Session: SPCOM-L13
Time: 3:30 - 5:30, Friday, May 11, 2001
Location: Room 251 A
Title: Channel/Source Coding and Quantization 2
Chair: Pierre Duhamel

3:30, SPCOM-L13.1
DECODING OF HALF-RATE WAVELET CODES; GOLAY CODE AND MORE
F. FEKRI, S. MCLAUGHLIN, R. MERSEREAU, R. SCHAFER
The primary goal of this paper is to give examples of the recently developed (finite-field) wavelet coding method by studying the encoder and decoder for some half-rate codes. We propose a decoding methodology based on estimating the polyphase components of the channel error pattern. To demonstrate the striking computational savings of the wavelet coding method over alternatives, we show that bounded-distance decoding of the (24,12,8) Golay code requires only weight computations (or at the worst case, it needs a cyclic lookup table of table size 12). The simplicity and computational savings that finite field wavelets offer for the encoding and decoding of wavelet block codes indicate their powerful capacities for error control coding applications. More information can be found at http://www.ee.gatech.edu/users/fekri.

3:50, SPCOM-L13.2
SOURCE AND CHANNEL CODING FOR REMOTE SPEECH RECOGNITION OVER ERROR-PRONE CHANNELS
A. ALWAN, A. BERNARD
This paper presents source and channel coding techniques for remote recognition systems. As a case study, line spectral pairs (LSP) extracted from the 6th order all-pole perceptual linear prediction (PLP) spectrum are transmitted and speech recognition features are then obtained. The LSPs, quantized using first-order predictive vector quantization (VQ) at 300 bps, provide recognition accuracy comparable to that of the baseline system with no quantization. A new soft decision channel decoding scheme appropriate for remote recognition is presented. The scheme outperforms commonly-used hard decision decoding in terms of error correction and error detection. The source and channel coding system operates at 500 bps and provides good recognition performance over a wide range of channel conditions.

4:10, SPCOM-L13.3
RATE DISTORTION OPTIMAL SIGNAL COMPRESSION BY SECOND ORDER POLYNOMIAL APPROXIMATION
R. NYGAARD, A. KATSAGGELOS
In this paper we present a time domain signal compression algorithm based on the coding of line segments which are used to approximate the signal. These segments are fit in a way that is optimal in the rate distortion sense. The approach is applicable to many types of signal, but in this paper we focus on the compression of ElectroCardioGram (ECG) signals. As opposed to traditional time-domain algorithms, where heuristics are used to extract representative signal samples from the original signal, an optimization algorithm is formulated in [1,2,3] for sample selection using graph theory, with linear interpolation applied to the reconstruction of the signal. In this paper the algorithm in [1,2,3] is generalized by using second order polynomial interpolation for the reconstruction of the signal from the extracted signal samples. The polynomials are fitted in a way that guarantees minimum reconstruction error given an upper bound on the number of bits. The method achieves good performance compared both to the case where linear interpolation is used in reconstruction of the signal and to other state-of-the-art ECG coders.

4:30, SPCOM-L13.4
A STABLE ADAPTIVE STRUCTURE FOR DELTA MODULATION WITH IMPROVED PERFORMANCE
M. ALDAJANI, A. SAYED
In this paper, we propose and study an adaptive delta modulator that has improved SNR performance and better robustness in tracking highly varying signals. The step-size adaptation used in this modulator is based on information about the absolute value of the input to the quantizer. The modulator is shown to be free of zero-input limit cycles and is BIBO stable.

4:50, SPCOM-L13.5
THE MISMATCH-NOISE PSD FROM A TREE-STRUCTURED DAC IN A SECOND-ORDER DELTA-SIGMA MODULATOR WITH A MIDSCALE INPUT
J. WELZ, I. GALTON
Mismatch-shaping DACs have become widely used in high-performance delta-sigma data converters in recent years. Nevertheless, no theoretical results have been published to date that quantify their performance, so designers have been forced to rely on simulation-based analyses. This paper presents the first theoretical performance analysis of a mismatch-shaping DAC. Specifically, the PSD of the mismatch noise introduced by a first-order tree-structured DAC within a second-order ADC delta-sigma modulator with a midscale constant input signal is derived. This particular mismatch-shaping DAC and delta-sigma modulator configuration was chosen for analysis because it has been demonstrated experimentally to achieve state-of-the-art ADC performance. The choice of a constant midscale input was made because simulation and experimental results suggest that it yields the worst-case performance.

5:10, SPCOM-L13.6
A LAGRANGIAN FORMULATION OF HIGH RATE QUANTIZATION
J. SHIH, A. AIYER, R. GRAY
The asymptotic optimal performance of variable-rate vector quantizers of fixed dimension and large rate was first developed in a rigorous fashion by Paul Zador. Subsequent design algorithms for such compression codes used a Lagrangian formulation in order to generalize Lloyd's classic quantizer optimization algorithm to variable rate codes. This formulation has been subsequently adopted in a variety of practical systems including rate-optimized streaming video. We describe a Lagrangian formulation of Zador's variable-rate quantization results and apply it to estimate Zador's constant using the generalized Lloyd algorithm.