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检索条件"主题词=linear predictive coding"

共 2456 条记录，以下是271-280 订阅

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Delay-free lossy audio coding using shelving pre- and post-filters

Delay-free lossy audio coding using shelving pre- and post-f...

引用

33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Holters, Martin Zoelzer, Udo Helmut Schmidt Univ Hamburg Germany

ISBN: (纸本)9781424414833

A delay-free audio coding scheme based on ADPCM with adaptive pre- and post-filtering is presented. The pre-/post-filters are realized as a cascade of shelving filters, designed to match the characteristics of human perception. The pre- and post-filters are adapted by dynamic compression of the respective sub-bands. The adaption is backward-adaptive, i.e. is fed by the reconstructed signal, which eliminates the need to transmit the filter coefficients and allows delay-free operation. This pre- and post-filtering significantly improves the audio quality compared to a plain ADPCM codec, as underlined by objective measurements. Since the base ADPCM used is also delay-free, the resulting coding system works without any algorithmic delay.

关键词： audio coding linear predictive coding adaptive filters

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Voiced/Unvoiced Classification Recovery in the Speech Decoder Based on GMM

Voiced/Unvoiced Classification Recovery in the Speech Decode...

引用

9th International Conference on Signal Processing

作者： Wei Xuan Dang Xiaoyan Cui Huijuan Tang Kun Tsinghua Univ Dept Elect Engn State Key Lab Microwave & Digital Commun Beijing 100084 Peoples R China

ISBN: (纸本)9781424421787

Voiced/Unvoiced (V/U) classification is an important parameter in low bit-rate speech coding algorithms. An algorithm that recovers the V/U classification from the linear prediction coding (LPC) coefficients and the gain in the speech decoder is proposed. Two Gaussian mixture models (GMM) are employed to model the joint probability of these parameters and to perform the V/U estimation. Experiments show the performance improvements of the proposed algorithm over the V/U classifier used in mixed excitation LPC vocoder (MELP). The proposed algorithm operates only at the receiving end and saves all the bits originally used for V/U quantization.

关键词： Decoding linear predictive coding Speech coding Databases Pattern recognition Quantization Speech synthesis Cepstral analysis Probability distribution Laboratories

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Variable dimension matrix quantization of LSP parameters for very low bit rate vocoder below 300bps

Variable dimension matrix quantization of LSP parameters for...

引用

9th International Conference on Signal Processing

作者： Min Gang Yang Ji-bin Chen Yan-pu Zhang Xiong-wei Institute of communications engineering People''s Liberation Army University of Science and Technology China Xian Communication Institute China

ISBN: (纸本)9781424421787

This paper examines the efficient quantization of LSP parameters for very low bit rate vocoder below 300bps, a new quantization scheme called variable dimension matrix quantization (VDMQ) is presented In the VDMQ scheme, the extracted LSP parameters matrix with variable dimension is quantized directly without dimension conversion. Based on the distance measure definition between low LSP matrices with different dimension, the optimal codeword is deduced Theoretical analysis and experiment results show that the VDMQ scheme performs better than the segment quantization and matrix quantization scheme at very low bit rate. Also, the codebook storage is almost reduced by 90%. The VDMQ scheme provides a new effective approach for efficient LSP parameters quantization at very low bit rate.

关键词： Bit rate Vocoders Speech Vector quantization linear predictive coding Programmable logic arrays Performance analysis Wireless communication Artificial satellites Underwater communication

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New Feature Extraction Methods Using DWT and LPC for Isolated Word Recognition

New Feature Extraction Methods Using DWT and LPC for Isolate...

引用

IEEE Region 10 Conference (TENCON 2008)

作者： Nehe, N. S. Holambe, R. S. SGGS Inst Engn & Technol Nanded MS India

ISBN: (纸本)9781424424085

In this paper a new feature extraction methods, which utilize reduced order linear predictive coding (LPC) coefficients for speech recognition, have been proposed The coefficients have been derived from the speech frames decomposed using Discrete Wavelet Transform (DWT). In the literature it is assumed that the speech frame of size 10 msec to 30 msec is stationary, however, in practice different parts of the speech signal may convey different amount of information (hence may not be perfectly stationary). LPC coefficients derived from subband decomposition of speech frame provide better representation than modeling the frame directly. Experimentally it has been shown that, the proposed approaches provide effective (better recognition rate) and efficient (reduced feature vector dimension) features. The speech recognition system using the continuous Hidden Markov Model (HMM) has been implemented. The proposed algorithms are evaluated using NIST TI-46 isolated-word database.

关键词： Feature Extraction linear predictive coding Discrete Wavelet Transform

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PHONETIC AND INTONATION ENHANCEMENT OF SPEECH FOR JAPANESE HEARING-IMPAIRED SPEAKERS

PHONETIC AND INTONATION ENHANCEMENT OF SPEECH FOR JAPANESE H...

引用

IEEE International Conference on Multimedia and Expo (ICME 2008)

作者： Watanabe, Shogo Moriyama, Tsuyoshi Ozawa, Shinji Keio Univ Dept Informat & Comp Sci Yokohama Kanagawa 2238522 Japan

ISBN: (纸本)9781424425709

We propose a system that is capable of improving intelligibility of speech by those with hearing impairment. We have found the speech often had problems in the intonation, the duration of the unvoiced consonants, and the tone of the voiced phonemes. The system systematically compensates problematic components of the speech using the counterparts in normal speech. It corrects the intonation using TD-PSOLA and elongates consonants by repeating the original waveform using inverting technique for making the result continuously connected until the duration reaches a threshold. Experimental results show that the proposed method successfully improves the intelligibility of the speech from 28% to 35% by making phonemes more articulated, and making double consonants, V of syllabic consonant, and unvoiced consonants perceived clearer.

