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检索条件"主题词=Audio coding"

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Binaural reproduction for Directional audio coding

Binaural reproduction for Directional Audio Coding

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Mikko-Ville Laitinen Ville Pulkki Department Signal Processing and Acoustics Helsinki University of Technology Espoo Finland

Directional audio coding (DirAC) is a recently proposed method for spatial sound reproduction. So far it has been used only in loud-speaker reproduction, and a method for headphone reproduction is presented in this article. In principle, the method uses virtual loud-speakers simulated with head related transfer functions (HRTFs), and head tracking. The method was evaluated subjectively, and the results are presented. The results show that a plausible spatial impression can be reproduced using the binaural realization of DirAC.

关键词： audio coding Loudspeakers Microphones Frequency Signal analysis Acoustic signal processing Acoustic applications Signal synthesis Conferences Headphones

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Low delay moving-horizon multiple-description audio coding forwireless hearing aids

Low delay moving-horizon multiple-description audio coding f...

引用

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

作者： Jan Ostergaard Daniel E. Quevedo Jesper Jensen Department of Electronic Systems University of Aalborg Aalborg Denmark School of Electrical Engineering and Computer Science University of Newcastle Australia Oticon Copenhagen Denmark

ISBN: (纸本)9781424423538

In this work, we construct a novel scheme for efficient perceptual coding of audio for robust communication between encoders and wireless hearing aids. To limit the physical size of the hearing aids and to reduce power consumption and thereby increase the lifetime expectancy of the batteries, the hearing aids are constrained to be of low complexity. We therefore provide an asymmetric strategy where most of the computational load is placed at the encoding side. We make use of multiple-description coding. This combats possible erasures on the wireless link between the encoder and the hearing aids without introducing significant delay. Furthermore, we employ psychoacoustically optimized noise-shaping quantizers based on the moving-horizon principle, which exploits a finite prediction horizon.

关键词： Delay audio coding Hearing aids Psychoacoustic models Decoding Robustness Quantization Filters Batteries Noise shaping

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SPATIAL audio coding BY SQUEEZING: ANALYSIS AND APPLICATION TO COMPRESSING MULTIPLE SOUNDFIELDS

SPATIAL AUDIO CODING BY SQUEEZING: ANALYSIS AND APPLICATION ...

引用

European signal processing conference

作者： Bin Cheng Christian Ritz Ian Burnett Whisper Laboratories School of Electrical Computer and Telecommunications Engineering University of Wollongong Wollongong NSW Australia 2500 School of Electrical and Computer Engineering Royal Melbourne Institute of Technology Melbourne VIC Australia 3000

ISBN: (纸本)9781617388767

Spatially Squeezed Surround audio coding (S^AC) proposed by the authors provides efficient compression of multi-channel surround audio. Compression is achieved by exploiting human sound localisation blur to save the surround soundfleld information in a squeezed stereo soundfleld. In this paper, the localisation loss during the S^AC analysis/synthesis is evaluated and the minimum size of the S3AC squeezed soundfleld is derived in a frequency dependent form. Results from perceptual listening tests show that, compared with standard squeezing from a 360° surround soundfleld to a 60°stereo soundfleld, a more intensive squeezing method, such as from 360°-to-5°, does not introduce audible localisation distortion. This leads to a further application of^AC for compressing more than one surround soundfleld into a single stereo downmix for applications such as spatialised teleconferencing. This application is also described and perceptually evaluate.

关键词： minimum dimension audio audio compression audio coding COPPER ALLOY NO 360 MULTICHANNEL Sound Localization Teleconference

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A perceptually enhanced Scalable-to-Lossless audio coding scheme and a trellis-based approach for its optimization

A perceptually enhanced Scalable-to-Lossless audio coding sc...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

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

Scalable-to-lossless (SLS) audio compression, as standardized by MPEG, provides a lossy base layer compatible with the advanced audio coding (AAC) format, ensuring state-of-the-art quality in the base layer, and additional fine grained enhancements that eventually provide a lossless compressed version of the signal. While SLS offers highly efficient lossless compression, the perceptual quality of its intermediate lossy layers has been observed to be suboptimal. This paper proposes a modified SLS audio coding scheme that provides enhanced perceptual quality at an intermediate bit-rate, at the expense of an additional parameter per frequency band as side-information. This scheme when coupled with a trellis-based optimization algorithm is demonstrated to outperform, in terms of quality at the intermediate bit-rate, both standard SLS and a recent perceptually enhanced variant, with minimal degradation in lossless coding performance.

关键词： audio coding Laser sintering Rate-distortion Encoding Performance loss MPEG 4 Standard Discrete transforms Discrete cosine transforms Conferences Application software

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Spatial filtering using directional audio coding parameters

Spatial filtering using directional audio coding parameters

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

作者： Markus Kallinger Giovanni Del Galdo Fabian Kuech Dirk Mahne Richard Schultz-Amling Fraunhofer Institute of Integrated Circuits and Systems Erlangen Germany

ISBN: (纸本)9781424423538

Directional audio coding (DirAC) is a recent method for spatial audio processing, based on a perceptually motivated representation of spatial sound. Due to its efficiency, DirAC has already been proposed for spatial audio teleconferencing scenarios. Modern hands-free communication systems usually include beamforming techniques to improve speech intelligibility by suppressing diffuse background noise and interfering sources. In this paper, we propose a novel spatial filtering method which can be integrated into the DirAC spatial codec. It uses a spectral weighting of the recorded audio signal, where the design of the corresponding spatial filter transfer function is based on the DirAC parameters, i. e., direction-of-arrival and diffuseness of the sound field. Simulation results show that compared to a standard beamformer the novel technique offers significantly higher interference attenuation, while introducing similar distortion of the desired signal.

