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International Conference on Signal Processing Proceedings (ICSP)

作者： Jian Lu N. Uemi Gang Li T. Ifukube Research Institute for Electronic Science Hokkaido University Sapporo Hokkaido Japan Tianjin University TJU China

In this paper, a digital processing method is described for modifying tone contrast that was defined as the difference in frequencies between peaks and valleys of pitch curves in natural utterances. Speech signals with modified tones were presented to hearing-impaired Chinese listeners who were asked to identify four alternative Mandarin words. Employing this method, it was found that modified speech with enhanced tone contrast contributed moderate gains in the percentage correct word identification when compared to unmodified speech, while reducing tone contrast generally reduced the percentage correct identification. These findings therefore offer support to the assertion that a hearing aid with tone modifications is indeed effective for hearing-impaired Chinese.

关键词： Auditory system Frequency Speech enhancement Signal processing Natural languages Speech processing linear predictive coding Vocoders Speech recognition

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ARMA lattice modeling for isolated word speech recognition

ARMA lattice modeling for isolated word speech recognition

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Midwest Symposium on Circuits and Systems (MWSCAS)

作者： H.K. Kwan T.X. Li Electrical and Computer Engineering Faculty of Engineering University of Windsor Windsor ONT Canada

In this paper, we introduce an auto-regressive moving average (ARMA) lattice model for speech modeling. The speech characteristics are modeled and expressed in the form of lattice reflection coefficients for classification. Self Organization Map (SOM) is used to build codebooks for classification and recognition of the lattice reflection coefficients. Experimental results based on an isolated word speech database of 10 words/names indicate that the ARMA lattice model achieves superior recognition performance as compared to those of the conventional auto-regressive (AR) model.

关键词： Lattices Speech recognition Speech analysis Reflection Filters Poles and zeros Databases linear predictive coding Resonance Central Processing Unit

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A new BSP excitation signal for linear prediction-based speech codecs

A new BSP excitation signal for linear prediction-based spee...

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International Conference on Signal Processing Proceedings (ICSP)

作者： Deng Lixin Yang Zhen Zheng Baoyu Information Engineering Department Nanjing University of Posts and Telecommunications China

We present in this paper a new binomial sine pulse (BSP) excitation signal used in linear prediction-based speech codecs. The structure of the BSP excitation signal is actually a sine wave whose amplitude is modulated by a binomial signal. The binomial signal describes the various trends of excitation signals in a pitch period, and the pulsatance of the BSP excitation signal coincides with the vibration frequency of vocal folds. In experiments, processing is going on frame by frame and the same excitation signal is placed at every pitch excitation moment in a frame. Speech codecs based on this new BSP excitation have the advantages of low complexity and low delay. Experiment results prove that such a new speech codec can provide highly intelligible synthesized speech below 3 kbps.

关键词： Speech codecs Speech synthesis Speech coding Signal synthesis Signal processing Bit rate Shape linear predictive coding Filters Telecommunications

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MPEG VBR slice layer model using linear predictive coding and generalized periodic Markov chains

MPEG VBR slice layer model using linear predictive coding an...

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IEEE International Conference on Performance, Computing and Communications (IPCCC)

作者： M.R. Izquierdo D.S. Reeves Network Hardware Division IBM Corporation NC USA Electrical and Computer Engineering North Carolina State University Raleigh NC USA

We present an MPEG slice layer model for VBR encoded video using linear predictive coding (LPC) and generalized periodic Markov chains. Each slice position within an MPEG frame is modeled using an LPC autoregressive function. The selection of the particular LPC function is governed by a generalized periodic Markov chain; one chain is defined for each I, P, and B frame type. The model is sufficiently modular in that sequences which exclude B frames can eliminate the corresponding Markov chain. We show that the model matches the pseudo-periodic autocorrelation function quite well. We present simulation results of an asynchronous transfer mode (ATM) video transmitter using a FIFO queue and measure the average cell delay. Simulation results showed good agreement with results obtained using actual traces as sources.

