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检索条件"主题词=Linear predictive coding"

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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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New algorithm for spectral smoothing and envelope modification for LP-PSOLA synthesis

New algorithm for spectral smoothing and envelope modificati...

引用

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

作者： F.M. Gimenez de los Galanes M.H. Savoji J.M. Pardo Dpto. Ingeniería Electrónica U. Politécnica de Madrid. E.T.S.I.Telecommunicatión Ciudad Universitaria Madrid Spain Dpto. Electrónica University de Castilla-La Mancha Santander Spain

The final quality of a concatenation synthesis system is directly related to the continuity of the spectrum at the concatenation point. Due to the subjective auditory masking, if we minimize the spectral distortion in the formant frequencies, the quality will increase significantly. We present, along with results concerning pitch marking, an algorithm capable of modifying the LPC envelope in a flexible way which is the heart of a spectral smoothing module for a diphone-based linear prediction pitch-synchronous overlap-add (LP-PSOLA) concatenation system.< >

关键词： Smoothing methods linear predictive coding Signal synthesis Speech synthesis Distortion Signal processing Filtering Heart Frequency measurement Time measurement

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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...

引用

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

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

引用

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 New Algorithm for Calculating LSP Parameters of Speech Signal

A New Algorithm for Calculating LSP Parameters of Speech Sig...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： Juanjuan Li Yibiao Yu Xianyi Rui School of Electronic and Information Engineering Soochow University Suzhou China

Line spectrum pairs (LSP) representation of linear predictive coding coefficients is widely used in speech coding, speech recognition and other domains due to its desirable interpolation and quantization properties. Several methods proposed for calculating LSP parameters have been complicated by high computation complexity. This paper proposed an effective and efficient algorithm APF using Aitken iterative method and polynomial synthesis division. LSP parameters were estimated by obtaining a root of N-order nonlinear equation by Aitken iterative method at first, then decreasing degrees with polynomial synthesis division, and finally calculating quartic equation using Ferrari's solution. Theoretic analysis and experiment results show that the proposed algorithm has not only high precision but also low calculation complexity

关键词： Iterative algorithms Iterative methods Polynomials Nonlinear equations linear predictive coding Speech coding Speech recognition Interpolation Quantization Speech synthesis

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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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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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Low cost voice compression for mobile digital radios

Low cost voice compression for mobile digital radios

引用

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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Generalized autoregressive spectral estimation

Generalized autoregressive spectral estimation

引用

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

作者： J. Tsao W.-J. Shyu Department of Electrical Engineering National Taiwan University Taipei Taiwan

An autoregressive spectral estimation method is developed to reduce the noise effect in prediction coefficient estimation. This method solves the prediction coefficients from a generalized Yule-Walker equation which is formed by the data and its generalized autocorrelation sequence. This method provides several control parameters for the spectral estimator to combat the unmodeled additive noise in the linear least square sense. Through the efficient use of information by this method, data size will be directly helpful in noise suppression.< >

关键词： Autocorrelation linear predictive coding Noise reduction Least squares approximation Difference equations Random processes Additive noise Spectral analysis White noise Councils

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