关键词： Filtering audio coding Teleconferencing Array signal processing Speech enhancement Background noise Codecs Signal design Spatial filters Transfer functions

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Efficient merging of multiple audio streams for spatial sound reproduction in Directional audio coding

Efficient merging of multiple audio streams for spatial soun...

引用

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

作者： Giovanni Del Galdo Fabian Kuech Markus Kallinger Richard Schultz-Amling Audio Department Fraunhofer IIS Erlangen Germany

ISBN: (纸本)9781424423538

Directional audio coding (DirAC) is an efficient technique to capture and reproduce spatial sound. The analysis step outputs a mono DirAC stream, comprising an omnidirectional microphone pressure signal and side information, i.e., direction of arrival and diffuseness of the sound field expressed in time-frequency domain. This contribution proposes a method to merge two or more mono DirAC streams for a joint playback at the reproduction side. This problem arises in applications such as immersive spatial audio teleconferencing. With respect to a trivial direct merging, the proposed method is more efficient as it does not require the synthesis step. From this follows the benefit that the loudspeaker setup at the reproduction side does not have to be known in advance. Simulations and informal listening tests confirm the absence of any artifacts and that the proposed method is practically indistinguishable from the ideal merging.

关键词： Merging Streaming media audio coding MONOS devices Information analysis Signal analysis Microphones Time frequency analysis Teleconferencing Loudspeakers

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G.722 ANNEX D AND G.711.1 ANNEX F - NEW ITU-T STEREO CODECS

G.722 ANNEX D AND G.711.1 ANNEX F - NEW ITU-T STEREO CODECS

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： David Virette Yue Lang Lei Miao Wenhai Wu Balazs Kovesi Claude Lamblin Stephane Ragot Huawei Technologies China France Telecom Orange France

ISBN: (纸本)9781479903573

This paper presents the two new ITU-T Recommendations G.722 Annex D and G.711.1 Annex F, which are stereo extensions of the wideband codecs ITU-T G.722 and G.711.1 and their superwideband extensions (G.722 Annex B and G.711.1 Annex D). An embedded scalable structure is used to add stereo extension layers on top of the wideband or superwideband core coding. Wideband stereo modes are supported at the bit rates of 64/80 and 96/128 kbit/s for G.722 and G.711.1 (respectively), while superwideband stereo modes are supported at 80/96/112/128 and 112/128/144/160 kbit/s. The parametric stereo coding model is based on a frequency domain downmix, wideband interchannel differences estimation, quantization and synthesis, low complexity coherence analysis and synthesis, stereo transient detection and stereo post-processing. An overview of formal ITU-T characterization listening tests illustrates the performance of these codecs.

关键词： speech coding audio coding parametric stereo coding G.722 Annex D G.711.1 Annex F

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Multi-channel audio Compression Method Based on ITU-T G.719 Codec

Multi-channel Audio Compression Method Based on ITU-T G.719 ...

引用

International Conference on Intelligent Information Hiding and Multimedia Signal Processing (IIHMSP)

作者： Wenjin Jiang Jing Wang Yi Zhao Baoguang Liu Xuan Ji School of Information and Electronics Beijing Institute of Technology Beijing China Institute of Acoustics Chinese Academy of Sciences Beijing China Beijing Institute of Technology Beijing Beijing CN

Through exploiting the human perception of spatial sound, a new approach for compression coding of multi-channel audio signal based on ITU-T G.719 codec is put forward in this paper. Multi-channel input signals are converted to a down-mixed signal plus spatial perceptual parameters by use of down-mix and up-mix step-by-step techniques in frequency domain. The algorithm can significantly reduce the coding rate under the premise of an acceptable sound quality in combination with the G.719 audio codec. The paper presents the implementation of the algorithm and describes in detail the calculation and features of the selected spatial parameters. Finally some experiments are done to evaluate the algorithm from the perspective of the compression ratio, the reconstructed sound quality, and the algorithm complexity.

关键词： Codecs Complexity theory Quantization (signal) Decoding audio coding audio compression

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Technologies for speech and audio coding

Technologies for speech and audio coding

引用

IEEE International Symposium on Consumer Electronics (ISCE)

作者： Takehiro Moriya NTT Communication Science Laboratories Nippon Telegraph and Telephone Corporation Atsugi Japan

This paper reviews some important technologies for speech and audio coding, contrasting time domain and frequency domain processing. These technologies have contributed to the standardization activities in ITU-T and ISO/IEC MPEG, respectively. It is interesting that the standardization targets for speech and audio have started to converge recently. The MPEG USAC (Unified Speech and audio coding) scheme, which is being actively developed, may make use of most of the important technologies for speech and audio that have been developed so far.

关键词： Speech coding audio coding Frequency domain analysis Standardization Speech processing Narrowband Bit rate Delay Bandwidth Linear predictive coding

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Distributed audio coding with Efficient Source Correlation Extraction

Distributed Audio Coding with Efficient Source Correlation E...

引用

IEEE Digital Signal Processing Workshop

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

ISBN: (纸本)9781424436767

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 encode MDCT coefficients of the second source. Based on conditional entropy calculations, exploitation of this correlation results in potential improvements of 54-58% in coding efficiency. Using a multi-context adaptive arithmetic coder, an actual bitrate reduction of 38% is achieved.

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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