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IEEE International Conference on Consumer Electronics - Berlin (ICCE-Berlin)

作者： Karlheinz Brandenburg Fraunhofer Institute for Digital Media Technology TU Ilmenau DE

Dr. Karlheinz Brandenburg has been a driving force behind some of today's most innovative digital audio technology, notably the MP3 and MPEG audio standards. He is acclaimed for seminal work in digital audio coding and perceptual measurement techniques, Wave Field Synthesis (WFS) and psycho-acoustics. Karlheinz Brandenburg is a full professor at the Institute for Media Technology at Technische Universität Ilmenau. At the same time he is the director of the Fraunhofer Institute for Digital Media Technology IDMT in Ilmenau.

关键词： Brandenburg Germany Research Institutes Digital audio coding audio coding Digital audio WFS1 gene DIRECTORS Digital media Digital audio players helium Moving Pictures Experts Group DRIVING FORCE

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Packet-loss concealment technology advances in EVS

Packet-loss concealment technology advances in EVS

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： J. Lecomte T. Vaillancourt S. Bruhn H. Sung K. Peng K. Kikuiri B. Wang S. Subasingha J. Faure Fraunhofer IIS VoiceAge Corp. Ericsson AB Samsung Electronics Co. Ltd. ZTE Corporation NTT DOCOMO Inc. Huawei Technologies Co. Ltd Qualcomm Technologies Inc. Orange Labs

ISBN: (纸本)9781467369985

EVS, the newly standardized 3GPP Codec for Enhanced Voice Services (EVS) was developed for mobile services such as VoLTE, where error resilience is highly essential. The presented paper outlines all aspects of the advances brought during the EVS development on packet loss concealment, by presenting a high level description of all technical features present in the final standardized codec. Coupled with jitter buffer management, the EVS codec provides robustness against late or lost packets. The advantages of the new EVS codec over reference codecs are further discussed based on listening test results.

关键词： Concealment EVS VoLTE audio coding speech coding

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Harmonic Vector Quantization

Harmonic Vector Quantization

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： V. Grancharov S. Sverrisson E. Norvell T. Toftgard J. Svedberg H. Pobloth Ericsson Res. Ericsson AB Stockholm Sweden

ISBN: (纸本)9781467369985

audio coding of harmonic signals is a challenging task for conventional MDCT coding schemes. In this paper we introduce a novel algorithm for improved transform coding of harmonic audio. The algorithm does not deploy the conventional scheme of splitting the input signal into a spectrum envelope and a residual, but models the spectral peak regions. The presented coding scheme is part of the recently standardized 3GPP EVS codec.

关键词： audio coding EVS MDCT VQ

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Perceptual coding of High-Quality Digital audio

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PROCEEDINGS OF THE IEEE 2013年第9期101卷 1905-1919页

作者： Brandenburg, Karlheinz Faller, Christof Herre, Juergen Johnston, James D. Kleijn, W. Bastiaan Fraunhofer Inst Digitale Medientechnol Fraunhofer IDMT D-98693 Ilmenau Germany Tech Univ Ilmenau D-98693 Ilmenau Germany Illusonic GmbH CH-8610 Uster Switzerland Ecole Polytech Fed Lausanne CH-1015 Lausanne Switzerland Int Audio Labs Erlangen D-91058 Erlangen Germany Victoria Univ Wellington Sch Engn & Comp Sci Wellington 6140 New Zealand Delft Univ Technol Dept Intelligent Syst NL-2628 Delft Netherlands

This paper introduces high-quality audio coding using psychoacoustic models. This technology is now abundant, with gadgets named after a standard (mp3 players) and the ability to play high-quality audio from literally billions of devices. The usual paradigm for these systems is based on filterbanks, followed by quantization and coding, controlled by a model of human hearing. The paper describes the basic technology, theoretical framework to apply to check for optimality, and the most prominent standards built on the basic ideas and newer work.

关键词： audio coding

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Frequency-domain Comfort Noise Generation for Discontinuous Transmission in EVS

Frequency-domain Comfort Noise Generation for Discontinuous ...

