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The MPEG Surround audio coding Standard

IEEE SIGNAL PROCESSING MAGAZINE 2009年第1期26卷 148-152页

作者： Hilpert, Johannes Disch, Sascha Fraunhofer Inst Integrated Circuits Audio Dept Grp Audiocoding Erlangen Germany Fraunhofer Inst Integrated Circuits Audio Dept Multimediasoftware Grp Erlangen Germany Leibniz Univ Hannover Hannover Germany

This article provides a compact overview of the history, technology, and performance of MPEG Surround. The technology of MPEG Surround is based on the spatial audio coding (SAC) principle: In the encoder, a mono- or stereophonic down- mix is generated from the multichannel input signal, and additional parametric side information is extracted to guide the subsequent up-mix procedure in the *** Moving Pictures Expert Group (MPEG) Surround, a data-rate efficient coding scheme for high-quality multichannel sound using novel parametric coding techniques has been standardized.

关键词： audio coding Bit rate Digital video broadcasting Digital audio broadcasting Signal processing Decoding Material storage DVD Standardization MPEG standards

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A new bitplane coder for scalable transform audio coding

A new bitplane coder for scalable transform audio coding

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

作者： Thi Minh Nguyet Hoang Stephane Ragot Marie Oger Marc Antonini TECH/SSTP AV.Pierre Marzin Orange Laboratories Lannion France Orange Laboratories France Laboratory I3S-UMR 6070 CNRS CNRS and University of Nice-Sophia Antipolis Sophia-Antipolis France

This paper proposes a new bit plane coding method for signed integer sequences. This method consists in mapping successive bit planes onto quinary symbols (+, -, 0, 1, EoP), where the symbol ldquoEoPrdquo stands for ldquoEnd of Planerdquo, and applying arithmetic coding. Sign bits are efficiently coded in combination with the corresponding most significant bit of non-zero integers. Moreover, bit planes are scanned and coded in a non-sequential manner to exploit the correlation between successive planes. Results for conversational transform coding of wideband speech and audio signals - sampled at 16 kHz - show that the performance/complexity of the proposed bitplane coder is near equivalent to non-embedded coding (stack-run coding), while offering additional flexibility (bitstream scalability).

关键词： audio coding Image coding Speech coding Entropy coding Arithmetic Scalability MPEG 4 Standard Transform coding Speech analysis Switches

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Nested microphone array processing for parameter estimation in Directional audio coding

Nested microphone array processing for parameter estimation ...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Giovanni Del Galdo Oliver Thiergart Fabian Kuech Fraunhofer Institute of Integrated Circuits and Systems Erlangen Germany

Directional audio coding (DirAC) is an efficient technique to capture and reproduce spatial sound on the basis of a downmix signal and side information, i. e., direction of arrival and diffuseness of the sound field expressed in time-frequency domain. The main drawback of using arrays of omnidirectional microphones to obtain these parameters is that reliable estimates are available only in a certain frequency range, which depends on the aperture. To overcome this problem and cover large audio bandwidths nested microphone arrays can be used. In this paper, we derive optimal estimators which combine the microphone signals to achieve the best possible estimation of the DirAC parameters with respect to the mean squared error.

关键词： Array signal processing Parameter estimation audio coding Microphone arrays Frequency estimation Bandwidth Direction of arrival estimation Time frequency analysis Loudspeakers Layout

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ITU-T G.719: A new low-complexity full-band (20 kHZ) audio coding standard for high-quality conversational applications

ITU-T G.719: A new low-complexity full-band (20 kHZ) audio c...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Minjie Xie Peter Chu Anisse Taleb Manuel Briand Huawei Technologies Company Limited USA Polycom Inc. Burlington MA USA Ericsson Research-Multimedia Technologies Kista Sweden

This paper describes a new low-complexity full-band (20 kHz) audio coding algorithm which has been recently standardized by ITU-T as Recommendation G.719. The algorithm is designed to provide 20 Hz - 20 kHz audio bandwidth using a 48 kHz sample rate, operating at 32 - 128 kbps. This codec features very high audio quality and low computational complexity and is suitable for use in applications such as videoconferencing, teleconferencing, and streaming audio over the Internet. Subjective test results from the optimization/characterization phase of G.719 are also presented in the paper.

关键词： audio coding Codecs Bandwidth Teleconferencing Signal processing algorithms Computational complexity Testing Quantization Conferences Acoustic signal processing

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Low bitrate audio coding using generalized adaptive gain shape vector quantization across channels

Low bitrate audio coding using generalized adaptive gain sha...

