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检索条件"主题词=Audio coding"

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Noiseless coding of quantized spectral components in MPEG-2 Advanced audio coding

Noiseless coding of quantized spectral components in MPEG-2 ...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： S.R. Quackenbush J.D. Johnston AT and T Laboratories Murray Hill NJ USA

Advanced audio coding (AAC), part of ISO/MPEG-2, issued as an international standard in April, 1997. It supports single or multiple channel audio programs and delivers excellent audio quality at or below 64 kbps/channel by exploiting the compression capabilities of a high-resolution filterbank, backward-adaptive prediction, joint channel coding, nonlinear quantizers and noiseless (Huffman) coding. This paper describes the flexible Huffman coding algorithm used in AAC and discusses the compression provided by this component of the standard.

关键词： audio coding Transform coding Frequency Filter bank Acoustic noise Signal resolution Sampling methods Quantization Noise shaping Decoding

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Perceptual suppression of quantization noise in low bitrate audio coding

Perceptual suppression of quantization noise in low bitrate ...

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Asilomar Conference on Signals, Systems & Computers

作者： Y.H. Lam R.W. Stewart Signal Processing Division Department of Electrical & Electronic Engineering University of Strathclyde Glasgow UK

In this paper a generalized perceptual filter which aims to reduce the audible quantization noise in low bitrate audio coding is presented. The perceptual filter is based on a human auditory perception model whereby its derivation takes the highly nonlinear nature of the human ears into account and thus attempts to model the psychoacoustic behaviour of the ear. The results show the perceptual filter can reduce the audible quantization noise in a subband coded audio signal under a subjective analysis. The proposed filter has straightforward implementation, thus making it suitable for incorporation into future coder designs.

关键词： Quantization Bit rate Filters Acoustic noise Noise reduction Humans Psychoacoustic models Ear audio coding Psychology

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An improved stereophonic coding scheme compatible to the ISO/MPEG audio coding algorithm

An improved stereophonic coding scheme compatible to the ISO...

引用

International Conference on Communication Systems

作者： Do-Hui Teh Ah-Peng Tan A/V Information Research Centre Asia Matsushita Electric (S) Private Limited Singapore

ISBN: (纸本)0780308034

Stereophonic coding has been applied to the ISO/MPEG audio coding algorithm to improve the quality of stereo signals at very low bit rates. While reasonable quality has been achieved at bit rates of 2*64 kbit/s or lower, it was found that the proposed scheme could not achieve consistent good quality sounds for all types of audio signals. An improved stereophonic coding scheme is presented. This improved scheme employs the use of error minimisation and power equalization to obtain the improvements. Simulations of the scheme at a bit rate of 128 kbit per stereophonic channel produced a significant improvement in the quality of the audio over the original scheme. A quality that is close to transparency can be obtained at a bit rate of 128 kbit/s.< >

关键词： audio coding Bit rate Frequency Testing HDTV ISO standards IEC standards Masking threshold Ear Asia

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A fast noise-scaling algorithm for uniform quantization in audio coding schemes

A fast noise-scaling algorithm for uniform quantization in a...

引用

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

作者： C.A. Serantes A.S. Pena N.G. Prelcic E.T.S.E. Telecomunicación Universidad de Vigo Vigo Spain

A new bit assignment algorithm is presented. Its goals are the simultaneous assignment on all subbands in a few steps of an iterative calculus, the use of memory to achieve a better speed of convergence and the consideration of a deformable error curve. The basis of the algorithm is discussed and also other considerations that are likely to arise in practice. Finally, an example of its performance is given.

关键词： audio coding Bit rate Equations Solids Convergence Piecewise linear techniques

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Low bitrate audio coding using generalized adaptive gain shape vector quantization across channels

Low bitrate audio coding using generalized adaptive gain sha...

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

作者： Sanjeev Mehrotra Wei-ge Chen Kishore Kotteri Microsoft Research Microsoft Corporation Redmond WA USA

ISBN: (纸本)9781424423538

audio coding at low bitrates suffers from artifacts due to spectrum truncation. Typical audio codecs code multi-channel sources using transforms across the channels to remove redundancy such as middle (mid) - side (M/S) coding. At low bitrates, the spectrum of the coded channels is truncated and the spectrum of the channels with lower energy, such as the side channel, is truncated severely, sometimes entirely. This results in a muffled sound due to truncation of all coded channels beyond a certain frequency. It also results in a loss of spatial image even at low frequencies due to severe truncation of the side channel. Previously we have developed a low bitrate coding method to combat the loss of higher frequencies caused by spectrum truncation. In this paper, we present a novel low bitrate audio coding scheme to mitigate the loss of spatial image. Listening tests show that the combination of the two low bitrate coding methods results in a audio codec that can get good quality even at bitrates as low as 32 kbps for stereo content with low decoder complexity.

