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Data Compression Conference (DCC 2002)

作者： Creusere, CD New Mexico State Univ Klipsch Sch Elect & Comp Engn Las Cruces NM 88003 USA

ISBN: (纸本)0769514774

In this paper we study coding artifacts in MPEG-compressed scalable audio. Specifically, we consider the MPEG advanced audio coder (AAC) using bit slice scalable arithmetic coding (BSAC) as implemented in the MPEG 4 reference software. First, we perform human subjective testing using the comparison category rating (CCR) approach, quantitatively comparing the performance of scalable BSAC with the nonscalable TwinVQ and AAC algorithms. This testing indicates that scalable BSAC performs very poorly relative to TwinVQ at the lowest bitrate considered (16 kb/s) largely because of an annoying and seemingly random mid-range tonal signal that is superimposed onto the desired output. In order to better understand and perceptually quantify the various forms of distortion introduced into compressed audio at low bit rates, we apply two analysis techniques: Reng probing and time-frequency decomposition. The Reng probing technique is capable of separating the Linear time-invariant component of a multirate system from its nonlinear and periodically time-varying components. Using this technique, we conclude that aliasing is probably not the cause of the annoying tonal signal;instead, time-frequency analysis indicates that its cause is most likely suboptimal bit allocation.

关键词： Arithmetic audio coding Bit rate Humans MPEG 4 Standard Nonlinear distortion Performance evaluation Testing Time frequency analysis Time varying systems

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Investigating the impact of audio instruction and audio-visual biofeedback for lung cancer radiation therapy

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Medical Physics 2005年第10期32卷 3229-3229页

作者： George, Rohini Department of Biomedical Engineering Virginia Commonwealth University Richmond Virginia 23220 United States

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Study of the audio coding algorithm of the MPEG-4 AAC standard and comparison among implementations of modules of the algorithm

Study of the audio coding algorithm of the MPEG-4 AAC standa...

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作者： Hoffmann, Gustavo André

audio coding is used to compress digital audio signals, thereby reducing the amount of bits needed to transmit or to store an audio signal. This is useful when network bandwidth or storage capacity is very limited. audio compression algorithms are based on an encoding and decoding process. In the encoding step, the uncompressed audio signal is transformed into a coded representation, thereby compressing the audio signal. Thereafter, the coded audio signal eventually needs to be restored (e.g. for playing back) through decoding of the coded audio signal. The decoder receives the bitstream and reconverts it into an uncompressed signal. ISO-MPEG is a standard for high-quality, low bit-rate video and audio coding. The audio part of the standard is composed by algorithms for high-quality low-bit-rate audio coding, i.e. algorithms that reduce the original bit-rate, while guaranteeing high quality of the audio signal. The audio coding algorithms consists of MPEG-1 (with three different layers), MPEG-2, MPEG-2 AAC, and MPEG-4. This work presents a study of the MPEG-4 AAC audio coding algorithm. Besides, it presents the implementation of the AAC algorithm on different platforms, and comparisons among implementations. The implementations are in C language, in Assembly of Intel Pentium, in C-language using DSP processor, and in HDL. Since each implementation has its own application niche, each one is valid as a final solution. Moreover, another purpose of this work is the comparison among these implementations, considering estimated costs, execution time, and advantages and disadvantages of each one. ...

关键词： Processamento : Sinais MPEG-4 AAC audio coding Processamento de sinais acústicos Perceptual coders Psychoacoustics PC MMX DSP VHDL Dissertação

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audio coding and electronic distribution of music

Audio coding and electronic distribution of music

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International Conference on Web Delivering of Music (WEDELMUSIC)

作者： K. Brandenburg Technische Universität Ilmenau and Fraunhofer AEMT Germany

audio coding originally has been developed as a means for making digital radio possible and for distributing audio via phone lines. The most apparent application today is electronic distribution of music via the Internet. While legal services and copy protection methods have been around for seven years, the public attention and the bulk of actual use is still on unauthorized copying of music files. The talk will introduce the basic techniques for high quality coding of music and discuss some possibilities to reduce the unauthorized copying of music.

关键词： audio coding Filter bank Encoding Decoding Signal analysis Psychoacoustic models Masking threshold Quantization ISO IEC standards

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Hidden Markov tree based transient estimation for audio coding

Hidden Markov tree based transient estimation for audio codi...

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IEEE International Conference on Multimedia and Expo (ICME)

作者： S. Molla B. Torresani LAP CMI Marseilles France

A new approach for transients detection and estimation in the context of hybrid audio coding is presented. The basic idea is to use an orthogonal dyadic wavelet expansion, followed by hidden Markov tree modeling of wavelet coefficients. Coefficients may be cast as "transient type" or "residual type", and the estimated transient is reconstructed from the transient type coefficients only. The estimation procedure involves the classical two steps of hidden Markov models: parameters estimation and state estimation. The implementation of those two steps in the case of wavelet coefficient trees is discussed in some detail, and numerical results are given. The application to audio signal encoding is also discussed.

