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Multiple length adaptive windowing of audio signals

ELECTRONICS LETTERS 1999年第14期35卷 1133-1135页

作者： Vanelli-Coralli, A Corazza, GE Orsini, A Univ Bologna Dept Elect Comp Sci & Syst I-40136 Bologna Italy

audio coders generally operate on a per-frame basis to tackle signal non-stationarity. The choice of frame length is crucial to system performance since it trades off coding efficiency and pre-echo avoidance. A new algorithm based on multiple length adaptive windowing is proposed as an enhanced front-end for audio coders. Overhead reductions in excess of 40% are reported.

关键词： audio signals coding efficiency audio coding frame length multiple length adaptive windowing overhead reductions enhanced front-end pre-echo avoidance audio coders Speech and audio coding system performance adaptive signal processing signal nonstationarity

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audioBIFS: Describing audio Scenes with the MPEG-4 Multimedia Standard

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IEEE TRANSACTIONS ON MULTIMEDIA 1999年第3期1卷 237-250页

作者： Scheirer, Eric D. Vaananen, Riitta Huopaniemi, Jyri MIT Machine Listening Grp Media Lab Cambridge MA 02139 USA Aalto Univ Lab Acoust & Audio Signal Proc Helsinki Finland Nokia Res Ctr Speech & Audio Syst Lab Helsinki Finland

We present an overview of the audioBIFS system, part of the Binary Format for Scene Description (BIFS) tool in the MPEG-4 International Standard. audioBIFS is the tool that integrates the synthetic and natural sound coding functions in MPEG-4. It allows the flexible construction of soundtracks and sound scenes using compressed sound, sound synthesis, streaming audio, interactive and terminal-dependent presentation, three-dimensional (3-D) spatialization, environmental auralization, and dynamic download of custom signal-processing effects algorithms. MPEG-4 sound scenes are based on a model that is a superset of the model in VRML 2.0, and we describe how MPEG-4 is built upon VRML and the new capabilities provided by MPEG-4. We discuss the use of structured audio orchestra language, the MPEG-4 SAOL, for writing downloadable effects, present an example sound scene built with audioBIFS, and describe the current state of implementations of the standard.

关键词： audio coding MPEG-4 SAOL SNHC audio 3-D audio

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Usage of short wavelets for scalable audio coding

Usage of short wavelets for scalable audio coding

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Conference on Wavelet Applications in Signal and Image Processing V

作者： Kudumakis, P Sandler, M Signals Circuits and Systems Research Group Department of Electronic and Electrical Engineering King's College London Strand London WC2R 2LS United Kingdom

ISBN: (纸本)0819425915

In this paper, an in depth investigation and comparison of the performance obtainable with short wavelet filters for low bit rate perceptual audio coding is presented. This a priori knowledge of the short wavelet filters performance evaluation open new horizons in their usage, especially, when combined with the Moving Pictures Expert Group (MPEG-4) requirements for segmental signal to noise ratio (SSNR) scalable audio coding.

关键词： audio coding scalable audio coding wavelets regularity

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audio coding using sinusoidal excitation representation

Audio coding using sinusoidal excitation representation

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

作者： Wen-Whei Chang De-Yu Wang Li-Wei Wang Department of Communication Engineering National Chiao Tung university Hsinchu Taiwan

ISBN: (纸本)0818679190

Most LPC-based audio coders employ simplistic noise-shaping operations to perform psychoacoustic control of quantization noise. In this paper, we report on new approaches to exploiting perceptual masking in the design of adaptive quantization of LPC excitation parameters. Due to its localized spectral sensitivity, sinusoidal excitation representation is preferred to spectrally flat signals for use in excitation modeling. Simulation results indicate that the proposed multisinusoid excited coder can deliver high quality audio reproduction at the rate of 72 kb/s.

关键词： audio coding Linear predictive coding Quantization Psychoacoustic models Psychology Filters Frequency Noise shaping Bit rate Signal processing

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Wavelets versus conventional filters for low bit rate audio coding

Wavelets versus conventional filters for low bit rate audio ...

