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linear PREDICTION USING L(1) NORM IN ORTHOGONAL VECTOR-SPACE

ELECTRONICS LETTERS 1995年第6期31卷 430-431页

作者： HU, HT Department of Electronic Engineering National I-Lan Institute of Agriculture and Technology I-Lan Republic of China

linear prediction is formulated in a vector space by means of the orthogonal transformation, with which the L(1) criterion can be easily incorporated to yield an efficient iterative algorithm. An improvement over the ... 详细信息

关键词： linear predictive coding SPEECH ANALYSIS AND PROCESSING

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Speech recognition method using quantised LSP parameters in CELP-type coders

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ELECTRONICS LETTERS 1998年第2期34卷 156-157页

作者： Choi, SH Kim, HK Lee, HS Gray, RM Korea Adv Inst Sci & Technol Dept Informat & Commun Engn Dongdaemun Gu Seoul 130012 South Korea Samsung Adv Inst Technol Informat Proc Sector Human & Comp Interact Lab Yongin 449712 Kyungki Do South Korea Stanford Univ Dept Elect Engn Informat Syst Lab Stanford CA 94305 USA

A low-complexity speech recognition method applicable to digital communication networks is proposed. A feature set suitable for speech recognition is obtained from quantised LSP parameters in CELP-type coders without reconstructing the speech signals. The authors present the effects of the speech coder on speaker-independent recognition performance. and show that the recognition accuracy of the proposed method is better than that of the recogniser using reconstructed speech signals.

关键词： Speech processing techniques Speech recognition low-complexity recognition method linear predictive coding CELP-type coders digital communication digital communication networks feature set speaker-independent recognition performance Codes speech coding line spectral pair Speech and audio signal processing speech recognition recognition accuracy quantised LSP parameters

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Signal detection assisted from channel weight prediction for delay-sensitive orthogonal frequency-division multiplexing communications

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IET COMMUNICATIONS 2014年第10期8卷 1745-1750页

作者： Lv, Changwei Lin, Jia-Chin Hou, Shujuan Mei, Wenbo Beijing Inst Technol Sch Informat & Elect Beijing 100081 Peoples R China Natl Cent Univ Dept Commun Engn Jhongli Taiwan Chongqing Commun Inst Dept Informat Technol Chongqing 400035 Peoples R China

In signal coherent detection, data-aided (DA) method has high detection accuracy, but unavoidably suffers from a large processing delay. Meanwhile, a decision-directed (DD) method incurs no processing delay, but suffers from error propagation problem. Motivated by the above problems of DA and DD methods, the authors present a novel signal detection algorithm for delay-sensitive applications employing orthogonal frequency-division multiplexing systems. The proposed detection method provides similar detection accuracy to the DA method and incurs no processing delay. This method employs channel estimation followed by linear prediction and interpolation to obtain the channel-state information for symbol detection. To achieve a better prediction performance, they perform the linear prediction on the channel taps in the time domain, and conduct the interpolation using a raised cosine filter. Simulations on physical channel models verify the improvements of the proposed method.

关键词： SIGNAL detection ORTHOGONAL frequency division multiplexing linear predictive coding INTERPOLATION COSINE function

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MATRIX QUANTIZER DESIGN FOR LPC SPEECH USING THE GENERALIZED LLOYD ALGORITHM

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1985年第3期33卷 537-545页

作者： TSAO, C GRAY, RM Information Systems Laboratory Department of Electrical Engineering University of Stanford Stanford CA USA

Rate-distortion theory provides the motivation for using data compression techniques on matrices of N LPC vectors. This leads to a simple extension of speech coding techniques using vector quantization. The effects of using the generalized Lloyd algorithm on such matrices using a summed Itakura-Saito distortion measure are studied, and an extension of the centroid computation used in vector quantization is presented. The matrix quantizers so obtained offer substantial reductions in bit rates relative to full-search vector quantizers. Bit rates as low as 150 bits/s for the LPC matrix information (inclusive of gain, but without pitch and voicing) have been achieved for a single speaker, having average test sequence and codebook distortions comparable to those in the equivalent full-search vector quantizer operating at 350 bits/s. Preliminary results indicate that higher quality or lower bit rates may be achieved with enough computational resources.

关键词： Algorithm design and analysis linear predictive coding Speech coding Bit rate Rate-distortion Vector quantization Information systems Data compression Distortion measurement Testing

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

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ELECTRONICS LETTERS 2000年第12期36卷 1082-1083页

作者： Hacioglu, K Eastern Mediterranean Univ Elect Elect Engn Dept G Magosa 10 Mersin Turkey

For original paper, see D.P. Palomar et al., ibid., vol. 35, no. 13, pp. 1058-9 (1999). Palomar et al. addressed the question of LPC filter re-optimisation and proposed a recursive algorithm that they considered promising. The present commentator shows that the iterative algorithm employed by Palomar is unnecessarily complex because of poor initialisation. The algorithm that uses available speech data at the very first step is shown to perform well. The results of the one-shot algorithm can be used to initiate Paolmar's method to significantly reduce the number of iterations

关键词： filtering theory minimisation one-shot algorithm linear predictive coding iterative methods Filtering methods in signal processing iterative algorithm recursive algorithm Optimisation techniques speech coding initialisation Speech and audio coding multi-pulse coded excitation speech data Interpolation and function approximation (numerical analysis) LPC filter re-optimisation

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RoIP Compression Method in Burst Noise Upstream Environment for CATV Network

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IEEE TRANSACTIONS ON BROADCASTING 2021年第1期67卷 351-355页

