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1995 IEEE Workshop on Speech Coding for Telecommunications, SCFT 1995

作者： Das, Amitava Gersho, Allen Signal Compression Laboratory Department of Electrical and Computer Engineering University of California Santa BarbaraCA93106 United States

We present a low bit rate multimode speech coding algorithm which applies a suitable spectral model and a custom quantization scheme to each frame according to the selected mode. For unvoiced speech, the spectrum is characterized by a fixed-dimension spectral shape vector (SSV), while the spectra of 13 different voiced modes are specified by the pitch and a variable dimension SSV. A variable dimension vector quantization technique offers efficient spectral quantization and improves overall speech quality. A variable rate (1.7 kb/s avg. rate;2.4 kb/s max. rate) implementation of the multimode algorithm yields speech quality comparable to the 4.8 kb/s FS 1016 CELP coder and the 4.15 kb/s Inmarsat-MIMBE coder. © 1995 Proceedings - 1995 IEEE Workshop on Speech Coding for Telecommunications: Speech Coding for Interoperable Global Communications, SCFT 1995. All rights reserved.

关键词： Speech coding

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MORPHOLOGICAL REPRESENTATION OF ORDER-STATISTICS FILTERS

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IEEE TRANSACTIONS ON IMAGE PROCESSING 1995年第6期4卷 838-845页

作者： CHARIFCHEFCHAOUNI, M SCHONFELD, D Signal and Image Research Laboratory Department of Electrical Engineering and Computer Science University of Illinois Chicago IL USA

In this correspondence, we propose a comprehensive theory for the morphological bounds on order-statistics filters (and their repeated iterations). Conditions are derived for morphological openings and closings to serve as bounds (lower and upper, respectively) on order-statistics filters (and their repeated iterations). Under various assumptions, morphological open-closings and close-openings are also shown to serve as (tighter) bounds (lower and upper, respectively) on iterations of order-statistics filters. Simulations of the application of the results presented to image restoration are finally provided.

关键词： Image coding Fractals signal processing algorithms Speech processing Decoding Pixel Heuristic algorithms signal processing Application software Digital images

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ON THE CONVERGENCE AND ROOTS OF ORDER-STATISTICS FILTERS

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IEEE TRANSACTIONS ON signal PROCESSING 1995年第8期43卷 1990-1993页

作者： CHARIFCHEFCHAOUNI, M SCHONFELD, D Signal and Image Research Laboratory Department of Electrical Engineering and Computer Science University of Illinois Chicago Chicago IL USA

In this correspondence, we propose a comprehensive theory of the convergence and characterization of roots of order-statistics filters. Conditions for the convergence of iterations of order-statistics filters are prop... 详细信息

关键词： Convergence Filters signal processing Image processing Statistics Image restoration signal restoration Image converters Computational efficiency Filtering theory

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Adaptive wavelet subband coding for music compression

Adaptive wavelet subband coding for music compression

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Data compression Conference (DCC)

作者： K. Ferens W. Kinsner Department of Electrical & Computer Engineering Signal and Data Compression Laboratory University of Manitoba Winnipeg MAN Canada

This paper describes modelling of the coefficient domain in wavelet subbands of wideband audio signals for low-bit rate and high-quality compression. The purpose is to develop models of the perception of wideband audio signals in the wavelet domain. The coefficients in the wavelet subbands are quantized using a scheme that adapts to the subband signal by setting the quantization step size for a particular subband to a size that is inversely proportional to the subband energy, and then, within a subband, by modifying the energy determined step size as inversely proportional to the amplitude probability density of the coefficient. The amplitude probability density of the coefficients in each subband is modelled using learned vector/scalar quantization employing frequency sensitive competitive learning. The source data consists of 1-channel, 16-bit linear data sampled at 44.1 kHz from a CD containing major classical and pop music. Preliminary results show a bit-rate of 150 kbps, rather than 705.6 kbps, with no perceptual loss in quality. The wavelet transform provides better results for representing multifractal signals, such as wide band audio, than do other standard transforms, such as the Fourier transform.

关键词： Multiple signal classification Wavelet transforms Psychoacoustic models Wideband Frequency Fourier transforms signal resolution Data compression Wavelet domain Fractals

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Transient analysis and genetic algorithms for classification

Transient analysis and genetic algorithms for classification

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Communications, Power and Computing Conference: WESCANEX

作者： J. Toonstra W. Kinsner Department of Electrical & Computer Engineering Signal and Data Compression Laboratory University of Manitoba Winnipeg MAN Canada

This paper presents a method of generating unique fingerprints of radio transmitter turn-on transients. The fingerprinting system consists of the application of multiresolution wavelet analysis used to characterize the features contained in the transient followed by the use of a genetic algorithm to extract the wavelet coefficients that represent critical features of the transient. To measure the ability of the system to generate efficient and unique fingerprints, a neural network is used to classify the transients by their fingerprints. To test the noise sensitivity of the system, noisy transients were applied to a trained neural network, the network was able to positively classify noisy transients with 20 dB signal to noise ratios (SNR) and up. Experiments with real radio transients show that the system is able generate uniqiue fingerprints for absolute classification by a neural network for radios of differing model type as well as radios of the same model type.

