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检索条件"主题词=Audio coding"

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EFFICIENT CONTEXT ADAPTIVE ENTROPY coding FOR REAL-TIME APPLICATIONS

EFFICIENT CONTEXT ADAPTIVE ENTROPY CODING FOR REAL-TIME APPL...

引用

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

作者： Fuchs, Guillaume Subbaraman, Vignesh Multrus, Markus Fraunhofer Inst Integrierte Schaltungen D-91058 Erlangen Germany

ISBN: (纸本)9781457705397

Context based entropy coding has the potential to provide higher gain over memoryless entropy coding. However serious difficulties arise regarding the practical implementation in real-time applications due to its very high memory requirements. This paper presents an efficient method for designing context adaptive entropy coding while fulfilling low memory requirements. From a study of coding gain scalability as a function of context size, new context design and validation procedures are derived. Further, supervised clustering and mapping optimization are introduced to model efficiently the context. The resulting context modelling associated with an arithmetic coder was successfully implemented in a transform-based audio coder for real-time processing. It shows significant improvement over the entropy coding used in MPEG-4 AAC.

关键词： Entropy coding audio coding context modelling

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RE-ENGINEERING ITU-T G.722: LOW DELAY AND COMPLEXITY SUPERWIDEBAND coding AT 64 KBIT/S WITH G.722 BITSTREAM WATERMARKING

RE-ENGINEERING ITU-T G.722: LOW DELAY AND COMPLEXITY SUPERWI...

引用

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

作者： Kovesi, Balazs Ragor, Stephane Lamblin, Claude Miao, Lei Liu, Zexin Hu, Chen France Telecom Orange France Huawei Technologies China

ISBN: (纸本)9781457705397

This paper presents the lowest bitrate mode (64 kbit/s) of the new superwideband (SWB, 50-14000 Hz) coder, recently standardized as ITU-T G.722 Annex B. This mode provides a superwideband extension of G.722 at 56 kbit/s with one 8 kbit/s enhancement layer divided in two sub-layers. The resulting bitstream is compatible with ITU-T G.722 at 64 kbit/s and can be viewed as watermarking G.722 least significant bits (LSBs). The novel technologies in this mode include G.722 enhancements (noise feedback coding, scalable quantization in G.722 higher band), as well as multimode bandwidth extension (BWE). Selected ITU-T characterization test results and additional informal test results show that the 64 kbit/s mode of G.722 Annex B gives high SWB quality with low delay and complexity.

关键词： Speech coding audio coding G.722B superwideband bandwidth extension watermarking

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The Application of Nonuniform Fast Fourier Transform in audio coding

The Application of Nonuniform Fast Fourier Transform in Audi...

引用

2008 International Conference on audio，Language and Image Processing(2008国际声音、语言、图像过程大会)

作者： Zheng Deng Jing Lu Institute of Acoustics Key Lab of Modern Acoustics Nanjing University Nanjing 210093

In this paper, we introduce nonuniform fast Fourier transform (NUFFT) into audio coding. NUFFT benefits from the flexible resolution in different frequency range which is unable to be achieved by the common uniform transforms such as FFT, SDFT, and MDCT. So it matches better to the nonlinear property of human auditory system. As a result, the precision of psychoacoustic model and coding efficiency will be improved. To further increase the calculation efficiency, a new inverse transform method is proposed in this paper. Although some calculation errors are inevitable, fortunately the errors exist only in high frequency range, to which human ear is insensitive. Comparing with FFT, it is shown that NUFFT has better stability to the quantization errors especially at low and middle frequency range. Experiments prove the superiority of NUFFT in both the numerical precision and the objective audio quality, especially at low bit rates.

关键词： audio coding Transforms Bit rate Frequency domain analysis Bandwidth Time domain analysis Quantization

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Algebraic quantization of transform coefficients for embedded audio coding

Algebraic quantization of transform coefficients for embedde...

