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Scalable audio coding for compression and loss resilient streaming

IEE PROCEEDINGS-VISION IMAGE AND SIGNAL PROCESSING 2006年第3期153卷 331-339页

作者： Sandler, M. Black, D. Queen Mary Univ London Dept Elect Engn Ctr Digital Mus London E1 4NS England

Current popular Internet audio streaming solutions impose a division between source coding (provided, for example, by MPEG layer III) and channel coding, typically accomplished in the server by means of packet re-transmission. A novel joint source and channel coder that provides packet-loss recovery and continuous bitrate scalability is presented. These functionalities are well suited to streaming audio over third and future generation wireless broadband networks.

关键词： packet-loss recovery loss resilient streaming scalable audio coding continuous bitrate scalability compression future generation wireless broadband networks third generation wireless broadband networks media streaming Internet streaming audio coding Speech and audio coding Multimedia communications combined source-channel coding joint source channel coder

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A Dynamic Fine-Grain Scalable Compression Scheme With Application to Progressive audio coding

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2011年第1期19卷 14-23页

作者： Strahl, Stefan Hansen, Heiko Mertins, Alfred Carl von Ossietzky Univ Oldenburg D-26111 Oldenburg Germany Med Univ Lubeck Inst Signal Proc D-23538 Lubeck Germany

This paper studies the fine-grain scalable compression problem with emphasis on 1-D signals such as audio signals. Like in the successful 2-D still image compression techniques embedded zerotree wavelet coder (EZW) and set partitioning in hierarchical trees (SPIHT), the desired fine-granular scalability and high coding efficiency are benefited from a tree-based significance mapping technique. A significance tree serves to quickly locate and efficiently encode the important coefficients in the transform domain. The aim of this paper is to find such suitable significance trees for compressing dynamically variant 1-D signals. The proposed solution is a novel dynamic significance tree (DST) where, unlike in existing solutions with a single type of tree, a significance tree is chosen dynamically out of a set of trees by taking into account the actual coefficients distribution. We show how a set of possible DSTs can be derived that is optimized for a given (training) dataset. The method outperforms the existing scheme for lossy audio compression based on a single-type tree (SPIHT) and the scalable audio coding schemes MPEG-4 BSAC and MPEG-4 SLS. For bitrates less than 32 kbps, it results in an improved perceived audio quality compared to the fixed-bitrate MPEG-2/4 AAC audio coding scheme while providing progressive transmission and finer scalability.

关键词： audio coding embedded coding progressive compression significance tree

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Single-Channel and Multi-Channel Sinusoidal audio coding Using Compressed Sensing

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2011年第5期19卷 1382-1395页

作者： Griffin, Anthony Hirvonen, Toni Tzagkarakis, Christos Mouchtaris, Athanasios Tsakalides, Panagiotis Fdn Res & Technol Hellas FORTH ICS Inst Comp Sci GR-70013 Iraklion Crete Greece Univ Crete Dept Comp Sci GR-70013 Iraklion Crete Greece

Compressed sensing (CS) samples signals at a much lower rate than the Nyquist rate if they are sparse in some basis. In this paper, the CS methodology is applied to sinusoidally modeled audio signals. As this model is sparse by definition in the frequency domain (being equal to the sum of a small number of sinusoids), we investigate whether CS can be used to encode audio signals at low bitrates. In contrast to encoding the sinusoidal parameters (amplitude, frequency, phase) as current state-of-the-art methods do, we propose encoding few randomly selected samples of the time-domain description of the sinusoidal component (per signal segment). The potential of applying compressed sensing both to single-channel and multi-channel audio coding is examined. The listening test results are encouraging, indicating that the proposed approach can achieve comparable performance to that of state-of-the-art methods. Given that CS can lead to novel coding systems where the sampling and compression operations are combined into one low-complexity step, the proposed methodology can be considered as an important step towards applying the CS framework to audio coding applications.

关键词： audio coding compressed sensing (CS) signal reconstruction signal sampling sinusoidal model

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audio coding based on rate-distortion and perceptual optimization

Audio coding based on rate-distortion and perceptual optimiz...

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Conference on Wavelet Applications VII

作者： Erne, M Moschytz, G Swiss Fed Inst Technol Inst Signal & Informat Proc CH-8092 Zurich Switzerland

ISBN: (纸本)0819436828

The time-frequency tiling, bit allocation and the quantizer of most perceptual coding algorithms is either fixed or controlled by a perceptual model. The large variety of existing audio signals, each exhibiting different coding requirements due to their different temporal and spectral fine-structure suggests to use a signal-adaptive algorithm. The framework which is described in this paper makes use of a signal-adaptive wavelet filterbank which allows to switch any node of the wavelet-packet tree individually. Therefore each subband can have an individual time-segmentation and the overall time-frequency tiling can be adapted to the signal using optimization techniques. A rate-distortion optimality can be defined which will minimize the distortion for a given rate in every subband, based on a perceptual model. Due to the additivity of the rate and distortion measure over disjoint covers of the input signal, an overall cost function including the switching cost for the filterbank switching can be defined. By the use of dynamic programming techniques, the wavelet-packet tree can be pruned based on a top-down or bottom-up "split-merge" decision in every node of the wavelet-tree. Additionally we can profit from temporal masking due to the fact that each subband can have an individual segmentation in time without introducing time domain artifacts such as pre-echo distortion.

