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Midwest Symposium on Circuits and Systems (MWSCAS)

作者： Do Hyoung Kim Dong Hyun Kim Jae Ho Chung Dept. of Electronic Engr. Inha Univ. Inchon KOREA

ISBN: (纸本)078037150X

MPEG-4 General audio (GA) Encoder/Decoder has inherited MPEG-2 AAC (Advanced audio coding) and added TVQ and other algorithms as scalable options. In this paper, the authors deal with the method of optimization of a MPEG-4 AAC scalable encoder. An original MPEG-4 VM source takes 15 times to 30 times the encoding time of the original sample playing time as its options, although this can be changed for varying CPU and memory conditions. The authors mainly used the reduction of unnecessary loop or the loop-unrolling technology to improve the encoder performance. For this work, firstly they performed the profiling based on general PC and checked the frequency or complexity of each function, and then selected the target function to be optimized.

关键词： MPEG 4 Standard Encoding Frequency Transform coding Tiles Decoding audio coding Optimization methods Virtual manufacturing Compression algorithms

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New recursive algorithms for the forward and inverse MDCT

New recursive algorithms for the forward and inverse MDCT

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IEEE Workshop on Signal Processing Systems (SIPS)

作者： V. Nikolajevic G. Fettweis Mannesmann Mobilfunk Chair for Mobile Communications Systems Dresden University of Technology Dresden Germany

Forward and inverse MDCT (modified discrete cosine transform) are two of the most computationally intensive operations in the MPEG audio coding standard. We derive sinusoidal recursive formulas for transforming kernels of the MDCT and IMDCT. Then we efficiently implement general length MDCT and IMDCT using the regressive structure derived from the sinusoidal recursive formulas. The proposed regular structure is particularly suitable for parallel VLSI realization. Our solution requires less hardware than a recently proposed one.

关键词： audio coding Very large scale integration Discrete cosine transforms Forward contracts Kernel Hardware Decoding Mobile communication Communication standards Filters

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Scalable and progressive audio codec

Scalable and progressive audio codec

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

作者： M.S. Vinton L.E. Atlas Department of Electrical Engineering University of Washington Seattle WA USA

A source coding technique for variable, bandwidth-constrained channels such as the Internet must do two things: offer high quality at low data rates, and adapt gracefully to changes in available bandwidth. Here we propose an audio coding algorithm that is superior on both counts. It is inherently scalable, meaning that channel conditions can be matched without the need for additional computation. Moreover, it is compact: in subjective tests our algorithm, coded at 32 kb/s/channel, outperformed MPEG-1 Layer 3 (MP3) coded at 56 kb/s/channel (both at 44.1 kHz). We achieve this simultaneous increase in compression and scalability through use of a two-dimensional transform that concentrates relevant information into a small number of coefficients.

关键词： Codecs audio coding Scalability Frequency modulation Bandwidth Bit rate Source coding Testing Transform coding Digital audio players

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Fast encoding algorithms for MPEG-4 TwinVQ audio tool

Fast encoding algorithms for MPEG-4 TwinVQ audio tool

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

作者： N. Iwakami T. Moriya A. Jin T. Mori K. Chikira NTT Laboratories Musashino Tokyo Japan

The ISO/IEC MPEG-4 audio standard includes the TwinVQ encoding tool. This tool is suitable for low-bit-rate general audio coding, but a drawback is the computational complexity of the encoder. To develop a faster TwinVQ encoder, new fast vector quantization algorithms - area localized pre-selection and hit zone masking - are introduced. These algorithms exploit the pre- and main-selection procedure scheme of the conjugate structure vector quantization which is used in the TwinVQ. The improvement is evaluated by measuring the encoding speed and the sound quality of reproduction.

关键词： Encoding MPEG 4 Standard Distortion measurement Vector quantization ISO standards IEC standards audio coding Decoding Computational complexity Laboratories

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A class of biorthogonal nonuniform cosine-modulated filter banks with lower system delay

A class of biorthogonal nonuniform cosine-modulated filter b...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： X.M. Xie S.C. Chan T.I. Yuk Department of Electrical and Electronic Engineering University of Hong Kong Hong Kong China

ISBN: (纸本)0780366859

In this paper, the theory and design of a class of perfect-reconstruction (PR) biorthogonal nonuniform cosine-modulated filter banks (CMFB) is proposed, It is based on a recombination or merging structure previously proposed by the authors [2000]. By relaxing the original CMFB and the recombination transmultiplexer (TMUX) to be biorthogonal, nonuniform CMFBs with lower system delay can be obtained. This increases the possible choices of the prototype filters and greatly reduces the overall system delay of the nonuniform filter bank. A new method is also introduced to suppress the spurious response resulting from the mismatch in the transition bands of the two biorthogonal CMFBs. A design example shows that nonuniform CMFBs with good stopband attenuation, lower system delay and implementation complexity can be realized using the proposed method.

关键词： Channel bank filters Delay systems Filter bank Merging Prototypes Attenuation Design engineering Spontaneous emission audio coding Nonlinear filters

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Frequency selectivity via the SpEnt methodology for wideband speech compression

Frequency selectivity via the SpEnt methodology for wideband...

