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International Conference on Electrical, Electronics, Communication, Computer, and Optimization Techniques (ICEECCOT)

作者： Unnibhavi, Anand H. Jangamshetti, D. S. Basaveshwara Engn Coll Dept E&C Bagalkot Karnataka India Basaveshwara Engn Coll Dept E&E Bagalkot Karnataka India

ISBN: (纸本)9781538623619

Automatic speech recognition (ASR) is most promising area which is of special attention today. This paper proposes speech recognition system for Kannada vowels. The proposed framework consists of preprocessing unit and classification unit. The preprocessing unit segments the speech signal into proper frames and extracts the features using linear predictive coding (LPC). The Kannada vowels are classified into the proper classes in the classification unit where Euclidean distance is used. The recognition accuracy obtained is about 40 %.

关键词： linear predictive coding Mel frequency cepstral coefficients Isolated Word Recognition

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Digital waveguide mesh modeling of the vocal tract acoustics

Digital waveguide mesh modeling of the vocal tract acoustics

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IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

作者： Mullen, J Howard, DM Murphy, DT Univ York Dept Elect York YO10 5DD N Yorkshire England

ISBN: (纸本)0780378504

The Digital Waveguide Mesh is a technique used in the modelling of room acoustics and musical instruments. This paper details a project that applies the theory of waveguide mesh acoustic modelling to the production of human-like vowel sounds. A 2D software mesh model is created that approximates the shape of the vocal tract in different vowel positions, and a glottal flow input is applied. The resulting signal bears similar resonant frequencies or formants to that of recorded speech. Recommendations are made towards extending the model to include some of the more complex features of the mouth, potentially constructing an acoustical model of the human vocal tract capable of creating speech sounds of increased naturalness.

关键词： Acoustic propagation Acoustic waveguides Acoustic waves Instruments Lifting equipment linear predictive coding Shape Speech synthesis Waveguide components Waveguide junctions

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ULTRA-LOW LATENCY AUDIO coding BASED ON DPCM AND BLOCK COMPANDING

ULTRA-LOW LATENCY AUDIO CODING BASED ON DPCM AND BLOCK COMPA...

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14th International Workshop on Image Analysis for Multimedia Interactive Services (WIAMIS)

作者： Simkus, Gediminas Holters, Martin Zoelzer, Udo Helmut Schmidt Univ Univ Fed Armed Forces Dept Signal Proc & Commun Hamburg Germany

ISBN: (纸本)9781479908332

A low delay audio coding scheme with good perceptual audio quality for a desired limited bit rate is presented. The proposed audio coding scheme is based on differential pulse code modulation (DPCM) and block companded (BC) quantization. Prediction is realized as a FIR filter in lattice structure. DPCM performs in feedback manner, therefore no transmission of prediction filter coefficients is needed. The incorporation of BC quantization in the DPCM relies on a prediction error recalculation scheme. The use of BC quantization in the DPCM allows to accurately follow the prediction error signal. This improves the perceptual audio quality significantly compared to a plain DPCM with an adaptive quantizer. An algorithmic delay below a half millisecond and an overhead of less than a half bit per sample is introduced due to the short fixed block length of the BC quantizer. Therefore, a real time bidirectional audio application is achievable.

关键词： Audio coding linear predictive coding block compander

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How steady are vowel steady-states?

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CLINICAL LINGUISTICS & PHONETICS 1998年第5期12卷 405-415页

作者： Blomgren, M Robb, M Univ Connecticut Dept Commun Sci Storrs CT 06269 USA

The duration of vowel steady-states (VSS) was examined acoustically in the speech production of 40 normal young adults. VSS was assessed according to formant frequency changes in sustained /i/ productions and consonant + /i/ + /d/(/Cid/) productions. The duration of the VSS was measured for the first and second formants (F1 and F2) by incorporating a fixed rate-of-change criterion. Results indicated no significant differences in VSS duration according to gender or vowel context. VSS duration based on F1 was significantly longer than F2 VSS duration. The duration of VSS was also found to be correlated to the overall vowel duration in /Cid/ contexts. Discussion focuses on the analysis and application of VSS in acoustic studies of normal and disordered speech production.

