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Fast algorithm for computing in MPEG the forward and inverse MDCT audio coding

SIGNAL PROCESSING 2006年第5期86卷 1055-1060页

作者： Truong, TK Chen, PD Cheng, TC I Shou Univ Dept Informat Engn Coll Elect & Informat Engn Ta Hsu Hsiang 84008 Kaohsiung Cty Taiwan Univ So Calif Dept Elect Engn Electrophys Los Angeles CA 90089 USA

The modified discrete cosine transform (MDCT) is always employed in transform-coding schemes as the analysis/ synthesis filter bank. In this paper, an efficient algorithm for MDCT and inverse MDCT (IMDCT) computation for MPEG-1 audio layer III and MPEG-2 international audio-coding standards is proposed, using only the type-II DCT. Finally, the proposed algorithm is compared to the similar algorithms in this paper. (C) 2005 Elsevier B.V. All rights reserved.

关键词： MDCT/IMDCT audio coding type-II DCT MPEG

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A novel audio coding scheme using warped linear prediction model and the discrete wavelet transform

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2006年第6期14卷 2039-2048页

作者： Deriche, Mohamed Ning, Daryl King Fahd Univ Petr & Minerals Dept Elect Engn Dhahran 31261 Saudi Arabia Mathworks Sydney NSW Australia

In this paper, we present a novel audio coder using the discrete wavelet transform (DWT) and warped linear prediction (WLP). In contrast to conventional LP, WLP allows for the control of frequency resolution to closely match the response of the human auditory system. The structure of the system is similar to the transform coded excitation techniques used in wideband speech coding, where LP has been replaced with WLP, and the residual is analyzed by a wavelet filterbank designed to approximate the critical bands. The inherent shaping of the WLP synthesis filter, and a controlled bit allocation to the wavelet coefficients helps minimise the perceptually significant noise due to the quantization error in the residual. For monophonic signals sampled at 44.1 kHz, the coder achieves near transparent to transparent quality for a variety of speech and music signals at an average bitrate of about 64 kb/s. Tests also show that the coder (in its initial implementation) delivers superior quality to the MPEG layer III and comparable quality to the MPEG2-AAC codec when operating at the same bitrate.

关键词： audio coding subband coding transform coding warped linear prediction

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ATSC video and audio coding

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PROCEEDINGS OF THE IEEE 2006年第1期94卷 60-76页

作者： Davidson, GA Isnardi, MA Fielder, LD Goldman, MS Todd, CC Dolby Labs San Francisco CA 94103 USA Sarnoff Corp Princeton NJ 08543 USA Tandberg TV Bedford NH 03110 USA

In recent decades, digital video and audio coding technologies have helped revolutionize the ways we create, deliver and consume audiovisual content. This is exemplified by digital television (DTV), which is emerging as a captivating new program and data broadcasting service. This paper provides an overview of the video and audio coding subsystems of the Advanced Television Systems Committee (ATSC) DTV standard. We first review the motivation for data compression in digital broadcasting. The MPEG-2 video and AC-3 audio compression algorithms are described, with emphasis on basic concepts, system features, and coding performance. Next-generation video and audio codecs currently under consideration for advanced services are also presented.

关键词： audio coding bandwidth compression codecs consumer electronics data compression digital communication digital TV HDTV transform coding TV broadcasting video coding video signal processing

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Genesis of the MP3 audio coding standard

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 2006年第3期52卷 1043-1049页

作者： Musmann, Hans Georg Univ Hannover Dept Elect Engn & Comp Sci D-30167 Hannover Germany

In 1988 the International Standardization Organization (ISO) established the Motion Picture Expert Group (MPEG) to develop a digital coding standard for video and audio signals in order to enable interactive video and audio signals on digital storage media. The MPEG audio Group started with members from 14 research institutions in order to develop a digital audio coding standard guided by a chairman. As a result the MPEG-1, Layer I, Layer II and Layer III coding standards have been developed and proposed for coding of stereo audio signals at 2 x 192 kbit/s, 2 x 128 kbit/s and 2 x 64 kbit/s in 1992. Later, the abbreviation "mp3" or "MP3" was introduced in order to substitute the long name of the successful MPEG-1, Layer III coding standard. This paper describes the development of the MP3 coding standard and its essential components as contributed by the members of the MPEG audio Group.

关键词： audio coding MPEG standards MP3

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A trellis-based optimal parameter value selection for audio coding

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2006年第2期14卷 623-633页

作者： Aggarwal, A Regunathan, SL Rose, K Harman Consumer Grp Northridge CA 93129 USA Microsoft Corp Redmond WA 98052 USA Univ Calif Santa Barbara Dept Elect & Comp Engn Santa Barbara CA 93106 USA

This paper considers the problem of selecting a set of parameter values from a given parameter space, in order to perform rate-distortion optimization in the context of audio compression. Due to interdependencies between parameters, separate optimization of parameter values is inherently suboptimal, yet a straightforward brute-force joint search involves prohibitive computational complexity. This work proposes a new method for joint rate-distortion optimization, while accounting for interparameter dependencies. The optimal solution is achieved, at significantly reduced complexity as compared to a brute-force search, by employing a Viterbi search over a trellis. Two objective distortion metrics are specifically considered: the average, and the maximum noise-to-mask ratio. Subjective (AB/MOS) and objective (average/maximum noise-to-mask ratio) tests demonstrate considerable gains at low bit rates of 16 kbps per channel for a 44.1-kHz sampled audio signal using the proposed approach.

