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检索条件"主题词=Linear predictive coding"

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Maximum-take-precedence ACELP: a low complexity search method

Maximum-take-precedence ACELP: a low complexity search metho...

引用

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

作者： Fu-Kun Chen Jar-Ferr Yang Department of Electrical Engineering National Cheng Kung University Tainan County Taiwan

The ACELP method makes use of multipulse structure to represent the excitation pulses of residual signal. With the purpose of computational complexity reduction, this paper provides the maximum-take-precedence ACELP (MTP-ACELP) search method under the acceptable degradation in performance. Because the maximum of target signal is preferentially compensated, the degradation of performance would be diminished. By predicting the locations of pulses, the computational complexity would be reduced. We not only reduce the possible pulse combinations in the search procedure but also avoid the computation of useless correlation functions before the search procedure. Furthermore, the proposed method is compatible to any ACELP type vocoder, e.g. the G.723.1, G.729, GSM- EFR standards.

关键词： Search methods Computational complexity linear predictive coding Speech coding Stochastic processes Speech analysis Covariance matrix Degradation Speech codecs Signal synthesis

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MVDR based feature extraction for robust speech recognition

MVDR based feature extraction for robust speech recognition

引用

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

作者： S. Dharanipragada B.D. Rao Human Language Technologies IBM Thomas J. Watson Research Center Yorktown Heights NY USA Department of Electrical and Computer Engineering University of California San Diego San Diego CA USA

Describes a robust feature extraction method for continuous speech recognition. Central to the method is the minimum variance distortionless response (MVDR) method of spectrum estimation and a feature trajectory smoothing technique for reducing the variance in the feature vectors. The above method, when evaluated on continuous speech recognition tasks in a stationary and moving car, gave an average relative improvement in WER of greater than 30%.

关键词： Feature extraction Robustness Speech recognition Spectral analysis Gaussian processes Band pass filters Humans Cepstral analysis Mel frequency cepstral coefficient linear predictive coding

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Gauss mixture vector quantization

Gauss mixture vector quantization

引用

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

作者： R.M. Gray Department of Electrical Engineering Information Systems Laboratory Stanford CA USA

Gauss mixtures are a popular class of models in statistics and statistical signal processing because they can provide good fits to smooth densities, because they have a rich theory, and because they can be well estimated by existing algorithms such as the EM (expectation maximization) algorithm. We here extend an information theoretic extremal property for source coding from Gaussian sources to Gauss mixtures using high rate quantization theory and extend a method originally used for LPC (linear predictive coding) speech vector quantization to provide a Lloyd clustering approach to the design of Gauss mixture models. The theory provides formulas relating minimum discrimination information (MDI) for model selection and the mean squared error resulting when the MDI criterion is used in an optimized robust classified vector quantizer. It also provides motivation for the use of Gauss mixture models for robust compression systems for general random vectors.

关键词： Gaussian processes Vector quantization Signal processing algorithms Clustering algorithms linear predictive coding Robustness Statistics Source coding Speech coding predictive models

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Matrix quantization based speech coder at 1200 bps

Matrix quantization based speech coder at 1200 bps

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： S. Ozaydin B. Baykal Department of Electrical & Electronics Engineering Gazi University Ankara Turkey Department of Electrical & Electronics Engineering Middle East Technical University Ankara Turkey

ISBN: (纸本)0780366859

This paper presents a new 1200 bps speech coder designed with a tree searched multi stage matrix quantization scheme. With the new matrix quantization method, spectral distortion about 1 dB is achieved using rates as low as 18 bits/frame. In the proposed coder, LSF parameters of two consecutive frames are grouped into a superframe and jointly quantized. For other speech parameters, quantization is made for each frame. New techniques for improving performance include joint quantization of pitch and voiced/unvoiced/mixed decisions, gain interpolation and residual LSF quantization. For the new matrix quantization based speech coder (MQBC), the listening tests have proven that an efficient and high quality coding has been achieved at bit rate 1200 bps. Test results are compared with the 2400 bps LPC10e coder and the new 2400 bps MELP coder chosen as the new 2400 bps Federal Standard.

关键词： Bit rate Testing Vector quantization Interpolation Speech coding linear predictive coding Design engineering Performance gain Code standards Computer networks

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Formant weighted cepstral feature for LSP-based speech recognition

Formant weighted cepstral feature for LSP-based speech recog...

