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In AM radio it is of interest to increase the perceived loudness of transmitted speech in order to increase the effective range of the radio broadcasts. In particular, shortwave radio broadcasts are subject to a variety of natural and men-made interference and noise which can severely limit the signal-to-noise ratio of the received signal. Often, broadcasting is done using very powerful and expensive transmitters and it is of interest to determine whether or not signal processing techniques could provide a more cost-effective increase of the broadcast range. In addition to developing digital signal processing (DSP) algorithms [1,2] for enhancing the perceived loudness of speech, we are developing a facility to evaluate the effectiveness of the audio preprocessing algorithms. ]his evaluation will eventually include subjective measures of processed speech quality, quantitative measures of speech intelligibility, and measurement of the peak/rms ratio of the processed speech. ]his paper describes a system that has been developed for automatically measuring the enhancement in the peak/rms ratio that a given audio processor provides, and gives some preliminary results on the performance of three commercial audio processors and two rather simple processors. An important outcome of the results is the establishment of an engineering goal for an improved digital signal processing (DSP) speech enhancement algorithm. Author Signal Processing; Signal to Noise Ratios; Broadcasting; Dynamic Range 20060001697 Zoran Corp., Santa Clara, CA, USA A Novel VLSI Digital Signal Processor Architecture for High-Speed Vector and Transform Operations Taylor, David M.; Retter, Rafi; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP ‘87); Volume 1; 1987, pp. 13.5.1 - 13.5.4; In English; See also 20060001583; Copyright; Avail.: Other Sources This paper presents a new concept in monolithic digital signal processing integrated circuits. The device is called a Vector Signal Processor (VSP) and is manufactured by Zoran Corporation. It achieves extremely high signal processing throughput in both the time and frequency domains for many standard DSP operations. It introduces a number of architectural features unique to digital signal processing components. Among these are: a ‘high-level’ instruction set which allows programming at the functional level, embedded hardware to support block floating-point arithmetic, and the ability to easily parallel multiple devices to achieve signal processing throughput even higher than that attainable with a single device. The combination of these features and a powerful applications development environment make the VSP easy to both use and program for developing sophisticated, high-performance applications. Author Signal Processing; Very Large Scale Integration; Architecture (Computers) 20060001699 Georgia Inst. of Tech., Atlanta, GA, USA Iterative Speech Enhancement with Spectral Constraints Hansen, John H.; Clements, Mark A.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP ‘87); Volume 1; 1987, pp. 6.7.1 - 6.7.4; In English; See also 20060001583; Copyright; Avail.: Other Sources A new and improved iterative speech enhancement technique based on spectral constraints is presented in this paper. The iterative technique, originally formulated by Lim and Oppenheim, attempts to solve for the maximum likelihood estimate of a speech waveform in additive white noise. The new approach applies inter- and intra-frame spectral constraints to ensure convergence to reasonable values and hence improve speech quality. An extremely efficient technique for applying these constraints is in the use of line spectral pair (LSP) coefficients. The inter-frame constraints ensures more speech-like formant trajectories than those found in the unconstrained approach. Results from speech degraded by additive white Gaussian noise show noticeable quality improvement. Author Speech Recognition; Maximum Likelihood Estimates; Convergence; Spectra; Augmentation 20060001700 GTE Labs., Inc., Waltham, MA, USA Enhancement of Block-Coded Speech Veeneman, D. E.; Mazor, B.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP ‘87); Volume 1; 1987, pp. 6.9.1 - 6.9.4; In English; See also 20060001583; Copyright; Avail.: Other Sources This paper describes a very effective method for reducing framing noise associated with block-by-block speech coding algorithms. It is a post-coding speech enhancement procedure designed specifically to combat this type of noise. The procedure is based on a fully adaptive comb filter which is adapted in synchrony with the coding algorithm, s frame boundary. An optimal (minimum mean-squared-error sense) linear prediction approach is applied to determine the filter coefficients and 58
adapt them to the short-time statistics of the speech. A formal subjective listening trial has shown the method to significantly improve the perceived quality of block-coded speech. Author Voice Data Processing; Coding; Noise Reduction; Adaptive Filters; Linear Prediction; Mean Square Values 20060001705 ITT Defense Communications Div., San Diego, CA, USA Techniques for Suppression of an Interfering Talker in Co-Channel Speech Naylor, J. A.; Boll, S. F.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP ‘87); Volume 1; 1987, pp. 6.12.1-6.12.4; In English; See also 20060001583; Copyright; Avail.: Other Sources The problem addressed by this study is the suppression of an undesired talker when two talkers are communicating simultaneously on the same monophonic channel (co-channel speech). Two different applications are considered, improved intelligibility for human listeners, and improved performance for automatic speech and speaker recognition (ASR) systems. For the human intelligibility problem, the desired talker is the weaker of the two signals with voice-to-voice power ratios (Power desired / Power interference), or VVRs, as low as -18dB. For ASR applications, the desired talker is the stronger of the two signals, with VVRs as low as 5dB. Signal analysis algorithms have been developed which attempt to separate the co-channel spectrum into components due to the two different (stronger and weaker) talkers. Author Speech Recognition; Communicating; Human Performance; Signal