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检索条件"主题词=Audio coding"

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Low bitrate audio coding-state-of-the-art, challenges and future directions

Low bitrate audio coding-state-of-the-art, challenges and fu...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： K. Brandenburg Fraunhofer IIS Arbeitsgruppe Elektronische Medientechnologie Ilmenau Technical University Ilmenau Germany

Perceptual encoding of high quality audio has found its way to many applications including digital radio, electronic music distribution (EMD) systems and portable audio devices. An overview on the basics of high quality low bitrate audio coding is followed by a look into currently widely used and newer, state-of-the-art coding systems like MP3 and MPEG-2 Advanced audio coding (AAC). The rapid deployment of older (1992) technologies (like MP3) followed by the news of new and improved algorithms (like AAC) raises the question about future improvements. The paper analyses some candidates for such improvements and provide a view of some current research activities.

关键词： Bit rate audio coding Encoding Decoding Filter bank Digital audio broadcasting Digital video broadcasting Digital audio players audio compression Standardization

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Computationally Efficient Amplitude Modulated Sinusoidal audio coding Using Frequency-Domain Linear Prediction

Computationally Efficient Amplitude Modulated Sinusoidal Aud...

引用

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

作者： M.G. Christensen S.H. Jensen Department of Communication Technology University of Aalborg Denmark

A method for amplitude modulated sinusoidal audio coding is presented that has low complexity and low delay. This is based on a sub-band processing system, where, in each subband, the signal is modeled as an amplitude modulated sum of sinusoids. The envelopes are estimated using frequency-domain linear prediction and the prediction coefficients are quantized. As a proof of concept, we evaluate different configurations in a subjective listening test, and this shows that the proposed method offers significant improvements in sinusoidal coding. Furthermore, the properties of the frequency-domain linear prediction-based envelope estimator are analyzed

关键词： Amplitude modulation Chirp modulation Modulation coding audio coding Delay Frequency domain analysis Frequency estimation Rate-distortion Distortion measurement Predictive coding

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Lossless OLS-NLMS algorithm for audio coding

Lossless OLS-NLMS algorithm for audio coding

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International Conference on Signals and Electronic Systems, ICSES

作者： Grzegorz Ulacha Ryszard Stasiński Department of Computer Science West Pomeranian University of Technology Szczecin Poland Faculty of Electronics and Communications Poznań University of Technology Poznań Poland

In the paper a lossless audio coding backward adaptation algorithm consisting of cascaded OLS and NLMS sections is presented. The technique performance is compared to that of 4 other lossless codecs, including MPEG-4 ALS one, and it is shown that indeed, the new method is the best one. Entropy coder is an advanced adaptive Golomb one supported by arithmetic coders reducing redundancy of its output.

关键词： Transform coding Prediction algorithms Entropy audio coding Image coding Speech

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Subjective evaluation of LSF quantization in conventional and warped LP based audio coding

Subjective evaluation of LSF quantization in conventional an...

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European Signal Processing Conference (EUSIPCO)

作者： Markus Vaalgamaa Aki Härmä Nokia Mobile Phones Acoustic Technology Research Team P.O. Box 407 FIN-00045 Nokia group Finland Helsinki University of Technology Acoustics and Audio Signal Processing Lab. P.O. Box 3000 FIN-02015 HUT Finland

The objective of this study is to evaluate subjective quality of spectral envelope quantization in audio coding. In this paper, the spectral envelope is modeled using Linear Prediction (LP) and Warped Linear Prediction (WLP). The advantage of WLP compared to LP is that the frequency resolution of analysis can be modified so that it follows perceptual frequency scale, like the Bark- or ERB-scale. The coefficients in both cases are transformed into Line Spectral Frequency (LSF) parameters and quantized. The quantization of these parameters is studied and the subjective and objective qualities are compared.

关键词： Quantization (signal) audio coding Distortion measurement Speech coding Speech Vectors

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A Unified Computing Kernel for MDCT/IMDCT in Modern audio coding Standards

A Unified Computing Kernel for MDCT/IMDCT in Modern Audio Co...

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2007 International Symposium on Communications and Information Technologies (ISCIT 2007), vol.2

作者： Tan Li Reed Zhang Robert Yang Heyun Huang Fuhuei Lin Spreadtrum Communications Inc. China

MDCT/IMDCT has been widely adopted to transform an audio sequence from time domain to transform domain by various audio codec standards. This paper presents a unified algorithm and its hardware design for cosine modulated filter bank. In algorithm derivation, both MDCT and IMDCT are first converted to a half size DCT-IV with input or output permutation. The DCT-IV is then realized by a fast DCT-II algorithm Both radix-3 and radix-2 DCT-II algorithms are analyzed and a unified computing kernel is concluded.

关键词： Kernel audio coding Codecs Portable media players Code standards Information technology Discrete cosine transforms MPEG 4 Standard Fast Fourier transforms Hardware

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Near-optimal selection of encoding parameters for audio coding

Near-optimal selection of encoding parameters for audio codi...