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： A. Lombard S. Wilde E. Ravelli S. Dohla G. Fuchs M. Dietz Fraunhofer IIS Erlangen Germany

ISBN: (纸本)9781467369985

Discontinuous Transmission (DTX) is an efficient way to drastically reduce the transmission rate of a communication codec in the absence of voice input. In this mode, most frames that are determined to consist of background noise only are dropped from transmission and replaced by some Comfort Noise Generation (CNG) in the decoder. In this paper, we propose a novel CNG approach combining information gained about the actual background noise at both encoder and decoder side. It is able to better reproduce background noise types showing a pronounced spectral tilt, which is difficult for traditional schemes based on a linear prediction model. The proposed technique operates in the frequency domain. It is part of the Enhanced Voice Services (EVS) codec, where it is known as FD-CNG. Listening tests show the superior quality of FD-CNG over existing approaches for certain background noise such as car noise.

关键词： CNG DTX EVS audio coding speech coding

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R-TTT Module with Modified Residual Signal for Improving Multichannel audio Signal Accuracy

R-TTT Module with Modified Residual Signal for Improving Mul...

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International Conference on Automation, Cognitive Science, Optics, Micro Electro-Mechanical System, and Information Technology

作者： Ikhwana Elfitri Amirul Luthfi Fitrilina Department of Electrical Engineering Faculty of Engineering Andalas University

ISBN: (纸本)9781467374095

Spatial audio coding is a technique that capable of representing multichannel audio signals as a lower number of audio channels accompanied by spatial parameters and residual signal which will be useful for recreating the original multi-channel audio signals. Moving Picture Expert Group (MPEG) Surround, an international standard developed based on spatial audio coding, specifies Reverse Two-To-Three (R-TTT) module to extend stereo audio, consisted of left and right channels, into three audio channels: left, centre, and right channels based on Channel Prediction Coefficient (CPC) as spatial parameter and residual signal. In this paper, a modified residual signal is proposed to provide a better audio waveform reconstruction in the decoder side by minimising distortion caused by quantisation of CPC. Our experiments show that the waveform accuracy in terms of Signal-to-Noise Ratio (SNR) gets improved as high as 11 dB while the subjective test shows that the proposed method does not reduce perceptual quality, in terms of Subjective Difference Grade (SDG) score, of the reconstructed audio signals.

关键词： MPEG Surround Spatial audio coding Multichannel audio Signals residual signal audio coding audio signals audio channels Moving Pictures Experts Group Subjective testing Quantization

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Overview of the EVS codec architecture

Overview of the EVS codec architecture

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： M. Dietz M. Multrus V. Eksler V. Malenovsky E. Norvell H. Pobloth L. Miao Z. Wang L. Laaksonen A. Vasilache Y. Kamamoto K. Kikuiri S. Ragot J. Faure H. Ehara V. Rajendran V. Atti H. Sung E. Oh H. Yuan C. Zhu Consultant for Fraunhofer IIS Fraunhofer IIS VoiceAge Ericsson AB Huawei Technologies Co. Ltd. Nokia Technologies Finland Nokia Technologies Niooon Telegraph and Telephone Corp. NTT DOCOMO INC. Orange Panasonic Qualcomm Technologies Inc. India Qulcornm-Technologies Inc. Samsung Electronics Co. Ltd. ZTE Corporation

ISBN: (纸本)9781467369985

The recently standardized 3GPP codec for Enhanced Voice Services (EVS) offers new features and improvements for low-delay real-time communication systems. Based on a novel, switched low-delay speech/audio codec, the EVS codec contains various tools for better compression efficiency and higher quality for clean/noisy speech, mixed content and music, including support for wideband, super-wideband and full-band content. The EVS codec operates in a broad range of bitrates, is highly robust against packet loss and provides an AMR-WB interoperable mode for compatibility with existing systems. This paper gives an overview of the underlying architecture as well as the novel technologies in the EVS codec and presents listening test results showing the performance of the new codec in terms of compression and speech/audio quality.