Analysis; Intelligibility 20060001720 Massachusetts Univ., Amherst, MA, USA Segmentation of Noisy Textured Images Using Simulated Annealing SunWon, Chee; Derin, Haluk; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP ‘87); Volume 1; 1987, pp. 563-566; In English; See also 20060001583 Contract(s)/Grant(s): NSF ECS-8403685; ONR N00014-85-K-0561; Copyright; Avail.: Other Sources This paper presents a segmentation algorithm for noisy textured images. To represent noisy textured images, we propose a hierarchical stochastic model that consists of three levels of random fields: the region process, the texture processes and the noise. The hierarchical model also includes local blurring and nonlinear image transformation as results of the image corrupting effects. Having adopted statistlc31 model, the maximum a posteriori (MAP) estimation is used to find the segmented regions through the restored(noise-free) textured image data. Since the joint a posteriori distribution at hand is a Gibbs distribution, we use simulated annealing as a maximization technique. The simulated annealing based segmentation algorithm presented in this paper can also be viewed as a two step iterative algorithm in the spirit of the EM algorithm. Author Electromagnetic Noise; Algorithms; Mathematical Models; Segments 20060001722 University of Southern Illinois, IL, USA Invariant Planar Shape Recognition Using Dynamic Alignment Gupta, L.; Srinath, M. D.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP ‘87); Volume 1; 1987, pp. 7.3.1-7.3.4; In English; See also 20060001583; Copyright; Avail.: Other Sources A technique for classifying closed planar shapes is described in which a shape is characterized by an ordered sequence that represents the Euclidean distance between the centroid and all contour pixels of the shape. Shapes belonging to the same class have similar sequences, hence a procedure for classifying shapes is based on the degree of similarity between these sequences. In order to determine the similarity between sequences, a non-linear alignment process is developed to find the best correspondence between the sequences. Optimum alignment is obtained by expanding segments of the sequences to minimize a dissimilarity function between the sequences. Normalization with respect to scaling and rotation is described and an example illustrating the use of dynamic alignment for the classification of noisy shapes is presented. Author Shapes; Pixels; Alignment; Classifications; Euclidean Geometry; Contours 20060001734 Texas A&M Univ., College Station, TX, USA Jump Detection and Fast Parameter Tracking for Piecewise AR Processes Using Adaptive Lattice Filters Li, Shiping; Dickinson, Bradley W.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP ‘87); Volume 1; 1987, pp. 9.2.1 - 9.2.4; In English; See also 20060001583 Contract(s)/Grant(s): NSF ECS-84-05460; AFOSR-84-0381; Copyright; Avail.: Other Sources 59
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Life Sciences 51 Life Sciences (Gen
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Microstructure Fabrication and Perf
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RNP SAAAR procedures have been prop
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concept was originally developed to
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Bacillus anthracis. Anthrax will be
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were evaluated throughout the exper
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20060001236 Institute of Space Medi
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20060001245 Institute of Space Medi
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properties of hyperbolic Dirichlet
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This paper had been prepared as a t
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20060002350 BERG2 Micro-Computer Es
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Exploratory Learning; Neurobiologic
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operation of LSI is pursuited to ac
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industrial tools, and increased int
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20060001694 Motorola, Inc., Schaumb
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enchmarks in this document is meant
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contemporary team theory, adapted a
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toolkit), each of which we built to
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developmental effort. This paper il
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signal-processing algorithms, allow
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Scheduling (TIES) for efficient net
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safety, and real time performance o
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20060001877 New York Univ., New Yor
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domain. dRBAC utilizes PKI to ident
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difference in the synchronous frequ
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This paper presents an analysis of
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method relies on two mathematical t
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This paper considers the behaviour
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20060001709 Pennsylvania State Univ
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The feasibility of real time, micro
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a product of two Toeplitz data matr
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estimators, and investigate the eff
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65 STATISTICS AND PROBABILITY Inclu
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A new approach to making voiced/unv
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near-optimal strategy with DBN serv
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non-linear system design. Finally,
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The aim of this three year project
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extraction, and object recognition.
