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检索条件"主题词=Linear predictive coding"

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A new Chinese text-to-speech system with high naturalness

A new Chinese text-to-speech system with high naturalness

引用

International Conference on Spoken Language, ICSLP

作者： Ren-Hua Wang Qinfeng Liu Difei Tang University of Science and Technology Hefei China

ISBN: (纸本)0780335554

Introduces a new Chinese text-to-speech system that produces far more natural and intelligible synthesized speech than existing systems. There are two distinguishing features in this system. One is the perfect prosodic rules that are made up from the linguistic knowledge and statistical results from a standard Chinese database. These rules are successfully used to modify the elemental synthesis units to get high naturalness while concatenating them into a sentence. The other feature is that the log-magnitude approximate (LMA) filter is used as the synthesis filter in the system. With the LMA filter, the prosody of the synthesized speech can be modified in a wide range while maintaining high intelligence and naturalness. In this paper, the formulated prosodic rules are presented and the LMA filter-based speech synthesis is described in detail.

关键词： Speech synthesis Control system synthesis Databases Filters Synthesizers Natural languages linear predictive coding Electrical capacitance tomography Libraries

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Towards optimal LPC spectral parameters estimation in standard CELP 4800 b/s speech coder

Towards optimal LPC spectral parameters estimation in standa...

引用

IEEE Digital Signal Processing Workshop

作者： M. Markovic M. Obradovic Institute of Applied Mathematics and Electronics Belgrade Serbia Faculty of Technical Sciences University of Novi Sad Novi Sad Yugoslavia

The influence of nonuniform scalar quantization and the specific interpolation procedure of LSP parameters, as proposed in USA FED STD 1016 CELP 4800 b/s speech coder, to the LPC spectra of speech obtained by using four standard LPC methods: autocorrelation with Hamming window, covariance, modified covariance and lattice, is considered. Comparative experimental analysis is done referring to the results of three different spectral measures related to the RMS LOG spectral measure: likelihood ratio, cosh measure and cepstral distance. The experimental analysis shows that, in the sense of the global LPC spectral representation, the best results are obtained by using the modified covariance method. In this way, the orientation to the autocorrelation method with Hamming window, proposed in USA FED STD 1016 CELP coder, is not justified.

关键词： linear predictive coding Parameter estimation Code standards Cepstral analysis Quantization Autocorrelation Interpolation Lattices Speech coding Filters

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Interaction of speech disorders with speech coders: effects on speech intelligibility

Interaction of speech disorders with speech coders: effects ...

引用

International Conference on Spoken Language, ICSLP

作者： D.G. Jamieson L. Deng M. Price V. Parsa J. Till Hearing Health Care Research Unit University of Western Ontario London ONT Canada Department of Electrical and Computer Engineering University of Waterloo Waterloo ONT Canada Audiology and Speech Pathology Service VA Medical Center Long Beach CA USA

Modern speech coding schemes have been developed to address the demand for economical spoken language telecommunication of acceptable quality. A variety of speech coding algorithms have been described, which compress speech to facilitate efficient transmission of spoken language over communication networks ((J.R. Deller Jr., 1993; P.E. Papamichalis, 1987). Most such speech coding algorithms are lossy in the sense that the processed speech is not identical to the original speech. As a result, some distortion is invariably introduced with any lossy speech coding strategy. For this reason, candidate coders undergo detailed evaluation to ensure that the associated speech output is of acceptable quality (S.R. Quackenbush et al., 1988). Three different coding algorithms were investigated relative to unprocessed speech: the Codebook Excited linear Prediction (CELP), the Global System for Mobile Communications (GSM) algorithm which is a standardized speech coding algorithm in Europe, and the linear predictive coding (LPC) algorithm. The specific coding schemes evaluated were MatLab implementations of NSA FS-1015 LPC-l0e; NSA FS-1016 CELP-v3.2; and ETSI GSM (A. Spanias, 1995). One of the goals of this study was to quantify the coding distortion using objective measures and to correlate these measures with speech intelligibility and subjective quality data, in the hope of identifying one or more measures that can predict the subjective results.

