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AN LSF QUANTIZER FOR THE NORTH-AMERICAN HALF-RATE SPEECH CODER

IEEE TRANSACTIONS ON VEHICULAR TECHNOLOGY 1994年第3期43卷 413-419页

作者： DEMARCA, JRB AT&T BELL LABS SPEECH PROC RES DEPTMURRAY HILLNJ 07974

The design of the half-rate coder for the North-American cellular communication standard poses a challenging problem. The desired speech quality is that of the full-rate coder, but the total bit rate specified for the half-rate is only 6.5 kb/s. Since the mobile communication channel is very noisy, the use of error-correcting codes is necessary, which leaves only about 4 kb/s for the actual speech information. This very restricted bit budget requires that fewer than 30 b be allotted to the quantization of the LSF parameters, which precludes the use of scalar quantization. In this work, a quantizer design procedure which is tailored for the transmission frame structure of the North-American (TIA) cellular communication standard is presented. It is basically a split vector quantizer which makes use of interframe prediction to lower the number of bits required for quantization. Since the prediction is performed only on every other frame, the error propagation is limited. The quantization technique yields good performance in the 26-27 b/frame range, and the complexity of its implementation is less than 0.5 Mip.

关键词： Quantization Frequency Communication standards linear predictive coding Bit rate Filters Mobile communication Error correction codes Speech coding predictive models

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Underwater target classification using wavelet packets and neural networks

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IEEE TRANSACTIONS ON NEURAL NETWORKS 2000年第3期11卷 784-794页

作者： Azimi-Sadjadi, MR Yao, D Huang, Q Dobeck, GJ Colorado State Univ Signal Image Proc Lab Dept Elect & Comp Engn Ft Collins CO 80523 USA NSWC Dahlgren Div Coastal Syst Stn Panama City FL 32407 USA

In this paper, a new subband-based classification scheme is developed for classifying underwater mines and mine-like targets From the acoustic backscattered signals. The system consists of a feature extractor using wavelet packets in conjunction with linear predictive coding (LPC), a feature selection scheme, and a backpropagation neural-network classifier, The data set used for this study consists of the backscattered signals from six different objects: two mine-like targets and four nontargets for several aspect angles. Simulation results on ten different noisy realizations and for signal-to-noise ratio (SNR) of 12 dB are presented. The receiver operating characteristic (ROC) curve of the classifier generated based on these results demonstrated excellent classification performance of the system, The generalization ability of the trained network was demonstrated by computing the error and classification rate statistics on a large data set, A multiaspect fusion scheme was also adopted in order to further improve the classification performance.

关键词： feature extraction linear predictive coding neural network underwater target classification wavelet packets

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A classification of reliable multicast protocols

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IEEE NETWORK 2004年第3期18卷 24-34页

作者： Atwood, JW Concordia Univ Montreal PQ Canada

The range of user requirements on multicast protocols is so wide that no single protocol will ever satisfy them. The set of multicast protocols can be classified using the user requirements, and the architectures, mechanisms, communications patterns, and policies used to satisfy these requirements. We provide such a classification, and illustrate it with several example protocols chosen to cover the range of features described.

关键词： Multicast protocols Internet Taxonomy Scalability Application software Unicast Telecommunication network reliability linear predictive coding Communication system control Data communication

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VECTOR QUANTIZATION OF THE ARTICULATORY SPACE

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1988年第12期36卷 1812-1818页

作者： LARAR, JN SCHROETER, J SONDHI, MM AT&T BELL LABS ACOUST RES DEPTMURRAY HILLNJ 07974

A technique is presented for quantizing the articulatory space, i.e. for replacing the continuum of all possible vocal tract shapes by a finite set of shapes that span the articulatory space. Vocal tract shapes are represented as vectors of parameters that control an articulatory model, so vector quantization are applicable for deriving this codebook of tract shapes. Samples of the articulatory space are generated by interpolating model parameters along paths between selected target configurations. These shapes constitute the training data that are clustered to form the codebook. Such a codebook of shapes has direct application when vocal tract shapes must be estimated from speech. One such application, which motivated the present study, is the use of the codebook to providing starting configurations for an optimizing articulatory speech synthesizer.< >

关键词： Vector quantization linear predictive coding Speech analysis Synthesizers Shape control Speech synthesis Tongue Spectral shape Speech coding Books