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20060001806 North Carolina State Un
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ate of the Booster, and provide a l
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The magnetic harmonic errors of the
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20060002069 Fermi National Accelera
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micrographs as triangular and trape
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plays a significant role in the dev
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20060002468 Lawrence Livermore Nati
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20060002492 Lawrence Livermore Nati
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parameters and propose some importa
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ATAL introduced a technique for dec
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The problem is addressed of evaluat
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20060001730 Naval Underwater System
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and tip-tilt natural guide star (NG
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oscillation, bursting, chirping and
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specifications automatically. Some
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discrimination information (or the
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matching algorithm is proposed for
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20060001768 Naval Postgraduate Scho
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20060001826 Massachusetts Univ., Am
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allows the user to specify a contex
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information from these resources, u
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such as ADSL and FTTH, we are now a
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Presented is the National Optical A
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20060000076 Gemini Observatory, Hil
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Telescope (HST) at optical waveleng
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Misadjustment Expressions for Infin
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A-4 Complexity Reduced Lattice Filt
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ASYMMETRY Anthrax Biosensor, Protec
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Regulation of hTERT Expression and
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CLIMATE C-Haendelser i Vinterklimat
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Neutronics Assessments for a RIA Fr
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CRYSTAL SURFACES High-Throughput Ap
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DIELECTRICS Theoretical and Experim
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Analysis of Depletion-Region Collec
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Scenario Customization for Informat
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GALLIUM ARSENIDES InGaAs/GaAs QD Su
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HUMAN RESOURCES The Management of P
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INFRARED SPECTRA Chemical Vapor Ide
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Adaptive Transversal Filters with D
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MATCHED FILTERS Simplified Class 1
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MILITARY TECHNOLOGY The USA Air For
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NONLINEAR FILTERS A Novel Approach
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Object Classification and Registrat
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Predicting Ductile Crack Growth in
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RECURSIVE FUNCTIONS Recursive Array
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Texture Segmentation Using a Class
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SITUATIONAL AWARENESS Aspects of Sh
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STABILITY Damping Transverse Instab
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Single Target Spin Asymmetries and
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On Generalized Feussner’s Princip
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WEAR Wear Prediction of Strip Seals
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Anderson, Robert H. The Vulnerabili
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Neutronics Assessments for a RIA Fr
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Chien, Ting-Li Navigation of a Mobi
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The Modeling and Segmentation of Sp
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Fomin, Pavel Test and Evaluation of
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Hann, P. R. Technical Equivalency D
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Irons, F. H. A Phase-Plane Approach
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Kawarada, Hiroshi A Hierarchical Re
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Lai, Feipei JADE: A Hierarchical Sw
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Lopez-Perez, Javier VHF Channel Occ
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Menard, J. Scaling of Kinetic Insta
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Ney, Hermann Dynamic Programming Sp
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Using Rhythms of Relationships to U
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Resch, C. Communications Technology
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Sedra, A. S. SlCOMP: A Silicon Comp
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Steffler, E. D. Predicting Ductile
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Torkelson, Mats A VLSI Sigma-Delta
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Wheeler, P. D. Practical Adaptive N
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Zong-ye, Wang Application of Multis
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