关键词： Speech coding GSM Distortion measurement Natural languages Speech processing linear predictive coding Economic forecasting Communication networks Speech analysis Europe

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WIDEBAND SPEECH COMPRESSION USING CELP AND WAVELET TRANSFORM

WIDEBAND SPEECH COMPRESSION USING CELP AND WAVELET TRANSFORM

引用

1996 3rd International Conference on Signal Processing(ICSP’96)

作者： Yao Li Ru-wci Dai Jian-jian Feng Yong-gui Xie A1 Lab. Automation Inst.C.A.S. Radio-Electronics Dep. Beijing Normal University

ISBN: (纸本)0780329120

＜正＞The paper presents a new scheme for wideband speech coding in sub-bands, where the wideband speech signal is sampled at 16kH/ and 16 bits / sample,and the subband decomposition is performed using discrete orthogonal *** wavelet transform uses bandpass filters to split the speech signal into two frequency *** order to attain low bit rates,high quality speech and less computational complexity,each subband is then coded independently with improved CELP algorithms at low frequency-band and adaptive differential pulse code modulation （ADPCM ） coding at high frequency-band. The overall effect is to produce a wideband speech coder that has almost the same computational complexity as a narrowband CELP coding algorithms and the coding rate at 12.8kbit/s.

关键词： data compression speech coding wavelet transforms transform coding signal sampling linear predictive coding differential pulse code modulation computational complexity band-pass filters filtering theory wideband speech compression CELP wavelet transform speech coding signal sampling subband decomposition discrete orthogonal wavelets bandpass filters low bit rate high quality speech computational complexity adaptive differential pulse code modulation ADPCM subband coding 12.8 kbit/s

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New Method for the Extraction of the Gate Voltage Dependence of the Series Resistance in CMOS Transistors

New Method for the Extraction of the Gate Voltage Dependence...

引用

European Conference on Solid-State Device Research (ESSDERC)

作者： H. Brut G. Ghibaudo A. Juge SGS-THOMSON Microelectronics Crolles France LPCS ENSERG Grenoble France

The resistance based extraction method for the determination of effective channel length and series resistance behaviour with gate bias is critically analysed. The impossibility of extracting the gate voltage variations of these parameters concurrently is demonstrated. Then both new approach and parameter extraction procedure are given and experimentally applied to several technologies from 1.2μm down to 0.35μm technologies.

关键词： Voltage CMOS technology Parameter extraction MOS devices Microelectronics linear predictive coding Design engineering Semiconductor device modeling Circuit synthesis CMOS process

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Low-delay subband CELP coding for wideband speech

Low-delay subband CELP coding for wideband speech

引用

IEEE Region 10 International Conference TENCON

作者： W. Tian W.C. Wong C. Tsao Department of Electrical Engineering National University of Singapore Singapore

In this paper we propose low-delay techniques for coding 7 kHz speech using subband coding together with code-excited linear predictive coding (CELP). We compare the use of separate as well as joint index codebooks. Specifically, the joint index subband CELP (JISBC) algorithm is found to provide good quality with processing delay in the range of 2.375 ms to 3.375 ms at corresponding bitrates of 16 kbit/s to 8 kbits/s.

关键词： Wideband Speech coding Codecs Shape Bit rate linear predictive coding Delay Stochastic processes Teleconferencing Internet telephony

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Dual-Pulse CS-CELP: a toll-quality low-complexity speech coder at 7.8 kbit/s

Dual-Pulse CS-CELP: a toll-quality low-complexity speech cod...

