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检索条件"主题词=Linear Predictive coding"

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An MPEG-4 Twin-VQ based high quality audio codec design

An MPEG-4 Twin-VQ based high quality audio codec design

引用

IEEE Workshop on Signal Processing Systems (SIPS)

作者： Yin-Tsung Hwang Nan-Jung Liu Ming-Chang Tsai Department of Electronic Engineering National Yunlin University of Science and Technology Yunlin Taiwan

This paper presents a high quality audio codec design based on a transform-domain weighted interleave vector quantization (Twin-VQ) scheme adopted in the MPEG-4 audio standard. Three novel techniques are employed in this scheme to compress the data, ie, (1) flattening of MDCT coefficients by the spectrum of linear predictive coding (LPC) coefficients; (2) further flattening of MDCT coefficients by the Bark envelope; and (3) weighted interleave vector quantization. This paper examines the related design issues in implementing an efficient Twin-VQ codec. Fast computation algorithms are derived for the computationally intensive modules. Design parameters of each module are determined and the codebooks for weighted interleave vector quantization are constructed. Experimental results show that the designed codec can compress natural audio efficiently and reproduce high quality outputs.

关键词： MPEG 4 Standard Codecs linear predictive coding Vector quantization Bit rate Audio coding Sampling methods Data engineering Design engineering Code standards

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Speech enhancement via frequency bandwidth extension using line spectral frequencies

Speech enhancement via frequency bandwidth extension using l...

引用

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

作者： S. Chennoukh A. Gerrits G. Miet R. Sluijter Philips Research Laboratories Eindhoven Netherlands Philips Consumer Communications Le Mans France

This paper contributes to narrowband speech enhancement by means of frequency bandwidth extension. A new algorithm is proposed for generating synthetic frequency components in the high-band (i.e., 4-8 kHz) given the low-band ones (i.e., 0-4 kHz) for wide-band speech synthesis. It is based on linear prediction (LPC) analysis-synthesis. It consists of a spectral envelope extension using efficiently line spectral frequencies (LSF) and a bandwidth extension of the LPC analysis residual using a spectral folding. The low-band LSF of the synthesis signal are obtained from the input speech signal and the high-band LSF are estimated from the low-band ones using statistical models. This estimation is achieved by means of four models that are distinguished by means of the first two reflection coefficients obtained from the input signal linear prediction analysis.

关键词： Speech enhancement Bandwidth Speech synthesis linear predictive coding Narrowband Frequency synthesizers Wideband Signal synthesis predictive models Reflection

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A hybrid coder based on a new phase model for synchronization between harmonic and waveform coded segments

A hybrid coder based on a new phase model for synchronizatio...

引用

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

作者： N. Katugampala A. Kondoz Center for Communication Systems Research University of Surrey Guildford UK

This paper presents a hybrid coder with a new phase model to synchronize harmonic and waveform coded segments, with a target bit rate of 4 kbps. The coder also employs a new technique based on analysis by synthesis to distinguish between stationary and transitional segments. Harmonic excitation is synchronized with the LPC residual by transmitting the location of the pitch pulse closest to the frame boundary and a phase value that represents the shape of the corresponding pitch pulse. The performance of this phase model and the classification technique is evaluated using a hybrid coder. The coder has three modes: scaled white noise excitation colored by LPC for unvoiced, ACELP for transitions, and harmonic excitation for stationary segments. Subjective listening tests show that the coder produces good quality speech and the switching between the modes is transparent.

关键词： Speech synthesis Pulse shaping methods Bit rate Shape linear predictive coding White noise Vocoders Electronic mail Testing Speech enhancement

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Ways of reducing emissions at coal-cleaning plant

Ways of reducing emissions at coal-cleaning plant

引用

International Scientific and Practical Conference of Students, Post-graduates and Young Scientists Modern Technique and Technologies

作者： L.I. Molodezhnikova O.V. Porutchikova V.E. Trofimov Tomsk Polytechnic University

The analysis of the harmful wastes of the coal preparation mills is presented in this work. The main sources of these wastes are drying installations. Measures for protection of nature were made and measures for incre... 详细信息

ISBN: (纸本)0780363469

关键词： Milling machines Nitrogen Fuels Combustion Furnaces Atmosphere linear predictive coding Ash Flue gases Power engineering

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eX-CELP: a speech coding paradigm

eX-CELP: a speech coding paradigm

引用

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

作者： Yang Gao A. Benyassine J. Thyssen Huan-yu Su E. Shlomot Conexant Systems Inc. USA

This paper presents the core technology of novel enhancements to achieve toll quality at 4 kbps, our experiments and test results traditional CELP coding, coined eXtended CELP (eX-CELP). It is showed that this technology is also successful and suitable for centered on a combined and selective usage of closed-loop/open-both high and medium bit rates. Fig. I and Fig.2 illustrate the basic loop approach, and variant algorithm structure concept. The above structure of the eX-CELP encoder and decoder. two concepts are complemented by new features and refined One of the main themes of the eX-CELP technology is the existing technologies. The eX-CELP paradigm was used in judicious combination of the closed-loop approach and the open-several speech coding systems. It is the core technology of the loop approach, together with a careful selective usage of them. recently chosen candidate for the 3G-CDMA speech codec This mechanism is coined COLA, and its main objective is to standard. It was the best candidate for ITU-T 4 kbps codec intelligently employ the most appropriate approach for different qualification test, and became the basis technology for a types of input signals in order to preserve the perceptually consortium candidate to the ITU-T 4 kbps speech coding important contents. Another important feature in the eX-CELP competition.

