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检索条件"主题词=Audio coding"

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Estimating spatial cues for audio coding in MDCT domain

Estimating spatial cues for audio coding in MDCT domain

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： Shuixian Chen Ruimin Hu Shuhua Zhang Computer School Wuhan University of China Wuhan China National Engineering Research Center of Multimedia Software Wuhan University of China Wuhan China Department of Electronic Engineering Tsinghua University Beijing China

Although widely used otherwise, MDCT is excluded in the current scheme for spatial cues representation, due to its lacking of phase information and energy conservation. But combining MDCT with MDST overcomes the difficulties. Moreover, MDST spectra can be built perfectly from neighboring MDCT spectra. The MDCT-MDST conversion, in matrix form, is approximating to a banded sparse matrix. When applied to spatial audio coding using MDCT based core coders, this method avoids separate transforming for cues representation and saves significant computation. Listening tests also show that it has same audio quality as other complex transform based methods.

关键词： audio coding Discrete transforms Power generation economics Delay Testing Energy conservation Sparse matrices Computational complexity Mirrors Filter bank

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Binaural reproduction for Directional audio coding

Binaural reproduction for Directional Audio Coding

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Mikko-Ville Laitinen Ville Pulkki Department Signal Processing and Acoustics Helsinki University of Technology Espoo Finland

Directional audio coding (DirAC) is a recently proposed method for spatial sound reproduction. So far it has been used only in loud-speaker reproduction, and a method for headphone reproduction is presented in this article. In principle, the method uses virtual loud-speakers simulated with head related transfer functions (HRTFs), and head tracking. The method was evaluated subjectively, and the results are presented. The results show that a plausible spatial impression can be reproduced using the binaural realization of DirAC.

关键词： audio coding Loudspeakers Microphones Frequency Signal analysis Acoustic signal processing Acoustic applications Signal synthesis Conferences Headphones

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An efficient transient audio coding algorithm based on DCT and Matching Pursuit

An efficient transient audio coding algorithm based on DCT a...

引用

Congress on Image and Signal Processing, CISP

作者： Jing Wang Hailong He Ning Ning Jingming Kuang Department of Electronic Engineering Beijing Institute of Technology Beijing China

The paper proposes an efficient transient audio coding algorithm based on Discrete Cosine Transform (DCT) and sine extraction with Matching Pursuit (MP), which can be used conveniently in the sinusoidal parametric audio coder. The overlapped input frame is firstly classified into transient and non-transient signal. The transient detection is designed based on the energy ratio of the overlapped subframe to the whole frame. The transient signal can be represented with sinusoidal model in the DCT domain. After DCT processing, the sinusoids are extracted based on MP algorithm. The transient coding parameters contain a series of amplitudes, frequencies and phases in the DCT domain. Finally, a 15.5kbps wide-band transient audio coder is designed and the synthesized transient signal performs a little better than the one coded by MPEG-4 sinusoidal parametric coder.

关键词： Transient analysis Discrete cosine transforms Matching pursuit algorithms audio coding Algorithm design and analysis Noise

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Implementation of the MPEG-4 advanced audio coding encoder on ADSP-21060 SHARC

Implementation of the MPEG-4 advanced audio coding encoder o...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Dong-Yan Huang Xuesong Gong Daqing Zhou T. Miki S. Hotani Institute of Microelectronics Singapore Multimedia Signal Processing Laboratory Multimedia Laboratories NTT Mobile Communications Network Inc. Yokosuka Kanagawa Japan

In this paper, we present the implementation of the MPEG-4 advanced audio coding (MPEG-4 AAC) encoder on a floating-point DSP (ADSP-21060 SHARC) based hardware. We discuss how selective optimization of the encoder verification model (VM) software structure allows robust performance using limited resources, highlight some of the problems inherent in its real-time implementation, and point towards further investigations of real-time implementation and advantages of DSP in audio processing.

关键词： MPEG 4 Standard audio coding Transform coding Digital signal processing Noise shaping Quantization Sampling methods Bit rate Filter bank Kernel

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Very low bitrate spatial audio coding with dimensionality reduction

Very low bitrate spatial audio coding with dimensionality re...

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Christian Rohlfing Jeremy E. Cohen Antoine Liutkus Institut fur Nachrichtentechnik RWTH Aachen University Germany Department of Image and Signal-processing Gipsa-lab CNRS Grenoble France Inria speech processing team Villers-les-Nancy France

ISBN: (纸本)9781509041183

In this paper, we show that tensor compression techniques based on randomization and partial observations are very useful for spatial audio object coding. In this application, we aim at transmitting several audio signals called objects from a coder to a decoder. A common strategy is to transmit only the downmix of the objects along some small information permitting reconstruction at the decoder. In practice, this is done by transmitting compressed versions of the objects spectrograms and separating the mix with Wiener filters. Previous research used nonnegative tensor factorizations in this context, with bitrates as low as 1 kbps per object. Building on recent advances on tensor compression, we show that the computation time for encoding can be extremely reduced. Then, we demonstrate how the mixture can be exploited at the decoder to avoid the transmission of many parameters, permitting bitrates as low as 0.1 kbps per object for comparable performance.

关键词： Spatial audio object coding informed source separation higher order SVD dimensionality reduction dimensionality reduction Bit rate audio coding decoders computational time TRANSMITTING higher-order ENCODER Tensor

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A Novel Bit-Plane Shifting Algorithm for Scalable audio coding

A Novel Bit-Plane Shifting Algorithm for Scalable Audio Codi...

