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检索条件"主题词=Linear predictive coding"

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A simple and efficient two-band speech coder at 2.4 kbit/s for real-time implementation on a single low-cost DSP

A simple and efficient two-band speech coder at 2.4 kbit/s f...

引用

Mediterranean Electrotechnical Conference (MELECON)

作者： F. Beritelli S. Casale F. Spataro Istituto di Informatica e Telecomunicazioni Universita Catania Catania Italy

In the new generation of very low bit rate speech coding models, multiband excitation (MBE) represents one of the most promising techniques. The paper proposes a new low-complexity two-band coding scheme capable of producing good speech quality at 2.4 kbit/s. Two new solutions are proposed to guarantee pitch continuity and establish the frequency of transition between the voiced and unvoiced portions of band. During informal listening tests we have seen that the speech coder proposed maintains a natural quality, perfect intelligibility and speaker identity. Finally, to demonstrate the simplicity of the algorithms proposed, the paper presents an evaluation of the speech coder complexity for a real-time implementation on a single low-cost digital signal processor.

关键词： Bit rate Speech coding Frequency Vocoders Speech analysis Speech processing Digital signal processors Speech synthesis linear predictive coding Fourier transforms

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QD-an algorithm for non-linear inverse filtering

QD-an algorithm for non-linear inverse filtering

引用

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

作者： P. Hedelin G. Hult Department of Information Theory Chalmers University of Technology Gothenburg Sweden

Statistical methods for parameter estimation are often based on a Gaussian assumption. In this paper we describe a signal model based on an assumption of uniform distributions. This signal model leads to a simple, non-linear algorithm, called QD, for parameter estimation. The QD algorithm can be applied to adaptive, inverse filtering of speech signals. The technique used is to sequentially estimate the reflection coefficients of a lattice structure. Some of the results of one such application are discussed and compared to those obtained with other methods.

关键词： Filtering algorithms Parameter estimation linear systems Lattices linear predictive coding Maximum likelihood estimation Statistical distributions Gaussian distribution Filters Information theory

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A text-to-speech system for Arabic using neural networks

A text-to-speech system for Arabic using neural networks

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International Joint Conference on Neural Networks (IJCNN)

作者： S.B. Sassi R. Braham A. Belghith Ecole Nat. des Sci. de l'Inf. Tunisia Ecole Nationale des Sciences de I''Informatique Tunisia

Text-to-speech synthesis is of great interest and its applications are several. For this reason, it has interested many researchers for decades. Two methods are usually used: synthesis by rule and synthesis by concatenation of pre-recorded sounds. But these methods have some disadvantages such as difficulty to be adapted to a new speaker or to a new language. Recently, neural networks (NN) have been used with nonconventional problems where a traditional solution seems impossible. Text-to-speech appears as one of these problems. In this field, it has been shown that NN don't work well when they are directly fed with speech samples. Therefore, works have been done to explore and evaluate different parametric forms of speech based on linear predictive coding (LPC), used for training, and found that LSP produced the best results. However, these methods don't take into account residual signal and speech produced was machine-like and not natural. We propose in this paper to drive the NN with codebook-excited linear prediction (CELP), which provides high quality speech, to perform Arabic speech synthesis.

关键词： Speech synthesis Neural networks Network synthesis Natural languages Loudspeakers Speech analysis linear predictive coding Humans Auditory system Books

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Temperature Dependence of Fowler-Nordheim Emission Tunneling Current in MOS Structures

Temperature Dependence of Fowler-Nordheim Emission Tunneling...

引用

European Conference on Solid-State Device Research (ESSDERC)

作者： R. Kies C. Papadas G. Pananakakis G. Ghibaudo LPCS URA C NRS ENSERG Grenoble France Thomson Microelectronics S.G.S. Microelettronica S.p.A Grenoble France

The temperature dependence of the Fowler-Nordheim tunnel current in a MOS structure is investigated both theoretically and experimentally between 25° C and 400°C. The F-N current is found to increase substantially with temperature. The existence of effective pre-exponential and exponential F-N coefficients up to 400°C is demonstrated and validated by numerical simulations based on the general F-N current model.

关键词： Temperature dependence Tunneling Temperature distribution Electrodes linear predictive coding Microelectronics EPROM Electron emission Silicon Statistics

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A DFT-based front-end for word recognition systems

A DFT-based front-end for word recognition systems

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

作者： R. Geppert P. Schwartau Philips GmbH Forschungslaboratorium Hamburg Hamburg Germany

Combining a custom LSI chip-set for DFT computation and user-programmable signal processors a versatile front-end for word recognition systems has been set up. Depending on the complexity of the required processing algorithms up to four signal processors can be employed in a pipelined architecture. The Multibus-compatible board performs functions like spectral analysis, spectral smoothing, intensity normalisation, and filter bank realisation on speech signals sampled at rates of up to 8 kHz.

关键词： Speech analysis Signal processing Spectral analysis Signal processing algorithms Filter bank Signal analysis Large scale integration Computer architecture Digital filters linear predictive coding

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LPC-VQ based Hidden Markov Models for Similarity Searching in DNA Sequences

LPC-VQ based Hidden Markov Models for Similarity Searching i...

