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检索条件"主题词=Audio coding"

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A hybrid parametric-waveform approach to bit stream scalable audio coding

A hybrid parametric-waveform approach to bit stream scalable...

引用

Asilomar Conference on Signals, Systems & Computers

作者： F. Riera-Palou A.C. den Brinker A.J. Gerrits Philips Research Laboratories Eindhoven Netherlands

In this paper we present a wideband (44.1 kHz sampling rate) audio and speech coder that combines two different strategies, namely, parametric and waveform coding. It is shown how this approach can be used to design a layered bit stream scalable coder offering a wide variety of decoding bit rates with little scalability loss. Moreover, the bit rates associated with the different layers are competitive, in terms of quality, to those of standardized coders (MP3, AAC) tuned at a particular bit rate.

关键词： Streaming media audio coding Bit rate Decoding Scalability Digital audio players Speech coding Internet Laboratories Wideband

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Lowdelay combined wideband speech and audio coding with scalable bit rates

Lowdelay combined wideband speech and audio coding with scal...

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IEEE International Conference on Multimedia and Expo (ICME)

作者： M. Farrugia A.M. Kondoz Telecom Modus Limited Surrey UK Multimedia Communications Group CCSR University of Surrey Guildford Surrey UK

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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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A new orthonormal wavelet packet decomposition for audio coding using frequency-varying modulated lapped transforms

A new orthonormal wavelet packet decomposition for audio cod...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： M. Purat P. Noll Institut fuür Fernmeldeteechnik Technische Universität Berlin Berlin Germany

Wavelet packet decompositions based on tree structured 2-channel filter banks with conjugate quadrature filters (CQF) have found many applications in the area of audio coding. Their time-frequency tiling is the dual of the time-varying modulated lapped transforms (MLT). We present a new orthonormal wavelet packet basis, which is constructed by the frequency-varying MLT. These can be viewed as the direct analogy to time-varying transforms. In contrast to the classical decomposition, the new transform shows a good bandpass behaviour without strong spectral sidelobes. Hence, it is particularly useful for the coding of audio signals. For the signals examined the coding gain is higher than that obtained with cascaded CQF filter banks. A fast algorithm for the new wavelet packet transform is possible with a polyphase implementation for the required modulated lapped transforms. This algorithm exceeds a fast CQF based wavelet packet decomposition, especially with the high frequency resolution, which is necessary in audio coding.

关键词： Wavelet packets audio coding Frequency modulation Modulation coding Filter bank TV Wavelet transforms Time frequency analysis Signal analysis Discrete Fourier transforms

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Card-sized portable audio player using high quality audio coding technology TwinVQ

Card-sized portable audio player using high quality audio co...

引用

IEEE International Conference on Consumer Electronics (ICCE)

作者： T. Takahashi T. Morita T. Shimoda T. Kaneko T. Moriya Kobe Steel Company Limited Kobe Japan NTT Cyber Space Laboratories Tokyo Japan

This paper presents an audio player using TwinVQ, an advanced high quality coding technology, which can compress 44.1 kHz sampled audio data at a very low bit rate of 40 kbit/s/ch. The player is business-card-sized of... 详细信息

关键词： audio coding Bit rate Decoding Space technology Digital signal processing Quantization Solid state circuits Flash memory Steel Laboratories

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Comparison of windowing in speech and audio coding

Comparison of windowing in speech and audio coding

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Tom Bäckström International Audio Laboratories Erlangen Erlangen Germany

Speech and audio coding have during the last decade converged to an increasingly unified technology. This contribution discusses one of the remaining fundamental differences between speech and audio paradigms, namely, windowing of the input signal. audio codecs generally use lapped transforms and apply a perceptual model in the transform domain, whereby temporal continuity is achieved by windowing and overlap-add. Speech codecs on the other hand achieve temporal continuity by using linear predictive filtering, whereby windowing is applied in the residual domain. Despite these fundamental differences, we demonstrate that the two windowing approaches, combined with perceptual modeling, perform very similarly both in terms of perceptual quality and theoretical properties.

关键词： Transforms Speech Speech coding Codecs audio coding Delays Quantization (signal)

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Low Delay Filterbanks for Enhanced Low Delay audio coding

Low Delay Filterbanks for Enhanced Low Delay Audio Coding

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Markus Schnell Ralf Geiger Markus Schmidt Markus Multrus Michael Mellar Jurgen Herre Gerald Schuller Fraunhofer IIS Erlangen Germany Fraunhofer IDMT Ilmenau Germany

Low delay perceptual audio coding has recently gained wide acceptance for high quality communication. While common schemes are based on the well-known Modified Discrete Cosine Transform (MDCT) filterbank, this paper describes novel coding algorithms that, for the first time, make use of dedicated low delay filterbanks, thus achieving improved coding efficiency while maintaining or even reducing the low codec delay. The MPEG-4 Enhanced Low Delay AAC (AAC-ELD) coder currently under development within ISO/MPEG combines a traditional perceptual audio coding scheme with spectral band replication (SBR), both running in a delay-optimized fashion by using low delay filterbanks.

关键词： audio coding Codecs Delay systems Filter bank Design methodology MPEG 4 Standard Delay effects Conferences Acoustic signal processing Acoustic applications

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Efficient large-scale multichannel audio coding

Efficient large-scale multichannel audio coding

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EUROMICRO Conference

作者： F.E. Sandnes Dept. Computer Science Faculty of Engineering Oslo University College Oslo Norway

Systems employing multichannel audio are becoming increasingly common. Multichannel audio is used in domestic surround sound systems, multitrack audio recording and new applications are emerging. Digital multichannel audio systems often employ non-standard formats that complicate the exchange of audio data between different systems. Typical multi-track recording environments consist of a large number of channels. Each channel has its own acoustic configuration with respect to equalisation, levels and effects. Another characteristic of multitrack recording is that only a small subset of channels are non-silent at any one time. A family of lossless, compact and general multichannel audio formats are proposed that support a large number of audio channels and the embedding of non-audio information. Non-silent tracks are combined using Reed Solomon codes. The strategy can therefore be implemented efficiently using one of the many existing Reed Solomon encoder/decoder hardware designs, and several configurations are illustrated.

关键词： Large-scale systems audio coding audio recording Streaming media Reed-Solomon codes Hardware Computational modeling Computer simulation Computer science Acoustical engineering

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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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An adaptive tree search algorithm with application to multiresolution based perceptive audio coding

An adaptive tree search algorithm with application to multir...

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IEEE-SP International Symposium on Time-Frequency and Time-Scale Analysis

作者： N.G. Prelcic A.S. Pena Depart ament o de Tecnoloxías das Comunicacións Universidade de Vigo Vigo Spain

Search algorithms for selecting signal decompositions based on the minimization of a cost functional have been proposed in the literature, in addition to several additive and non-additive information costs. We introduce a new cost function and a search algorithm built from a perceptual criterion. Their efficiency is demonstrated with results showing higher quality compressed audio signals than preliminary approaches at similar bit-rates.

关键词： audio coding Cost function Frequency Bandwidth Signal resolution Wavelet packets Distortion Signal analysis Auditory system Minimization methods

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