关键词： Speech coding Bit rate Delay Speech codecs Code standards Qualifications System testing Decoding Appropriate technology linear predictive coding

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Distributed speech recognition with codec parameters

Distributed speech recognition with codec parameters

引用

IEEE Workshop on Automatic Speech Recognition and Understanding

作者： B. Raj J. Migdal R. Singh Mitsubishi Electric Research Laboratories Inc. Cambridge MA USA Massachusetts Institute of Technology Cambridge MA USA Carnegie Mellon University Pittsburgh PA USA

Communication devices which perform distributed speech recognition (DSR) tasks currently transmit standardized coded parameters of speech signals. Recognition features are extracted from signals reconstructed using these on a remote server. Since reconstruction losses degrade recognition performance, proposals are being considered to standardize DSR-codecs which derive recognition features, to be transmitted and used directly for recognition. However, such a codec must be embedded on the transmitting device, along with its current standard codec. Performing recognition using codec bitstreams avoids these complications: no additional feature-extraction mechanism is required on the device, and there are no reconstruction losses on the server. We propose an LDA-based method for extracting optimal feature sets from codec bitstreams and demonstrate that features so derived result in improved recognition performance for the LPC, GSM and CELP codecs. For GSM and CELP, we show that the performance is comparable to that with uncoded speech and standard DSR-codec features.

关键词： Speech recognition Codecs Feature extraction GSM Propagation losses Degradation Performance loss Proposals Code standards linear predictive coding

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HMM/NN hybrids for continuous speech recognition

HMM/NN hybrids for continuous speech recognition

引用

National Conference on Radio Science (NSRC)

作者： O.A. Abdel Alim N. Elboghdadly N.M. El Shaar Faculty of Engineering EE . Department Aexaadria University Egypt

ISBN: (纸本)9775031680

The main goal of automatic speech recognition (ASR) is to produce a machine which will recognize accurately normal human speech from any speaker. The recognition system may be classified as speaker-dependent or speaker-independent and isolated-word or connected word. There are three approaches to research in automatic speech recognition (ASR); the acoustic-phonetic approach, the pattern recognition approach, and the database statistical approach. Two approaches of this kind: hidden Markov model (HMM) and artificial neural network (ANN) are presented in this paper.

关键词： Hidden Markov models Neural networks Speech recognition Automatic speech recognition Pattern recognition linear predictive coding Artificial neural networks Speech analysis Spectral analysis Loudspeakers

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Multi stage matrix quantization for very low bit rate speech coding

Multi stage matrix quantization for very low bit rate speech...

引用

IEEE Workshop on Signal Processing Advances in Wireless Communications (SPAWC)

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)0780367200

Efficient quantization of linear predictive coding (LPC) filter coefficients play an essential role in very-low-bit-rate speech coding systems. This paper examines a new suboptimal matrix quantization scheme for LPC parameters, called multi-stage matrix quantization (MSMQ), which operates at bit rates between 400 and 800 bit/s. With the new matrix quantization method, using a 22.5 ms LPC analysis frame, spectral distortion about 1 dB is achieved at 800 bit/s. In the proposed coder, line spectral frequency (LSF) parameters of multiple consecutive frames are grouped into a superframe and jointly quantized. The new residual LSF vector quantization scheme gives a bit rate reduction in MSMQ without any additional complexity or storage. The new MSMQ leads into several schemes of various computational complexity/storage characteristics.

关键词： Bit rate Speech coding linear predictive coding Nonlinear filters Vector quantization Spectral analysis Frequency Large scale integration Computational complexity Wireless communication

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A class of reject-first possibilistic classifiers

A class of reject-first possibilistic classifiers

引用

Conference of the North American Fuzzy Information Processing Society - NAFIPS

作者： L. Mascarilla C. Frelicot L3i Université de La Rochelle La Rochelle France

The authors address the problem of defining a general class of reject-first possibilistic classifiers. It relies on fuzzy XOR operators based on dual triples (t-norm, t-conorm, complement). Such a classifier operates in two sequential steps. It starts with testing for exclusive classification by thresholding the fuzzy XOR combination of membership degrees to the different classes. If the pattern has to be rejected, the classifier continues by testing for the kind of rejection encountered (i.e. ambiguity or distance) using another threshold on the fuzzy OR combination.

关键词： Testing Labeling Pattern classification Hypercubes Mathematical model linear predictive coding Input variables

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Voice over IP with JVOIPLIB and JRTPLIB

Voice over IP with JVOIPLIB and JRTPLIB

引用

Conference on Local Computer Networks (LCN)

作者： J. Liesenborgs W. Lamotte F. Van Reeth Expertisecentrum Digitale Media Limburgs Universitair Centrum Diepenbeek Belgium

ISBN: (纸本)0769513212

Voice over IP (VoIP) can be used in a wide variety of applications, all having different requirements. We present JVOIPLIB and JRTPLIB, a VoIP library and an RTP library respectively. Together they make it possible to easily add VoIP to various types of applications. Both libraries are written in an object-oriented style in C++, are open-source and are both very extensible. Several measures have been taken to allow good synchronization between the communicating parties.

关键词： Internet telephony Virtual environment Delay linear predictive coding Software libraries Open source software Speech processing Signal processing Jitter Degradation

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