关键词： Speech analysis Speech intelligibility Speech enhancement Speech synthesis linear predictive coding

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Matrix-based linear predictive compression of multi-channel surface EMG signals

Matrix-based linear predictive compression of multi-channel ...

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33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Carotti, Elias S. G. de Martin, J. C. Merletti, Roberto Farina, Dario Politecn Torino DAUIN Corso Duca Abruzzi 24 I-10129 Turin Italy Politecn Torino DELEN I-10129 Turin Italy Aalborg Univ Ctr Sensory Motor Interact Dept Hlth Sci & Tech Aalborg Denmark

ISBN: (纸本)9781424414833

We propose a linear predictive coding technique for multi-channel electromyographic (EMG) recordings. The signals are acquired using two-dimensional grid of electrodes which generate strongly correlated signals. Previous work only considered spectral redundancy across the signal matrix. In this paper we exploit the correlation present in the residual signals, i.e., the signals after the short term prediction. The proposed technique achieves a compression ratio of about 1 divided by 9, i.e., slightly better than spectral-only decorrelation methods, but with a strong increase of approximately 3.2 dB SNR in the quality of the reconstructed waveform.

关键词： data compression linear predictive coding electromyography

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Doppler-variant modeling of the vocal tract

Doppler-variant modeling of the vocal tract

引用

33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Heim, Axel Sorger, Uli Hug, Florian Univ Ulm Inst Telecommun & Appl Informat Theory D-89069 Ulm Germany Univ Luxembourg Comp Sci & Commun Luxembourg Luxembourg

ISBN: (纸本)9781424414833

A common technique to deploy linear prediction to non-stationary signals is time segmentation and local analysis. Variations of a process within such a segment cause inaccuracies. In this paper, we model the temporal changes of linear prediction coefficients (LPCs) as a Fourier series. We obtain a compact description of the vocal tract model limited by the predictor order and the maximum Doppler frequency. Filter stability is guaranteed by all-pass filtering, deploying the human ear's insensitivity to absolute phase. The periodicity constraint induced by the Fourier series is counteracted by oversampling in the Doppler domain. With this approach, the number of coefficients required for the vocal tract modeling is significantly reduced compared to a LPC system with block-wise adaptation while exceeding its prediction gain. As a by-product it is found that the Doppler frequency of the vocal tract is in the order of 10 Hz. A generalization of the algorithm to an auto-regressive moving average model with time-correlated filter coefficients is straight forward.

关键词： speech coding linear predictive coding Fourier series

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Wavelet scalable speech coding using algebraic quantization

Wavelet scalable speech coding using algebraic quantization

引用

33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： De Meuleneire, M. Taddei, H. Pastor, D. Nokia Siemens Networks COO RTP PT HWCT Comp Techn SDE Otto Hahn Ring 6 D-81739 Munich Germany ENST Bretagne Technop Brest Irois Dept SC F-29238 Brest 3 France

ISBN: (纸本)9781424414833

This paper proposes a new structure for a scalable codec. Our proposed codec works with 10 ms input frame for wideband speech and audio signals at bit rates ranging from 8 to 32 kbit/s. The core layer is the ITU-T G.729 at 8 kbit/s producing a narrowband output. The first enhancement layer is a band-witdh extension providing a wideband output with 2 kbit/s. The second enhancement layer is based on algebraic quantization of wavelet packet coefficients and improves gradually the synthesized signal as the bitrate increases. For speech signals, at bitrates of 24 and 32 kbit/s, the codec is shown to be equivalent to the ITU-T G.722 codec at 56 and 64 kbit/s, respectively. Moreover, the codec at 32 kbit/s is assessed to be equivalent to the recently standardized embedded codec ITU-T G.729.1 at the same bitrate with a lower algorithmic delay.

关键词： embedded coding linear predictive coding bandwidth extension wavelets algebraic quantization

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Fast vector quantization search using multipath tree in the structuralized codebook of LSF coefficients

Fast vector quantization search using multipath tree in the ...

引用

International Conference on Signals and Electronic Systems, ICSES

作者： Sawicki Andrzej Petrovsky Alexander Department of Real Time Systems Bialystok Technical University Bialystok Poland

For mobile communication systems computational complexity and memory requirements are serious problems in real-time digital signal processing of speech signal. In this article we proposed new structuralization algorithm intended to split vector quantizer codebook of LSF coefficients. Fast search procedure, based on structure of codebook and description tree, allows reduce the entire quantity of comparisons over searching the codebook. Our approach allows us to eliminate in search procedure codevectors with minimal probability of belonging to solution and create fast codebook search algorithm with significant decrease of complexity.

关键词： Vector quantization Speech coding Signal processing algorithms linear predictive coding Mobile communication Computational complexity Frequency Real time systems Digital signal processing Speech processing

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Vocal tract area based formant tracking using particle filter

Vocal tract area based formant tracking using particle filte...

引用

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

作者： Kaustubh Kalgaonkar Mark Clements School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA

This paper presents a novel method for estimating formant frequencies and bandwidths based on an underlying vocal tract model. A novel statistical model for vocal tract cross-sectional areas is developed which allows computation of full likelihood functions. Modifications to the basic particle filter algorithm have also been developed to help combat both diversity depletion and convergence problems. The performance of the method is evaluated against hand labeled formant database (L. Deng et al., 2003).

关键词： Particle tracking Particle filters Random variables Frequency estimation Resonance linear predictive coding Bandwidth Filtering State-space methods Speech analysis

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