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检索条件"主题词=Linear Predictive coding"

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Experimental evaluation of an active speech control method

Experimental evaluation of an active speech control method

引用

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

作者： K. Kondo K. Nakagawa Department of Electrical Engineering Yamagata University Yonezawa Yamagata Japan

We experimentally evaluated an active speech control scheme which reduces unnecessary speech radiated into the surrounding space. The intended application of this system, typically cellular phones, does not require speech to be radiated into the surrounding space, but only into the microphone. We previously proposed to reduce speech by generating phase-inverted predicted speech from a secondary loudspeaker. We used LPC recursively to predict samples ahead of the associated processing delay, which could go up to a few milliseconds. First, predicted samples of recorded speech were prepared off line. Then, both the original and the phase-inverted predicted samples were played out simultaneously from two loud speakers. It was found that: 1) speech cancellation of 10 dB is possible, but is highly speaker dependent; 2) the secondary loudspeaker should be oriented in the same direction as the primary source, i.e., the mouth for maximum cancellation.

关键词： Speech analysis Delay Mouth Cellular phones linear predictive coding predictive models Microphones Prediction methods Loudspeakers Position measurement

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Sinusoidal + all-pole modification based spectral smoothing for concatenative speech synthesis

Sinusoidal + all-pole modification based spectral smoothing ...

引用

IEEE International Conference on Natural Language Processing and Knowledge Engineering (NLP-KE)

作者： Heng Kang Wenju Liu National Laboratory of Pattern Recognition Chinese Academy and Sciences Beijing China

The conventional LPC spectral smoothing algorithm causes an evident degradation in the speech quality when the smoothing amount is large. To improve speech quality of the smoothed speech, we proposed a new spectral smoothing algorithm. The source LPC spectral envelopes are first interpolated to generate the smoothed target spectra. Then the sinusoidal + all-pole modification is performed on the source speech to get the spectra of the modified speech which will coincide with the target spectra. Experimental results show that this method can get smooth spectral envelope even if the speech boundaries have large spectral distance. Experimental results prove that this algorithm is effective on avoiding degradation in quality of smoothed speech.

关键词： Smoothing methods Speech synthesis linear predictive coding Frequency Degradation Piecewise linear techniques Signal synthesis Speech analysis Laboratories Pattern recognition

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A study of auditory modeling and processing for speech signals

A study of auditory modeling and processing for speech signa...

引用

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

作者： Woojay Jeon Biing-Hwang Juang School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA

ISBN: (纸本)0780388747

We study a modified version of a computational model of the human peripheral and central auditory system (Wang, K. and Shamma, S.A., 1995; Yang, X. et al., 1992), and examine the validity of its output from two practical perspectives. One considers the well-known Mel-frequency cepstral coefficients (MFCC) as an approximate representation of the physiology-based early auditory processing result. The other allows the derivation of feature vectors from the dimension expanded cortical response of the central auditory system for use in a conventional phoneme recognition task. In addition to confirming the relevancy of the model under an existing statistical speech recognition framework, we conduct a preliminary study of the cortical response in connection with known physiological studies, to find new possibilities in using the auditory model to perform cognitive functions based on a better understanding of the human auditory system. In particular, the cortical response may be a place-coded data set where sounds are categorized according to the regions containing their most distinguishing features. The results of this study encourage us to develop hierarchical, detection-based methods in which this mechanism may be utilized to simulate a variety of human perceptual and cognitive functions.

关键词： Speech processing Signal processing Humans Auditory system Mel frequency cepstral coefficient Computational modeling Speech recognition Automatic speech recognition Cognition linear predictive coding

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Modified silence suppression algorithms and their performance tests

Modified silence suppression algorithms and their performanc...

引用

Midwest Symposium on Circuits and Systems (MWSCAS)

作者： Wei Chu M.O. Ahmad M.N.S. Swamy Department of Electrical and Computer Engineering McGill University Montreal QUE Canada Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

In packet voice communication systems, silence suppression algorithms are employed to achieve bandwidth efficiency by suppressing the inactive part of the speech. In this paper, we consider a Nortel fixed-point silence suppression algorithm to investigate means of reducing the computational complexity with a little or no loss in the performance. In view of the compact structure and superior performance of the existing Nortel algorithm, certain modifications are introduced without altering the structure of the original design. Based on these modifications, two algorithms are proposed. Results from objective and subjective quality tests show that the performance of each of the two modified algorithms is comparable to that of the Nortel algorithm. As for the average computational complexity, simulation tests based on TMS320C5402 DSK show that reduction of about 40% and 22% can be achieved for the two modified algorithms.

关键词： Computational complexity System testing Bandwidth Performance loss Algorithm design and analysis Speech synthesis Engines Noise generators Signal generators linear predictive coding

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linear predictive Analysis for Targeting the Basal Ganglia in Deep Brain Stimulation Surgeries

Linear Predictive Analysis for Targeting the Basal Ganglia i...