关键词： Bit rate audio coding Shape Vector quantization Codecs Discrete transforms Frequency Decoding Redundancy Testing

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

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

引用

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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a Family of Random Waveform Models for audio coding

a Family of Random Waveform Models for Audio Coding

引用

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

作者： M. Kowalski B. Torresani LATP CMI Marseilles France

We study the behavior of hybrid random waveform models for audio signals, involving sparse random series of waveforms, with random coefficients. Similar approaches have been considered in the recent years. However, these do generally not rely on explicit models, and are of more "algorithmical" nature. The models we propose allow us to analyze mathematical properties of such signals and corresponding estimators, and derive estimation algorithms, which do not rely on complex optimization techniques

关键词： audio coding Yield estimation Dictionaries Parameter estimation Mathematical model Signal analysis Algorithm design and analysis Random variables Humans Contracts

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Comparison of filterbanks for high quality audio coding

Comparison of filterbanks for high quality audio coding

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： K. Brandenburg E. Eberlein J. Herre B. Edler Lehrstuhl für Technische Elektronik Universität Erlangen Nürnberg Erlangen Germany Institut für integrierte Schooltumgen Fraunhofer Gesellschaft e. V. Erlangen Germany Institut für Theoretische Nachrichtentechuik und Informationsverarbeitung Universität Hannover Hanover Germany

Perceptual coding techniques have recently been applied successfully to high-quality coding of digital audio signals. The basic perceptual coding system uses an analysis/synthesis system to map the time-domain data into a number of frequency-domain channels. A perceptual model is used to estimate the amount of noise shaping needed in order to avoid any audible noise due to the quantization of the frequency domain data. Different filterbanks have been applied to perceptual coding. A comparison of different filterbanks shows that there is no performance penalty for hybrid filterbanks compared to other solutions.< >

关键词： audio coding Filter bank Channel bank filters Digital filters Frequency domain analysis Noise figure Masking threshold Quantization Narrowband Computational complexity

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Algebraic quantization of transform coefficients for embedded audio coding

Algebraic quantization of transform coefficients for embedde...

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

作者： M. De Meuleneire H. Taddei D. Pastor Nokia Siemens Networks COO RTP PT HWCT Computing Techn SDE Munich Germany Department SC ENST Bretagne Technopole Brest Iroise Brest France

This paper proposes a new quantization for transform coefficients based on algebraic quantization. The coefficients are represented by a few pulses multiplied by a unique amplitude. The coefficients to be transmitted are selected by optimizing an error criterion, that determines the signs, positions and amplitudes of the pulses. This simple quantization has been implemented in a wavelet-based wideband scalable coder, and has been proved to provide a perceptually better quality than SPIHT on speech signal and music.

关键词： audio coding Bit rate Transform coding Decoding Bandwidth Time frequency analysis Wavelet transforms Huffman coding Vector quantization Image reconstruction

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A Comparative Study of Lossless audio coding Schemes

A Comparative Study of Lossless Audio Coding Schemes

引用

International Conference on Computer Engineering and Systems

作者： Hend Ali Elsayed Masoud Alghoniemy Mohammed El-Banna Electrical Engineering Department Alexandria University Alexandria Egypt

ISBN: (纸本)1424402719;1424402727

In this paper, we present a comparative study of different lossless audio coding schemes, which are implemented using different integer transforms. The audio signal under consideration, which is assumed to be integer-valued, as in the case of fixed-point implementation, is first decorrelated using the appropriate integer transform. The resulting integer coefficients are then entropy-coded to produce the output stream. Several integer transforms have been considered, such as the integer wavelet transform (IntWT) with different decomposition levels as well as different filters, the integer discrete cosine transform (IntDCT), and the integer Walsh Hadamard transform (IntWHT). Arithmetic and Huffman coding have been used for entropy coding. The performed simulation provides insight on the performance of the different integer transforms in the lossless audio coding context

关键词： audio coding Discrete wavelet transforms Discrete transforms Decorrelation Streaming media Filters Discrete cosine transforms Arithmetic Huffman coding Entropy coding

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