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

作者： A. Aggarwal S.L. Regunathan K. Rose Department of Electrical and Computer Engineering University of California Santa Barbara CA USA

We address the issue of optimizing side information rate for efficient audio coding. In coders such as the MPEG-4 AAC, at rates around 16 kbps to 48 kbps, the side information rate forms a substantial part of the total rate. The parameter search procedure in the verification model optimizes each band separately and results in poor performance at low rates. We propose to jointly optimize the encoding parameter of all the bands. The near-optimal solution using a brute force search has drastic computational complexity. However, the same solution is obtained at a much reduced complexity using a Viterbi search through a trellis. The search procedure is developed and evaluated for two objective measures, the average and the maximum noise-mask ratio. For both the measures, the trellis-based search yields substantially better solutions. In particular, trellis-based optimization of maximum noise-mask ratio greatly improves the performance of AAC at low rates. The resulting bit stream is standard-compatible, and the additional complexity due to the proposed optimization is only incurred at the encoder.

关键词： Encoding audio coding Bit rate Signal to noise ratio Information rates MPEG 4 Standard Noise measurement Decoding Virtual manufacturing Viterbi algorithm

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Wide-Band Perceptual audio coding Based on Frequency-Domain Linear Prediction

Wide-Band Perceptual Audio Coding Based on Frequency-Domain ...

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

作者： Petr Motlicek Vijay Ullal Hynek Hermansky Idiap Research Institute Martigny Switzerland International Computer Science Institute Berkeley CA USA école Polytechnique Fédérale de Lausanne Switzerland

In this paper we propose an extension of the very low bit-rate speech coding technique, exploiting predictability of the temporal evolution of spectral envelopes, for wide-band audio coding applications. Temporal envelopes in critically band-sized sub-bands are estimated using frequency domain linear prediction applied on relatively long time segments. The sub-band residual signals, which play an important role in acquiring high quality reconstruction, are processed using a heterodyning-based signal analysis technique. For reconstruction, their optimal parameters are estimated using a closed-loop analysis-by-synthesis technique driven by a perceptual model emulating simultaneous masking properties of the human auditory system. We discuss the advantages of the approach and show some properties on challenging audio recordings. The proposed technique is capable of encoding high quality, variable rate audio signals on bit-rates below 1 bit/sample.

关键词： Wideband audio coding Speech coding Frequency estimation Frequency domain analysis Signal processing Signal analysis Parameter estimation Humans Auditory system

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STEREO AND MULTICHANNEL audio coding WITH ROOM RESPONSE COMPENSATION FOR IMPROVED coding TRANSPARENCY

STEREO AND MULTICHANNEL AUDIO CODING WITH ROOM RESPONSE COMP...

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European Signal Processing Conference

作者： Maciej Bartkowiak University of Pisa

The paper presents a very simple enhancement of joint coding of stereo and surround channels within a perceptual audio codec. Moreover, the paper proposes two improvements to standard parametric stereo and spatial audio compression in order to avoid the smearing of transients in the process of channel downmixing. The improvements consist in compensating of inter-channel delays prior to mixdown, as well as additional encoding of the room response for realistic reconstruction of the stereo ambience.

关键词： audio coding compensated audio MULTICHANNEL Transparency audio compression

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Linear AM decomposition for sinusoidal audio coding

Linear AM decomposition for sinusoidal audio coding

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

作者： M.G. Christensen A. Jakobsson S.V. Andersen S.H. Jensen Department of Communication Technology University of Aalborg Denmark Department of Electrical Engineering Karlstad University Sweden

We present a novel decomposition for sinusoidal audio coding using amplitude modulation of sinusoids via a linear combination of arbitrary basis vectors. The proposed method, which incorporates a perceptual distortion measure, is based on a relaxation of a non-linear least squares minimization. It offers benefits in the modeling of transients in audio signals. We compare the decomposition to constant-amplitude sinusoidal coding using rate-distortion curves and listening tests. Both indicate that, at the same bit-rate, perceptually significant improvements can be achieved using the proposed decomposition.

关键词： audio coding Amplitude modulation Distortion measurement Chirp modulation Vectors Nonlinear distortion Rate-distortion Modulation coding Frequency dependence Signal resolution

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JOINT OPTIMIZATION OF THE PERCEPTUAL CORE AND LOSSLESS COMPRESSION LAYERS IN SCALABLE audio coding

JOINT OPTIMIZATION OF THE PERCEPTUAL CORE AND LOSSLESS COMPR...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Emmanuel Ravelli Vinay Melkote Tejaswi Nanjundaswamy Kenneth Rose Department of Electrical and Computer Engineering University of California Santa Barbara CA 93106-9560 USA

ISBN: (纸本)9781424442959;9781424442966

MPEG-4 High-Definition Advanced audio coding (HD-AAC) enables scalable-to-lossless (SLS) audio coding with an Advanced audio coding (AAC) base layer, and fine-grained enhancements based on the MPEG SLS standard. While the AAC core offers better perceptual quality at lossy bit-rates, its inclusion has been observed to compromise the ultimate lossless compression performance as compared to the SLS 'non-core' (i.e., without an AAC base layer) codec. In contrast, the latter provides excellent lossless compression but with significantly degraded audio quality at low bit-rates. We propose a trellis-based approach to directly optimize the trade-off between the quality of the AAC core and the lossless compression performance of SLS. Simulations to test the effectiveness of the approach demonstrate the capability to adjust the trade-off to match application specific needs. Moreover, such optimization can in fact achieve an AAC core of superior perceptual quality while maintaining state-of-the-art (and surprisingly sometimes even better) lossless compression, all this in compliance with the HD-AAC standard.

关键词： audio coding lossless coding AAC SLS rate-distortion optimization audio coding lossless compression automatic amplitude control Selective laser sintering Rate distortion lossless coding audio core

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