关键词： acoustics vowel steady-state formant frequency linear predictive coding

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New Feature Extraction Methods Using DWT and LPC for Isolated Word Recognition

New Feature Extraction Methods Using DWT and LPC for Isolate...

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IEEE Region 10 Conference (TENCON 2008)

作者： Nehe, N. S. Holambe, R. S. SGGS Inst Engn & Technol Nanded MS India

ISBN: (纸本)9781424424085

In this paper a new feature extraction methods, which utilize reduced order linear predictive coding (LPC) coefficients for speech recognition, have been proposed The coefficients have been derived from the speech frames decomposed using Discrete Wavelet Transform (DWT). In the literature it is assumed that the speech frame of size 10 msec to 30 msec is stationary, however, in practice different parts of the speech signal may convey different amount of information (hence may not be perfectly stationary). LPC coefficients derived from subband decomposition of speech frame provide better representation than modeling the frame directly. Experimentally it has been shown that, the proposed approaches provide effective (better recognition rate) and efficient (reduced feature vector dimension) features. The speech recognition system using the continuous Hidden Markov Model (HMM) has been implemented. The proposed algorithms are evaluated using NIST TI-46 isolated-word database.

关键词： Feature Extraction linear predictive coding Discrete Wavelet Transform

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Tree-based Context Clustering Using Speech Recognition Features for Acoustic Model Training of Speech Synthesis 12

Tree-based Context Clustering Using Speech Recognition Featu...

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International Conference on Electrical Engineering/Electronics, Computer, Telecommunications and Information Technology (ECTI-CON)

作者： Chanjaradwichai, Supadaech Suchato, Atiwong Punyabukkana, Proadpran Chulalongkorn Univ Fac Engn Dept Comp Engn Bangkok Thailand

ISBN: (纸本)9781479979615

Tree based context clustering processes reduce the sizes of acoustic models of Hidden Markov Model (HMM) speech synthesis systems as well as eliminate problems arising from unseen sound units. Representations of speech units in speech synthesis systems are often LPC or MCEP features whose characteristics promote speech reconstruction rather than discrimination among different sound units. In this paper, MFCC features, successfully utilized in speech recognition, were selected as features for generating context clustering trees applied to LPC/MCEP-based speech synthesis. On average, the collective size of acoustic models was 29% smaller than ones of typical cases while spectral features generated from a speech synthesis system using each type of clustering trees did not significantly deviate from features extracted from actual spoken utterances. Applying MFCC-based clustering tree did not significantly affect the resulting pitch and duration models of the system. We concluded that MFCC-based clustering tree can reduce the overall size of acoustic models while synthetic sound quality is maintained.

关键词： speech synthesis context clustering mel-frequency cepstral coefficients linear predictive coding Mel Cepstrum

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JOINT ESTIMATION OF SHORT-TERM AND LONG-TERM PREDICTORS IN SPEECH CODERS

JOINT ESTIMATION OF SHORT-TERM AND LONG-TERM PREDICTORS IN S...

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Giacobello, Daniele Christensen, Mads Graesboll Dahl, Joachim Jensen, Soren Holdt Moonen, Marc Aalborg Univ Dept Elect Syst ES MISP Aalborg Denmark Katholieke Univ Leuven Dept Elect Engn ESAT SCD Leuven Belgium

ISBN: (纸本)9781424423538

In low bit-rate coders, the near-sample and far-sample redundancies of the speech signal are usually removed by a cascade of a short-term and a long-term linear predictor. These two predictors are usually found in a sequential and therefore suboptimal approach. In this paper we propose an analysis model that jointly finds the two predictors by adding a regularization term in the minimization process to impose sparsity constraints on a high order predictor. The result is a linear predictor that can be easily factorized into the short-term and long-term predictors. This estimation method is then incorporated into an Algebraic Code Excited linear Prediction scheme and shows to have a better performance than traditional cascade methods and other joint optimization methods, offering lower distortion and higher perceptual speech quality.

