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Re-optimisation of LPC filters for multi-pulse coded excitation

ELECTRONICS LETTERS 1999年第13期35卷 1058-1059页

作者： Palomar, DP Price, M Sandler, M Univ Politecn Cataluna Dept Signal Theory & Commun Barcelona 08034 Spain Univ London Kings Coll Dept Elect & Elect Engn London WC2R 2LS England

A method for optimising LPC filters in linear prediction based speech coders is described. The optimisation process compensates for errors incurred through coding the excitation signal, providing an improvement in the... 详细信息

关键词： optimisation filtering theory linear predictive coding Filtering methods in signal processing Optimisation techniques decoded speech quality improvement speech coding error compensation filter optimisation excitation signal coding linear prediction based speech coders Speech and audio coding multi-pulse coded excitation LPC filters Signal processing theory re-optimisation

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1.6-KBIT/S LP VOCODER USING TIME ENVELOPE

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ELECTRONICS LETTERS 1995年第7期31卷 517-519页

作者： ATKINSON, IA KONDOZ, AM EVANS, BG Centre for Satellite Engineering Speech Coding Group University of Surrey Guildford United Kingdom

The authors present a linear prediction (LP) based vocoder in which speech waveforms are considered as having a time envelope, the shape of which contains important perceptual information. By ensuring that the time envelope of the synthetic speech closely matches that of the original, natural sounding synthetic speech can be achieved at 1.6kbit/s.

关键词： linear predictive coding VOCODERS

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CODEBOOK GENERATION FOR VECTOR QUANTIZATION

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ELECTRONICS LETTERS 1995年第7期31卷 522-523页

作者： CHEN, CQ KOH, SN SIVAPRAKASAPILLAI, P School of Electrical and Electronic Engineering Nanyang Technological University Singapore Republic of Singapore

A novel codebook generation scheme for vector quantisation is presented. The proposed scheme is of comparable computational complexity to the Linde-Buzo-Gray (LBG) algorithm, but its performance is shown to be superior.

关键词： VECTOR QUANTIZATION linear predictive coding

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Digital linear pre-compensation technique to enhance predistortion performance in multicarrier DVB-S2 satellite communication systems

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ELECTRONICS LETTERS 2014年第13期50卷 957-958页

作者： Allegue-Martinez, M. Kelly, N. Zhu, A. UCD Sch Elect Elect & Commun Engn RF & Microwave Res Grp Dublin Ireland

Nonlinearity in communication satellite payloads often severely degrades the downlink signal quality in such systems. A digital pre-compensation technique to enhance digital predistortion performance in multicarrier satellite communication systems is presented. The proposed technique has been validated by simulations with an appropriate platform where excellent performance improvements have been measured. In addition, the implementation feasibility of this technique has been briefly demonstrated by means of a real-time FPGA implementation.

关键词： ARTIFICIAL satellites in telecommunication TELECOMMUNICATION systems COMMUNICATION REMOTE broadcasting linear predictive coding

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A SIMPLE NONITERATIVE SPEECH EXCITATION ALGORITHM USING THE LPC RESIDUAL

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1985年第2期33卷 432-434页

作者： ALEXANDER, ST Department of Electrical and Computer Engineering Center for Communications and Signal Processing North Carolina State University Raleigh NC USA

This paper provides an analytical derivation of a simple noniterative technique for extracting a multiple impulse excitation model for synthesized speech directly from the LPC residual sequence. While suboptimal with respect to "multipulse" techniques, this method is very applicable for speech enhancement where processor capability is limited. The results suggest an additional "orthogonality" requirement between the excitation sequence and the resulting prediction error, which aids in the intuitive understanding of the method.

关键词： linear predictive coding Speech synthesis Speech enhancement Bit rate Detectors Degradation Speech analysis Filters Signal processing algorithms Switches

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Modular neural predictive coding for discriminative feature extraction

Modular neural predictive coding for discriminative feature ...

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

作者： M. Chetouani B. Gas J. Zarader Laboratoire des Instruments et Systémes dIle-De-France Université Paris IV Paris France

We present an architecture called the modular neural predictive coding architecture (Modular NPC). The Modular NPC is used for discriminative feature extraction (DFE). It provides an architecture based on phonetics knowledge applied to phoneme recognition. The phonemes are extracted from the Darpa-Timit speech database. Comparisons with coding methods (LPC, MFCC, PLP) are presented: they put in obviousness an improvement of the recognition rates.

关键词： predictive coding Feature extraction linear predictive coding predictive models Speech Mel frequency cepstral coefficient Cepstral analysis Context modeling Instruments Spatial databases

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AN MOS-LSI AUTOCORRELATOR FOR linear PREDICTION OF SPEECH

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IEEE JOURNAL OF SOLID-STATE CIRCUITS 1984年第6期19卷 1022-1029页

作者： HURST, PJ BRODERSEN, RW Department of Electrical Engineering and Computer Sciences University of California Berkeley CA USA

An autocorrelator for speech analysis has been integrated in CMOS technology. The CMOS integrated circuit is composed of analog, switched-capacitor, and digital circuitry. The autocorrelator uses a time-multiplexed multiplying digital-to-analog converter and three multiplexed switched-capacitor filter sections. Each multiplexed filter section consists of ten channels so that ten autocorrelation lags are computed. Practical problems associated with the integration of the multiplexed filters are discussed, and experimental data are given for the filters and the complete autocorrelator.

