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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Se-Woon Jeon Dongil Hyun Jeongil Seo Young-Cheol Park Dae-Hee Youn School of Electrical & Electronic Eng. Yonsei Univ. Seoul Korea Broadcasting Media Technology Department ETRI Daejeon Korea Computer & Telecommunications Eng. Division Yonsei Univ. Wonju Gangwon Korea

ISBN: (纸本)9781424442959;9781424442966

In this paper, we propose a method of restoring principal to ambient energy ratio (PAR) at the decoder in the principal component analysis (PCA)-based parametric audio coding. The conventional approach applies the post-scaling at the decoder using the energy information extracted from the input signal at the encoder. However, this approach has a problem that the relative energy of principal source in the reconstructed signal is smaller than the original signal and also affected by the panning angle of principal source. To restore the PAR at the decoder, the proposed method estimates the post-scaling factors using parametric information extracted from PCA-based formulation. The objective and subjective results verify the performance improvement of proposed method.

关键词： Parametric audio coding PCA stereo audio coding principal components analysis decoders Passive Cutaneous Anaphylaxis Energy information audio Energy sources ambient environment input signal Visible Light JTB gene performance boost original signal principal Principal

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Performance Analysis and Evaluation of AVS-M audio coding

Performance Analysis and Evaluation of AVS-M Audio Coding

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2010 International Conference on audio,Language and Image Processing(2010年音频、语言与图像处理国际会议 ICALIP 2010)

作者： Zhang Wen Zhang Tao Zhao Liang Zhang Hai School of Electronic and Information Engineering Tianjin University China

ISBN: (纸本)9781424458653

AVS-M audio standard, which targets at wireless network and mobile equipment, is now independently drawn up in China. Its framework is similar to that of AMR-WB+. The performance of AVS-M audio core algorithms is analyzed in this paper. In order to analyze its complexity fixed-point version of AVS-M codec are implemented on DSP platform. At last, a performance evaluation between AVS-M and AMR-WB+ is discussed. The result is that AVS-M audio performance is no worse than AMR-WB+ on average.

关键词： AMR-WB Mobile devices Performance Evaluation audio coding audio Wireless networks

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SINUSOIDAL SPATIAL audio coding FOR LOW-BITRATE BINAURAL REPRODUCTION

SINUSOIDAL SPATIAL AUDIO CODING FOR LOW-BITRATE BINAURAL REP...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Toni Hirvonen Athanasios Mouchtaris Department of Computer Science University of Crete and Institute of Computer Science Foundation for Research and Technology - Hellas (FORTH-ICS) Heraklion Crete Greece

ISBN: (纸本)9781424442959;9781424442966

A binaural audio synthesis system based on sinusoidal modeling is proposed for spatial, low-bitrate audio coding utilized for example in teleconference applications. The system transmits monaural sinusoidal parameters of a downmix signal, from which the left and right binaural signals are synthesized according to the directional metadata at the receiver. Typical sinusoidal synthesis methods, as well as the effectiveness of a monaural frequency masking model, are evaluated in binaural context. Furthermore, a method for binaural noise residual synthesis and efficiency improvements for HRTF parameter acquisition are suggested. Tests utilizing speech signals indicate that sinusoidal modeling is an attractive technique for applications such as the proposed one.

关键词： Binaural synthesis sinusoidal coding spatial audio coding audio coding Sinusoidal audio monaural copying Reproduction signals head-related transfer function

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MULTICHANNEL audio coding USING SINUSOIDAL MODELLING AND COMPRESSED SENSING

MULTICHANNEL AUDIO CODING USING SINUSOIDAL MODELLING AND COM...

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European Signal Processing Conference

作者： Anthony Griffin Toni Hirvonen Athanasios Mouchtaris Panagiotis Tsakalides Institute of Computer Science Foundation for Research and Technology - Hellas (FORTH-ICS) and Department of Computer Science University of Crete

This paper explores the potential of applying compressed sensing (CS) to multichannel audio coding. In this context, we consider how sinusoidally-modelled multichannel audio signals might be encoded using compressed sensing, as opposed to directly encoding the sinusoidal parameters (amplitude, frequency, phase) as current state-of-the-art methods do. The results, obtained from listening tests using 80 sinusoids per frame with no residual noise signal, show that such a model can achieve equal or better performance to that of the state-of-the-art methods. Given that CS can lead to novel coding systems where the sampling and compression operations are combined into one low-complexity step, this can be considered as an important step towards applying the CS framework to audio coding applications.

关键词： audio coding Compressed sensing audio signals audio MULTICHANNEL COMPRESSED carbon monosulfide CONTAINMENT SPRAY Cesium noise signal conditioned stimulus

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A New Efficient Method of Computing MDCT in MP3 audio coding

A New Efficient Method of Computing MDCT in MP3 Audio Coding

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International Conference on Multimedia Information Networking and Security (MINES)

作者： Guixing Wu En-hui Yang School of Software Engineering Suzhou Institute for Advanced Technology USTC Suzhou Jiangsu China Department of Electrical and Computer Engineering University of Waterloo Waterloo ONT Canada

In this paper, a new efficient method of computing modified discrete cosine transform (MDCT) in MP3 audio coding is proposed. We first define a scaled discrete cosine transform (SDCT), then convert N-point MDCT to N/2-point SDCT. Next we factor N/2-point SDCT matrix mathematically. The factored SDCT matrices are used to develop a fast MDCT algorithm for realizing N-point MDCT. We show that the proposed MDCT algorithm outperforms all fast MDCTalgorithms developed in the literature in terms of computational complexity.

