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检索条件"主题词=linear predictive coding"

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Snap-drift learning for phrase recognition

Snap-drift learning for phrase recognition

引用

International Joint Conference on Neural Networks (IJCNN)

作者： Sin Wee Lee D. Palmer-Brown Computational Intelligence Group School of Computing Leeds Metropolitan University Leeds UK School of Computing and Technology University of East London London UK

This paper presents a new application of the snap-drift algorithm by S. W. Lee, et al. (2004): phrase recognition using a set of phrases from the Lancaster parsed corpus (LPC) by R. Garside, et al. (1987). The learning algorithm is the classifier version of snap-drift. In this version, along with the complementary concepts of fast minimalist learning (snap) and slow drift towards the input pattern, each node of the snap-drift neural network (SDNN) swaps between snap and drift modes when declining performance is indicated on that particular node. This method enables the SDNN to learn at node level, in the sense that each node has its learning mode toggled independently of the other nodes. Learning on each node is also reinforced by enabling learning with a probability that decreases with increasing performance. The simulations demonstrate that learning is stable, and the results have consistently shown similar classification performance and advantages in terms of speed in comparison with a multilayer perceptron (MLP) and back-propagation by J. Topper, et al. (2002), D. E. Rumelhart, et al. (1986) applied to the same problem.

关键词： Subspace constraints Neural networks linear predictive coding Resonance Fuzzy set theory Computer network management Speech Computational modeling Silicon compounds Computational intelligence

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Decomposition of the speech signal into short-time waveforms using spectral segmentation

Decomposition of the speech signal into short-time waveforms...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： C. d'Alessandro J.-S. Lienard LIMSI-CNRS Orsay France

Representation of the speech signal by a set of discrete elements which respect its acoustical and perceptive structures is considered. The signal is pre-analyzed frame by frame, and the spectral envelope obtained for each frame is segmented into regions comprising a single peak. The signal is then filtered in each region, and the elementary waveforms are spotted in the time domain. The problem of grouping the waveforms in adjacent channels is thus circumvented. The resulting representation is satisfactory, as is the signal reconstruction, except for some modeling problems remaining in the lowest part of the spectrum.< >

关键词： Speech processing Signal analysis Filters Speech analysis linear predictive coding Acoustic pulses Signal processing Signal reconstruction Energy resolution Signal resolution

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A model of LPC excitation in terms of eigenvectors of the autocorrelation matrix of the impulse response of the LPC filter

A model of LPC excitation in terms of eigenvectors of the au...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： B.S. Atal Acoustics Research Department AT and T Bell Laboratories Inc. Murray Hill NJ USA

A description is presented of a framework for developing compact and accurate representation of LPC (linear predictive coding) excitation. In this representation, the excitation waveform is expressed as a linear combination of the eigenvectors of the autocorrelation matrix of the LPC filter's impulse response. This representation allows a systematic study of changes in the filter excitation on the speech output. The author presents results on the precision that is necessary in the excitation without producing perceptible distortion in the output speech signal and an estimate of the minimum number of bits necessary for accurate reproduction of the excitation.< >

关键词： linear predictive coding Autocorrelation Speech synthesis Bit rate Vocoders Nonlinear filters Encoding Human voice Frequency Stochastic resonance

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Enhanced Lossless coding Tools of LPC Residual for ITU-T G.711.0

Enhanced Lossless Coding Tools of LPC Residual for ITU-T G.7...

引用

Data Compression Conference (DCC)

作者： Takehiro Moriya Yutaka Kamamoto Noboru Harada NTT Communication Science Laboratories Nippon Telegraph and Telephone Corporation Japan

ISBN: (纸本)9781424464258;9780769539942

Motivated by the rapid increase of VoIP services with G.711 for telephone speech, a new ITU-T recommendation, G.711.0 (frame-wise stateless lossless compression scheme for G.711 log PCM symbols), has been standardized. The standard scheme has several coding parts, each of which is adaptively selected depending on the characteristics of the input. Among them, the mapped domain prediction part is the one most frequently activated for normal speech signals. This part consists of linear prediction in the mapped domain and variable length coding of the prediction residual. It is useful for log-compressed/expanded signal, such as ITU-T G.711. This paper describes three newly devised enhancement tools for the coding of prediction residual signals: progressive order prediction, quantized prediction order, and adaptive and sub-frame base coding for separation parameters. The design criterion is the maximization of the averaged FoM (figure of merit) over frame lengths of 40, 80, 160, 240, and 320 samples. The first tool, progressive order prediction associated with the adaptive modification of the separation parameter for the first and second samples, enhances the compression ratio by 0.5 % with a negligible increase of the complexity. The second tool, quantized prediction order, improves the compression ratio by 0.2 % with even reduced complexity. The third tool, sub-frame base adaptive coding of separation parameters, gives a 0.2 % improvement in the compression ratio with comparable complexity. All three schemes are consistently and independently effective for improving the compression ratio, although the amount of improvement with each tool is small. At the same time, none of the tools have any significant impact on computational complexity. Therefore, all the devised tools improve the FoM and have been adopted in the mapped domain prediction part of the ITU-T G.711.0 standard.

关键词： linear predictive coding Internet telephony Data compression Laboratories Telegraphy Speech enhancement Speech coding Phase change materials Adaptive coding Computational complexity

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Vehicle license plate recognition in marketing application

Vehicle license plate recognition in marketing application

引用

IEEE Symposium on Intelligent Vehicle

作者： A. Taleb-Ahmed D. Hamad G. Tilmant LASL Université du Littoral Cote d'Opale Calais cedex LASL Université du Littoral Cote d'Opale Calais cedex France Heltys Roubaix Roubaix France

The paper presents an artificial vision system able to analyze the image of a car given by a camera, to locate the registration plate and to recognize its registration number. It describes practical problems encountered in implementing this application and the proposed solutions. The system has been designed using a modular approach. Sub-modules can be upgraded and or substituted independently, thus making the system potentially suitable in a large variety of vision applications. Performances of the system, have been evaluated in real situation.