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1997 International Broadcasting Convention IBS 97

作者： P. Kudumakis M. Sandler King''s College London UK

Wavelet filtering is a promising tool for use in audio signal compression. What is still lacking, however, is a thorough understanding of wavelet filters performance relative to the more sophisticated examples of conventional filters. So, as we seek to apply wavelet filters to low bit rate audio coding, attention must be focused not only to the bit-rate/signal-quality trade-off, but also the complexity and processing delay should not be underemphasised. The results presented in this paper attempt to clarify these issues. To assess the coding gain of wavelet and conventional filters, various codec models have been designed and implemented based on a wavelet packet algorithm, an auditory perception model and entropy noiseless coding. The wavelet packet based coding approach is compared to the MPEG-audio international standard in terms of objective and subjective measurements and is shown to be superior to MPEG-audio layer I and competitive with layer II.

关键词： audio coding

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Multiple-description coding (MDC) of speech with an invertible auditory model

Multiple-description coding (MDC) of speech with an invertib...

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IEEE Workshop on Speech coding Proceedings

作者： G. Kubin W.B. Kleijn University of Technology Vienna Vienna Austria Department of Speech Music and Hearing KTH Royal Institute of Technology Stockholm Sweden

Network signal processing aspects dominate in speech and audio coding applications such as Internet telephony or packet radio networks. We demonstrate that our approach to speech coding in a perceptual domain provides an implicit forward error concealment mechanism to handle random erasures of the channel. To this end, the individual acoustic subchannels of our auditory model are grouped into different transport subchannels or packets. Due to the strongly overlapping, redundant filterbank structure of the model, reconstruction of speech without audible degradation becomes possible even if a significant percentage of channels is erased (e.g., up to 40% in a 50-channel auditory model for narrowband speech). We discuss this result both from a hearing-physiology and a frame-theoretic perspective.

关键词： Speech coding Filter bank audio coding Signal resolution Signal representations Quantization Speech analysis Signal processing Speech processing Internet telephony

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Perception-based bit-allocation algorithms for audio coding

Perception-based bit-allocation algorithms for audio coding

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： S. Voran U.S. Department of Commerce Institute of Telecommunication Sciences Boulder CO USA

We describe six algorithms for bit-allocation in audio coding. Each algorithm stems from the minimization of a different perceptually-motivated objective function. Three of these objective functions are extensions of existing ones, and three are new. Closed-form bit-allocation equations result in five cases, and an iterative approach is required in the sixth.

关键词： audio coding Quantization Signal processing Bit rate Acoustic noise Nuclear magnetic resonance Iterative algorithms Frequency domain analysis Signal resolution Closed-form solution

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Continuously signal-adaptive filterbank for high-quality perceptual audio coding

Continuously signal-adaptive filterbank for high-quality per...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： J. Herre J.D. Johnston Fraunhofer Institute Integrated Circuits FhG US AT and T Research Laboratories USA

Historically, the choice of the optimum filterbank has been the subject of much research and discussion in the development of perceptual audio coders. Desirable properties of a good filterbank include both a good extraction of the signal's redundancy and effective utilization of that redundancy while maintaining control over perceptual demands. Often, there is a conflict between the use of perceptual constraints and the redundancy extraction, in that a filterbank with good resolution in both time and frequency is needed. Recently, a method for performing temporal noise shaping (TNS) of the error signal of a perceptual audio coder has been proposed, providing control over both the time and frequency structure of the coding noise. This paper focuses on the core part of the scheme, forming a continuously adaptive filterbank, and discusses its theoretical background, properties and limitations.

关键词： Filter bank audio coding Redundancy Signal resolution Noise shaping Frequency synthesizers Laboratories Error correction codes Shape control Performance analysis

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Combined speech and audio coding using non-linear adaptations

Combined speech and audio coding using non-linear adaptation...

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IEEE Workshop on Speech coding For Telecommunications

作者： Cheung-Fat Chan Department of Electronic Engineering City University of Hong Kong Kowloon Hong Kong China

ISBN: (纸本)0780340736

A combined speech and audio coder is proposed. The coder structure resembles a low-delay CELP coder, however, the excitation gain is adapted non-linearly in a sample-by-sample fashion by using a trained neural network, and the spectral parameters are derived from backward non-linear prediction based on a second-order Volterra filter. A perceptual weighting filter derived from psychoacoustic analysis in the spectral domain is used to shape the coding noise. The proposed non-linear adaptation schemes significantly improve the effectiveness of using an analysis-by-synthesis model for coding audio signal. Simulation results show that transparent coding of wideband (7 kHz) speech and audio at 24 kbps is achieved.

关键词： Speech coding audio coding Filters Psychoacoustic models Neural networks Psychology Spectral analysis Noise shaping Signal analysis Wideband

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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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