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IEEE Sensor Array and Multichannel Signal Processing Workshop

作者： Lee, Jonghwa Lee, Chulhee Yonsei Univ Dept Elect & Elect Engn Seoul 120749 South Korea

ISBN: (纸本)9781424422401

This paper presents a multi-channel audio coding method compatible with the standard stereo audio coding method. The encoding method consists of three parts: spatial-temporal analysis of sound source location, compressing multi-channel audio into stereo audio and encoding stereo signals. The proposed method is simpler and more effective than conventional methods in terms of audio data size with small additional information. With proposed method, it is possible to transmit or store multi-channel audio signals as stereo audio signals.

关键词： Encoding ISO audio coding Position measurement Standards Transform coding Digital audio broadcasting

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Efficient implementation of a class of MDCT/IMDCT filterbanks for speech and audio coding applications

Efficient implementation of a class of MDCT/IMDCT filterbank...

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33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Chivukula, Ravi K. Reznik, Yuriy A. QUALCOMM Inc San Diego CA 92121 USA Univ Texas Arlington Elect Engn Dept Arlington TX 76019 USA

ISBN: (纸本)9781424414833

In this paper, an efficient algorithm for implementing MDCT/IMDCT of lengths N = 5.2(m) (m >= 2) is presented. Transforms for such lengths are of interest for speech and audio coding applications, such as recently issued and/or emerging standards G.729.1, ***-VBR, and EVRC-WB. In our design we utilize a mapping of MDCT of size N into N/2-point DCT-IV and DCT-II with isolated pre-multiplications, which are subsequently moved in the windowing stage. We show that such a modified window is piece-wise symmetric, and can be stored using N/2 words. In our algorithm we also use an efficient factorization of 5-point DCT-II which requires only 4 multiplications by irrational factors. We compare our proposed algorithm with several alternative implementations and show that our design offers practically appreciable reduction in complexity and memory usage.

关键词： speech coding audio coding FFT DCT MDCT factorization fast algorithms

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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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33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Holters, Martin Zoelzer, Udo Helmut Schmidt Univ Hamburg Germany

ISBN: (纸本)9781424414833

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： audio coding linear predictive coding adaptive filters

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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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Workshop on Hands-Free Speech Communication and Microphone Arrays

作者： Kallinger, Markus Kuech, Fabian Schultz-Amling, Richard Del Galdo, Giovanni Ahonen, Jukka Pulkki, Ville Fraunhofer Inst Integrated Circuit IIS Erlangen Germany Aalto Univ Lab Acoust & Audio Sig Proc Helsinki Finland

ISBN: (纸本)9781424423378

Modem 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： spatial audio coding microphone array spatial filters audio coding planar arrays

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The diffuse sound field in energetic analysis

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JOURNAL OF THE ACOUSTICAL SOCIETY OF AMERICA 2012年第3期131卷 2141-2151页

作者： Del Galdo, Giovanni Taseska, Maja Thiergart, Oliver Ahonen, Jukka Pulkki, Ville Fraunhofer Inst Integrated Circuits Audio Dept D-91058 Erlangen Germany Aalto Univ Sch Elect Engn Dept Signal Proc & Acoust FI-00076 Helsinki Finland

Measuring the degree of diffuseness of a sound field is crucial in many modern parametric spatial audio techniques. In these applications, intensity-based diffuseness estimators are particularly convenient, as the sound intensity can also be used to obtain, e.g., the direction of arrival of the sound. This contribution reviews different diffuseness estimators comparing them under the conditions found in practice, i.e., with arrays of noisy microphones and with the expectation operators substituted by finite temporal averages. The estimators show a similar performance, however, each with specific advantages and disadvantages depending on the scenario. Furthermore, the paper derives an estimator and highlights the possibility of using spatial averaging to improve the temporal resolution of the estimates. (C) 2012 Acoustical Society of America. [DOI: 10.1121/1.3682064]

关键词： acoustic field acoustic intensity acoustic signal processing acoustic wave scattering audio coding direction-of-arrival estimation

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An MPEG-4 AAC decoder FPGA implementation for the Brazilian digital television

An MPEG-4 AAC decoder FPGA implementation for the Brazilian ...

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Southern Conference on Programmable Logic

作者： Adriano Renner Altamiro Amadeu Susin PPGEE UFRGS Porto Alegre Brazil

This paper presents an MPEG-4 AAC decoder described in VHDL language and compliant with the Brazilian Digital Television standard (SBTVD). It has been synthesized to an Altera Cyclone II 2C35 FPGA using 26549 logic elements and 248704 memory bits. The implemented architecture has been verified using an Altera DE2 prototyping board, being capable of decoding stereo signals coded as MPEG-4 AAC Low Complexity audio objects. The minimum operating frequency required for real time decoding of a stereo audio stream with a sampling rate of 48 kHz is 4 MHz and the implemented decoder is capable of running at 56 MHz, meeting the requirements. This decoder design is intended to be integrated with a system on chip for the SBTVD set-top box.

