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检索条件"主题词=Audio coding"

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A new approach to low bit rate audio coding using a combined harmonic-multiband-wavelet representation

A new approach to low bit rate audio coding using a combined...

引用

International Symposium on Signal Processing and Its Applications (ISSPA)

作者： M. Deriche D. Ning S. Boland Signal Processing Research Centre School of Electrical and Electronic Systems Engineering Queensland University of Technology Brisbane QLD Australia

In this paper, we present updated results for the combined harmonic-wavelet based audio coder. This coder first approximates the audio frame as a sum of several sinusoids using a harmonic analysis-synthesis scheme. The reconstructed harmonic signal is then subtracted from the original to give a residual, which is analysed using a wavelet filtering scheme. After each step (harmonic analysis and wavelet filtering), the perceptually relevant parameters are quantised and encoded. The total least squares (TLS)-Prony algorithm is used for the harmonic analysis scheme, and cascaded M-band wavelet transforms are used for analysing the residual. At encoding bit rates of 60-70 kbit/s, this audio coder is capable of delivering better results than the MPEG layer II audio coder operating at the much higher bit rate of 128 kbit/s.

关键词： Bit rate audio coding Harmonic analysis Wavelet analysis Power harmonic filters Filtering Least squares approximation Signal analysis Least squares methods Wavelet transforms

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Implementation of the MPEG-4 advanced audio coding encoder on ADSP-21060 SHARC

Implementation of the MPEG-4 advanced audio coding encoder o...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Dong-Yan Huang Xuesong Gong Daqing Zhou T. Miki S. Hotani Institute of Microelectronics Singapore Multimedia Signal Processing Laboratory Multimedia Laboratories NTT Mobile Communications Network Inc. Yokosuka Kanagawa Japan

In this paper, we present the implementation of the MPEG-4 advanced audio coding (MPEG-4 AAC) encoder on a floating-point DSP (ADSP-21060 SHARC) based hardware. We discuss how selective optimization of the encoder verification model (VM) software structure allows robust performance using limited resources, highlight some of the problems inherent in its real-time implementation, and point towards further investigations of real-time implementation and advantages of DSP in audio processing.

关键词： MPEG 4 Standard audio coding Transform coding Digital signal processing Noise shaping Quantization Sampling methods Bit rate Filter bank Kernel

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High quality audio coding using a novel hybrid WLP-subband coding algorithm

High quality audio coding using a novel hybrid WLP-subband c...

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Asia-Pacific Conference on Communications

作者： Yu Rongshan Ko Chi Chung Digital System and Application Lab Department of Electrical Engineering National University of Singapore Singapore

ISBN: (纸本)1864354518

A high quality audio coding scheme based on a novel hybrid algorithm combining warped linear prediction (WLP) and subband coding is proposed. The proposed codec is capable of providing a high quality audio output at a low bit-rate. Subjective tests have shown that the proposed codec is able to provide a performance comparable to that of the MPEG audio layer II while operating at a lower bit-rate.

关键词： audio coding Frequency Masking threshold Quantization Linear predictive coding Filters Low-frequency noise Codecs Humans Auditory system

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Best wavelet-packet bases for audio coding using perceptual and rate-distortion criteria

Best wavelet-packet bases for audio coding using perceptual ...

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

作者： M. Erne G. Moschytz C. Faller Signal and Information Processing Laboratory Swiss Federal Institute of Technology Zurich Switzerland

This paper presents a new approach to the adaptation of a wavelet filterbank based on perceptual and rate-distortion criteria. The system makes use of a wavelet-packet transform where each subband can have an individual time-segmentation. Boundary effects can be avoided by using overlapping blocks of samples and therefore switching bases is possible at every tree-level without affecting other subbands. A modified psychoacoustic model using perceptual entropy can control the switching of the wavelet filterbank and the individual time-segmentation of every subband allows one to take advantage of temporal masking. Additionally a rate-distortion measure can control the filterbank for lossless audio coding applications or in cases where large coding gains can be achieved without using perceptual criteria. The weight of the perceptual measure as well as the weight of the rate-distortion measure can be selected individually, enabling to trade lossless coding versus perceptual coding.

关键词： Wavelet packets audio coding Rate-distortion Filter bank Loss measurement Auditory system Wavelet transforms Humans Time frequency analysis Energy resolution

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MPEG-4 parametric audio coding and its implementation

MPEG-4 parametric audio coding and its implementation

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Annual Conference of Industrial Electronics Society

作者： Heng-Ming Tai Shufei Jiang Dept. of Electr. Eng. Tulsa Univ. OK USA

audio coding at very low bit rates requires very efficient evaluation of characteristics of the signal sources and of perceptual properties of the human ear. MPEG-4 aims to provide an audio-visual coding standard for very low bit rate channel, such as those found in Internet, mobile, and multimedia applications. MPEG-4 audio coding schemes consist of the parametric, CELP and time/frequency coding algorithms. This paper describes the parametric audio coding scheme and its implementation on a DSP.