引用

IEEE International Conference on Systems, Man and Cybernetics

作者： Tuan D. Pham Hong Yan Bioinformatics Applications Research Center School of Information Technology James Cook University Townsville QLD Australia Department of Electronic Engineering City University of Hong Kong Kowloon Hong Kong China School of Electrical and Information Engineering University of Sydney NSW Australia

Given a newly found gene of some particular genome and a database of sequences whose functions have been known, it must be very helpful if we can search through the database and identify those that are similar to the particular new sequence. The search results may help us to understand the functional role, regulation, and expression of the new gene by the inference from the similar database sequences. This is the task of any methods developed for biological database searching. In this paper we present a new application of the theories of linear predictive coding, vector quantization, and hidden Markov models to address the problem of DNA sequence similarity search where there is no need for sequence alignment. The proposed approach has been tested and compared with some existing methods against real DNA and genomic datasets. The experimental results demonstrate its potential use for such purpose.

关键词： Hidden Markov models DNA Sequences Databases Bioinformatics linear predictive coding Genomics Vector quantization Dynamic programming Cybernetics

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A generalization of isolated word recognition using vector quantization

A generalization of isolated word recognition using vector q...

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

作者： D. Burton J. Shore J. Buck Information Technology Division Naval Research Laboratory Computer Science and Systems Washington D.C. DC USA

The use of vector quantization (VQ) for isolated-word speech recognition of a 20-word vocabulary was shown in previous work to achieve more than 99% accuracy for speaker-dependent recognition and 87% accuracy for speaker-independent recognition. Separate VQ codebooks were designed for each word in the recognition vocabulary, and input words were classified by performing VQ and finding the codebook that achieves the smallest average distortion. The method obviates time-normalization and makes no use of time-sequence information. This paper presents results for a generalization that incorporates time-sequence information. The generalization, which was motivated by work of Martinez, Riveria, and Buzo, is more accurate and faster than the previous method. Words in the training and input sequences are normalized linearly to the same length and then divided into sections. Separate VQ codebooks are designed for each section of each vocabulary word. Each vocabulary word is then represented by a "multisection" codebook - a time-dependent sequence of section-codebooks. New words are classified by performing VQ and finding the multi-section codebook that achieves the smallest average distortion. Initial tests on a twenty-word vocabulary resulted in accuracies greater than 97% for speaker-independent recognition.

关键词： Vector quantization Vocabulary linear predictive coding Distortion measurement Speech recognition Speech coding Shape measurement Computer science Information technology Laboratories

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A composite source model for speaker and isolated word recognition

A composite source model for speaker and isolated word recog...

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

作者： R. Fontana M. Fox Department of Electrical Engineering Carnegie Mellon University Pittsburgh PA USA

A composite source is an indexed family of random processes (subsources) together with a switch which chooses from among these processes in a stochastic fashion. Such a source has often been proposed as a model for speech and other processes having piece-wise, or quasi, stationary behavior. Until recently, however, very little has been known about such models from either a theoretical or a practical perspective. In this paper, we consider a speaker/isolated word recognition system derived from a composite source model for speech production. In particular, estimates of the underlying subsources are obtained using a modified data compression algorithm. Switch sequences are then derived from these estimates for each utterance. Finally, switch sequences are compared in the time domain (using Levenshtein's metric) and from a statistical point of view (via variation distance). Both modes of comparison are seen to be highly correlated and produce a recognition procedure with very encouraging results.

关键词： Switches Stochastic processes Speech recognition Data compression Random processes Speech processing Speech analysis linear predictive coding Performance analysis Production systems

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Investigation of series resistances, interface coupling and irradiation effects on the transconductance of fully-depleted SOI-MOSFETs

Investigation of series resistances, interface coupling and ...

引用

IEEE SOI-3D-Subthreshold Microelectronics Technology Unified Conference (S3S)

作者： T. Ouisse S. Cristoloveanu G. Borel Thomson-TMS Saint-Egreve France LPCS Grenoble France

The operation of fully-depleted silicon-on-insulator (SOI) MOS transistors is far more complex than in bulk Si MOSFETs due to a strong coupling between the front and back interfaces. This means that the front channel current can substantially be influenced by the back gate bias and vice-versa. The authors propose a dual-channel transconductance model which depicts and clarifies various experimental situations. It is concluded that the different effects, i.e., interface coupling, short-channel, series resistances, and irradiation, are interacting in SOI. The series resistance influence is enhanced when the back channel is activated either by increasing the substrate bias or by irradiation. The interest of the above model is two-fold: to describe the transconductance behavior by sorting out the different effects and to show how the interface parameters of thin SOI structures can be evaluated.< >

关键词： Transconductance Silicon on insulator technology MOSFETs Threshold voltage Degradation linear predictive coding Conductive films Shape Virtual colonoscopy Equivalent circuits

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On the Use of Band-Passed Excitation Codebooks for CELP coding of Speech

On the Use of Band-Passed Excitation Codebooks for CELP Codi...

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IEEE Workshop on Speech coding For Telecommunications

作者： Ping Zlieng P. Mermelstein INRS-Telecommunications Universite du Quebec Canada INRS-Télécommunications Université du Quàbec Verdun QUE Canada

This paper explores the benefits of using multiple band-passed codebooks to replace the single full-band excitation codebook of classical code-excited linear prediction for coding high quality speech in the range of 8-16 kb/s. In the classical method, the residual signal in the LPC frame is divided temporally into subframes and each subframe is coded using a single full-band excitation code book. In this study, we divide the residual signal in the LPC frame into its band-passed components and match each component with an entry from a band-passed code book. Our results indicate objective and subjective advantages for using three codebooks, each with the gain and shape components. In particular, the codebooks required for the frequencies above 1 kHz are very small. Also, by subsampling, the large low frequency (0-1 kHz) codebook can be searched very rapidly resulting in a significant complexity advantage.

关键词： Speech coding linear predictive coding Frequency Delay Books Bit rate Filters Business Shape Redundancy

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