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检索条件"主题词=linear predictive coding"

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A new approach of LPC analysis based on the normalization of vocal-tract length

A new approach of LPC analysis based on the normalization of...

引用

International Conference on Pattern Recognition

作者： Huang Ze-Zhen Yan Xiag-Jun Department of radio electronics Tsinghua University Beijing China Dept. of Radio Electron. Tsinghua Univ. Beijing China

ISBN: (纸本)0818608781

An approach to linear prediction coefficient (LPC) analysis based on the normalization of vocal-tract length is presented. The approach is of significance for speech recognition of arbitrary speakers. In this approach, the ratio of two vocal-tract lengths corresponding to a new speaker and a reference one is first estimated from the training speech data of several typical vowels. The LPC parameters normalized on this ratio can then be calculated for any speech data. Compared with previous methods of speech parameter normalization, this approach does not need to estimate formant frequencies and is simple and reliable in theory. Limited experiments on the recognition of nine Chinese vowels for four speakers to indicate that this new approach can achieve 5% to 20% improvements of correct recognition rate.< >

关键词： linear predictive coding Autocorrelation Shape Frequency estimation Fourier transforms Speech Testing Zirconium Maximum likelihood estimation Sampling methods

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Recognition of 24 Thai spoken Vowels Using the coefficients of 3rdOrder Polynomial Regression on the Voice Energy and Spectrum of LPC on the Bark Scale

Recognition of 24 Thai spoken Vowels Using the coefficients ...

引用

International Symposium on Wireless Pervasive Computing (ISWPC)

作者： K. Songwatana S. Sriratanapaprat P. Kultap K. Sittiprasert N. Suktangman Faculty of Engineering King Mongkut''s Institute of Technology Ladkrabang Bangkok Thailand

This paper presents a vowel recognition for Thai spoken language. The Thai language consists of 9 short unmixed vowels (a, i,ω,u, o, e, ε, γ, [unk]); 9 long unmixed vowels (aa, ii, ωω, uu, oo, ee, £ εε, γγ, [unk][unk]); 3 short mixed vowels (ia, ωa, ua); and 3 long mixed vowels (i:a:, ω:a:, u:a:). We proposed uses 3-stage decision making: step 1 distinguishes long and short vowels using coefficients of third order polynomial regression of signal energy as features set and 5-NN as classification method; step 2 classifies each voice segment (frame) into 9 basic vowels using 18 critical band intensities as feature set and 9-NN as classification method; finally step 3 decides whether each frame contains mixed or unmixed vowel via thresholding method. This solution is different from the conventional speech recognition mainly because decision making in this method is done for each frame, while conventional speech recognition chooses the best decision for a sequence of frames forming a word or a sentence. Evaluation is done by applying the algorithm to 3024 voice samples of male and female subjects. Each step of the algorithm is evaluated successively.

关键词： Speech recognition Polynomials linear predictive coding Equations Decision making Signal processing Natural languages Lungs Humans Ear

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Line spectrum pair (LSP) and speech data compression

Line spectrum pair (LSP) and speech data compression

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

作者： F. Soong B. Juang Acoustic Research Department Bell Laboratories Murray Hill NJ USA

Line Spectrum Pair (LSP) was first introduced by Itakura [1,2] as an alternative LPC spectral representations. It was found that this new representation has such interesting properties as (1) all zeros of LSP polynomials are on the unit circle, (2) the corresponding zeros of the symmetric and anti-symmetric LSP polynomials are interlaced, and (3) the reconstructed LPC all-pole filter preserves its minimum phase property if (1) and (2) are kept intact through a quantization procedure. In this paper we prove all these properties via a "phase function." The statistical characteristics of LSP frequencies are investigated by analyzing a speech data base. In addition, we derive an expression for spectral sensitivity with respect to single LSP frequency deviation such that some insight on their quantization effects can be obtained. Results on multi-pulse LPC using LSP for spectral information compression are finally presented.

关键词： Data compression linear predictive coding Polynomials Quantization Frequency Speech analysis Dynamic range Information filtering Information filters Stability

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TD-PSOLA versus harmonic plus noise model in diphone based speech synthesis

TD-PSOLA versus harmonic plus noise model in diphone based s...

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

作者： A. Syrdal Y. Stylianou L. Garrison A. Conkie J. Schroeter AT and T Research Laboratories Florham Park NJ USA AT and T Bell Laboratories Inc. Holmdel NJ USA

In an effort to select a speech representation for our next generation concatenative text-to-speech synthesizer, the use of two candidates is investigated; TD-PSOLA and the harmonic plus noise model, HNM. A formal listening test has been conducted and the two candidates have been rated regarding intelligibility, naturalness and pleasantness. Ability for database compression and computational load is also discussed. The results show that HNM consistently outperforms TD-PSOLA in all the above features except for computational load. HNM allows for high-quality speech synthesis without smoothing problems at the segmental boundaries and without buzziness or other oddities observed with TD-PSOLA.

关键词： Speech synthesis Testing Speech enhancement linear predictive coding Speech analysis Acoustic noise Synthesizers Spatial databases Smoothing methods Man machine systems

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Low cost voice compression for mobile digital radios

Low cost voice compression for mobile digital radios

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

作者： J. Omura S. Moon Department of Elec. Eng University of California Los Angeles CA USA Department of Electronics Kyungpook National University Daegu South Korea

This paper presents a new technique for low cost robust voice compression at rate of 9.6 K bps or less. Our approach is based on applying two new concepts in voice compression; State-variable digital biquads and Punctured tree search algorithm. The digital biquads are used for real time spectral analysis of speech. The simplified multipulse excitation is generated by the punctured tree search algorithm that combines the conventional (M,L) tree search algorithm and data compression.