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

作者： M. De Meuleneire H. Taddei D. Pastor Nokia Siemens Networks COO RTP PT HWCT Computing Techn SDE Munich Germany Department SC ENST Bretagne Technopole Brest Iroise Brest France

This paper proposes a new quantization for transform coefficients based on algebraic quantization. The coefficients are represented by a few pulses multiplied by a unique amplitude. The coefficients to be transmitted are selected by optimizing an error criterion, that determines the signs, positions and amplitudes of the pulses. This simple quantization has been implemented in a wavelet-based wideband scalable coder, and has been proved to provide a perceptually better quality than SPIHT on speech signal and music.

关键词： audio coding Bit rate Transform coding Decoding Bandwidth Time frequency analysis Wavelet transforms Huffman coding Vector quantization Image reconstruction

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Delay-free lossy audio coding using shelving pre- and post-filters

Delay-free lossy audio coding using shelving pre- and post-f...

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

作者： Martin Holters Udo Zolzer Helmut Schmidt University Hamburg Hamburg Germany

A delay-free audio coding scheme based on ADPCM with adaptive pre- and post-filtering is presented. The pre-/post- filters are realized as a cascade of shelving filters, designed to match the characteristics of human perception. The pre- and post-filters are adapted by dynamic compression of the respective sub-bands. The adaption is backward-adaptive, i.e. is fed by the reconstructed signal, which eliminates the need to transmit the filter coefficients and allows delay-free operation. This pre- and post-filtering significantly improves the audio quality compared to a plain ADPCM codec, as underlined by objective measurements. Since the base ADPCM used is also delay-free, the resulting coding system works without any algorithmic delay.

关键词： Delay audio coding Codecs Decoding Quantization Finite impulse response filter Cutoff frequency Lattices Humans Adaptive filters

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Efficient correlation extraction for distributed audio coding

Efficient correlation extraction for distributed audio codin...

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Asilomar Conference on Signals, Systems & Computers

作者： Sandeep Matta Charles D. Creusere Klipsch School of Electrical and Computer Engineering New Mexico State University Las Cruces NM USA

Distributed source coding is one of the enabling technologies for efficient bandwidth utilization in wireless sensor networks and is consequently of great current interest. This paper studies its application to audio signals, using a transform weighted interleaved vector quantization (TWIN-VQ) framework and allowing a sensor node to passively receive and use information from neighboring sensors that is being transmitted to the common joint decoder. Specifically, it uses the linear predictor coefficients generated as side information by TWIN-VQ for one source to determine its frame-by-frame correlations with another source and then conditionally encodes MDCT coefficients of the second source. Based on conditional entropy calculations, exploitation of this correlation results in improvements of 54-58% in coding efficiency.

关键词： audio coding Decoding Source coding Bandwidth Computational complexity Energy consumption Data mining Sensor systems Sensor arrays Wireless sensor networks

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ENHANCED coding OF HIGH-FREQUENCY TONAL COMPONENTS IN MPEG-D USAC THROUGH JOINT APPLICATION OF ESBR AND SINUSOIDAL MODELING

ENHANCED CODING OF HIGH-FREQUENCY TONAL COMPONENTS IN MPEG-D...

引用

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

作者： Zernicki, Tomasz Bartkowiak, Maciej Domanski, Marek Appl Res Ctr Telcordia Poland Sp Zoo Umultowska 85 Poznan Poland

ISBN: (纸本)9781457705397

The new eSBR tool of MPEG-D Universal Speech and audio coding offers a great advantage in compression of high frequency content, however it produces audible artifacts for sounds whose pitch frequencies are strongly variable or exceeding the split frequency of eSBR. We propose an extension of the forthcoming standard by adding a high frequency sinusoidal tool. This tool introduces additional parametric information to the data bitstream in order to encode the challenging tonal components which are excluded from eSBR processing. Listening tests demonstrate benefits of the proposed approach for test items of strongly tonal character.