关键词： Hidden Markov models audio coding Wavelet coefficients State estimation Encoding Tree data structures Parameter estimation Context modeling Integrated circuit modeling Random variables

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MDCT analysis of sinusoids: Exact results and applications to coding artifacts reduction

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 2004年第3期12卷 302-312页

作者： Daudet, L Sandler, M Univ Paris 06 Lab Acoust Musicale F-75015 Paris France Queen Mary Univ London Dept Elect Engn London E1 4NS England

Although widely used in many coding applications, the Modified Discrete Cosine Transform (MDCT) has the drawback of being sensitive to time shifts. With the popular choice a sine window, we show that it is possible to compute an explicit formulation of this time dependency. Starting from the exact MDCT of a pure sine and a simple interpretation in terms of combined modulations, we propose a regularization method that computes a "pseudo-spectrum" for the set of MDCT coefficients. This pseudo-spectum is shown to provide, at a low computational, cost, a good approximation of the local spectrum of the signal, with an improved behavior with respect to frequency and phase than the classical MDCT spectrum, ie. the absolute value of the coefficients. Amongst other applications, this procedure can be used to reduce some of the artifacts that appear in MDCT-based audio coders at low bit-rates.

关键词： audio coding lapped transforms sound artifacts time-frequency transforms

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Application layer error-correction coding for rate-distortion optimized streaming to wireless clients

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IEEE TRANSACTIONS ON COMMUNICATIONS 2004年第10期52卷 1675-1687页

作者： Chakareski, J Chou, PA Stanford Univ Dept Elect Engn Informat Syst Lab Stanford CA 94305 USA Microsoft Corp Redmond WA 98052 USA

This paper addresses the problem of streaming packetized media over a lossy packet network to a wireless client, in a rate-distortion optimized way. We introduce an incremental redundancy error-correction scheme that combats the effects of both packet loss and bit errors in an end-to-end fashion, without support from the underlying network or from an intermediate base station. The scheme is employed within an optimization framework that enables the sender to compute which packets it should send, out of all the packets it could send at a given transmission opportunity, in order to meet an average transmission-rate constraint while minimizing the average end-to-end distortion. Experimental results show that our system is robust and maintains quality of service over a wide range of channel conditions. Up to 8 dB performance gains are registered over systems that are not rate-distortion optimized, at bit-error rates as large as 10(-2).

关键词： application layer audio coding channel coding error correction Internet Markov processes multimedia communication optimal control protocols rate distortion video coding

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An analysis of perceptual artifacts in MPEG scalable audio coding

An analysis of perceptual artifacts in MPEG scalable audio c...

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Data Compression Conference (DCC)

作者： C.D. Creusere Klipsch School of Electrical & Computer Engineering New Mexico State University Las Cruces NM USA

ISBN: (纸本)0769514774

We study coding artifacts in MPEG-compressed scalable audio. Specifically, we consider the MPEG advanced audio coder (AAC) using bit slice scalable arithmetic coding (BSAC) as implemented in the MPEG 4 reference software. First, we perform human subjective testing using the comparison category rating (CCR) approach, quantitatively comparing the performance of scalable BSAC with the nonscalable TwinVQ and AAC algorithms. This testing indicates that scalable BSAC performs very poorly relative to TwinVQ at the lowest bitrate considered (16 kb/s), largely because of an annoying and seemingly random mid-range tonal signal that is superimposed onto the desired output. In order to understand better and quantify perceptually the various forms of distortion introduced into compressed audio at low bit rates, we apply two analysis techniques: Reng probing and time-frequency decomposition. The Reng probing technique is capable of separating the linear time-invariant component of a multirate system from its nonlinear and periodically time-varying components. Using this technique, we conclude that aliasing is probably not the cause of the annoying tonal signal; instead, time-frequency analysis indicates that its cause is most likely suboptimal bit allocation.

关键词： audio coding Bit rate Performance evaluation Testing Time frequency analysis Arithmetic MPEG 4 Standard Humans Nonlinear distortion Time varying systems

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Lossless audio coding with MPEG-4 structured audio

Lossless audio coding with MPEG-4 structured audio

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International Conference on Web Delivering of Music (WEDELMUSIC)

作者： N. Vasiloglou R.W. Schafer M.C. Hans Center for Signal and Image Processing Georgia Institute of Technology Atlanta GA USA Mobile & Media Systems Laboratory Hewlett Packard Laboratories Palo Alto CA USA

MPEG-4 structured audio (SA) has been proposed as a flexible standard for generalized audio coding. Originating out of Netsound software developed at MIT SA is based on MIDI-synthesis of sound, but it is enriched with DSP algorithms so as to allow emulation of other types of coders designed for speech and audio signals. We have investigated the use of structured audio for lossless coding of audio signals, and have found that certain limitations of structured audio make implementations of lossless coders less straightforward than might be desired. In particular we have used SA to implement an MPEG-4 compliant version of the lossless audio coder audioPaK. To implement and validate our new coder we used the software system Sfront, which translates MPEG-4 SA files into efficient C programs that render the audio signal.

关键词： audio coding MPEG 4 Standard Digital signal processing Software algorithms Signal processing Image processing Laboratories Emulation Signal design Algorithm design and analysis

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Non-stationary signal parametric modeling techniques with an application to low bit rate audio coding

Non-stationary signal parametric modeling techniques with an...

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International Conference on Signal Processing Proceedings (ICSP)

作者： R. Boyer S. Essid N. Moreau Department of Signal and Image Processing ENST Paris France

Low bit rate audio coding often relies on Fourier representation despite its limitations for transient signal modeling. This study proposes alternative decompositions and expansion strategies that lead to more accurate modeling. Two classes of methods are considered, subspace decomposition methods, and atomic decomposition methods and their performances are compiled to propose an audio modeling scheme amenable to low bit rate coding.

关键词： Parametric statistics Bit rate audio coding Matching pursuit algorithms Signal processing Image coding Transient analysis Signal analysis Fading Acoustic noise

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