引用

International IEEE/EMBS Conference on Neural Engineering, CNE

作者： J. Pukala J.C. Sanchez J.C. Principe F.J. Bova M.S. Okun Department of Electrical and Computer Engineering University of Florida Gainesville FL USA Department of Pediatrics University of Florida Gainesville FL USA Departments of Neurology and Neurosurgery University of Florida Gainesville FL USA

Intra-operative automated recognition of deep brain stimulation (DBS) targets from microelectrode recordings would improve the safety, efficiency, standardization, and accuracy of the surgical procedure. Our approach to the cellular classification problem is from a speech recognition perspective where linear predictive coefficient (LPC) analysis is used to model segments of thalamic and subthalamic nucleus cellular activity. We then cluster the linear prediction coefficients for three Parkinson's disease patients and develop discriminant surfaces with an artificial neural network to generate the target classes. The methods presented here yielded a significant separation of the cell types within a two-dimensional prediction coefficient data space. The results indicate that LPC analysis for DBS targeting warrants additional study for a larger variety of deep brain structures and patients

关键词： Basal ganglia Brain stimulation Surges Satellite broadcasting linear predictive coding Target recognition Microelectrodes Safety Standardization Surgery

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Multitaper Covariance Estimation and Spectral Denoising

Multitaper Covariance Estimation and Spectral Denoising

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Asilomar Conference on Signals, Systems & Computers

作者： N. Erdol T. Gunes Electrical Engineering Florida Atlantic University Boca Raton FL USA

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Comparison of autoregressive parameter estimation algorithms for speech processing and recognition

Comparison of autoregressive parameter estimation algorithms...

引用

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

作者： R.W. Morris J.A. Arrowood M.A. Clements Nexidia Inc. Atlanta GA USA Center for Signal & Image Processing Georgia Institute of Technology Atlanta GA USA

Noise mitigation systems for speech coding and recognition have primarily focused on spectral subtraction techniques due to their well understood behavior and computational simplicity. As computation complexity becomes a smaller constraint, understanding the characteristics of different estimation schemes becomes more important. The merits of two algorithms based on direct estimation of the linear prediction spectrum of a speech signal are explored. These algorithms are maximum likelihood (ML) and minimum mean square error estimation (MMSE) of the autoregressive speech spectrum. The MMSE algorithm is able to improve objective quality effectively at low SNRs while also improving the speech recognition accuracy by 20-30% on the Aurora2 test set at the cost of requiring two orders of magnitude more operations than the ML method. Because of these improvements, autoregressive based algorithms should be considered in the future for noise robust speech processing tasks.

关键词： Parameter estimation Speech processing Speech recognition Iterative algorithms Maximum likelihood estimation Image recognition Speech coding linear predictive coding Signal processing Image processing

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Real-time implementation and optimization of ITU's G.728 on TMS320C64X DSP

Real-time implementation and optimization of ITU's G.728 on ...

引用

IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： M. Akabri B.V. Vahdat K. Nayebi Department of Electrical Engineering Sharif University of Technology Iran

The objective of this paper is to illustrate the details of optimization and real-time implementation of ITU's G.728 on C64x DSPs. First using pseudo codes provided by CCITTs published documents, we implemented the algorithm in C language. This implementation was performed for G and H (12.8 and 9.6 kbit/s) annexes. Next we optimized the written codes for implementation on DSP. At first stage, using different techniques based on DSPs' hardware characteristics, we rewrote and changed the most time-consuming parts of our codes in order to reduce their execution time. At second stage, we balanced the computational load of G.728 coder algorithm by splitting the Durbin's recursion for synthesis filter between different input speech vectors. In each stage, we verified the correctness of our implementation by testing our codes against testing vectors provided by ITU. Applying the above mentioned methods enabled us to optimize the C codes into 22.7 MIPS in worst case. At the end we also implemented the optimized codes in real-time on a DSK6416

关键词： Digital signal processing Signal processing algorithms Speech analysis Hardware Testing linear predictive coding Speech synthesis Compression algorithms Speech codecs Delay

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A system that helps learning Web services

A system that helps learning Web services

引用

International Conference on Advanced Learning Technologies (ICALT)

作者： Jinhyun Ahn Seongbin Park Department of Computer Science Education Korea University Seoul South Korea

A Web service is a software system interoperating with each other in a platform independence manner. COM (component object model) enables components to interoperate with each other. Based on the similarities between a Web service and COM, we provide a system that helps users who get started learning the concept of Web services by experiencing and seeing the process involved in using Web services.

关键词： Web services Simple object access protocol Containers linear predictive coding Computer science education Software systems Programming profession Standards publication Sun Java

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Speech analysis by estimating perceptually relevant pole locations

Speech analysis by estimating perceptually relevant pole loc...

引用

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

作者： V. Atti A. Spanias Department of Electrical Engineering Ira A. Fulton School of Engineering Arizona State University Tempe AZ USA

An approach for estimating the perceptually-relevant pole locations is described. These "perceptual poles" are determined by using an auditory excitation pattern-matching method. The estimated perceptual poles are then used to construct a perceptually-motivated all-pole (PMAP) filter for use in speech analysis/synthesis. The proposed PMAP approach is compared against some of the existing perceptually-based linear prediction (LP) methods, i.e., the perceptual LP and the warped LP. The PMAP approach compares well against the perceptual LP and the warped LP in terms of speech reconstruction quality and estimation of the formant frequencies.

关键词： Speech analysis Psychoacoustic models Frequency estimation Psychology Speech synthesis linear predictive coding Filter bank State estimation predictive models Speech coding

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