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

作者： R. Geiger Y. Yokotani G. Schuller Fraunhofer IIS Erlangen Germany Fraunhofer IDMT Ilmenau University of Technology Ilmenau Germany

This paper describes high data-rate audio data hiding using the IntMDCT. The IntMDCT is an integer approximation of the MDCT with perfect reconstruction. Based on this transform, we describe a straight-forward way to embed data and extract it in a bit-exact manner while perceptual transparency is maintained. Since the IntMDCT spectrum can be used to obtain a closer approximation of the masking threshold compared to time domain approaches, it is possible to achieve higher data rates. In a simple experimental implementation, we found we could embed data at rates up to about 140 kb/s without introducing audible distortions

关键词： Data encapsulation Masking threshold MPEG 4 Standard Data mining audio coding Quantization Image reconstruction Discrete cosine transforms Humans Watermarking

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MDCT IP core generator

MDCT IP core generator

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International Conference on Design and Technology of Integrated Systems in Nanoscale Era (DTIS)

作者： M. Balaz P. Malik T. Pikula M. Simlastik Institute of Informatics Slovak Academy of Sciences Bratislava Slovakia

Digital compression of audio signals has become increasingly more important with the advent of fast and inexpensive microprocessors and digital signal processors. Several compression schemes were developed and well established. Most of them adopt MDCT/IMDCT. This paper presents a new MDCT IP core generator. The software tool generates several MDCT architectures with adjustable parameters for FPGA-based design

关键词： audio coding audio compression Software tools Computer architecture Digital audio players Transform coding Discrete cosine transforms Discrete transforms Filter bank Application software

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AVS Digital audio Processing Technology

AVS Digital Audio Processing Technology

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International Conference on Innovative Computing, Information and Control (ICICIC)

作者： Cong Zhang RuiMin Hu HaoJun Ai Key Laboratory of Multimedia & Network Communications Engineering Wuhan University of China Wuhan China Wuhan University Wuhan Hubei CN

During the last decade, new digital audio applications have emerged for network multimedia, wireless multimedia communication, and multimedia computing. In response of this need, considerable research has been devoted to the development of high-fidelity digital audio. Several international or commercial digital audio standards are also presented, such as the ISO/MPEG family and the Dolby AC-3. Although these standards are publicly available, it does not mean they are free. Therefore, China own audio and video compression standard (AVS) is presented. In this paper, AVS audio coding and decoding technologies are discussed in some detail and concludes with a brief discussion of future research directions

关键词： Decoding audio coding Multimedia computing Switches Multimedia communication Psychoacoustic models Phase change materials ISO standards Video compression Standards development

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Lossless scalable audio coding and quality enhancement

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IEICE TRANSACTIONS ON INFORMATION AND SYSTEMS 2003年第3期E86D卷 425-429页

作者： Moriya, T Jin, A Mori, T Ikeda, K Kaneko, T NTT Corp NTT Cyber Space Labs Musashino Tokyo 1808585 Japan

This paper proposes a lossless scalable audio coding scheme and quality enhancement processing at the decoder to compensate for some missing scalable units of information. The bit rate scalability is achieved by combining high-compression coding, such as MPEG-4, and horizontal bit slicing of the PCN4-coded error signal between the original waveform and the locally, reconstructed MPEG-4 signal. The horizontally sliced stream may be transported through an IP network with priority. Even if some units are missing at the decoder, reasonable quality waveform can be reconstructed by means of preserving the important packets. In addition, quality enhancement procedures including scale adjustment and post-processing have been proposed. The scale adjustment eliminates unnecessary zero's, and the post-processing recovers the spectral envelope characteristics of the original input signal. As a result of objective quality evaluation, the two techniques are confirmed to be useful for quality enhancement when lower priority packets are lost. This scheme enables graceful degradation by supporting lossless, near lossless, and high-compression coding within a single scalable framework, and is useful for narrowband to broadband audio streaming.

关键词： lossless audio coding scalable MPEG-4

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Warped ARMA filters in high quality audio coding

Warped ARMA filters in high quality audio coding

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International Symposium on Signal Processing and Its Applications (ISSPA)

作者： M. Deriche D. Ning Department of Electrical Engineering King Fahd University of Petroleum and Minerals Dhahran Saudi Arabia Queensland University of Technology Brisbane Australia

In this paper, a novel audio coder using the discrete wavelet transform (DWT) and a warped linear prediction (WLP) model, is proposed. In contrast to conventional LP, WLP allows for the control of frequency resolution to closely match the response of the human auditory system. The residual from the inverse WLP filtering is analyzed by a wavelet filterbank designed to approximate the critical bands. For monophonic signals sampled at 44.1 kHz, the coder achieves near transparent quality at an average bit-rate of 64 kb/s.

关键词： Filters audio coding Discrete wavelet transforms Predictive models Control systems Frequency Signal resolution Humans Auditory system Filtering

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A new sinusoidal modelling approach for parametric speech and audio coding

A new sinusoidal modelling approach for parametric speech an...