关键词： MPEG 4 Standard audio coding Bit rate Decoding Speech synthesis Frequency Digital signal processing Speech coding Scalability Internet

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On the utilization of overshoot effects in low-delay audio coding

On the utilization of overshoot effects in low-delay audio c...

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

作者： A. Harma U.K. Laine M. Karjalainen Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Espoo Finland

In low-delay audio coding (coding delay <5 ms) there is no time for detailed spectral modeling in the case of brief percussive sounds, e.g., the castanets, and onsets of music or speech sounds. On the other hand, it is known from psychoacoustic experiments that the ear is not accurate near the onset of a wideband sound. We study the audibility of coding errors near the onsets of musical sounds in a simulated low-delay audio codec based on frequency-warped linear prediction. It is suggested that for many musical transients it is sufficient to reproduce a rough temporal and spectral envelope of the original signal during the first 5-10 ms. Preliminary listening tests support this idea. It is proposed that the overshoot effect of hearing could be utilized efficiently in enhancing the performance of a low-delay audio coding scheme.

关键词： audio coding Psychoacoustic models Delay effects Music Speech coding Psychology Ear Wideband Predictive models Codecs

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Digital sound synthesis by physical modelling

Digital sound synthesis by physical modelling

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International Symposium on Image and Signal Processing and Analysis

作者： R. Rabenstein L. Trautmann Telecommunications Laboratory University Erlangen-Nuünberg Erlangen Germany

ISBN: (纸本)9539676940

After recent advances in coding of natural speech and audio signals, the synthetic creation of musical sounds is also gaining importance. Various methods for waveform synthesis are currently used in digital instruments and software synthesizers. A family of new synthesis methods is based on physical models of vibrating structures (string, drum etc.) rather than on descriptions of the resulting waveforms. This article describes various approaches to digital sound synthesis in general and discusses physical modelling methods in particular. Physical models in the form of partial differential equations are presented. Then it is shown, how to derive discrete-time models which are suitable for real-time DSP implementation. Applications to computer music are given as examples.

关键词： audio coding Application software Instruments Music audio recording Signal synthesis Speech synthesis Synthesizers Microcomputers Disk recording

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Combined spectral envelope normalization and subtraction of sinusoidal components in the ODFT and MDCT frequency domains

Combined spectral envelope normalization and subtraction of ...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： A.J.S. Ferreira INESC Porto School of Engineering University of Porta Portugal

Recent research in high-quality audio coding seeks not only improved coding gains but also new functionalities such as easy semantic access to compressed audio material and audio modification in the compressed domain. These objectives imply the decomposition of the audio signal into several components of specific semantic value, such as sinusoidal components, that take advantage of selective coding and parametrization tools. We presume an MDCT based audio coding environment and present a new technique combining spectral envelope normalization with accurate subtraction of sinusoidal components in the MDCT frequency domain. It is shown how a parametrization of L stationary sinusoids in the complex ODFT spectrum can lead to the effective subtraction, in the real MDCT spectrum, of 3L spectral lines. A demonstration of the implementation of the technique is available on the Internet (see http://***//spl sim/ajf/waspaa01/***.).

关键词： audio coding Signal analysis Frequency domain analysis Signal synthesis Internet Delay Resilience Scalability MPEG 7 Standard Signal generators

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Multiscale overlap-add sinusoidal modeling using matching pursuit and refinements

Multiscale overlap-add sinusoidal modeling using matching pu...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： M.M. Goodwin Creative Advanced Technology Center Scotts Valley CA USA

Sinusoidal models have proven useful for speech and audio analysis, modification, synthesis, and compression. We describe an adaptive multiscale overlap-add sinusoidal model based on the matching pursuit algorithm and various refinements such as conjugate pursuit and orthogonal subspace projection. The model is useful for scalable audio coding and signal modification.

关键词： Matching pursuit algorithms Signal synthesis Dictionaries Time domain analysis Speech synthesis Speech analysis audio coding Interpolation Pursuit algorithms Speech coding

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An MPEG-4 Twin-VQ based high quality audio codec design

An MPEG-4 Twin-VQ based high quality audio codec design

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IEEE Workshop on Signal Processing Systems (SIPS)

作者： Yin-Tsung Hwang Nan-Jung Liu Ming-Chang Tsai Department of Electronic Engineering National Yunlin University of Science and Technology Yunlin Taiwan

This paper presents a high quality audio codec design based on a transform-domain weighted interleave vector quantization (Twin-VQ) scheme adopted in the MPEG-4 audio standard. Three novel techniques are employed in this scheme to compress the data, ie, (1) flattening of MDCT coefficients by the spectrum of linear predictive coding (LPC) coefficients; (2) further flattening of MDCT coefficients by the Bark envelope; and (3) weighted interleave vector quantization. This paper examines the related design issues in implementing an efficient Twin-VQ codec. Fast computation algorithms are derived for the computationally intensive modules. Design parameters of each module are determined and the codebooks for weighted interleave vector quantization are constructed. Experimental results show that the designed codec can compress natural audio efficiently and reproduce high quality outputs.

关键词： MPEG 4 Standard Codecs Linear predictive coding Vector quantization Bit rate audio coding Sampling methods Data engineering Design engineering Code standards

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