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IEEE Southeastcon

作者： S. Nagarajan R. Sankar University of South Florida Tampa FL US

In linear predictive coding (LPC) analysis, the linear predictors are computed using the classical Levinson-Durbin algorithm. But the Levinson-Durbin algorithm is a processing bottleneck as it involves the addition of inner products in the calculation of the reflection coefficients. This paper develops and reports on modifications to the algorithm used in various speech processing and coding applications for efficient implementations.

关键词： linear predictive coding Prediction algorithms Signal processing algorithms Equations Autocorrelation Speech analysis Reflection Algorithm design and analysis Digital signal processing Mathematical model

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Vector quantization based on Gaussian mixture models

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 2000年第4期8卷 385-401页

作者： Hedelin, P Skoglund, J Chalmers Dept Signals & Syst Informat Theory Grp SE-41296 Gothenburg Sweden

In this paper, we model the underlying probability density function of vectors in a database as a Gaussian mixture (GM) model. The model is employed for high rate vector quantization analysis and for design of vector quantizers. It is shown that the high rate formulas accurately predict the performance of model-based quantizers. We propose a novel method for optimizing GM model parameters for high rate performance, and an extension to the EM algorithm for densities having bounded support is also presented. The methods are applied to quantization of LPC parameters in speech coding and we present new high rate analysis results for band-limited spectral distortion and outlier statistics. In practical terms, we find that an optimal single-stage VQ can operate at approximately 3 bits less than a state-of-the-art LSF-based 2-split VQ.

关键词： Vector quantization Databases Probability density function predictive models Optimization methods linear predictive coding Speech coding Speech analysis Statistical analysis Gaussian distribution

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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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Perceptual coding of digital audio

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PROCEEDINGS OF THE IEEE 2000年第4期88卷 451-513页

作者： Painter, T Spanias, A Arizona State Univ Ctr Telecommun Res Dept Elect Engn Tempe AZ 85287 USA

During the last decade, CD-quality digital audio has essentially replaced analog audio. Emerging digital audio applications for network, wireless, and multimedia computing systems face a series of constraints such as reduced channel bandwidth, limited storage capacity, and low rest. These new applications have created a demand for high-quality digital audio delivery at low bit rates. In response to this need, consider-able research has been devoted to the development of algorithms for perceptually transparent coding of high,fidelity (CD-quality) digital audio. As a result, many algorithms have been proposed, and several have non become international and/or commercial product standards. This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. This paper is organized as follows. First, psychoacoustic principles are described, with the MPEG psychoacoustic signal analysis model I discussed in some detail. Next, filter bank design issues and algorithms are addressed, with a particular emphasis placed on the modified discrete cosine transform, a perfect reconstruction cosine-modulated filter bank that has become of central importance in perceptual audio coding. Then, we review methodologies that achieve perceptually transparent coding of FM- and CD-quality audio signals, including algorithms that manipulate transform components, subband signal decompositions, sinusoidal signal components, and linens prediction parameters, as well as hybrid algorithms that make use of more than one signal model. These discussions concentrate on architectures and applications of those techniques that utilize psychoacoustic models to exploit efficiently masking characteristics of the human receiver. Several algorithms that have become international and/or commercial standards receive in-depth treatment, including the ISO/IEC MPEG family (-1, -2, -4), the Lucent Technologies PAC/EPAC/MPAC, the Dolby(

关键词： AC-2 AC-3 advanced audio coding (AAC) MPEG ATRAC audio coding audio coding standards audio signal processing data compression digital audio radio (DAR) digital broadcast audio (DBA) filter banks high-definition TV (HDTV) linear predictive coding lossy compression modified discrete cosine transform (MDCT) MP3 MPEG MPEG-1 MPEG-2 MPEG-4 MPEG audio multimedia signal processing perceptual audio coding (PAC) perceptual coding perceptual model pseudoquadrature mirror filter (PQMF) psychoacoustic model psychoacoustics SDDS signal compression signal-processing applications sinusoidal coding subband coding transform coding

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R/D optimal linear prediction

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 2000年第6期8卷 646-655页

作者： Prandoni, P Vetterli, M Ecole Polytech Fed Lausanne Lab Commun Audio Visuelle CH-1015 Lausanne Switzerland Univ Calif Berkeley Dept Elect Engn & Comp Sci Berkeley CA 94720 USA

A common technique to extend linear prediction to nonstationary signals is time segmentation: the signal is split into small portions and the modelization is carried out locally, The accuracy of the analysis is, however dependent on the window size and on the signal characteristics, so that the problem of finding a good segmentation is crucial to the entire modeling scheme, In this paper, we will present an algorithm which determines the optimal segmentation with respect to a cost function relating prediction error to modeling cost. The proposed approach casts the problem in a rate/distortion (R/D) framework, whereby the segmentation is implicitly computed while minimizing the modelization distortion for a given modelization cost. The algorithm is implemented by means of dynamic programming and takes the form of a trellis-based Lagrangian minimization, The optimal linear predictor, when applied to speech coding, dramatically reduces the number of bits per second devoted to the modeling parameters in comparison to fixed-window schemes.