关键词： MPEG audio coding perceptual model wavelet signal-adaptivity

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audio coding WITH POWER SPECTRAL DENSITY PRESERVING QUANTIZATION

AUDIO CODING WITH POWER SPECTRAL DENSITY PRESERVING QUANTIZA...

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Li, Minyue Klejsa, Janusz Ozerov, Alexey Kleijn, W. Bastiaan KTH Royal Inst Technol Sch Elect Engn Stockholm Sweden Technicolor F-35576 Cesson Sevigne France Victoria Univ Wellington Sch Engn & Comp Sci Victoria New Zealand

ISBN: (纸本)9781467300469

The coding of audio-visual signals is generally based on different paradigms for high and low rates. At high rates the signal is approximated directly and at low rates only signal features are transmitted. The recently introduced distribution preserving quantization (DPQ) paradigm provides a seamless transition between these two regimes. In this paper we present a simplified scheme that preserves the power spectral density (PSD) rather than the probability distribution. In a practical system the PSD must be estimated. We show that both forward adaptive and backward adaptive PSD estimation are possible. Our experimental results confirm that preservation of PSD at finite precision leads to a unified coding paradigm that provides effective coding at both high and low rates. An audio coding application shows the perceptual benefits of PSD preserving quantization.

关键词： Quantization audio coding predictive coding forward adaptation backward adaptation

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MDCTNet: A Hybrid Approach to Neural audio coding

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IEEE JOURNAL OF SELECTED TOPICS IN SIGNAL PROCESSING 2024年第8期18卷 1506-1516页

作者： Villemoes, Lars Vinton, Mark Ekstrand, Per Lu, Lie Davidson, Grant Zhou, Cong Dolby Sweden AB Adv Technol Grp S-11330 Stockholm SE Sweden Dolby Labs Inc Adv Technol Grp San Francisco CA 94103 USA Anuttacon Santa Clara CA 95054 USA

We describe and evaluate a hybrid neural audio coding system consisting of a perceptual audio encoder and a generative model, MDCTNet. By applying recurrent layers (RNNs) we capture correlations in both time and frequency directions in a perceptually weighted adaptive modified discrete cosine transform (MDCT) domain. By training MDCTNet on a diverse set of full-range monophonic audio signals at 48 kHz sampling, we achieve performance competitive with state-of-the-art audio coding at 24 kb/s variable bitrate (VBR). We also quantify the effect of the generative model-based decoding at lower and higher bitrates and discuss some caveats of the use of data driven signal reconstruction for the audio coding task.

关键词： Decoding Psychoacoustic models Transforms Codecs Bit rate audio coding Entropy Distortion Training Time-frequency analysis Perceptual audio coding deep learning generative models neural networks

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WIDE-BAND SPEECH AND audio coding

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IEEE COMMUNICATIONS MAGAZINE 1993年第11期31卷 34-44页

作者： NOLL, P Technical University Berlin Germany

Typical parameters of wideband speech and audio signals, including digitized versions of each, potential applications, and available transmission media, are described. Facts about human auditory perception that are exploited in audio coding and quality measures that play an important role in coder evaluations and designs are reviewed. Techniques for efficient coding of wideband speech and audio signals, with an emphasis on existing standards, are discussed. The audio coding standard developed by the Moving Pictures Expert Group within the International Organization for standardization (ISO/MPEG) is covered in some detail, since it will be used in many application areas, including digital storage, transmission, and broadcasting of audio-only signals and audiovisual applications such as videotelephony, videoconferencing, and TV broadcasting. Ongoing research and standardization work is outlined.< >

关键词： Wideband Speech coding audio coding ISO standards TV broadcasting Humans Standards development Standards organizations MPEG standards Digital video broadcasting

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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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A warped linear-prediction-based subband audio coding algorithm

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 2002年第1期10卷 1-8页

作者： Yu, RS Ko, CC Natl Univ Singapore Dept Elect & Comp Engn Singapore 117576 Singapore

In this paper, a novel audio coding algorithm is proposed where the warped linear prediction (WLP) technique is employed to construct a perceptual pre- and post-filter for subband audio coding. A modified signal-to-mask ratio (SMR) calculation is given for subband coding of the WLP residuals of audio Signals. The concept of perceptual entropy (PE) is extended to sub-band coding, resulting in the subband perceptual entropy (SPE), which gives a short time estimate of the lowest possible bit rate for transparent subband audio coding. Two WLP models with frequency responses approximating the spectral shape of the masking threshold are investigated and it is found that the residual signals of both models contain less SPE compared with that of the original audio signals. Subjective tests show that the proposed audio codec operating at 56 kbps has a perceptual quality comparable to MPEG-1 audio Layer II operating at 64 kbps.

关键词： audio coding perceptual entropy subband coding warped linear prediction

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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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