关键词： Decoding Transform coding Standards Transforms audio coding Filter banks Quantization

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Detection of tampering in lossy compressed digital audio recordings

Detection of tampering in lossy compressed digital audio rec...

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Signal Processing Algorithms, Architectures, Arrangements and Applications (SPA)

作者： Rafał Korycki Institute of Radioelectronics Warsaw University of Technology Warsaw Poland

This paper addresses the problem of tampering detection and discusses one of the methods used for authenticity analysis of digital audio recordings. Presented approach is based on checking frame offsets in audio files compressed by using perceptual audio coding. This method can be further improved by applying histogram analysis of modified discrete cosine transform spectrum and detection of maximum number of nonzero spectral coefficients. The influence of compression algorithms' parameters on detection of forgeries are presented by applying AAC and Ogg Vorbis encoders as examples. The effectiveness of tampering detection algorithms proposed in this paper is tested on a predefined music database and compared graphically using ROC-like curves.

关键词： Decision support systems audio recording Abstracts audio coding Histograms Discrete cosine transforms Compression algorithms

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Tutorial on Critical Listening of Multi-Channel audio Codec Performance

Tutorial on Critical Listening of Multi-Channel Audio Codec ...

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SMPTE Technical Conference

作者： Sunil Bharitkar Grant Davidson Louis Fielder Poppy Crum Dolby Laboratories Inc. 100 Potrero Avenue San Francisco CA 94103

ISBN: (纸本)9781622765386

Listening for impairments introduced by multichannel audio codecs is an important task. Classical objective methods are not adequate in assessing audio coding schemes. Accordingly, the ITU-R BS.1116 & 1534 recommendations provide guidelines for subjective evaluation of codecs. This paper provides a tutorial on the proper conditions to do reliable codec testing. Several key components covered are, proper experimental design, selection of listening panel and training of listeners, developing the test methodology, selecting balanced program material, loudspeaker/room and sound-field requirements, listening for artifacts, and statistical analysis. This paper addresses these various components including the sound-field requirements since, as per the ITU: “The characteristics of the reference sound field at the listening area are most important for the subjective perception of, or the quality assessment of, auditory events and their reproducibility at other listening places or rooms. These characteristics result from the interaction of the loudspeaker(s) and the listening room”.

关键词： audio codec evaluation ITU-R Listening tests BS.1116 Critical listening setup Subjective testing Auscultation Listening Tutorials audio coding acoustic field Quality assessment codecs Loudspeakers International Telecommunication Union Intensive Care Units subjective evaluation Listeners design and selection physical impairment

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Sound Field Model Violations in Parametric Spatial Sound Processing

Sound Field Model Violations in Parametric Spatial Sound Pro...

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IWAENC 2012;International Workshop on Acoustic Signal Enhancement

作者： Oliver Thiergart Emanuel A. P. Habets International Audio Laboratories Erlangen

Describing a sound field with a single audio signal and parametric side information offers an efficient way of storing, transmitting, or reproducing the recorded spatial sound. In many approaches, the parametric representation is based on a simple sound field model assuming for each time-frequency instance a single plane wave and diffuse sound. When multiple sources are active, we thus have to assume that the source signals do not overlap much in the time-frequency domain so that the single-wave model is still satisfied. In this paper, we show that a single-wave model is easily violated even for sparse signals such as speech. When the parametrized spatial sound is reproduced, these violations degrade the spatial quality. We further study whether the well-known measure of approximate W-disjoint orthogonality can be applied to detect the relevant model violations.

关键词： Speech Time frequency analysis Microphones audio coding Loudspeakers Indexes Acoustics

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Efficient transform coding of two-channel audio signals by means of complex-valued stereo prediction

Efficient transform coding of two-channel audio signals by m...

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

作者： Christian R. Helmrich Pontus Carlsson Sascha Disch Bernd Edler Johannes Hilpert Matthias Neusinger Heiko Purnhagen Nikolaus Rettelbach Julien Robilliard Lars Villemoes Fraunhofer Institut für Integrierte Schaltungen Erlangen Germany Dolby Sweden AB Stockholm Sweden

ISBN: (纸本)9781457705380

Traditional MDCT-based perceptual audio coding schemes employ mid/side and intensity stereo techniques to allow efficient joint coding of the two channels of a stereophonic signal. These techniques, however, provide only little coding gain for critical stereo signals characterized by spectral components with a distinct level or phase difference between the channels. To overcome this deficiency, we propose an extension to the mid/side coding paradigm that utilizes complex-valued inter-channel linear prediction in the MDCT spectral domain. The required imaginary spectrum (MDST) is calculated in a computationally efficient manner without additional algorithmic delay. A formal listening test conducted in the course of the ISO/MPEG standardization of the unified speech and audio codec USAC illustrates that the proposed stereo prediction approach pro vides significant improvements in coding efficiency and shows that at 96 kb/s, excellent quality can be obtained even for critical signals.

关键词： Decoding Transforms Complexity theory audio coding Transform coding Compaction

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