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International Symposium on Image and Signal Processing and Analysis

作者： P. Vera-Candeas N. Ruiz-Reyes J. Curpian-Alonso M. Rosa-Zurera Electronic Department University of Jaén Polytechnical School Alfonso X el Sabio 28 Linares Jaén Spain Signal Theory and Communications Department University of Alcalá Polytechnical School Alcalá de Henares Madrid Spain

In this paper we propose a new approach based on energy-adaptive matching pursuits to improve sinusoidal modelling for parametric speech and audio coding. To reduce the complexity of the algorithm, an over-complete dictionary composed of complex exponentials is used. An analysis-synthesis windows scheme that avoids overlapping is proposed. For efficient quantization of sinusoidal model parameters, a new algorithm that largely reduces the side information required by the decoder is described. Experimental results show evidence of the advantages of the proposed method to be integrated into multi-pans models for parametric speech/audio coding.

关键词： Speech coding audio coding Matching pursuit algorithms Psychoacoustic models Signal processing algorithms Decoding Parameter estimation Pursuit algorithms Signal analysis Dictionaries

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Individualization in spatial-audio coding

Individualization in spatial-audio coding

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： C. Jin N. van Schaik V. Best S. Carlile School of EIE University of Sydney Sydney NSW Australia Department of Physiology University of Sydney Sydney NSW Australia

A novel spatial-audio coding technique that allows individualized 3D audio presentation is described. It exploits the dichotomous roles of the low-frequency interaural timing and level difference cues versus the high-frequency spectral cues in human sound localization. The high-frequency spectral cues are modified to match the acoustics of the listener's outer ears, while preserving the original low-frequency interaural cues. The theory behind the coding technique is described and sound localization data are shown demonstrating the fidelity of the coding technique. Issues relating to the frequency resolution of directional frequency bands are explored.

关键词： Humans Ear Frequency audio coding Auditory system Australia Filters Acoustic measurements Transfer functions Timing

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Soft decision unequal error protection scheme for MPEG advanced audio coding

Soft decision unequal error protection scheme for MPEG advan...

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IEEE International Conference on Multimedia and Expo (ICME)

作者： T.H. Yeo W.C. Wong D.Y. Huang Department of Electrical Engineering National University of Singapore Singapore Institute for Infocomm Research National University of Singapore Singapore

In this paper, a system using soft decision decoding of convolutional codes is investigated for MPEG-4 advanced audio coding. Under severe channel random bit error rates of 4.00x10/sup -2/ and above, the proposed scheme is able to achieve above 90% improvement in residual bit error rate performance over the current scheme. The average percentage overhead incurred using the proposed scheme is about 21%, which is the same as that of the original UEP scheme. An analysis with the theoretical bounds shows that within experimental error, the proposed scheme can perform to near optimum as compared to the current scheme.

关键词： Error correction codes audio coding Codecs Decoding Convolutional codes MPEG 4 Standard Bit error rate Robustness Redundancy Forward error correction

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A comparison of differential schemes for low-rate sinusoidal audio coding

A comparison of differential schemes for low-rate sinusoidal...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： J. Jensen R. Heusdens Department of Mediamatics Delft University of Technnology Delft Netherlands

Sinusoidal coding plays an important part in low rate audio coding. Typically, differential techniques are employed to reduce the bit rate for the representation of sinusoidal parameters. We compare several schemes for time-differential (TD) and frequency-differential (FD) representation of the sinusoidal model parameters. We show, through simulation experiments, that bit rates obtained with FD techniques are competitive with those achieved using TD techniques, provided that a variable length segmentation algorithm is first applied to the input signal. This result is important because FD techniques (in contrast to TD techniques) do not rely on the presence of previous segments for correct decoding, and therefore offer robust performance in a lossy packet channel environment.

关键词： audio coding Encoding Bit rate Decoding Signal processing algorithms Frequency Speech codecs Noise robustness Performance loss audio compression

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Fine grain scalable perceptual and lossless audio coding based on IntMDCT

Fine grain scalable perceptual and lossless audio coding bas...

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

作者： R. Geiger A. Herre G. Schuller T. Sporer Fraunhofer AEMT Ilmenau Germany Fraunhofer IIS A Erlangen Germany

This papers presents an embedded fine grain scalable perceptual and lossless audio coding scheme. The enabling technology for this combined perceptual and lossless audio coding approach is the integer modified discrete cosine transform (IntMDCT), which is an integer approximation of the MDCT based on the lifting scheme. It maintains the perfect reconstruction property and therefore enables efficient lossless coding in the frequency domain. The close approximation of the MDCT also allows us to build a perceptual coding scheme based on the IntMDCT. In this paper a bitsliced arithmetic coding technique is applied to the IntMDCT values. Together with the encoded shape of the masking threshold a perceptually hierarchical bitstream is obtained, containing several stages of perceptual quality and extending to lossless operation when transmitted completely. A concept of encoding subslices is presented in order to obtain a fine adaptation to the masking threshold especially in the range of perceptually transparent quality.

关键词： audio coding Discrete cosine transforms MPEG 4 Standard Scalability Frequency domain analysis Arithmetic Masking threshold Codecs Noise shaping Shape

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