关键词： predictive models linear predictive coding Video compression Asynchronous transfer mode Delay Computer networks Bit rate Transform coding Stochastic processes Streaming media

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LPC analysis incorporating spectral interpolation for speech coding

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ELECTRONICS LETTERS 1999年第3期35卷 200-201页

作者： Lee, MS Kim, HK Lee, HS Korea Adv Inst Sci & Technol Dongdaemun Gu Seoul 130012 South Korea SK Telekom Cent R&D Ctr Yusong Gu Taejon 305348 South Korea

A linear predictive coding (LPC) analysis scheme which is applicable to speech coding is proposed. The analysis method, called interpolative LPC (ILPC) analysis, estimates the spectral envelope by incorporating the interpolation characteristics into the LPC analysis. The ILPC analysis reduces average spectral distortion and the percentage of outlier frames, compared with the conventional LPC analysis followed by linear interpolation.

关键词： LPC analysis Speech processing techniques interpolation linear predictive coding speech coding spectral envelope estimation Speech and audio coding Interpolation and function approximation (numerical analysis) Signal processing theory spectral interpolation interpolative LPC

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

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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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On polynomial reduction in the computation of LSP frequencies

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1999年第5期7卷 592-594页

作者： Rothweiler, J Sanders Nashua NH 03061 USA

This work describes a new version of the decimation-in-degree (DID) transformation used by Wu and Chen [1] as part of their procedure for computing line spectrum pair (LSP) coefficients. This new version eliminates all nontrivial multiplications, requires fewer total operations (additions and multiplications), and is performed in place-eliminating a memory buffer.

关键词： Chebyshev functions linear predictive coding polynomials

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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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Pitch synchronous extended excitation in multimode CELP

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IEEE COMMUNICATIONS LETTERS 1999年第9期3卷 275-276页

作者： Tian, WS Wong, WC Law, CY Tan, AP Natl Univ Singapore Dept Elect Engn Singapore 119260 Singapore Panason Singapore Labs Pte Ltd Singapore 534415 Singapore

This letter proposes a 4-kb/s multimode code-excited linear prediction (CELP) coder with pitch synchronous extended excitation, Three modes are used for the short-term excitation, namely algebraic, extended, or stochastic excitations, together with an adaptive codebook for the long-term excitation. Comparisons with the FS-1016 and ITU-T G.723.1 coders show a performance level between these standards.

关键词： Speech Bit rate linear predictive coding Signal synthesis Stochastic processes Code standards Nonlinear filters Vectors Filtering theory Laboratories

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Adaptive feature mapping for underwater target classification

Adaptive feature mapping for underwater target classificatio...

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International Joint Conference on Neural Networks (IJCNN)

作者： De Yao M.R. Azimi-Sadjadi G.J. Dobeck Department of Electrical and Computer Engineering Colorado State University Fort Collins CO USA Naval Warfare Systems Center Coastal Systems Station Panama FL USA

In this paper, a new feature mapping scheme is presented to cope with environmental and target signature changes for underwater target classification. A wavelet packet-based feature extraction scheme is used in conjunction with the linear prediction coding (LPC) scheme as the front-end processor. The core of the system is the adaptive feature mapping subsystem that minimizes the classification error of the classifier. The extracted feature vector is mapped by the resultant transformation matrix in such a way that the mapped version remains invariant to the environmental and sensory changes. The feedback to the adaptation mechanism is provided by a k-nearest neighbor classifier. The test results on 40 kHz linear FM acoustic backscattered data collected for six different objects are presented The effectiveness of the adaptive system vs. nonadaptive one is demonstrated for several signal-to-noise ratio (SNR) conditions.

关键词： Feature extraction Adaptive systems Wavelet packets linear predictive coding Data mining Vectors Feedback Acoustic testing System testing Signal to noise ratio

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