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检索条件"主题词=Audio coding"

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Superwideband Extension of G.718 and G.729.1 Speech Codecs

Superwideband Extension of G.718 and G.729.1 Speech Codecs

引用

11th Annual Conference of the International-Speech-Communication-Association 2010

作者： Laaksonen, Lasse Tammi, Mikko Malenovsky, Vladimir Vaillancourt, Tommy Lee, Mi Suk Yamanashi, Tomofumi Oshikiri, Masahiro Lamblin, Claude Kovesi, Balazs Miao, Lei Zhang, Deming Gibbs, Jon Francois, Holly Nokia Finland VoiceAge Canada ETRI Korea Republic of Panasonic Corp. Japan France Telecom France Huawei Technologies China Motorola United Kingdom

ISBN: (纸本)9781617821233

This communication presents the recently standardized superwideband (SWB) extensions of ITU-T G.718 and G.729.1. These extensions were standardized as G.718 annex B and G.729.1 annex E. The SWB functionality is implemented using embedded scalable layers on top of the wideband (WB) core codecs, and it extends the bit rate of the codecs to 48 and 64 kbit/s for the G.718 and G.729.1, respectively. The main technology is a two-mode SWB coding method of the high frequencies. In addition, the G.729.1 SWB extension enhances the lower frequency range. The codec performance is illustrated with some listening test results extracted from the ITU-T Characterization phase.

关键词： speech coding audio coding embedded coding scalable coding superwideband bandwidth extension ITU

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Arithmetic coding of Sub-band Residuals in FDLP Speech/audio Codec

Arithmetic Coding of Sub-band Residuals in FDLP Speech/Audio...

引用

10th INTERSPEECH 2009 Conference

作者： Motlicek, Petr Ganapathy, Sriram Hermansky, Hynek Idiap Res Inst Martigny Switzerland Johns Hopkins Univ ECE Dept Baltimore MD 21218 USA

ISBN: (纸本)9781615676927

A speech/audio codec based on Frequency Domain Linear Prediction (FDLP) exploits auto-regressive modeling to approximate instantaneous energy in critical frequency sub-bands of relatively long input segments. The current version of the FDLP codec operating at 66 kbps has been shown to provide comparable subjective listening quality results to state-of-the-art codecs on similar bit-rates even without employing standard blocks such as entropy coding or simultaneous masking. This paper describes an experimental work to increase compression efficiency of the FDLP codec by employing entropy coding. Unlike conventional Huffman coding employed in current speech/audio coding systems, we describe an efficient way to exploit arithmetic coding to entropy compress quantized spectral magnitudes of the sub-band FDLP residuals. Such an approach provides 11% (similar to 3 kbps) bit-rate reduction compared to the Huffman coding algorithm (similar to 1 kbps).

关键词： audio coding Frequency Domain Linear Prediction (FDLP) Entropy coding Arithmetic coding Huffman coding

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Maximal Coherence Rotation for stereo coding

Maximal Coherence Rotation for stereo coding

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： Shuhua Zhang Weibei Dou Huazhong Yang Tsinghua National Laboratory of Information Science and Technology Department of Electronic Engineering Tsinghua University Beijing China

This paper presents a linear operation called Maximal Coherence Rotation (MCR) on paired vectors for stereo audio coding. Intrigued by the idea that stronger coherence between paired channels will lead to higher stereo coding efficiency, we develop MCR to maximize the coherence restricted by being invertible and energy-conserving. It results in equal energy, minimized difference, and always non-negative coherence for the pair of channels processed. In binaural hearing, this can be viewed as turning a physical sound source at any azimuth to a virtual one on the median plane. A prototype MCR stereo coder shows significantly higher quality for some test sequences than that of AMR-WB+, one of the best low bitrate stereo coders in the public domain. And as a preprocessing tool for MPEG-4 Parametric Stereo (PS), MCR avoids out-of-phase inter-channel cancellation during 2-to-1 channel downmixing without any additional bandwidth requirement, thanks to the maximized coherence.