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International Conference on Networking (ICN)

作者： Bo Zhang Weibei Dou Tsinghua National Laboratory for Information Science and Technology Department of Electronic Engineering Tsinghua University Beijing China

ISBN: (纸本)9781424460830

Bit-plane shifting is one of the strategies for providing Fine Grain Scalability (FGS) in scalable audio coding. It is still desirable to cater the network bandwidth, device capabilities and user preferences. In this paper, a Noise to Mask Ratio (NMR) based bit-plane shifting algorithm is proposed. It is different from just sequential bit-plane coding used in MPEG 4 scalable to lossless (SLS) audio coding, and also different from other existed strategies at lower bit-rate audio coding. It firstly transforms normal bit-planes corresponding with scalefactor bands to the priority bit-planes by a bit-plane shifting according to their NMR. And then, a simple sequential bit-plane coding is used. The priority bit-plane means that a bit-plane corresponding to a scalefactor band which have larger NMR takes a higher priority of coding. Both the objective and the subjective audio quality evaluations give us good scores by comparing with the sequential bit-plane coding of SLS and Just Noticeable Distortion (JND) method. Otherwise, no extra side information is needed and the computational complexity is reduced substantially. The proposed NMR-based bit-plane shifting algorithm obtains good performance in both subjective and objective assessments while leaving the computational complexity at a low level.

关键词： audio coding Laser sintering Bandwidth Nuclear magnetic resonance MPEG 4 Standard Scalability Signal to noise ratio Computational complexity Error correction codes Streaming media

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A Prony method for high-quality audio coding

A Prony method for high-quality audio coding

引用

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

作者： M. Paraskevas D. Tsoukalas J. Mourjopoulos Wire Communication Laboratory University of Patras Patras Greece Wire Commun. Lab. Patras Univ. Greece

A technique for high-quality audio signal coding is proposed. The technique follows Prony modeling of the audio signal. A new method for dynamic estimation of the model order is proposed, based on an analysis/synthesis process with psychoacoustic criteria. According to this approach the model order is iteratively estimated, until the distortion introduced by inaccurate order selection becomes inaudible. An average coding rate gain of 6:1 can be achieved in this way without loss in signal quality.< >

关键词： audio coding Psychoacoustic models Transform coding Discrete Fourier transforms Quantization Optimization methods Discrete cosine transforms Computational complexity Frequency Least squares approximation

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Balanced-delay filterbank for Closed-Loop Spatial audio coding

Balanced-delay filterbank for Closed-Loop Spatial Audio Codi...

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International Seminar on Intelligent Technology and Its Applications (ISITIA)

作者： Ikhwana Elfitri Heru Dibyo Laksono Al Kautsar Permana Department of Electrical Engineering Andalas University Padang Sumatera Barat Indonesia

Closed-loop configuration has been introduced to Spatial audio coding (SAC) which make it possible to minimise distortion introduced during the quantisation and encoding processes. Significant performance improvement has been shown and reported in some papers. However, implementation of closed-loop system directly to MPEG Surround (MPS) is still problematic due to the unbalanced-delay filterbank that is used in the MPEG standard which is not appropriate for a closed-loop system that needs synchronisation between the original audio signals with the target audio signals. In this paper, we investigate the delay characteristic of the Quadrature Mirror Filterbank (QMF) which is used in several MPEG standards. Based on this study, a balanced-delay QMF is proposed and tested in a closed-loop MPS system. The results of the experiments show that approximately 8 dB of SNR improvement is achieved when applying balanced-delay filterbank compared to the unbalanced-delay filterbank as specified in MPS.

关键词： Standards audio coding Signal to noise ratio Transform coding Delays Filter banks

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MPEG-4 ALS: an emerging standard for lossless audio coding

MPEG-4 ALS: an emerging standard for lossless audio coding

引用

Data Compression Conference (DCC)

作者： T. Liebchen Y.A. Reznik Technical University Berlin Berlin Germany Real NEtworks Inc. Seattle WA USA

This paper provides a brief overview of an emerging ISO/IEC standard for lossless audio coding, MPEG-4 ALS and explains the choice of algorithms used in its design, and compare it to current state-of-the-art algorithm... 详细信息

关键词： MPEG 4 Standard audio coding Data compression

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Low delay moving-horizon multiple-description audio coding forwireless hearing aids

Low delay moving-horizon multiple-description audio coding f...

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

作者： Jan Ostergaard Daniel E. Quevedo Jesper Jensen Department of Electronic Systems University of Aalborg Aalborg Denmark School of Electrical Engineering and Computer Science University of Newcastle Australia Oticon Copenhagen Denmark

ISBN: (纸本)9781424423538

In this work, we construct a novel scheme for efficient perceptual coding of audio for robust communication between encoders and wireless hearing aids. To limit the physical size of the hearing aids and to reduce power consumption and thereby increase the lifetime expectancy of the batteries, the hearing aids are constrained to be of low complexity. We therefore provide an asymmetric strategy where most of the computational load is placed at the encoding side. We make use of multiple-description coding. This combats possible erasures on the wireless link between the encoder and the hearing aids without introducing significant delay. Furthermore, we employ psychoacoustically optimized noise-shaping quantizers based on the moving-horizon principle, which exploits a finite prediction horizon.

关键词： Delay audio coding Hearing aids Psychoacoustic models Decoding Robustness Quantization Filters Batteries Noise shaping

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