关键词： advanced audio coder (AAC) audio coding bit allocation dynamic programming parameter selection side-information trellis Viterbi

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Amplitude modulated sinusoidal signal decomposition for audio coding

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IEEE SIGNAL PROCESSING LETTERS 2006年第7期13卷 389-392页

作者： Christensen, Mads Graesboll Jakobsson, Andreas Andersen, Soren Vang Jensen, Soren Holdt Aalborg Univ Dept Commun Technol DK-9220 Aalborg Denmark Karlstad Univ Dept Elect Engn SE-65188 Karlstad Sweden

In this letter, we present a decomposition for sinusoidal coding of audio, based on an amplitude modulation of sinusoids via a linear combination of arbitrary basis vectors. The proposed method, which incorporates a perceptual distortion measure, is based on a relaxation of a nonlinear least-squares minimization. Rate-distortion curves and listening tests show that, compared to a constant-amplitude sinusoidal coder, the proposed decomposition offers perceptually significant improvements in critical transient signals.

关键词： audio coding parametric coding signal representations sinusoidal

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Cascaded trellis-based rate-distortion control algorithm for MPEG-4 advanced audio coding

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2006年第3期14卷 998-1007页

作者： Yang, CH Hang, HM Natl Chiao Tung Univ Dept Elect Engn Hsinchu 300 Taiwan

In this paper, a few low-complexity and high-performance rate-distortion control algorithms for MPEG-4 Advanced audio coding (AAC) are proposed. One key element in producing good quality compressed audio particularly at medium and low rates is a high performance rate-distortion controller in the audio encoder. Although the trellis-based rate-distortion control algorithms previously proposed can achieve a praiseworthy performance, their computational complexity is extremely high. Therefore, for practical applications, it is very desirable to achieve a similar performance at a much lower complexity. Two types of techniques are proposed in this paper to reduce the computational burden of the trellis-based algorithms. One is splitting a very heavy calculation stage into two sequential steps with much less computation. The other is reducing the candidates in the trellis for parameter search. Together, when applicable, our approach achieves a similar coding performance (audio quality) but requires less than 1/1000 complexity in computation.

关键词： advanced audio coding (AAC) audio coding rate-distortion control trellis-based search

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Efficient bit-rate scalability for weighted squared error optimization in audio coding

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2006年第4期14卷 1313-1327页

作者： Aggarwal, Ashish Regunathan, Shankar L. Rose, Kenneth Harman Consumer Grp Northridge CA 91329 USA Microsoft Corp Redmond WA 98052 USA Univ Calif Santa Barbara Dept Elect & Comp Engn Santa Barbara CA 93106 USA

We propose two quantization techniques for improving the bit-rate scalability of compression systems that optimize a weighted squared error (WSE) distortion metric. We show that quantization of the base-layer reconstruction error using entropy-coded scalar quantizers is suboptimal for the WSE metric. By considering the compandor representation of the quantizer, we demonstrate that asymptotic (high resolution) optimal scalability in the operational rate-distortion sense is achievable by quantizing the reconstruction error in the compandor's companded domain. We then fundamentally extend this work to the low-rate case by the use of enhancement-layer quantization which is conditional on the base-layer information. In the practically important case that the source is well modeled as a Laplacian process, we show that such conditional coding is implementable by only two distinct switchable quantizers. Conditional coding leads to substantial improvement over the companded scalable quantization scheme introduced in the first part, which itself significantly outperforms standard techniques. Simulation results are presented for synthetic memoryless Laplacian sources with P-law companding, and for real-world audio signals in conjunction with MPEG AAC. Using the objective noise-mask ratio (NMR) metric, the proposed approaches were found to result in bit-rate savings of a factor of 2 to 3 when implemented within the scalable MPEG AAC. Moreover, the four-layer scalable coder consisting of 16-kb/s layers achieves performance close to that of the 64-kb/s nonscalable coder on the standard test database of 44.1-kHz audio.

关键词： AAC audio coding bit-rate scalability embedded transmission quantization

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Lossless coding of audio spectral coefficients using selective bitplane coding

Lossless coding of audio spectral coefficients using selecti...

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International Symposium on Communications and Information Technologies (ISCIT)

作者： Eunju Imm Hochong Park Department of Electronics Engineering Kwangwoon University Seoul South Korea

ISBN: (纸本)9781424445219

In this paper, a novel lossless coding method of spectral coefficients for audio codec is proposed. Conventional lossless coder directly codes the spectral coefficients based on their statistical characteristics, but does not provide the high coding efficiency due to its simple structure. To solve this limitation, a new lossless coding scheme consisting of bitplane coding and runlength coding is proposed. In the proposed scheme, the spectral coefficients are first transformed by bitplane coding to a bit stream, and the resulting bit stream is coded by runlength and finally entropy coded. In addition, the coding performance is further increased by applying the proposed bitplane coding selectively to spectral bands. The performance of proposed coding method is measured in terms of the ideal number of bits based on the entropy, which shows that the proposed method has better performance than the conventional lossless coder in AAC audio codec.

关键词： Performance loss Codecs Entropy Loss measurement audio coding Quantization Psychoacoustic models Huffman coding Arithmetic Streaming media

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Quantization of wavelet packet audio coding

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High Technology Letters 2006年第3期12卷 295-299页

作者：谭建国 Zhang Wenjun LiuPeilin Institute of Image Communication and Information Processing Shanghai Jiaotong University Shanghai 200240 P.R. China

Abstract The method of quantization noise control of audio coding in the wavelet domain is proposed. Using the inverse Discrete Fourier Transform （DFT）, it converts the masking threshold coming from MPEG psycho-acoustic model in the frequency domain to the signal in the time domain; the Discrete Wavelet Packet Transform （DWPF） is performed; the energy in each subband is regarded as the maximum allowed quantization noise energy. The experimental result shows that the proposed method can attain the nearly transparent audio quality below 64kbps for the most testing audio signals.

关键词： quantization wavelet packet audio coding DFT

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