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International Conference on Communication Technology (ICCT)

作者： J. Herre B. Grill Audio Department Fraunhofer Institute of Integrated Circuits and Systems Erlangen Germany

MPEG-4 is the most recent member of the well-known family of ISO/MPEG coding standards, and provides an integrated set of audio coders with attractive novel capabilities, including bit rate and bandwidth scalability. The paper gives an introduction into the underlying design concepts of MPEG-4 and provides an overview of MPEG-4 audio coding technology and its features. In particular, it discusses the technical provisions which make MPEG-4 audio very well suited for applications in the domain of mobile communications.

关键词： MPEG 4 Standard Mobile communication Bit rate audio coding Channel capacity ISO standards Code standards IEC standards Standards development Bandwidth

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Multiple frequency harmonics analysis and synthesis of audio signals

Multiple frequency harmonics analysis and synthesis of audio...

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

作者： A. Jbira A. Kondoz Centre for Communication Systems Research University of Surrey Guildford Surrey UK

In this paper we propose a multiple frequency harmonics model for analysis and synthesis of audio signals. The novelty of this model is that a composite sound can be represented by a few number (two or three) of frequency harmonics, each with time-varying fundamental frequency. We present the model and some key issues in tuning it. Results demonstrate that the model is valid and promising. It is convenient for time- and frequency-scale modifications and is of interest for low bit rate audio coding.

关键词： Frequency synthesizers Harmonic analysis Signal synthesis Signal analysis Bit rate audio coding Speech synthesis Fourier transforms Autocorrelation Tuning

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Structured coding for content based interactive audio

Structured coding for content based interactive audio

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IEEE International Conference on Multimedia Computing and Systems (ICMCS)

作者： K. Melih R. Gonzalez School of Information Technology Griffith University QLD Australia

The paper addresses the dual problem of audio coding for compression and retrieval. Because of the sheer volume of multimedia data, much research effort has focused on coding techniques for the compression of this data. However, being able to store large amounts of data in a compact fashion has limited application if there is no meaningful way to access and interact with them. Traditionally, these two issues have been treated separately resulting in many problems for multimedia data management. The authors have proposed a solution to these problems for audio by developing a compact, perceptually based, structured audio representation that provides explicit support for content based retrieval, browsing and other interactions. In addition, the representation offers potential for compressed data storage. The paper outlines a coding scheme based on the representation and reports the compression results.

关键词： Content based retrieval audio coding Data compression Information retrieval Music information retrieval Gold Postal services Australia Memory Transform coding

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A progressive algorithm for perceptual coding of digital audio signals

A progressive algorithm for perceptual coding of digital aud...

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Asilomar Conference on Signals, Systems & Computers

作者： Ye Shen Hongmei Ai C.-C.J. Kuo Integrated Media System Center Department of Electrical Engineering-System University of Southern California Los Angeles CA USA

In this paper, a progressive audio coding algorithm is presented. Distinctive from existing transform-based perceptual audio coding schemes, which allow either constant bit rate or multiple fixed bit rates, our proposed algorithm achieves a fully embedded audio coding scheme whose bit rate can be controlled depending on the user's requirement. This feature makes it extremely useful in audio networking applications, such as audio on-demand and audio broadcasting over the Internet. In the proposed audio coding scheme, efficient successive quantization and bit layer coding are applied to the compact audio signal representation obtained from traditional subband decomposition. Our research focuses on the improved successive approximation quantisation (ISAQ), which performs bit allocation for quantization based on psychoacoustic analysis of the input audio signal. Thus the progressive audio coding is achieved, while perceptually transparent audio coding is also maintained to the allowance of bit rate constraint. The performance measurement of our presented algorithm is demonstrated.

关键词： audio coding Bit rate Quantization Broadcasting IP networks Signal representations Psychology Signal analysis Performance analysis Measurement

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Real-time MPEG-1 audio coding aad decoding on a DSP chip

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1997年第1期43卷 40-47页

作者： Murphy, CD Anandakumar, K Moore School of Electrical Engineering University of Pennsylvania Philadelphia PA USA

The MPEG-1 audio standard (ISO/IEC 11172-3) establishes guidelines for the compression of high-quality digital audio signals [I]. The standard dictates the function of an encoder/decoder pair (codec), leaving form intentionally vague to allow for competing implementations. A typical approach to real-time operation is to design an application-specific integrated circuit (ASIC) dedicated to encoding, decoding [2], or both. We present an alternative codec that makes use of the general-purpose digital signal processing (DSP) chips that are now common in multimedia-capable workstations and personal computers. We discuss how selective optimization of codec structure allows robust performance using limited resources, highlight some of the problems inherent in translating the abstractions of the standard into assembly code, and point towards further investigations of real-time implementations of communications standards.