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

作者： M.G. Kokes J.D. Gibson Multimedia Communications and Networking Laboratory Department of Electrical engineering Southern Methodist University Dallas TX USA

In speech and audio coding, frequency selectivity of the basis functions is an important property of the codec. The more precise the frequency selectivity, the less chance there is for audible coding effects due to uncanceled aliasing. We use Campbell's (1960) coefficient rate and the spectral entropy (SpEnt) of the source random process as a guide to formulate adaptive nonuniform modulated lapped biorthogonal transforms (NMLBT). The use of the NMLBT allows for efficient implementation of a time-varying transform which possesses both good frequency and time resolution at all instances, without the need for transitional filters. By coupling the SpEnt methodology with modulated lapped biorthogonal transforms (MLBT), we develop band combining strategies to produce an adaptive NMLBT. Due to the nature of the SpEnt methodology, the new frequency selection process comprises a non-linear approximation method to determine the best n basis functions to represent the current speech frame. We implement a wideband speech compression scheme based on this strategy and verify its improved performance in coding speech and audio signals at 16 and 24 kbps.

关键词： Frequency Wideband Speech coding Speech codecs audio coding Entropy Random processes Filters Approximation methods Speech processing

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Perceptual coding of digital audio

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PROCEEDINGS OF THE IEEE 2000年第4期88卷 451-513页

作者： Painter, T Spanias, A Arizona State Univ Ctr Telecommun Res Dept Elect Engn Tempe AZ 85287 USA

During the last decade, CD-quality digital audio has essentially replaced analog audio. Emerging digital audio applications for network, wireless, and multimedia computing systems face a series of constraints such as reduced channel bandwidth, limited storage capacity, and low rest. These new applications have created a demand for high-quality digital audio delivery at low bit rates. In response to this need, consider-able research has been devoted to the development of algorithms for perceptually transparent coding of high,fidelity (CD-quality) digital audio. As a result, many algorithms have been proposed, and several have non become international and/or commercial product standards. This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. This paper is organized as follows. First, psychoacoustic principles are described, with the MPEG psychoacoustic signal analysis model I discussed in some detail. Next, filter bank design issues and algorithms are addressed, with a particular emphasis placed on the modified discrete cosine transform, a perfect reconstruction cosine-modulated filter bank that has become of central importance in perceptual audio coding. Then, we review methodologies that achieve perceptually transparent coding of FM- and CD-quality audio signals, including algorithms that manipulate transform components, subband signal decompositions, sinusoidal signal components, and linens prediction parameters, as well as hybrid algorithms that make use of more than one signal model. These discussions concentrate on architectures and applications of those techniques that utilize psychoacoustic models to exploit efficiently masking characteristics of the human receiver. Several algorithms that have become international and/or commercial standards receive in-depth treatment, including the ISO/IEC MPEG family (-1, -2, -4), the Lucent Technologies PAC/EPAC/MPAC, the Dolby(

关键词： AC-2 AC-3 advanced audio coding (AAC) MPEG ATRAC audio coding audio coding standards audio signal processing data compression digital audio radio (DAR) digital broadcast audio (DBA) filter banks high-definition TV (HDTV) linear predictive coding lossy compression modified discrete cosine transform (MDCT) MP3 MPEG MPEG-1 MPEG-2 MPEG-4 MPEG audio multimedia signal processing perceptual audio coding (PAC) perceptual coding perceptual model pseudoquadrature mirror filter (PQMF) psychoacoustic model psychoacoustics SDDS signal compression signal-processing applications sinusoidal coding subband coding transform coding

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Prolog to -: Perceptual coding of digital audio -: An introduction to the paper by Painter and Spanias

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PROCEEDINGS OF THE IEEE 2000年第4期88卷 449-450页

作者： Falk, H

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audio coding using transform-domain weighted interleave vector quantization (Twin VQ)

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ELECTRONICS AND COMMUNICATIONS IN JAPAN PART III-FUNDAMENTAL ELECTRONIC SCIENCE 1998年第3期81卷 1-9页

作者： Iwakami, N Moriya, T Miki, S Ikeda, K Jin, A NTT Corp Human Interface Labs Musashino Tokyo 180 Japan

Twin VQ (transform-domain weighted interleave vector quantization) is a method that encodes the wideband acoustic signal with a low bit rate. It is transform coding with a basic structure that transforms the input signal to the frequency domain by MDCT;vector quantization is applied after flattening. This encoding method has characteristic features such as weighted interleave vector quantization, normalization of the frequency characteristics by the linearly predicted spectrum, and interframe prediction in the frequency domain. Especially, high performance is realized for lower bit rates. Another feature is robustness against the error, since adaptive bit assignment is not applied. (C) 1998 Scripta Technica.

关键词： audio coding transform coding vector quantization linear prediction

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audio coding using variable-depth multistage quantization

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 1998年第2期6卷 186-189页

作者： Kossentini, F Macon, M Smith, MJT Univ British Columbia Dept Elect & Comp Engn Vancouver BC V6T 1Z4 Canada

An algorithm for high-quality coding of 48 kHz sampled audio signals is presented. The algorithm employs a perceptual transform and a variable-depth multistage quantizer. The resulting audio reproduction quality is better than that of the Motion Pictures Expert Group (MPEG) layer I coder and roughly equivalent to that of the MPEG layer II coder.

关键词： audio coding Quantization Frequency Psychoacoustic models Masking threshold Motion pictures Transform coding Auditory system Signal analysis Smoothing methods

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