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检索条件"主题词=linear predictive coding"
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...
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33rd IEEE International Conference on Acoustics, Speech and Signal Processing
作者: Holters, Martin Zoelzer, Udo Helmut Schmidt Univ Hamburg Germany
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 p... 详细信息
来源: 评论
Voiced/Unvoiced Classification Recovery in the Speech Decoder Based on GMM
Voiced/Unvoiced Classification Recovery in the Speech Decode...
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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
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 g... 详细信息
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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...
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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
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 sche... 详细信息
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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...
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IEEE Region 10 Conference (TENCON 2008)
作者: Nehe, N. S. Holambe, R. S. SGGS Inst Engn & Technol Nanded MS India
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 fra... 详细信息
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PHONETIC AND INTONATION ENHANCEMENT OF SPEECH FOR JAPANESE HEARING-IMPAIRED SPEAKERS
PHONETIC AND INTONATION ENHANCEMENT OF SPEECH FOR JAPANESE H...
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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
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 ... 详细信息
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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
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. Previ... 详细信息
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Doppler-variant modeling of the vocal tract
Doppler-variant modeling of the vocal tract
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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
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 ... 详细信息
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Wavelet scalable speech coding using algebraic quantization
Wavelet scalable speech coding using algebraic quantization
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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
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.7... 详细信息
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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 ...
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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 algorit... 详细信息
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Vocal tract area based formant tracking using particle filter
Vocal tract area based formant tracking using particle filte...
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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 ... 详细信息
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