关键词： audio coding Decoding Digital signal processing chips Codecs Transform coding IEC standards ISO standards Code standards Application specific integrated circuits Communication standards

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MPEG digital audio coding - Setting the standard for high-quality audio compression

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IEEE SIGNAL PROCESSING MAGAZINE 1997年第5期14卷 59-81页

作者： Noll, P Technische Universität Berlin Berlin Germany

The Moving Pictures Expert Group (MPEG) within the International Organization of Standardization (ISO) has developed a series of audio-visual standards known as MFEG-1 and MPEG-2. These audio-coding standards are the first international standards in the field of high-quality digital audio compression. MPEG-1 covers coding of stereophonic audio signals at high sampling rates aiming at transparent quality, whereas MPEG-2 also offers stereophonic audio coding at lower sampling rates. In addition, MPEG-2 introduces multichannel coding with and without backwards compatibility to MPEG-1 to provide an improved acoustical image for audio-only applications and for enhanced television and video-conferencing systems. MPEG-2 audio coding without backwards compatibility, called IMPEG-2 Advanced audio coding (AAC), offers the highest compression rates. Typical application areas for MPEG-based digital audio are in the fields of audio production, program distribution and exchange, digital sound broadcasting, digital storage, and various multimedia applications. We describe in some detail the key technologies and main features of MPEG-1 and MPEG-2 audio coders. We also present the MPEG-4 standard and discuss some of the typical applications for MPEG audio compression.

关键词： audio coding Transform coding ISO standards audio compression Image sampling Image coding MPEG standards Standardization Standards development Standards organizations

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The modulated lapped transform, its time-varying forms, and its applications to audio coding standards

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 1997年第4期5卷 359-366页

作者： Shlien, S Communications Rcscarch Centre Ottawa ONT Canada

The modulated lapped transform (MLT) is used in both audio and video data compression schemes, This paper describes its properties and how it can be used to generate a time-varying filterbank, Examples of its implemen... 详细信息

关键词： Modulation coding audio coding Discrete cosine transforms Transform coding Signal processing algorithms audio compression Discrete transforms Video compression Data compression Filter bank

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On fast algorithms for computing the inverse modified discrete cosine transform

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IEEE SIGNAL PROCESSING LETTERS 1999年第3期6卷 61-64页

作者： Fan, YH Madisetti, VK Mersereau, RM Georgia Inst Technol Ctr Signal & Image Proc Elect & Comp Engn Atlanta GA 30332 USA

Two new fast algorithms for computing the inverse modified discrete cosine transform (IMDCT) as used in the adaptive spectral entropy coder (ASPEC) [2] are proposed. A fixed-point error analysis is presented to determ... 详细信息

关键词： audio coding discrete cosine transform discrete Hartley transform discrete transform error analysis quantization signal reconstruction

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MPEG-2 AAC decoder on a fixed-point DSP

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1999年第4期45卷 1200-1205页

作者： Chen, J Tai, HM Univ Tulsa Dept Elect Engn Tulsa OK 74104 USA

This paper presents the implementation of MPEG-2 Advanced audio coding (AAC) decoder on a fixed-point DSP. AAC is a powerful coding algorithm and offers the high-duality multi-channel surround audio. We will also disc... 详细信息

关键词： Decoding Digital signal processing Bit rate Filter bank audio coding Transform coding Signal processing algorithms Internet Bandwidth Shape

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Structured audio and effects processing in the MPEG-4 multimedia standard

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MULTIMEDIA SYSTEMS 1999年第1期7卷 11-22页

作者： Scheirer, ED MIT Media Lab Machine Listening Grp Cambridge MA 02139 USA

While previous generations of the MPEG multimedia standard have focused primarily on coding and transmission of content digitally sampled from the real world, MPEG-4 contains extensive support for structured, synthetic and synthetic/natural hybrid coding methods. An overview is presented of the "Structured audio" and "audioBIFS" components of MPEG-4, which enable the description of synthetic soundtracks, musical scores, and effects algorithms and the compositing, manipulation, and synchronization of real and synthetic audio sources. A discussion of the separation of functionality between the systems layer and the audio toolset of MPEG-4 is presented, and prospects for efficient DSP-based implementations are discussed.

关键词： structured audio audio coding audio synthesis MPEG-4 multimedia standards

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