关键词： Coherence audio coding Azimuth Auditory system Transforms Ear

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Keynote address 1

Keynote address 1

引用

IEEE International Symposium on Consumer Electronics (ISCE)

Provides an abstract of the presentation and a brief professional biography of the presenter. The complete presentation was not made available for publication as part of the conference proceedings.

关键词： Vehicles Consumer electronics Electrical engineering USA Councils audio coding Industries Media

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MULTI-BAND PRE-ECHO CONTROL USING A FILTERBANK EQUALIZER

MULTI-BAND PRE-ECHO CONTROL USING A FILTERBANK EQUALIZER

引用

European Signal Processing Conference

作者： Bernd Geiser Martin Roggendorf Peter Vary Institute of Communication Systems and Data Processing (iNd) RWTH Aachen University

We propose a new method to mitigate so called "pre-echo artifacts" that occur with transform-based audio codecs owing to their comparatively large block lengths. Our algorithm operates entirely in the time domain. In contrast to previous time domain approaches, a frequency selective pre-echo control (PEC) is achieved by applying the concept of a filterbank equalizer (FBE) with non-uniform frequency resolution. As an example application, the new PEC approach has been used to enhance the ITU-T G.722.1C super-wideband codec. The performance of the algorithm has been objectively measured and optimized w.r.t. various FBE parameters. Considerable quality gains could be obtained which are confirmed by the subjective listening impression. The proposed algorithm can be easily extended to act as an adaptive spectro-temporal pre- and deemphasis filter.

关键词： TIME-DOMAIN BLOCK LENGTH algorithms frequency resolution codecs Standing Orders swine enteric calicivirus audio coding de emphasis Filter banks Equalizers

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audio encoding using Huang and Hilbert transforms

Audio encoding using Huang and Hilbert transforms

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International Symposium on Communications Control and Signal Processing (ISCCSP)

作者： K. Khaldi A.O. Boudraa B. Torrésani Th. Chonavel M. Turki IRENav (EA 3634) Ecole Navale BCRM Brest Brest France LATP CMI Université de Provence Marseilles France Institut Telecom Télécom Bretagne Brest France Unità Signaux et Systemes Tunis Tunisia

In this paper an audio coding scheme based on the Empirical Mode Decomposition (EMD) in association with the Hilbert transform is presented. The audio signal is decomposed adaptively into intrinsic oscillatory components called Intrinsic Mode Functions (IMFs) by EMD, and the associated instantaneous amplitudes and the instantaneous phases are calculated. The basic principle of the proposed approach consists in encoding the instantaneous amplitudes by linear prediction and the instantaneous phases by scalar quantization. The decoder recovers the original signal from IMFs reconstruction by demodulation and summation. The compression method is applied to different audio signals, and results are compared to MP3 a variable bit rate coder and to wavelet approaches.

关键词： Encoding Bit rate Frequency Signal processing Digital audio players Telecommunications Process control Communication system control audio coding Quantization

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audio object individual operation and its application to earphone leakage noise reduction

Audio object individual operation and its application to ear...

引用

International Symposium on Communications Control and Signal Processing (ISCCSP)

作者： Shota Suzuki Shigeki Miyabe Noriyoshi Kamado Hiroshi Saruwatari Kiyohiro Shikano Toshiyuki Nomura Nara Institute of Science and Technology Nara Japan University of Tokyo Tokyo Japan NEC Corporation Limited Kanagawa Japan

In this paper, we propose a new extension framework of multichannel audio coding based on temporal quantization of spatial information. In our previous study, multiple-audio-object signal can be encoded/decoded via prototypes of directional clustering for each audio object. This paper, first, pays attention to the fact that quantized information corresponds to the spatial image of each sound object, and is easily modified and operated by the end users. Next, as a novel application of the spatial image operation, we propose a new earphone leakage noise reduction method in which the mostly leaked sound object can be detected automatically based on the higher-order statistics, and eliminated. The effectiveness of the proposed CODEC is verified in the subjective evaluation, and leakage noise reduction method is assessed in the objective evaluation.