关键词： Digital audio players Signal processing algorithms audio coding Matrix converters Discrete cosine transforms Mirrors Matrix decomposition

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Rate-Distortion Modeling of Spherical Vector Quantization Performance in Transform Excitation audio coding

Rate-Distortion Modeling of Spherical Vector Quantization Pe...

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Electronics, Computer, Telecommunications and Information Technology International Conference on Electrical Engineering/Electronics, Computer, Telecommunications and Information Technology

作者： Wisarn Patchoo Thomas R. Fischer School of EECS Washington State University

ISBN: (纸本)9781424456062

A block-based Gaussian mixture model (GMM) is used to model the distribution of the transform audio data encoded using spherical vector quantization and lossless coding. The expectation-maximization algorithm is used to design the GMM to model the marginal density of the transform coefficients and the block energy density. A GMM-based rate-distortion function is derived and shown to closely match the observed spherical VQ performance.

关键词： Gaussian distribution audio coding Rate distortion theory And vector quantization

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An Improved Blind Bandwidth Extension Method for audio coding

An Improved Blind Bandwidth Extension Method for Audio Codin...

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WRI International Conference on Communications and Mobile Computing, CMC

作者： Hui Liu Shenghui Zhao Jing Wang Jingming Kuang School of Information and Electronics Beijing Institute of Technology Beijing China

ISBN: (纸本)9781424463275;9780769539898

Some existing blind bandwidth extension algorithms are restricted to linear extrapolation, such as linear spectrum replication. In this paper, we propose an improved blind bandwidth extension method using a non-linear extrapolation to reconstruct the missing high-frequency components. By training weights of the neural network in real time, the high-band signal can be reconstructed dynamically. Objective and subjective listening tests are used to evaluate the performance of the proposed algorithm, and the results show that it obtains higher quality signals.

关键词： Bandwidth audio coding Extrapolation Neural networks Cutoff frequency Testing Frequency domain analysis Mobile communication Mobile computing Prediction algorithms

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GAUSSIAN-MIXTURE MODELING OF LATTICE-BASED SPHERICAL VECTOR QUANTIZATION PERFORMANCE IN TRANSFORM audio coding

GAUSSIAN-MIXTURE MODELING OF LATTICE-BASED SPHERICAL VECTOR ...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Wisarn Patchoo Thomas R. Fischer School of EECS Washington State University Pullman WA 99163-2752 USA

ISBN: (纸本)9781424442959;9781424442966

A block-based Gaussian mixture model (GMM) is used to model the distribution of transform audio data to be encoded using lattice-based spherical vector quantization (LSVQ). The expectation-maximization algorithm is used to design the GMM to model the marginal density of the transform coefficients and the vector energy density. A GMM-based rate-distortion function is derived and shown to closely match the observed spherical VQ performance. The LSVQ transform audio coding performance is characterized for the best lattices known in 4, 8, 16, and 32 dimensions.

关键词： Gaussian distributions audio coding rate distortion theory vector quantization audio coding Vector quantization Rate distortion theory Gaussian distribution Sphericity Gm(m) audio Quantization

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An FPGA implementation of Dynamic Reconfigurable Lifting-Based Wavelet Packet Processor for Real-Time Scalable audio coding

An FPGA implementation of Dynamic Reconfigurable Lifting-Bas...

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2010 International Conference on audio,Language and Image Processing(2010年音频、语言与图像处理国际会议 ICALIP 2010)

作者： Maxim Rodionov Alexey Petrovsky Alexander Petrovsky Wanggen Wan Department of Computer Engineering Belarusian State University of Informatics and Radioelectronics School of Communication and Information Engineering Shanghai University

ISBN: (纸本)9781424458653

In this paper, dynamic algorithm transforms (DAT) for reconfigurable real-time processor for audio application based on the adaptive wavelet packet (WP) decomposition are presented. DAT techniques is to constrain a minimum cost sub-band decomposition of wavelet transform by maximinimizing the minimum masking threshold (which is limited by the perceptual entropy) in every sub-band for the given embedded processor architecture and temporal resolution. The processor architecture is based on the implementation of the wavelet transform by means of its factoring into lifting steps. Practical reconfiguration strategies for the given processor are presented.

关键词： Processor Architectures Wavelet packets PROCESSOR wavelet transforms audio coding configurability subband Field programmable gate arrays Digital audio tape

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Blind bandwidth extension of audio signals based on non-linear prediction and hidden Markov model

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APSIPA TRANSACTIONS ON SIGNAL AND INFORMATION PROCESSING 2014年第1期3卷 e8-e8页

作者： Liu, Xin Bao, Changchun Beijing Univ Technol Sch Elect Informat & Control Engn Speech & Audio Signal Proc Lab Beijing 100124 Peoples R China

The bandwidth limitation of wideband (WB) audio systems degrades the subjective quality and naturalness of audio signals. In this paper, a new method for blind bandwidth extension of WB audio signals is proposed based on non-linear prediction and hiddenMarkovmodel (HMM). The high-frequency (HF) components in the band of 7-14 kHz are artificially restored only from the low-frequency information of the WB audio. State-space reconstruction is used to convert the fine spectrum of WB audio to a multi-dimensional space, and a non-linear prediction based on nearest-neighbor mapping is employed in the state space to restore the fine spectrum of the HF components. The spectral envelope of the resulting HF components is estimated based on an HMM according to the features extracted from theWB audio. In addition, the proposed method and the reference methods are applied to the ITU-T G.722.1WB audio codec for comparison with the ITU-T G.722.1C superWB audio codec. Objective quality evaluation results indicate that the proposed method is preferred over the reference bandwidth extension methods. Subjective listening results show that the proposed method has a comparable audio quality with G.722.1C and improves the extension performance compared with the reference methods.

关键词： audio coding audio bandwidth extension Nearest-neighbor mapping Hidden Markov model

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