关键词： dynamic programming linear predictive coding nonstationary signal segmentation rate-distortion

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Re-optimisation of LPC filters for multi-pulse coded excitation - Comment

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ELECTRONICS LETTERS 2000年第12期36卷 1082-1083页

作者： Hacioglu, K Eastern Mediterranean Univ Elect Elect Engn Dept G Magosa 10 Mersin Turkey

For original paper, see D.P. Palomar et al., ibid., vol. 35, no. 13, pp. 1058-9 (1999). Palomar et al. addressed the question of LPC filter re-optimisation and proposed a recursive algorithm that they considered promising. The present commentator shows that the iterative algorithm employed by Palomar is unnecessarily complex because of poor initialisation. The algorithm that uses available speech data at the very first step is shown to perform well. The results of the one-shot algorithm can be used to initiate Paolmar's method to significantly reduce the number of iterations

关键词： filtering theory minimisation one-shot algorithm linear predictive coding iterative methods Filtering methods in signal processing iterative algorithm recursive algorithm Optimisation techniques speech coding initialisation Speech and audio coding multi-pulse coded excitation speech data Interpolation and function approximation (numerical analysis) LPC filter re-optimisation

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Computational improvement for G.729 standard

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ELECTRONICS LETTERS 2000年第13期36卷 1163-1164页

作者： Hwang, SH Ming Hsin Inst Technol Dept Management Informat Hsinchu Taiwan

G.729 is a CS-ACELP-based [1] standard for 8kbit/s speech compression and is defined by the ITU. In the G.729 encoding phase. the open-loop pitch T-op is estimated for each frame to reduce the computation required to search for the best adaptive codebook delay. However, > 25% CPU time is required for the open-loop pitch analysis. The weighted delta-LSP(WD-LSP) function is employed here to reduce the computation required for open-loop pitch analysis. An 81% saving is made in the computation of T-op. A total computational saving of 21% for the encoding phase Is achieved and the speech quality still sounds good (MOS = 3.5).

关键词： telecommunication standards open-loop pitch data compression linear predictive coding encoding phase speech quality CS-ACELP-based standard G.729 standard Codes speech coding open-loop pitch analysis 8 kbit/s Speech and audio coding ITU computational saving speech compression weighted delta-LSP adaptive codebook delay

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Lossless image coding via adaptive linear prediction and classification

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PROCEEDINGS OF THE IEEE 2000年第11期88卷 1790-1796页

作者： Motta, G Storer, JA Carpentieri, B Brandeis Univ Dept Comp Sci Waltham MA 02454 USA Univ Salerno Dipartimento Informat & Applicaz RM Capocelli I-84081 Baronissi Italy

In the past years, there have been several improvements in lossless image compression. All the recently proposed state-of-the-art lossless image compressors can be roughly divided into two categories: single and double-pass compressors. linear prediction is rarely used in the first category, while TMV [7], a state-of-the-art double-pass image compressor, relies on linear prediction for its performance. We propose a single-pass adaptive algorithm that uses context classification and multiple linear predictors, locally optimized on a pixel-by-pixel basis. Locality is also exploited in the entropy coding of the prediction error. The results we obtained on a test set of several standard images are encouraging. On the average, our ALPC obtains a compression ration comparable to CALIC [20] while improving on some images.

关键词： adaptive coding arithmetic codes data compression Golomb-Rice codes gradient methods image coding linear predictive coding

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A mixed harmonic excitation linear predictive speech coding for low bit rate applications

A mixed harmonic excitation linear predictive speech coding ...

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Asilomar Conference on Signals, Systems & Computers

作者： Suat Yeldner M.R. Baraniecki Commun. Technol. Div. COMSAT Lab. Clarksburg MD USA Communications Technology Division Comsat Laboratories Clarksburg MD USA

The mixed harmonic excitation linear predictive speech coder (MHE-LPC) is based on a technique derived from the multi-band excitation (MBE) and multi-band linear predictive coding (MB-LPC) types of speech coding algorithms. The MHE-LPC coder has the potential of producing high quality speech at 4.8 kb/s and below. This coder employs a new pitch estimation and a mixed voicing technique that utilizes a number of bands within the 4 kHz speech bandwidth (6-8 bands), each band having a voicing probability that defines voiced and unvoiced portions of the excitation spectrum. In addition to these, a new DCT based LPC and residual amplitude quantization technique has been developed and is described in this paper. The 3.6 kb/s MHE-LPC coder with 14th order LPC filter was found to produce better speech quality than various other low rate speech coding standards. During an informal listening test, the 3.6 kb/s MHE-LPC vocoder produced better performance than the 3.6 kb/s INMARSAT Mini-M coder. The MHE-LPC algorithm can be easily extended to bit rates between 1.2 and 8 kb/s depending on the applications.

关键词： Speech coding linear predictive coding Power harmonic filters Bandwidth Discrete cosine transforms Quantization Code standards Testing Vocoders Bit rate

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Exploiting simultaneously masked linear prediction in a WI speech coder

Exploiting simultaneously masked linear prediction in a WI s...

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7th IEEE Workshop on Speech coding

作者： Lukasiak, J Burnett, IS Univ Wollongong Whisper Labs TITR Wollongong NSW 2522 Australia

ISBN: (纸本)0780364163

This paper uses a method of incorporating simultaneous masking into the calculation of a linear predictive filter (SMLPC) as the front end to a 2kbps waveform interpolation (WI) speech coder. A modification to the masking threshold calculation used in SMLPC is proposed. This modification improves the performance of SMLPC in noise like sections by placing greater emphasis on strongly voiced speech. MOS test results reveal that the modified SMLPC improved the perceptual quality of the WI coder. The improvement is significant for female speakers whilst the quality for male speech is virtually unchanged. This result conflicts with previous results reported for SMLPC where only male speech was improved. The change is attributed to the modification of the masking threshold and confirms that adapting the masking threshold according to the pitch of the speech will allow SMLPC to remove more perceptually important information from all input speech than standard LPC.

关键词： Autocorrelation Filtering Frequency Interpolation Laboratories linear predictive coding Masking threshold Nonlinear filters Redundancy Speech coding

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