关键词： Speech analysis linear predictive coding

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Connectivity-guided predictive compression of dynamic 3d meshes

Connectivity-guided predictive compression of dynamic 3d mes...

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IEEE International Conference on Image Processing (ICIP 2006)

作者： Stefanoski, Nikolce Ostermann, Joern Leibniz Univ Hannover Ist Informationsverarbeitung Appelstr 9A D-30167 Hannover Germany

ISBN: (纸本)9781424404810

We introduce an efficient algorithm for real-time compression of temporally consistent dynamic 3D meshes. The algorithm uses mesh connectivity to determine the order of compression of vertex locations within a frame. Compression is performed in a frame to frame fashion using only the last decoded frame and the partly decoded current frame for prediction. Following the predictive coding paradigm, local temporal and local spatial dependencies between vertex locations are exploited. In this framework we present a novel angle preserving predictor and evaluate its performance against other state of the art predictors. It is shown that the proposed algorithm improves up to 25% upon the current state of the art for compression of temporally consistent dynamic 3D meshes.

关键词： mesh compression dynamic 3D mesh coding animation compression prediction methods linear predictive coding non-linear predictive coding

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An Intelligibility Evaluation of Several linear Prediction Vocoder Modifications

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IEEE Transactions on Acoustics, Speech, and Signal Processing 1978年第5期26卷 424-435页

作者： Wong, David Y. Markel, John D. Signal Technology Inc. Santa Barbara CA 93101 United States

Several linear prediction vocoder modifications and an evaluation of their effects on intelligibility are presented. Diagnostic rhyme test (DRT) comparisons among 1) the fixed filter order, fixed analysis frame rate vocoder, 2) various modified vocoders, and 3) the input speech, are implemented using two speakers, seven listeners, and a selected set of word pairs reflecting six phonemic attribute contrasts. Reducing the filter order from ten to four for unvoiced speech frames analyzed at a rate of 44.4 frames/s produces no significant decrease in the scores for all of the six features tested by the DRT. Updating the unvoiced analysis frames with shorter windows at twice the frame rate (88.8 frames/s) leads to a significant improvement in the scores for the sustention feature. Lower order coefficient representation and faster analysis update during unvoiced speech improves the sustention feature with little degradation to the other features, and almost no increase to transmission rate. Results of the DRT evaluation and considerations for implementing the test within ordinary laboratory facilities are discussed. © 1978 IEEE.

关键词： Vocoders Speech analysis linear predictive coding Filters Degradation Economic forecasting Laboratories System testing Large-scale systems Oral communication

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Robust Quantization of LPC Parameters for Speech Communication Over Noisy Channel

Robust Quantization of LPC Parameters for Speech Communicati...

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2nd International Conference on the Applications of Digital Information and Web Technologies

作者： Merouane, Bouzid USTHB Elect Fac Speech Commun & Signal Proc Lab Algiers 16111 Algeria

ISBN: (纸本)9781424444564

In this paper, an optimized trellis coded vector quantization (OTCVQ) system designed for efficient and robust coding of LSF spectral parameters is presented. The aim of this system, called at the beginning "LSF-OTCVQ Encoder", is to achieve a low bit rate transparent quantization of the FS1016 LSF parameters. Once the effectiveness of the LSF-OTCVQ encoder was proven in the case of ideal transmissions over noiseless channel, we were interested after in the improvement of its robustness for real transmissions over noisy channel. To protect implicitly the transmission indices of the LSF-OTCVQ encoder incorporated in the FS1016, we used a joint source-channel coding carried out by the channel optimized vector quantization.

关键词： Robustness Quantization linear predictive coding Oral communication Decision support systems Fiber reinforced plastics Virtual reality

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