关键词： Costs Digital communication Speech analysis linear predictive coding Noise robustness Spectral analysis Data compression Digital filters Speech synthesis Adaptive filters

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Voice morphing by 3-D waveform interpolation surface and lossless tube area function

Voice morphing by 3-D waveform interpolation surface and los...

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IEEE Convention of the Electrical and ELectronic Engineers in Israel

作者： G. Porat Y. Lavner Signal and Image Processing Laboratory (SIPL) Faculty of Electrical Engineering Technion-Israel Institute of Technology Haifa Israel Computer Science Tel-Hai Academic College Upper Galilee Israel

Voice morphing is the process of gradually transforming the voice of a given speaker to that of another. The ability to change the speaker's individual characteristics and produce high-quality voices can be used in many applications. For example, in multimedia and video entertainment, voice morphing is just like its visual counterpart: while seeing a face gradually changing from one person's to another's, we can simultaneously hear the voice changing as well. Another application could be in forensic voice identification: creating a voice-bank of different pitches, rates, and timbres, to assist in recognition of the suspect's voice. In this study we present a new technique, which enables the production of N intermediate voices that gradually change between voices of two speakers, or one voice signal that changes gradually. This technique is based on two components. One is creating a 3D prototype waveform interpolation (PWI) surface from the residual error ' signal, which is estimated from LPC analysis, to produce a new intermediate excitation signal. The second component is a representation of the vocal tract by a lossless tube area function, and interpolation of the two speakers' parameters.

关键词： Interpolation Surface waves Speech analysis Signal analysis Surface reconstruction Prototypes Filters linear predictive coding Speech coding Signal synthesis

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A text-to-speech system for italian

A text-to-speech system for italian

引用

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

作者： R. Delmonte G. Mian G. Tisato Centro linguistico InterfacoltàIst.Fisica Applicata University Venezia Italy Centro di Sonologia Computazionale Università di Padova Italy

A system for the automatic translation of any text of Italian into naturally fluent speech is presented. The system, planned for use in a reading machine for the blind, is build up around a Phonological Processor (hence FP) and synthesizes speech-by joining LPC coded diphones. The FP maps into prosodic structures the phonological rules of Italian. Structural information is provided by such hierarchical prosodic constituents as Syllable (S), Metrical Foot (MF), Phonological Word (PW), Intonationa T Group (IG). Onto these structures, phonological rules are applied such as the "letter-to-sound" rules, automatic word stress rules, internal stress hierarchy rules indicating secondary stress, external sandhi rules, phonological focus assignment rules, logical focus assignment rules.

关键词： Speech synthesis Internal stresses Computational modeling Natural languages Computer interfaces Speech processing linear predictive coding Foot Vocabulary Graphics

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Using a Vocal Tract Length Related Parameter for Speaker Recognition

Using a Vocal Tract Length Related Parameter for Speaker Rec...

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IEEE Odyssey, The Speaker and Language Recognition Workshop

作者： Stephan Grashey Christian GeiBler Siemens AG Corporate Technology Munich Germany

The use of a parameter related to the vocal tract length of humans is investigated for a speaker recognition system. This parameter is a warping factor for the frequency spectrum and it was determined from a vocal tract length normalization module of an automatic speech recognizer. A decision fusion approach WHS used to combine the output of the stand-alone speaker verification with a separate classifier for the warping factor. The fused system exhibits a relative improvement of 23% compared to the stand-alone system

关键词： Speaker recognition Mel frequency cepstral coefficient Cepstral analysis Automatic speech recognition Finite impulse response filter Feature extraction linear predictive coding Humans Forensics Discrete cosine transforms

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A unified model of threshold voltage, subthreshold slope and interface coupling in thin film SOI MOSFETs

A unified model of threshold voltage, subthreshold slope and...

引用

IEEE SOI-3D-Subthreshold Microelectronics Technology Unified Conference (S3S)

作者： A.M. Ionescu S. Cristoloveanu A. Rusu A. Chovet A. Hassein-Bey University Politehnica of Bucharest Bucharest Romania LPCS ENSERG Grenoble France

Although powerful device simulators are being developed, analytical models are still essential for depicting the underlying physical mechanisms. Recently, attention was paid to a "unified" approach able to account for MOSFET continuous operation from weak to moderate and strong inversion. In this paper, we propose an original model which applies not only to bulk Si and partially depleted SOI MOSFET's, but also to ultrathin film SOI transistors.< >

关键词： Threshold voltage Transistors Transconductance Analytical models MOSFET circuits Thin film devices linear predictive coding Conductive films Numerical simulation Data mining

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Adaptive speech modification by spectral warping

Adaptive speech modification by spectral warping

引用

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

作者： V. Goncharoff S. Chandran Department of Electrical Engineering and Computer Science University of Illinois Chicago Chicago IL USA

Experimental results are reported in the area of speech modification for the purpose of correcting vocal tract resonance disorders. The linear predictive coding (LPC) coefficients and residual corresponding to an input speech signal are calculated, and a combination of nonlinear frequency warping and amplitude scaling is applied to produce modified LPC coefficients and then synthesize an output speech signal from these modified coefficients and the residual signal. The spectral envelope modification parameters are designed to adapt to the first two formant frequencies of the input speech. Cubic polynomial warping functions and amplitude scaling factors are found at vowel formant locations and are interpolated from these points over F1, F2 space. The results are compared with those obtained from frame-by-frame nonlinear spectral warping using a dynamic programming algorithm.< >

关键词： Loudspeakers linear predictive coding Speech enhancement Resonance Filters Autocorrelation Frequency estimation Parameter estimation Postal services Speech processing

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