关键词： Autocorrelation Speech analysis linear predictive coding Transfer functions Equations Filters CMOS technology CMOS integrated circuits Switching circuits Signal processing

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THEORETICAL-ANALYSIS OF THE HIGH-RATE VECTOR QUANTIZATION OF LPC PARAMETERS

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1995年第5期3卷 367-381页

作者： GARDNER, WR RAO, BD University of California San Diego CA USA QUALCOMM Incorporated San Diego CA USA University of California San Diego San Diego CA USA

This paper presents a theoretical analysis of high-rate vector quantization (VQ) systems that use suboptimal, mismatched distortion measures, and describes the application of the analysis to the problem of quantizing the linear predictive coding (LPC) parameters in speech coding systems, First, it is shown that in many high-rate VQ systems the quantization distortion approaches a simple quadratically weighted error measure, where the weighting matrix is a ''sensitivity matrix'' that is an extension of the concept of the scalar sensitivity. The approximate performance of VQ systems that train and quantize using mismatched distortion measures is derived, and is used to construct better distortion measures, Second, these results are used to determine the performance of LPC vector quantizers, as measured by the log spectral distortion (LSD) measure, which have been trained using other error measures, such as mean-squared (MSE) or weighted mean-squared error (WMSE) measures of LPC parameters, reflection coefficients and transforms thereof, and line spectral pair (LSP) frequencies, Computationally efficient algorithms for computing the sensitivity matrices of these parameters are described. In particular, it is shown that the sensivity matrix for the LSP frequencies is diagonal, implying that a WMSE measure of LSP frequencies converges to the LSD measure in high-rate VQ systems, Experimental results to support the theoretical performance estimates are provided.

关键词： Vector quantization linear predictive coding Distortion measurement Frequency measurement Speech analysis Particle measurements Speech coding Weight measurement Reflection Estimation theory

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A HIGH-QUALITY DIGITAL FILTERBANK FOR SPEECH RECOGNITION WHICH RUNS IN REAL-TIME ON A STANDARD MICROPROCESSOR

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1986年第5期34卷 1064-1073页

作者： SILVERMAN, HF Laboratory for Engineering Man/Machine Systems (LEMS) Division of Engineering Brown University of Providence Providence RI USA

Asynchronous digital filterbank or LPC output is widely used for high-quality research and commercial speech recognition applications. Special-purpose hardware is usually applied. In this paper, a decimation/DFT filterbank system is described which may be implemented in real time on a standard microprocessor. Key features of the approach are a dithering technique, which allows output sampling at a rate lower than would be theoretically predicted as adequate, and the Winograd Fourier transform algorithm (WFTA) implementation of the DFT. The contributions of this paper are the dithering idea, the demonstration that it is feasible to implement a high-quality digital filter-bank on a standard microprocessor, and the discussion of starting-phase error. The nonlinear effects of dithering are discussed in detail, and it is shown that the dithering technique significantly reduces starting phase error, which can be important in very close discrete utterance recognition (DUR) situations.

关键词： Filter bank Speech recognition Microprocessors Discrete Fourier transforms linear predictive coding Hardware Real time systems Sampling methods Fourier transforms Digital filters

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Two time slots distributed time-reversal space-time block coding for single-carrier block transmissions

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IET COMMUNICATIONS 2013年第18期7卷 2026-2033页

作者： Wang, Qiucai Yuan, Chaowei Zhang, Jinbo Du, Jianhe Beijing Univ Posts & Telecommun Sch Informat & Commun Engn Beijing 100876 Peoples R China

Distributed space-time block coding (STBC) is a promising technique for future broadband wireless communication system, because of its substantially improving the reliability of wireless channel by exploiting cooperative spatial diversity. In this study, the authors propose a novel two time slots distributed time-reversal STBC scheme for amplify-and-forward relay-assisted single-carrier (SC) block transmissions over frequency-selective fading channel. They first exploit the discrete Fourier transform extended properties to construct a linear precoding matrix. They then employ a low-complexity suboptimal frequency domain decision feedback equalisation (FD-DFE) to collect potential multipath diversity at high signal-to-noise ratio. Simulation results demonstrate that the proposed scheme provides better performance than the conventional distributed SC-STBC scheme with minimum-mean-square error FD linear equalisation.

关键词： computational complexity decision feedback equalisers discrete Fourier transforms diversity reception fading channels frequency-domain analysis linear predictive coding matrix algebra mean square error methods radiocommunication telecommunication network reliability space-time block codes amplify and forward communication cooperative communication minimum-mean-square error FD linear equalisation conventional distributed SC-STBC scheme signal-to-noise ratio multipath diversity low-complexity suboptimal frequency domain decision feedback equalisation linear precoding matrix discrete Fourier transform extended property frequency-selective fading channel SC block transmissions amplify-and-forward relay-assisted single-carrier block transmissions distributed time-reversal STBC scheme cooperative spatial diversity wireless channel reliability broadband wireless communication system distributed space-time block coding distributed time-reversal space-time block coding

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