作者： Cui, Chengyu Park, Sungkwon Hanyang Univ Dept Elect & Comp Engn Seoul 04763 South Korea

A hybrid fiber coaxial (HFC) network uses both optical fibers and coaxial cables for cable TV networks. Coaxial cables are mainly used for access networks. Recently, fiber to the home (FTTH) is being widely deployed. For cable TV companies to utilize readily deployed FTTH, legacy coax network-based customer premise equipment (CPE) such as cable modems and set top boxes may have to be replaced with fiber network based CPE. In order to reduce the expenditure incurred in this process, radio over Internet Protocol (RoIP) technology has been proposed. In the RoIP system, RF signal for coax generated by legacy CPE toward the headend is first sampled, and digitized, and then encapsulated in IP packets, which are then sent to the headend using FTTH access technologies such as passive optical network (PON). In the process of digitizing RF signal, a huge amount of data much more than the base band data is generated. So, a data compression method is required to lower the upstream bit rate maintaining the signal quality. Most compression methods are significantly degraded due to the interference caused by burst noise in the upstream link. Therefore, this paper proposes a new compression method which combines linear predictive coding (LPC), block scaling and non-linear quantization for CATV upstream links. The proposed method is demonstrated to greatly reduce the impact of burst noise while maintaining the compression performance.

关键词： Quantization (signal) RF signals Uplink Cable TV linear predictive coding IP networks Hybrid fiber coaxial cables Hybrid fiber coaxial radio over IP compression method linear predictive coding block scaling

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SPEECH PROCESSING AND SYNTHESIS FOR ASSESSING VOCAL DISORDERS

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IEEE ENGINEERING IN MEDICINE AND BIOLOGY MAGAZINE 1990年第1期9卷 69-71页

作者： CHILDERS, DG Department of Electrical Engineering University of Florida USA

Techniques for the quantitative assessment and classification of vocal disorders are described. Models for vocal disorders using speech synthesis are examined. Methods for characterizing the electroglottography (EGG) ... 详细信息

关键词： Speech processing Speech synthesis Speech analysis Filters Larynx Diseases Contacts Resonance predictive models linear predictive coding

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Computational improvement for G.729 standard

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ELECTRONICS LETTERS 2000年第13期36卷 1163-1164页

作者： Hwang, SH Ming Hsin Inst Technol Dept Management Informat Hsinchu Taiwan

G.729 is a CS-ACELP-based [1] standard for 8kbit/s speech compression and is defined by the ITU. In the G.729 encoding phase. the open-loop pitch T-op is estimated for each frame to reduce the computation required to search for the best adaptive codebook delay. However, > 25% CPU time is required for the open-loop pitch analysis. The weighted delta-LSP(WD-LSP) function is employed here to reduce the computation required for open-loop pitch analysis. An 81% saving is made in the computation of T-op. A total computational saving of 21% for the encoding phase Is achieved and the speech quality still sounds good (MOS = 3.5).

关键词： telecommunication standards open-loop pitch data compression linear predictive coding encoding phase speech quality CS-ACELP-based standard G.729 standard Codes speech coding open-loop pitch analysis 8 kbit/s Speech and audio coding ITU computational saving speech compression weighted delta-LSP adaptive codebook delay

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A new extrapolation law for data-retention time-to-failure of nonvolatile memories

IEEE ELECTRON DEVICE LETTERS

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IEEE ELECTRON DEVICE LETTERS 1999年第5期20卷 197-199页

作者： De Salvo, B Ghibaudo, G Pananakakis, G Guillaumot, B Candelier, P Reimbold, G ENSERG LPCS F-38016 Grenoble France SGS Thomson Microelect F-38019 Grenoble France LETI CEA F-38054 Grenoble 9 France

In this letter, we demonstrate that the commonly assumed:Arrhenius law is inconsistent with extrapolation of data-retention time-to-failure of nonvolatile memories in highly accelerated life-tests. We argue that the retention time, namely log(t(R)), varies linearly with temperature T rather than with 1/T as commonly assumed, yielding an important reduction in the extrapolated time-to-failure. Extensive experimental results demonstrate the physical consistency of the new model. In particular, data-retention of EPROM devices and leakage current of interpoly dielectric and gate oxide have been investigated over a wide range of temperatures. Finally, it is shown that our model reconciles seemingly controversial activation energy data from the literature.

关键词： Extrapolation Nonvolatile memory EPROM Acceleration Leakage current Dielectric devices Temperature distribution Dielectric losses linear predictive coding Microelectronics

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On the duality between line-spectral frequencies and zero-crossings of signals

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 2001年第4期9卷 458-461页

作者： Kumaresan, R Wang, YD Univ Rhode Isl Dept Elect Engn Kingston RI 02881 USA

Line spectrum pairs (LSPs) are the roots (located in the complex-frequency or z-plane) of symmetric and antisymmetric polynomials synthesized using a linear prediction (LPC) polynomial. The angles of these roots, known as line-spectral frequencies (LSFs), implicitly represent the LPC polynomial and hence the spectral envelope of the underlying signal. BY exploiting the duality between the time and frequency domains, we define analogous polynomials in the complex-time variable zeta. The angles of the roots of these polynomials in zeta -plane now correspond to zero-crossing time instants. Analogous to the fact that the line-spectral frequencies represent the spectral envelope of a signal, these zero-crossing locations can be used to represent the temporal envelope of bandpass signals.

关键词： Polynomials linear predictive coding Signal synthesis Frequency synthesizers Speech synthesis Band pass filters Frequency domain analysis Speech coding Speech recognition Quantization

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