performed by this network is controlled by the network's coefficients. While you may be familiar with 8-10 signal processing functions performed by this digital network (because it appears, in various forms, inside many DSP applications), its list of possible signal processing functions is 23 (and growing).

 


Signal to Noise and Numeric Range issues for Direct Form I & II IIR Filters on Modern Analog Devices and TI Digital Signal Processors by Mark Allie (Mark Allie Consulting LLC)

The implementation of IIR filters using fixed point math has been studied in the past by Jon Dattorro, Rhonda Wilson, and others. There are many fine papers on this topic that have been published in the Journal of the Audio Engineering Society. In most cases the comparisons have been made between 16 bit and 24 bit fixed point Digital Signal Processors (DSPs). This was due to the low cost availability of the TI TMS320, AD 21xx and Motorola 56K series of DSPs. The greatest demands came from the professional audio community which choose the Motorola processor. The 56K family offered 24 bit fixed point math where most competing DSPs from TI and Analog Devices utilized 16 bit operands. The general conclusion is that the Direct Form I IIR is superior for fixed point implementations.

There is evidence that 24 bits is not always enough for the recursive data path. This in part has shifted market share to Analog Devices and TI. Both companies offer low cost 32 bit DSPs. This presentation compares IIR filters using Direct Form I or Direct Form II topologies with either a fixed point or floating point implementation. The Analog Devices and TI floating point DSP’s can be configured as native 32 bit fixed point processors and 32 bit IEEE floating point processors. The AD SHARC processor also has an additional 40 bit extended precision IEEE floating point capability. The low cost and extended abilities of these processors suggests there could be a different optimum set of accurate, quiet, efficient filter structures.

Powerpoint Presentation PowerPoint File


The Art of Debugging: What are Commercial Emulators and How are They Used by Dr. Mike Rosing (University of Wisconsin - Madison)

This presentation covers the art of using JTAG emulators, oscilloscopes, and logic analyzers in debugging digital systems. An "art form" is an activity that requires a long time to acquire proficiency, to the point where many operations are instinctive. This presentation will discuss the practical aspects of using the tools of DSP engineering to improve your skills (and instincts) in finding problems in both hardware and software, and why understanding the system as a whole is critical.

Powerpoint Presentation PowerPoint File


Active Noise Control - Application Architectures and Potentials by Shawn Steenhagen (Applied Signal Processing)

This presentation describes the different approaches and architectures for actively controlling random or periodic noise and vibration. The difference between signal identification and system identification techniques are discussed. An evaluation process is presented which helps determine the viability of Active Control as a solution to a noise problem in both an economical and a technical sense.

Presentation PDF
Powerpoint Presentation PowerPoint File


Fast Convolution (FFT) Filtering: From Basics to Filter Banks: Part I by Mark Borgerding (Xetron)

Fast Convolution filtering is a powerful technique with which every DSP engineer should be familiar. All but the shortest FIR filters are more efficiently implemented with FFTs rather than direct forms. The longer the filter; the greater the advantage. This presentation begins with the primary forms of fast convolution: overlap-add and overlap-save. These concepts lay the foundation for building parallel filters that perform mixing, filtering, and downsampling in the frequency domain.

Presentation PDF
Slides (1) PDF
Slides (2) PDF


Fast Convolution (FFT) Filtering: From Basics to Filter Banks: Part II by Mark Borgerding (Xetron)

Fast Convolution filtering is a powerful technique with which every DSP engineer should be familiar. All but the shortest FIR filters are more efficiently implemented with FFTs rather than direct forms. The longer the filter; the greater the advantage. This presentation begins with the primary forms of fast convolution: overlap-add and overlap-save. These concepts lay the foundation for building parallel filters that perform mixing, filtering, and downsampling in the frequency domain.


A Survey of Noise Reduction Techniques by Maurice Givens

This presentation provides a tutorial overview of the most widely used noise-reduction techniques. The presentation covers the least mean-square (LMS) method using the adaptive noise canceller (ANC) and adaptive line enhancer (ALE), and coefficient shrinkage with decomposition by FFT and wavelet transformations. In addition, spectral (sub-band) subtraction using FFT, wavelet, frequency sampling, and sub-band decomposition will be discussed. The presentation will show examples, including speech enhancement using several of the above noise-reduction techniques.

Powerpoint Presentation PowerPoint File


USB Fundamentals and Considerations for Digital Signal Processing by Greg Burk (J. Gordon Electronic Design)

Powerpoint Presentation PowerPoint File


Frequency Estimation Techniques by Peter Kootsookos (United Technologies Research Center)

This presentation will examine the frequency estimation and associated problems such as estimation performance, instantaneous frequency estimation, and use (or not) of the analytic signal. Particular techniques to be examined are maximum likelihood, Kay's estimator (and variants) and DFT interpolation.

Powerpoint Presentation PowerPoint File

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