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A VERY SIMPLE AND EFFICIENT WEIGHTING FILTER WITH APPLICATION TO A CELP CODER FOR HIGH-QUALITY SPEECH AT 4800 BITS/S

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SIGNAL PROCESSING 1992年第2期27卷 109-116页

作者： BOITE, R LEICH, H GAO, Y Laboratoire de Théorie des Circuits et Traitement des Signaux (T.C.T.S.) Faculté Polytechnique de Mons 33 Boulevard Dolez B-7000 Mons Belgium Lernout & Hauspie Speechproducts n.v. Rozendaalstraat 14 8900 Ieper Belgium

In this paper we propose a very simple and efficient weighting filter with which the computational complexity of CELP coders can be considerably reduced. Other algorithms using a weighting filter could also benefit from the advantages of this simplified weighting filter. The estimation of the long term prediction with the close-loop method is described. A binary codebook is used for the excitation vectors. It is shown how the excitation sequence can be obtained by a non-exhaustive method in two steps with a simplified algorithm and the simple weighting filter. Several coders have been implemented showing that the perceptual quality of the simplified algorithm is equivalent to that of the original CELP.

关键词： SIGNAL PROCESSING SPEECH coding linear predictive coding SPEECH CELP coding SPEECH PERCEPTION

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Smelting condition identification for a fused magnesium furnace based on an acoustic signal

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JOURNAL OF MATERIALS PROCESSING TECHNOLOGY 2017年 244卷 231-239页

作者： Fu, You Wang, Ninghui Wang, Zhen Wang, Zhiqiang Ji, Bing Wang, Xiaochen Dalian Univ Technol Sch Elect Engn Dalian 116024 Peoples R China Dalian Univ Technol Sch Innovat & Entrepreneurship Dalian 116024 Peoples R China Univ Leicester Dept Engn Univ Rd Leicester LE1 7RH Leics England Environm Engn Assessment Ctr Liaoning Prov Shenyang 110032 Peoples R China

To promote energy efficiency during *** furnace smelting, four smelting states were introduced in the smelting stage: an unmelted state, semi-molten state, molten state, and overheating state. A smelting identification system to distinguish these smelting states was developed through the use of linear predictive coding and a principal component analysis algorithm. A new smelting condition identification system was obtained. Corresponding pilot productions were conducted to compare the differences between employing the method and not employing the method. All of the pilot production data showed that feeding raw materials over time during the overheating state and decreasing current injection in the molten state could reduce energy consumption as well as increase crystal purity. (C) 2016 Published by Elsevier B.V.

关键词： Acoustic signal Fused magnesium furnace linear predictive coding Principal component analysis

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Closed-form posterior Cramer-Rao bounds for bearings-only tracking

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IEEE TRANSACTIONS ON AEROSPACE AND ELECTRONIC SYSTEMS 2006年第4期42卷 1198-1223页

作者： Brehard, T. Le Cadre, J. -P. CNRS IRISA F-35042 Rennes France

We address the classical bearings-only tracking problem (BOT) for a single object, which belongs to the general class of nonlinear filtering problems. Recently, algorithms based on sequential Monte-Carlo methods (particle filtering) have been proposed. As far as performance analysis is concerned, the posterior Cramar-Rao bound (PCRB) provides a lower bound on the mean square error. Classically, under a technical assumption named "asymptotic unbiasedness assumption," the PCRB is given by the inverse Fisher information matrix (FIM). The latter is computed using Tichavsky's recursive formula via Monte-Carlo methods. Two major problems are studied here. First, we show that the asymptotic unbiasedness assumption can be replaced by an assumption which is more meaningful. Second, an exact algorithm to compute the PCRB is derived via Tichavsky's recursive formula without using Monte-Carlo methods. This result is based on a new coordinate system named logarithmic polar coordinate (LPC) system. Simulation results illustrate that PCRB can now be computed accurately and quickly, making it suitable for sensor management applications.