关键词： Image recognition Cameras Licenses Image analysis Character recognition linear predictive coding Image processing Machine vision Image segmentation Vehicles

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A New Modulation Concept for Mixed Pseudo Analogue-Digital Speech and Audio Transmissioin

A New Modulation Concept for Mixed Pseudo Analogue-Digital S...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： C. Hoelper P. Vary RWTH Aachen University Aachen Germany

Current speech, audio, and video coding and transmission systems are either analogue or digital, with a strong shift from analogue systems to digital systems during the last decades. We have combined both digital and analogue schemes for the benefit of saving transmission bandwidth, complexity, and of improving the achievable quality at any given signal-to-noise ratio on the channel. The combination is achieved by transmitting pseudo analogue samples of the unquantized residual signal of a linear predictive digital filter which is called mixed pseudo analogue-digital (MAD) transmission. In this paper a new modulation scheme based on QPSK for digital information and an Archimedes spiral for the time discrete, pseudo analogue residual signal is introduced and evaluated.

关键词： Speech coding Bandwidth Digital systems linear predictive coding Noise shaping Adaptive filters Digital modulation Signal to noise ratio Quadrature phase shift keying Quantization

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Speaker independent isolated word automatic speech recognition using computer generated phonemes

Speaker independent isolated word automatic speech recogniti...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： H. Hinton L. Siegel Bell Laboratories Naperville IL USA School of Electrical Engineering Purdue University West Lafayette IN USA

A speaker independent isolated word speech recognition system is developed based on computer generated phonemes (CGP). A CGP is a vector of features that has been generated to represent a region of speech. The CGP creation algorithm looks for stable sounds in the incoming word through the use of a similarity measure. When a stable sound is detected a CGP is created to represent it. In addition to the creation of CGPs for stable vocal tract sounds, when unvoiced fricatives occur at either the beginning or end of a word a representative CGP is created. Using a heavily constrained dynamic time warping algorithm, the CGPs of the incoming word are then compared against reference templates, which consist of previously created strings of CGPs. The identity of the reference template which is closest in distance to the incoming test word is chosen as the estimate of the test word.

关键词： Automatic speech recognition Speech recognition Testing Hardware Speech processing Natural languages linear predictive coding Loudspeakers Time factors Heuristic algorithms

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A telephone speech recognition system using word spotting technique based on statistical measure

A telephone speech recognition system using word spotting te...

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： T. Kimura K. Niyada S. Hiraoka S. Morii T. Watanabe Matsushita Research Institute Tokyo Inc. Kawasaki Japan

A telephone speech recognition system for an isolated word which has sufficient performance in practical use is reported here. Word spotting technique is applied to the system so as to keep it relatively immune from noise. Word spotting is performed by a new time normalization algorithm based on linear time distortion pattern matching method named COLM(Continuous linear Matching). The method includes only simple operations such as look up tables and additions. The entire system is implemented in a small board including a single digital signal processor. An experiment proved 96.4% average recognition rate. The experiment was carried out using 10 Japanese numerals pronounced by 240 males and females through telephone lines.

关键词： Telephony Speech recognition Signal processing algorithms Pattern matching linear predictive coding Cepstral analysis Vectors Covariance matrix Gaussian distribution Digital signal processors

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Evaluation of a glottal ARMA model of speech production

Evaluation of a glottal ARMA model of speech production

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： A.P. Lobo W.A. Ainsworth Department of Communication and Neuroscience University of Keele Keele Staffordshire UK INRS-Télécommunications Nuns Island QUE Canada

In many speech analysis/synthesis schemes, the source for excitation for voiced speech is a train of impulses. However, the quality of speech that has been attained due to the introduction of a dynamically varying source, e.g., parametric source model, multipulse excitation has been found to be better than that using impulse excitation. The authors describe a pitch-synchronous glottal autoregressive moving average analysis/synthesis scheme in which a parametric voice source model is used in jointly estimating the source and vocal tract parameters from the speech signal. This method is then compared with closed-phase linear predictive coding (LPC), wherein covariance analysis is required to be carried out in the closed glottis interval, and with robust LPC, in which the analysis frame is insensitive to glottal closure. The superiority of the proposed scheme over the latter two methods is shown in terms of better formant/bandwidth tracking capability and efficiency of resynthesis.< >

关键词： Speech analysis Autoregressive processes Speech synthesis linear predictive coding Time varying systems Filtering Signal generators IIR filters Neuroscience Signal analysis

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Multiple frame block quantisation of line spectral frequencies using Gaussian mixture models

Multiple frame block quantisation of line spectral frequenci...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： K.K. Paliwal S. So School of Microelectronic Engineering Griffith University Brisbane Australia

In this paper, we present a Gaussian mixture model-based block quantiser for coding line spectral frequencies that uses multiple frames and mean squared error as the quantiser selection criterion. The efficiency gained from jointly coding multiple frames permits the use of the mean squared error distortion (MSE) criterion rather than the computationally expensive spectral distortion. The proposed coder encompasses improvements in both distortion performance and complexity with transparency achieved at 23 bits per frame when coding two frames jointly or 21 bits per frame when coding 3 frames.

关键词： Quantization Frequency linear predictive coding Speech coding Bit rate Distortion measurement Microelectronics Vectors Signal synthesis Speech synthesis

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