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IEEE SIGNAL PROCESSING MAGAZINE 2002年第4期19卷 32-41页

作者： Konstantinides, K Chen, CT Chen, TC Cheng, H Jeng, FC Cirrus Log Video Recording Grp Austin TX USA

An overview of the critical engineering concerns was presented for the design and implementation of an moving picture experts group (MPEG)-2 a/v codec integrated circuit (IC). The codec with unique and flexible reduced instruction set computing (RISC) controller and video input unit (VIU), a programmable audio digital signal processor (DSP), and hardwired video processing units was designed. The codec was implemented using a standard-cell library in 0.18 mu m complementary metal oxide semiconductor (CMOS) technology.

关键词： code standards codecs digital signal processing chips software architecture speech codecs telecommunication standards video codecs video recording MPEG a/v codec MPEG specifications MPEG-1 standard MPEG-2 standard audio coding hardware architecture real- code standards speech codecs audio coding system requirements video codecs Real number digital signal processing chips Video Recording hardware architecture Software architecture codecs

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An MPEG-2 CODEC LSI with an audio accelerator for camcorders

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 2002年第3期48卷 656-661页

作者： Nakamura, T Marumori, H Takahashi, M Fujii, Y Digital Media Syst. R&D Div. Hitachi Ltd. Tokyo Japan

An MPEG-2 CODEC LSI with the audio accelerator for MPEG-2 CODEC system has been developed The audio accelerator assists audio CODEC processing on the host CPU, and enables to use a low-frequency host CPU and contribut... 详细信息

关键词： code standards telecommunication standards large scale integration digital signal processing chips video coding audio coding video codecs video cameras MPEG-2 codec LSI audio accelerator camcorders audio codec processing LF CPU power con Video coding code standards Camcorders audio coding Video cameras video codecs digital signal processing chips audio Large scale integrate circuit codecs

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A unified fast MDCT/MDST computation in the evenly stacked analysis/synthesis system

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CIRCUITS SYSTEMS AND SIGNAL PROCESSING 2002年第4期21卷 415-426页

作者： Britanak, V Rao, KR Slovak Acad Sci Inst Informat Bratislava 84237 Slovakia Univ Texas Dept Elect Engn Arlington TX 76019 USA

The modified discrete cosine transform (MDCT) and modified discrete sine transform (MDST) are employed in subband/transform coding schemes as the analysis/synthesis filter banks based on time domain aliasing cancellation. Princen, Bradley, and Johnson defined two types of MDCT, specifically, for an evenly stacked and an oddly stacked analysis/synthesis system. A new unified fast algorithm for the forward and inverse MDCT/MDST computation in the evenly stacked system is proposed. It is based on the DCT of type II (DCT-II) and the DST of type IV (DST-IV), and uses real arithmetic only. The corresponding generalized signal flow graph is regular and structurally simple and enables one to compute alternate MDCT/MDST and their inverses in general for any N divisible by 4 (N being the length of a data sequence).

关键词： modified discrete cosine transform modified discrete sine transform audio coding

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Robust signal/noise discrimination for wideband speech and audio coding

Robust signal/noise discrimination for wideband speech and a...

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IEEE Workshop on Speech coding Proceedings

作者： M. Jelinek F. Labonte Department of Electrical and Computer Engineering University of Sherbrooke Sherbrooke QUE Canada Universite de Sherbrooke Sherbrooke QC CA

We present a robust discrimination method to separate information carrying signals from ambient noise for wideband coding applications. The method consists of a two-stage procedure. First, a local decision is made based on a set of extracted parameters to update the estimated noise level. The parameters have been chosen to reliably detect speech as well as music signals. Then, the final decision is made based only on a frequency dependent signal to noise ratio (SNR). The noise level update does not depend on the final decision to prevent the discriminator from locking when noise level changes suddenly. The performance is compared with the performance of the voice activity detector (VAD) of G.729, Annex G.

关键词： Noise robustness Wideband Speech enhancement Speech coding audio coding Noise level Signal to noise ratio Multiple signal classification Signal detection Signal processing

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Card-sized portable audio player using high quality audio coding technology TwinVQ

Card-sized portable audio player using high quality audio co...

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IEEE International Conference on Consumer Electronics (ICCE)

作者： T. Takahashi T. Morita T. Shimoda T. Kaneko T. Moriya Kobe Steel Company Limited Kobe Japan NTT Cyber Space Laboratories Tokyo Japan

This paper presents an audio player using TwinVQ, an advanced high quality coding technology, which can compress 44.1 kHz sampled audio data at a very low bit rate of 40 kbit/s/ch. The player is business-card-sized of... 详细信息

关键词： audio coding Bit rate Decoding Space technology Digital signal processing Quantization Solid state circuits Flash memory Steel Laboratories

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Technical advances in digital audio radio broadcasting

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PROCEEDINGS OF THE IEEE 2002年第8期90卷 1303-1333页