关键词： Transient analysis Genetic algorithms Fingerprint recognition signal to noise ratio Neural networks Radio transmitters signal resolution Wavelet analysis Algorithm design and analysis Wavelet coefficients

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Multimode Spectral Coding of Speech at 2400 bps and Below

Multimode Spectral Coding of Speech at 2400 bps and Below

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IEEE Workshop on Speech Coding For Telecommunications

作者： A.M. Das A. Gersho Signal Compression Laboratory Department of Electrical and Computer Engineering University of California Santa Barbara CA USA

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Reduced-search fractal block coding of images

Reduced-search fractal block coding of images

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Data compression Conference (DCC)

作者： W. Kinsner L. Wall Department of Electrical & Computer Engineering Signal and Data Compression Laboratory University of Manitoba Winnipeg MAN Canada

Summary form only given, as follows. Fractal based data compression has attracted a great deal of interest since Barnsley's introduction of iterated functions systems (IFS), a scheme for compactly representing intricate image structures. This paper discusses the incremental development of a block-oriented fractal coding technique for still images based on the work of Jacquin (1990). A brief overview of Jacquin's method is provided, and several of its features are discussed. In particular, the high order of computational complexity associated with the technique is addressed. This paper proposes that a neural network paradigm known as frequency sensitive competitive learning (FSCL) be employed to assist the encoder in locating fractal self-similarity within a source image. A judicious development of the proper neural network size for optimal time performance is provided. Such an optimally-chosen network has the effect of reducing the time complexity of Jacquin's original encoding algorithm from O(n/sup 4/) to O(n/sup 3/). In addition, an efficient distance measure for comparing two image segments independent of mean pixel brightness and variance is developed. This measure, not provided by Jacquin, is essential for determining the fractal block transformations. An implementation of fractal block coding employing FSCL is presented and coding results are compared with other popular image compression techniques. The paper also present the structure of the associated software simulator.

关键词： Fractals Block codes Image coding Neural networks Power capacitors Data compression Computational complexity Frequency Image segmentation Pixel

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Consonant characterization using correlation fractal dimension for speech recognition

Consonant characterization using correlation fractal dimensi...

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Communications, Power and Computing Conference: WESCANEX

作者： A. Langi W. Kinsner Department of Electrical and Computer Engineering Signal and Data Compression Laboratory and TRLabs (Telecommunications Research Laboratories) University of Manitoba Winnipeg MAN Canada

This paper presents a new and promising approach in characterizing speech consonants using a fractal model for speech recognition systems. Characterization of consonants has been a difficult problem because consonant waveforms may be indistinguishable in time or frequency domain. The approach views consonant waveforms as coming from a turbulent constriction (excitation) in a human speech production system, and thus exhibiting turbulent and noise like time domain appearance. However, it departs from the usual approach by modeling consonant excitation using chaotic dynamical systems capable of generating turbulent and noise-like excitations. The scheme employs correlation fractal dimension and Takens embedding theorem to measure fractal dimension from time-series observation of the dynamical systems. It uses linear predictive coding (LPC) excitation of twenty-two consonant waveforms as the time series. Furthermore, the correlation fractal dimension is calculated using a fast Grassberger algorithm. A preliminary observation shows encouraging results because every consonant results in a unique trend of fractal dimensions for different embedding dimensions and scales.

关键词： Fractals Speech recognition Chaos Noise generators Frequency domain analysis Humans Production systems Time domain analysis Linear predictive coding Data compression

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Partially adaptive radar signal processing: The cross-spectral approach 29

Partially adaptive radar signal processing: The cross-spectr...

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29th Asilomar Conference on signals, Systems and computers, ACSSC 1995

作者： Goldstein, J. Scott Kogon, Stephen M. Reed, Irving S. Williams, Douglas B. Holder, E. Jeff USAF Rome Laboratory RL/C3BA 525 Brooks Rd. Grifilss AFBNY13441-4505 United States Statistical Signal and Image Processing Laboratory Department of Electrical Engineering University of Southern California Los AngelesCA90089-2565 United States Digital Signal Processing Laboratory School of Electrical and Computer Engineering Georgia Institute of Technology AtlantaGA30332-0250 United States

ISBN: (纸本)0818673702

This paper is concerned with reducing the rank of the adaptive weight vector in radar array signal processing. The motivation for reducing the rank is that modern space-time processing requires many more weights than can be supported on airborne and spacesegment platforms. The loss incurred in partially adaptive radar processing is that the steady-state Wiener solution of the lower rank weight vector may not perform as well as the full rank solution. Hence, this paper will be concerned with the Wiener solution of partially adaptive radar arrays and compare the performance of principal component techniques with the recently developed crossspectral technique. © 1996 IEEE.

关键词： Array processing

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VLSI implementation of very-high-order FIR filters

VLSI implementation of very-high-order FIR filters

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： M.A. Soderstrand K. Al-Marayati Digital Signal Processing and Communications Laboratory Department of Electrical and Computer Engineering University of California Davis USA

Very-high-order FIR filters required for the new modulation schemes associated with wireless computer networks and cellular telephones can be implemented in VLSI circuitry using low-power CMOS technology and a novel application of Residue Number System (RNS) arithmetic. Through this approach 20-bit equivalent integer arithmetic can be obtained for filters with 8 to 256 taps with only a modest increase in hardware for filters above 8 taps. Simulations indicate that this new technique can increase dramatically the number of taps implemented on a single VLSI chip when compared with an FIR filter generated using FIRGEN.

关键词： Very large scale integration Finite impulse response filter CMOS technology computer networks Cellular networks Telephony Circuits Application software Digital arithmetic Hardware

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