关键词： Nonlinear equations Target tracking linear matrix inequalities linear predictive coding Trajectory Time measurement Filtering algorithms Performance analysis Mean square error methods Computational modeling

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ERROR CORRECTION MASKING FOR DIGITAL VOICE TRANSMISSION OVER THE LAND MOBILE SATELLITE SYSTEM

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IEEE TRANSACTIONS ON COMMUNICATIONS 1989年第3期37卷 309-314页

作者： BRYDEN, B SEGUIN, GE CONAN, J BHARGAVA, VK BRINDAMOUR, A ROYAL MIL COLL CANADA DEPT ELECT ENGN KINGSTON K7L 2W3 ONTARIO CANADA ECOLE POLYTECH DEPT ELECT ENGN MONTREAL H3C 3A7 QUEBEC CANADA UNIV VICTORIA DEPT ELECT ENGN VICTORIA V8W 2Y2 BC CANADA

The development of a 2400-b/s speech digitizer which provides an acceptable level of intelligibility and quality over land mobile satellite channels is described. Performance tests over simulated channels in the UHF band (800 MHz) are presented. The voice digitizer is a linear prediction (LPC) vocoder which uses a channel error correction and concealment procedure tailored to error statistics for a minimum-shift keyed (MSK) downlink to a moving vehicle. The error-handling technique is based on perceptual criteria and utilizes the parametric nature of LPC representation of speech. A single-error-correcting, single-burst-detecting (28, 20) fire code is shown to be the best choice for the application. The intelligibility of the vocoder is measured and compared to the standard LPC-10 algorithm. The major remaining sources of speech quality degradation due to channel errors are determined and ranked.< >

关键词： Error correction Speech linear predictive coding Vocoders Satellites Testing Error analysis Downlink Vehicles Fires

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Design and evaluation of a parametric model for cardiac sounds

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COMPUTERS IN BIOLOGY AND MEDICINE 2017年 89卷 170-180页

作者： Ibarra-Hernandez, Roilhi F. Alonso-Arevalo, Miguel A. Cruz-Gutierrez, Alejandro Licona-Chavez, Ana L. Villarreal-Reyes, Salvador Ctr Invest Cient & Educ Super Ensenada Div Fis Aplicada Dept Elect & Telecomunicac Carretera Ensenada Tijuana 3918 Ensenada 22860 Baja California Mexico Ctr Estudios Univ Xochicalco Fac Med Campus EnsenadaSan Francisco 1139 Ensenada 22830 Baja California Mexico

Heart sound analysis plays an important role in the auscultative diagnosis process to detect the presence of cardiovascular diseases. In this paper we propose a novel parametric heart sound model that accurately represents normal and pathological cardiac audio signals, also known as phonocardiograms (PCG). The proposed model considers that the PCG signal is formed by the sum of two parts: one of them is deterministic and the other one is stochastic. The first part contains most of the acoustic energy. This part is modeled by the Matching Pursuit (MP) algorithm, which performs an analysis-synthesis procedure to represent the PCG signal as a linear combination of elementary waveforms. The second part, also called residual, is obtained after subtracting the deterministic signal from the original heart sound recording and can be accurately represented as an autoregressive process using the linear predictive coding (LPC) technique. We evaluate the proposed heart sound model by performing subjective and objective tests using signals corresponding to different pathological cardiac sounds. The results of the objective evaluation show an average Percentage of Root-Mean-Square Difference of approximately 5% between the original heart sound and the reconstructed signal. For the subjective test we conducted a formal methodology for perceptual evaluation of audio quality with the assistance of medical experts. Statistical results of the subjective evaluation show that our model provides a highly accurate approximation of real heart sound signals. We are not aware of any previous heart sound model rigorously evaluated as our proposal.

关键词： Heart sounds modeling Biomedical signal processing Matching Pursuit Time-frequency dictionaries Time-frequency atoms linear predictive coding

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The 1974 origins of VoIP

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IEEE SIGNAL PROCESSING MAGAZINE 2005年第4期22卷 87-90页

作者： Gray, RM Massachusetts Institute of Technology University of Southern California

This paper brings light on the digital signal processing (DSP) roots of a modern concept, voice over IP (VoIP). An example is also provided in which developments in DSP - speech coding, in particular - had a profound impact on the early development of the ARPANET, the ancestor of the Internet. The author shows how packet speech, recently rediscovered and made popular as VoIP, was first successfully demonstrated in 1974 on the ARPANET and how the Internet protocol (IP) emerged largely as a result of that effort.

关键词： Internet telephony Digital signal processing History Speech coding linear predictive coding Speech processing Image coding Collaboration ARPANET predictive models

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