关键词： Headphones Noise reduction audio coding Quantization Decoding Prototypes Object detection Leak detection Higher order statistics Codecs

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NOISE-TO-MASK RATIO MINIMIZATION BY WEIGHTED NON-NEGATIVE MATRIX FACTORIZATION

NOISE-TO-MASK RATIO MINIMIZATION BY WEIGHTED NON-NEGATIVE MA...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Joonas Nikunen Tuomas Virtanen Department of Signal Processing Tampere University of Technology Korkeakoulunkatu 1 FI-33720 Tampere Finland

ISBN: (纸本)9781424442959;9781424442966

This paper proposes a novel algorithm for minimizing the perceptual distortion in non-negative matrix factorization (NMF) based audio representation. We formulate the noise-to-mask ratio audio quality criterion in a form where it can be used in NMF and propose an algorithm for optimizing the criterion. We also propose a method for compensating the spreading of the representation error in the synthesis filterbank. The objective perceptual quality produced by the proposed method is found to outperform all the reference methods. We also study the trade-off between the window length and the rank of factorization with a fixed data rate, and find that the best performance is obtained with window lengths between 10 and 30 ms.

关键词： Non-negative matrix factorization Noise-to-mask ratio audio coding Signal representations audio coding factorization signal representation matrix decomposition NMF protocol audio REFERENCE METHODS

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Different Speed Optimizations for MPEG AAC Encoder

Different Speed Optimizations for MPEG AAC Encoder

引用

International Conference on Advances in Computer Engineering (ACE)

作者： Vishakha N. Gor Jignesh N. Sarvaiya ECE Department SVNIT Surat India

ISBN: (纸本)9781424471546;9780769540580

The MPEG Advanced audio coding (AAC) algorithm is an example for a modern audio coding algorithm which is equipped with a number of coding tools, delivers unsurpassed audio quality at rates at or below 64 kbps/channel. AAC defined by ISO/IEC standard is very slow and it was impossible to perform real time encoding at any platform. Thus in this paper we carry out algorithm level platform independent optimizations of most time consuming blocks of MPEG AAC Encoder. These optimizations will be relevant to real time implementation of MPEG AAC Encoder. Experimental results show the optimized encoder can save the computational complexity and improve the speed.

关键词： ISO standards IEC standards Psychoacoustic models audio coding Digital signal processing Computational complexity Filter bank Code standards Quantization Iterative algorithms

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A CONTINUOUS MODULATED SINGLE SIDEBAND BANDWIDTH EXTENSION

A CONTINUOUS MODULATED SINGLE SIDEBAND BANDWIDTH EXTENSION

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Frederik Nagel Sascha Disch Stephan Wilde Fraunhofer IIS Erlangen Germany Leibniz Universitat Hannover Germany

ISBN: (纸本)9781424442959;9781424442966

Bandwidth extension (BWE) is an important parametric technique applied by modern audio coders in order to achieve efficient data rate compression at low bitrates. The perceptual quality of BWE enhanced signals is, however, often hampered by artifacts caused by inharmonicity. We propose hence a bandwidth extension method that avoids inharmonicity, and, at the same time, avoids the costly transmission of additional control parameters for frequency shifts. Harmonicity of the decoded signal is ensured by calculation of the autocorrelation function of the magnitude spectrum. The proposed bandwidth extension method is implemented by single sideband modulation (SSM). Fitting nicely in the general scheme of well established spectral band replication (SBR), the new method has some potential to replace the SBR patching algorithm. Potentially, the SSM can additionally govern the subsequent spectral shaping. Listening test results demonstrate an advantage of the novel scheme compared to SBR as used in high efficiency advanced audio coding (HE-AAC).

关键词： Bandwidth extension audio coding modulation spectral autocorrelation audio coding Autocorrelation function Bandwidth SBR BUCKET WHEEL EXCAVATORS SSM standard reference materials control parameter

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