关键词： Bandwidth extension audio coding sinusoidal modeling

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A wideband speech and audio coding candidate for ITU-T G.711WBE standardization

A wideband speech and audio coding candidate for ITU-T G.711...

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

作者： Yusuke Hiwasaki Takeshi Mori Shigeaki Sasaki Hitoshi Ohmuro Akitoshi Kataoka NTT Cyber Space Laboratories Musashino Tokyo Japan

This paper describes an ITU-T G.711 embedded wideband speech coder, submitted as a candidate to the ITU-T G.711 wideband extension standardization qualification phase. The codec generates a bitstream comprised of three layers: a G.711 core layer with noise shaping, a time-domain weighted vector quantized narrowband enhancement layer, and an MDCT-based higher band enhancement layer. Through subjective evaluations, the coder was found to meet all tested requirements and objectives set in terms of reference, with a low computational complexity at 10 WMOPS.

关键词： Wideband Speech coding audio coding Standardization Qualifications Codecs Noise shaping Time domain analysis Narrowband Testing

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Biologically-inspired neural coding of sound onset for a musical sound classification task

Biologically-inspired neural coding of sound onset for a mus...

引用

International Joint Conference on Neural Networks (IJCNN)

作者： Newton, Michael J. Smith, Leslie S. Univ Stirling Inst Comp Sci & Math Stirling FK9 4LA Scotland

ISBN: (纸本)9781424496365

A biologically-inspired neural coding scheme for the early auditory system is outlined. The cochlea response is simulated with a passive gammatone filterbank. The output of each bandpass filter is spike-encoded using a zero-crossing based method over a range of sensitivity levels. The scheme is inspired by the highly parallellised nature of the auditory nerve innervation within the cochlea. A key aspect of early auditory processing is simulated, namely that of onset detection, using leaky integrate-and-fire neuron models. Finally, a time-domain neural network (the echo state network) is used to tackle the what task of auditory perception using the output of the onset detection neurons alone. A set of interim results are presented.

关键词： Auditory system Encoding Instruments Neurons Reservoirs Sensitivity Time domain analysis audio coding auditory nerve innervation auditory perception band-pass filters bandpass filter biologically-inspired neural coding biology computing cochlea response early auditory system echo state network hearing leaky integrate-and-fire neuron models music musical sound classification neural nets onset detection passive gammatone filterbank sound onset spike encoding time-domain neural network zero-crossing based method

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Enhanced Direction Estimation Using Microphone Arrays for Directional audio coding

Enhanced Direction Estimation Using Microphone Arrays for Di...

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Hands-Free Speech Communication and Microphone Arrays, HSCMA

作者： Markus Kallinger Fabian Kuech Richard Schultz-Amling Giovanni del Galdo Jukka Ahonen Ville Pulkki Fraunhofer Institute of Integrated Circuits and Systems Erlangen Germany Laboratory of Acoustics and Audio Signal Processing Helsinki Finland

Modern home entertainment systems offer surround sound audio playback. This progress over known mono and stereo devices is also intended for high quality hands-free telephony to enhance intelligibility of speech in group conversation. Directional audio coding (DirAC) provides an efficient and well-established way to record and encode spatial sound and to render it at an arbitrary loudspeaker setup. On the recording site, DirAC is based on B-format microphone signals. These signals can be obtained by one omnidirectional and three figure-of-eight microphones pointing along the axes of a three-dimensional Cartesian coordinate system. However, a grid of omnidirectional microphones is more appropriate for consumer applications due to economic reasons. Arrays can provide the required figure-of-eight directionality only for a certain frequency range. However, in this contribution we show that a straightforward direction estimator is biased. After formulating the bias analytically we propose an unbiased estimator and derive the theoretical limits for unique direction estimation. The results are illustrated by means of simulations and measurements.

关键词： Microphone arrays audio coding Azimuth audio recording Loudspeakers Frequency Laboratories Acoustic devices MONOS devices Telephony

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