关键词： audio coding mobile communication speech coding

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Fast Algorithms for Low-Delay TDAC Filterbanks in MPEG-4 AAC-ELD

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IEEE-ACM TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2014年第12期22卷 1701-1712页

作者： Chivukula, Ravi K. Reznik, Yuriy A. Hu, Yanyan Devarajan, Venkat Jayendra-Lakshman, Mythreya Indian Sch Business Mohali 140306 Punjab India InterDigital Inc San Diego CA 92121 USA Univ Texas Arlington Dept Elect Engn Arlington TX 76019 USA Qualcomm Inc San Diego CA 92121 USA

The MPEG committee has completed development of a new audio coding standard called "MPEG-4 advanced audio coding-enhanced low delay" (AAC-ELD). AAC-ELD uses low delay spectral band replication (LD-SBR) technology together with a low delay time domain alias cancellation (LD TDAC) filterbank in the encoder to achieve both high coding efficiency and low algorithmic delay. In this paper, we present fast algorithms for implementing LD-TDAC filterbanks in AAC-ELD. Two types of fast algorithms are presented. In the first, we map LD-TDAC analysis and synthesis filterbanks to modified discrete cosine transform (MDCT) and inverse modified discrete cosine transform (IMDCT), respectively. Since MDCT/IMDCT are already extensively used in AAC and they have many fast algorithms, this mapping not only provides a fast implementation but also allows a common implementation of the filterbanks in AAC Low Complexity (AAC-LC), AAC Low Delay (AAC-LD) and AAC-ELD codecs. In the second algorithm, we provide a mapping to discrete Cosine transform of type II. The mapping to DCT-II allows the merger of the matrix operations with the windowing stage that precedes or follows them. This further reduces the number of multiplications and leads to an algorithm with the lowest known arithmetic complexity. For filterbanks of lengths 1024 and 960, we also present a new fast factorization of 15-point DCT-II that requires only 14 irrational multiplications, 3 dyadic rational multiplications and 67 additions.

关键词： AAC audio coding DCT factorization fast algorithms filterbanks low delay MDCT MPEG speech coding time domain alias cancellation

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A blind bandwidth extension method for audio signals based on phase space reconstruction

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EURASIP JOURNAL ON audio SPEECH AND MUSIC PROCESSING 2014年第1期2014卷 1页

作者： Bao, Chang-Chun Liu, Xin Sha, Yong-Tao Zhang, Xing-Tao Beijing Univ Technol Sch Elect Informat & Control Engn Speech & Audio Signal Proc Lab Beijing 100124 Peoples R China

Bandwidth extension is an effective technique for enhancing the quality of audio signals by reconstructing their high-frequency components. In this paper, a novel blind bandwidth extension method is proposed based on phase space reconstruction. Phase space reconstruction is introduced to convert the low-frequency modified discrete cosine transform coefficients of wideband audio to a multi-dimensional space, and the high-frequency modified discrete cosine transform coefficients of the audio signal are reconstructed by a non-linear prediction model. The performance of the proposed method was evaluated through objective and subjective tests. It is found that the proposed method achieves a better performance than the typical linear extrapolation method, and its performance is comparable to the conventional efficient high-frequency bandwidth extension method.

关键词： audio coding Bandwidth extension High-frequency reconstruction Phase space reconstruction

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The IEEE 1857 Standard: Empowering Smart Video Surveillance Systems

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IEEE INTELLIGENT SYSTEMS 2014年第5期29卷 30-39页

作者： Gao, Wen Tian, Yonghong Huang, Tiejun Ma, Siwei Zhang, Xianguo Peking Univ Sch Elect Engn & Comp Sci Beijing Peoples R China

The IEEE 1857 Standard for Advanced audio and Video coding was released as IEEE 1857-2013 in June 2013. Despite consisting of several different groups, the most significant feature of IEEE 1857-2013 is its Surveillance Groups, which can not only achieve at least twice the coding efficiency on surveillance videos as H.264/AVC High Profile, but it"s the most analysis-friendly video coding standard. This article presents an overview of IEEE 1857 Surveillance Groups, highlighting background model-based coding technology and analysis-friendly functionalities. IEEE 1857-2013 will present new opportunities and drive research in smart video surveillance communities and industries.

关键词： IEEE standards audio coding video coding video surveillance H.264-AVC IEEE 1857 standard audio coding background model-based coding technology smart video surveillance video coding audio coding Encoding Feature extraction IEEE 1857 Standards Video coding Video surveillance IEEE 1857 background modeling intelligent systems surveillance video processing video coding standard

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