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检索条件"主题词=Linear predictive coding"

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A 32-band sub-band/Transform coder incorporating vector quantization for dynamic bit allocation

A 32-band sub-band/Transform coder incorporating vector quan...

引用

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

作者： C. Heron R. Crochiere R. Cox Acoustic Research Department Bell Laboratories Murray Hill NJ USA

In this paper we report on a study of a technique for 32-band subband/transform coding at 16 kb/s. This approach occupies the middle range of algorithm complexities and frequency resolution between that of Sub-Band coding (SBC) and Adaptive Transform coding (ATC). Two designs for 16 kb/s 32-band coders have been simulated on a laboratory computer. The results of informal listening tests indicate that the new designs offer performance comparable to existing ATC techniques while having complexities roughly three times that of existing 4 and 5 band sub-band coders.

关键词： Vector quantization Bit rate linear predictive coding Filter bank Databases Clustering algorithms Iterative algorithms Speech analysis Distortion measurement Filtering

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Design of a toll-quality 4-kbit/s speech coder based on phase-adaptive PSI-CELP

Design of a toll-quality 4-kbit/s speech coder based on phas...

引用

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

作者： L. Mano NTT Human Interface Laboratories Musashino Tokyo Japan

This paper describes the design of a toll-quality 4-kbit/s speech coder based on phase-adaptive PSI-CELP. This adaptation method not only gives pitch periodicity to the random excitation but also synchronizes the basic point of the stored random vector with the pitch phase. We further improve the proposed coder by introducing a backward gain prediction scheme. In subjective evaluation experiments, there is no significant difference between the quality of ITU-T G.726 32-kbit/s coder and that of the proposed 4-kbit/s coder under the conditions of normal and low input levels, tandem connection for clean speech. In noisy environments, there are also no significant differences between G.726 and 4-kbit/s coders from MOS results of the ACR test.

关键词： Speech synthesis Speech analysis Signal synthesis Filters linear predictive coding Speech coding Bit rate Humans Laboratories Testing

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Line spectrum pair (LSP) and speech data compression

Line spectrum pair (LSP) and speech data compression

引用

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...

引用

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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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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Speech coding Based On Vector Quantization For A Tactile Vocoder Hearing Device

Speech Coding Based On Vector Quantization For A Tactile Voc...

引用

Annual International Conference of the IEEE Engineering in Medicine and Biology Society (EMBC)

作者： S. Pourmehdi J. Mouine M. Sawan J. Lajoie F. Duval Electrical Engineering Department Université de Sherbrook Sherbrooke QUE Canada

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Floating-body kinks and dynamic effects in fully depleted SOI MOSFETs

Floating-body kinks and dynamic effects in fully depleted SO...

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

作者： S. Krishnan J.G. Fossum P.C. Yeh O. Faynot S. Cristoloveanu J. Gautier University of Florida Gainesville FL USA LETI DMEL-CENG France LPCS-ENSERG France IBM NY USA

Fully depleted (FD) SOI CMOS is a contender for low-voltage IC applications. However, as FD/SOI MOSFETs are scaled, floating-body effects, which previously seemed insignificant, become important. In this paper, we report kinks in the measured subthreshold current-voltage characteristics of highly scaled FD/SOI MOSFETs, and we describe and model the underlying physical mechanism, showing how it differs from the familiar kink effect in partially depleted (PD) devices. The insight afforded qualifies the meaning of FD/SOI and implies new design issues for low-voltage SOI CMOS.

关键词： MOSFETs Impact ionization Body regions Back linear predictive coding Application specific integrated circuits CMOS integrated circuits Current measurement Current-voltage characteristics Semiconductor device modeling

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Real-time text processing for Italian speech synthesis

Real-time text processing for Italian speech synthesis

引用

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

作者： E. Vivalda S. Sandri C. Miotti CSELT Torino Italy

The paper describes the software architecture of an Italian text-to-speech synthesis system based on the joining of LPC coded diphones. The automatic voice response system is designed according to multichannel and real time criteria. For each output channel, the following operations are performed: pre-processing of the input string of characters, translation into the proper sequence of diphones, generation of prosodic contours and real-time control of a hardware speech synthesizer.

关键词： Text processing Speech synthesis Synthesizers Computer interfaces Signal synthesis Software architecture linear predictive coding Real time systems Character generation Automatic generation control

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A robust parametric method for signal reconstruction

A robust parametric method for signal reconstruction

引用

Asilomar Conference on Signals, Systems & Computers

作者： S.R. Pillai A. Antoniou Department of Electrical Engineering University of Victoria Victoria BC Canada

In the context of signal reconstruction and coding, a new robust parametric formulation to linear predictive coding (LPC) is introduced. The linear prediction filter coefficients are transformed into a set of weighted line frequencies. The positive weights play the dual role of a new set of parameters and simultaneously they exhibit the relative importance of the associated line frequencies. This new representation for LPC is shown to be always stable under quantization.

关键词： Robustness Signal reconstruction linear predictive coding Speech coding Speech analysis Distortion measurement Mean square error methods Polynomials Frequency Redundancy

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