引用

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

作者： Ho Young Hur Hyung Soon Kim Department of Electronics Engineering Pusan National University busan South Korea

In this paper, we propose a formant weighted cepstral feature for an LSP-based speech recognition system. The proposed weighting scheme is based on the well-known property of LSPs that the speech spectrum has a peak when adjacent LSFs come close. By applying this scheme to the pseudo-cepstrum (PCEP) conversion process (Kim et al. 1993), we can obtain formant weighted or peak enhanced cepstral features. Results of speech recognition experiments using QCELP coder output show that the proposed feature set outperforms the conventional features such as LSP or PCEP. Moreover its performance also exceeds that of the unquantized LPC cepstrum.

关键词： Cepstral analysis Speech recognition Frequency linear predictive coding Cepstrum Weight measurement Bit rate Power harmonic filters Speech processing Speech coding

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Blind deconvolution of reverberated speech signals via regularization

Blind deconvolution of reverberated speech signals via regul...

引用

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

作者： Juan Liu H. Malvar Beckman Institute University of Illinois Urbana IL USA Microsoft Research Limited Redmond WA USA

This paper explores blind deconvolution of reverberated speech signals in microphone array applications. Two regularization approaches are proposed based on available a priori knowledge. The regularized least-squares (LS) approach uses the speech signal characteristics and the lowpass nature of the reverberation channel; and the regularized cross correlation (CR) approach requires more precise knowledge of reverberation which can be obtained through training. The two methods are robust to the presence of noise.

关键词： Deconvolution Microphone arrays linear predictive coding Convolution Reverberation Acoustic noise Speech analysis Chromium Noise robustness Working environment noise

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PLP coefficients can be quantized at 400 bps

PLP coefficients can be quantized at 400 bps

引用

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

作者： W. Gunawan M. Hasegawa-Johnson Department of Electrical and Computer Engineering University of Illinois Urbana-Champaign USA

Previous work in wireless speech recognition has focused on two methods, namely, quantizing recognition features (e.g. MFCC) or performing recognition using speech coding parameters (e.g. LPC). All of this previous research assumes that the communication channel is only large enough to transmit either speech coding parameters or speech recognition parameters. By contrast, we propose that the speech recognition parameters can be quantized at a rate sufficiently low to allow transmission of both speech coding and speech recognition parameters over a standard cellular channel. In particular, the paper shows that the perceptual LPC (PLP) coefficients can be transmitted at 400 bps with an insignificant loss of digit recognition accuracy.

关键词： Speech recognition linear predictive coding Speech synthesis Speech coding Automatic speech recognition Cellular phones Base stations Decoding Humans Speech analysis

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Coupling between 2D and Quantum Confinement Effects in Ultra-Short Channel Double-Gate MOSFETs

Coupling between 2D and Quantum Confinement Effects in Ultra...

引用

European Conference on Solid-State Device Research (ESSDERC)

作者： M. Mouis A. Poncet UMR CNRS 5531 Grenoble France LPM UMR 5511 Villeurbanne France

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A committee of neural networks for automatic speaker recognition (ASR) systems

A committee of neural networks for automatic speaker recogni...

引用

International Joint Conference on Neural Networks (IJCNN)

作者： V. Moonasar G.K. Venayagamoorthy Department of Electronics Engineering M L Sultan Technikon Durban South Africa Applied Computational Intelligence Laboratory University of Missouri Rolla MO USA

ISBN: (纸本)0780370449

This paper describes how the results of speaker verification systems can be improved and made robust with the use of a committee of neural networks for pattern recognition rather than the conventional single-network decision system. It illustrates the use of a supervised learning vector quantization neural network as the pattern classifier. linear predictive coding and cepstral signal processing techniques are utilized to form hybrid feature parameter vectors to combat the effect of decreased recognition success with increased group size (number of speakers to be recognized).

关键词： Neural networks Speaker recognition Automatic speech recognition Pattern recognition Signal processing linear predictive coding Information security Feature extraction Robustness Vector quantization

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Excitation codebook design for coding of the singing voice

Excitation codebook design for coding of the singing voice

引用

IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

作者： Y.E. Kim MIT Media Laboratory Machine Listening Group. Cambridge MA USA

The technique of code excited linear prediction (CELP) has led to the development of voice coding systems that provide toll quality speech at very low bitrates. While speech and singing share many similarities in terms of production, standard speech coding implementations fall far short when transmitting the singing voice. This paper explores the reasons for this discrepancy and suggests new variations on CELP speech coders that specifically enhance the quality of encoded singing for individual singers. These modifications could be used in a low-bitrate singing voice codec which, in conjunction with multi-track structured coding schemes such as MPEG-4 structured audio, could provide a highly compressed yet high-quality representation of a complex audio scene.

关键词： Speech coding Bit rate Resonance Codecs linear predictive coding Filters Transmitters Tongue Teeth Speech enhancement

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