作者： Faller, C Juang, BH Kroon, P Lou, HL Ramprashad, SA Sundberg, CEW Bell Labs Murray Hill NJ 07974 USA

The move to digital is a natural progression taking place in all aspects of broadcast media applications from document processing in newspapers to video processing in television distribution. This is no less true for audio broadcasting which has taken a unique development path in the United States. This path has been heavily influenced by a combination of regulatory and migratory requirements specific to the U.S. market. In addition, competition between proposed terrestrial and satellite systems combined with increasing consumer expectations have set ambitious, and often changing, requirements for the systems. The result has been a unique set of evolving requirements on source coding, channel coding, and modulation technologies to make these systems a reality.. This paper outlines the technical development of the terrestrial wireless and satellite audio broadcasting systems in the U.S., providing details on specific source and channel coding designs and adding perspective on why specific designs were selected in the final systems. These systems are also compared to other systems such as Eureka-147, DRM, and Worldspace, developed under different requirements.

关键词： audio coding channel coding digital sound broadcasting

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Multichannel audio coding and its applications in DAB and DVB

Multichannel audio coding and its applications in DAB and DV...

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International Conference on Signal Processing Proceedings (ICSP)

作者： M. Bosi Digital Theater Systems Inc. Agoura Hills CA USA

In spite of a steady increase in storage media capacity and transmission bandwidth, high quality, multichannel audio creates a challenge for traditional and new delivery media. This paper presents an overview of the technologies adopted in multichannel audio coding. New trends, including increased audio sample resolution and sampling rates, are discussed. Different channel configurations, their advantages and disadvantages, and their impact on system data rates and capacity are examined. Emerging formats like DAB and DVB are introduced. Finally, applications of multichannel audio in DAB and DVB are examined.

关键词： audio coding Digital video broadcasting Loudspeakers Frequency Bandwidth Sampling methods Board of Directors Marine technology Signal to noise ratio Dynamic range

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HILN-the MPEG-4 parametric audio coding tools

HILN-the MPEG-4 parametric audio coding tools

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： H. Purnhagen N. Meine Lab. fur Informationstechnol. Hannover Univ. Germany Laboratorium für Informationstechnologie University of Hannover Hanover Germany

The MPEG-4 audio Standard combines tools for efficient and flexible coding of audio. For very low bitrate applications, tools based on a parametric signal representation are utilised. The parametric speech coding tools (HVXC) are already available in Version 1 of MPEG-4. The main focus of this paper is on the parametric audio coding tools "Harmonic and Individual Lines plus Noise" (KILN) which are included in Version 2 of MPEG-4. As already indicated by their name, the HILN tools are based on the decomposition of the audio signal into components which are described by appropriate source models and represented by model parameters. This paper gives an overview of the HILN tools, presents the recent advances in signal modelling and parameter coding, and concludes with an evaluation of the subjective audio quality.

关键词： MPEG 4 Standard audio coding Bit rate Speech coding Scalability Mobile communication Laboratories Signal representations Context Digital multimedia broadcasting

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Hybrid audio coding for speech and audio below medium bit rate

Hybrid audio coding for speech and audio below medium bit ra...

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IEEE International Conference on Consumer Electronics (ICCE)

作者： K. Makino J. Matsumoto Dev. Center Sony Corp. Tokyo Japan HomeNet Processing Laboratory Development CenterHome Network Company Sony Corporation USA

A novel a audio/speech coding algorithm, hybrid audio coding (HAC) is described. New features of the algorithm include window switching with generalized MDCT, an improved quantization scheme of the MDCT coefficients, and waveform normalization in the time domain. HAC provides a good quality at a bit rate of 8 to 16 kbps, and it is also proven that the developed algorithm is effective for both audio and speech signals.

关键词： audio coding Speech coding Bit rate Codecs Testing Linear predictive coding Quantization Multiple signal classification Decoding Smoothing methods

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Complexity scalable audio coding algorithm based on wavelet packet decomposition

Complexity scalable audio coding algorithm based on wavelet ...

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International Conference on Signal Processing Proceedings (ICSP)

作者： He Dongmei Gao Wen Wu Jiangqin Dept. of Comput. Sci. & Eng. Harbin Inst. of Technol. China Department of Computer Science and Engineering Harbin Institute of Technology Harbin

In this paper, a complexity scalable audio coding algorithm based on wavelet packet decomposition is proposed. The incomplete wavelet packet decomposition with the property of complexity scalability to the signals is described in this paper. The zerotree coding of the coefficients is implemented which uses the characteristics of human ears and the self-similarity among different subbands wavelet coefficients. This algorithm cannot only be used to achieve transparent quality reconstruction audio at low bit rate, but also enable to support complexity scalable coding and multirate scalable coding. audio encoding and decoding can be realized in real-time utilizing this algorithm on computers with different levels of computational power. The experimental results indicate that this algorithm is of practical value to the real-time multimedia applications.

关键词： audio coding Wavelet packets Scalability Humans Ear Wavelet coefficients Bit rate Encoding Decoding Application software

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