引用

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

作者： H. Ohmuro J. Ikedo T. Moriya A. Kataoka S. Hayashi K. Mano NTT Human Interface Laboratories Musashino Tokyo Japan NTT Wireless System Laboratories Yokosuka Kanagawa Japan

Low-cost highly-efficient speech coding is important for personal multimedia communications. In this paper we propose a new low-complexity speech coding method called "Dual-Pulse CS-CELP (DP-CS-CELP)" at 7.8 kbit/s. This method is based on ITU-T G.729. To reduce the complexity, we applied a new excitation model to the random code vectors and simplified the LPC coding, adaptive codebook search, perceptual weighting, and the other structures. The number of operations for this encoder is 3.80 MOPS, which achieves real-time speech coding and decoding on personal computers. Although the efficiency of this coder is a little lower than that of G.729, the MOS listening test showed that the subjective quality was equivalent to or a little better than that of G.726 (32-kbit/s ADPCM).

关键词： Speech coding Finite impulse response filter linear predictive coding Microcomputers Speech synthesis Multimedia communication Decoding Erbium Humans Multimedia systems

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linear prediction based on Teager-Kaiser energy function and application to speech modeling and spectral analysis

Linear prediction based on Teager-Kaiser energy function and...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： P. Rao Chitrapu Dialog semiconductor Limited Parsippany NJ USA

In this paper, the classic problem of determining the optimal linear FIR filter that minimizes an appropriate norm of the filter output is addressed, with the signal norm being the recently introduced Teager-Kaiser Energy norm. Normal equations are derived and conditions for the existence of a solution are given. This theory is then applied to LPC modeling of speech signals and compared with the standard LPC results, where the L/sup 2/-norm of the filter output is minimized. It is observed that the TK-LPC spectra have enhanced formant structure and expanded bandwidth, which could be useful in increasing the perceptual quality of TK-LPC coders. Finally, TK-LPC analysis was applied to spectral estimation, where it is observed that the TK-LPC spectra had higher resolution.

关键词： predictive models Speech analysis Finite impulse response filter Equations Signal analysis linear predictive coding Stochastic processes Autocorrelation Spectral analysis Nonlinear filters

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A model for efficient formant estimation

A model for efficient formant estimation

引用

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

作者： L. Welling H. Ney Lehrstuhl für Informatik VI RWTH Aachen University of Technology Aachen Germany

This paper presents a new method for estimating formant frequencies. The formant model is based on a digital resonator. Each resonator represents a segment of the short-time power spectrum. The complete spectrum is modeled by a set of digital resonators connected in parallel. An algorithm based on dynamic programming produces both the model parameters and segment boundaries that optimally match the spectrum. The main results of this paper are: (1) modeling formants by digital resonators allows a reliable estimation of formant frequencies; (2) digital resonators can be used efficiently in connection with dynamic programming; and (3) a recognition test with formant frequencies results in a string error rate of 4.8% on the adult corpus of the TI digit string database.

关键词： Frequency estimation Dynamic programming linear predictive coding Heuristic algorithms Resonant frequency Testing Databases Speech Polynomials Bandwidth

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Application of speaker modification techniques to phonetic vocoding

Application of speaker modification techniques to phonetic v...

引用

International Conference on Spoken Language, ICSLP

作者： C.M. Ribeiro I.M. Trancoso INESC/ISEL-CEDET INESC-Instituto de Engenharis de Sistemas e Computadores Lisboa Portugal

The goal of the work described in the paper is to develop a very low bit rate vocoding scheme. The vocoder is a typical LPC vocoder, whose parameters are post-processed on a phone-by-phone basis, resulting in a variable bit rate segment vocoder. Given the well known speaker recognizability problems presented by vocoders at such low bit rates, the authors have attempted to integrate a speaker modification method based on altering the formant frequencies and bandwidths of vowel segments. This is done by transmitting the mean value and standard deviation of the radius and angle of the poles corresponding to formant frequencies for each phone. In the decoder stage, the phone index is used to retrieve a set of normalized values from a codebook of 'typical' phones. This set is speaker adapted to preserve the static characteristics (average and standard deviation) but relies in the typical phone to represent the dynamic characteristics such as formant trajectories.

关键词： Vocoders Bit rate Speech recognition Loudspeakers linear predictive coding Frequency Bandwidth Hidden Markov models Statistics Polynomials

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