引用

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

作者： Sanjeev Mehrotra Wei-ge Chen Kishore Kotteri Microsoft Research Microsoft Corporation Redmond WA USA

ISBN: (纸本)9781424423538

audio coding at low bitrates suffers from artifacts due to spectrum truncation. Typical audio codecs code multi-channel sources using transforms across the channels to remove redundancy such as middle (mid) - side (M/S) coding. At low bitrates, the spectrum of the coded channels is truncated and the spectrum of the channels with lower energy, such as the side channel, is truncated severely, sometimes entirely. This results in a muffled sound due to truncation of all coded channels beyond a certain frequency. It also results in a loss of spatial image even at low frequencies due to severe truncation of the side channel. Previously we have developed a low bitrate coding method to combat the loss of higher frequencies caused by spectrum truncation. In this paper, we present a novel low bitrate audio coding scheme to mitigate the loss of spatial image. Listening tests show that the combination of the two low bitrate coding methods results in a audio codec that can get good quality even at bitrates as low as 32 kbps for stereo content with low decoder complexity.

关键词： Bit rate audio coding Shape Vector quantization Codecs Discrete transforms Frequency Decoding Redundancy Testing

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Fine-grain scalable audio coding based on envelope restoration and the SPIHT algorithm

Fine-grain scalable audio coding based on envelope restorati...

引用

International Conference on Digital Signal Processing (DSP)

作者： Heiko Hansen Stefan Strahl Alfred Mertins Department of Physics Carl von Ossietzky University슠of슠Oldenburg Oldenburg Germany Institute of Signal Processing University of Lübeck Lubeck Germany

We present strategies for perceptual improvements of embedded audio coding based on psychoacoustic weighting and spectral envelope restoration. The encoding schemes exhibit fine-grain bitrate scalability via the set partitioning in hierarchical trees (SPIHT) algorithm. Weighting factors and envelope parameters are transmitted under careful consideration of the amount of side information. For low bitrates, where the number of actually transmitted waveform coefficients is low, missing coefficients are shaped w.r.t. the spectral envelope. In our approach, the envelope information is transmitted in form of band-wise values of the l 1 -norm. Sets of standardized audio files as well as various audio data of contemporary music are encoded and the results are analyzed with objective measures of perceptual quality. The proposed coding scheme competes in perceptual quality with existing state-of-the-art fixed bitrate coders such as MPEG-2/4 AAC. For low bitrates, the proposed embedded coding envelope restoration (ECER) improves the perceptual audio quality notably.

关键词： audio coding Bit rate Masking threshold Quantization Partitioning algorithms Signal processing algorithms Scalability Psychoacoustic models Psychology Transform coding

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Unified speech and audio coding scheme for high quality at low bitrates

Unified speech and audio coding scheme for high quality at l...

引用

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

作者： M. Neuendorf P. Gournay M. Multrus J. Lecomte B. Bessette R. Geiger S. Bayer G. Fuchs J. Hilpert N. Rettelbach R. Salami G. Schuller R. Lefebvre B. Grill Fraunhofer IIS Erlangen Germany University of Sherbrooke Sherbrooke Canada Voice Age Corporation Montreal Canada Fraunhofer IDMT Ilmenau Germany

ISBN: (纸本)9781424423538

Traditionally, speech coding and audio coding were separate worlds. Based on different technical approaches and different assumptions about the source signal, neither of the two coding schemes could efficiently represent both speech and music at low bitrates. This paper presents a unified speech and audio codec, which efficiently combines techniques from both worlds. This results in a codec that exhibits consistently high quality for speech, music and mixed audio content. The paper gives an overview of the codec architecture and presents results of formal listening tests comparing this new codec with HE-AAC(v2) and AMR-WB+. This new codec forms the basis of the reference model in the ongoing MPEG standardization activity for unified speech and audio coding.

关键词： Speech coding audio coding Bit rate Speech codecs Psychoacoustic models Predictive models Multiple signal classification Testing Frequency domain analysis Quantization

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An embedded speech and audio coding method based on bit-plane coding and SQVH

An embedded speech and audio coding method based on bit-plan...

引用

IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： Mao-shen Jia Chang-chun Bao Xin Liu Speech and Audio Signal Processing Laboratory School of Electronic Information and Control Engineering Beijing University of Technology Beijing China

A multi-layer embedded speech and audio coding algorithm based on bit-plane coding and Scalar Quantized Vector Huffman coding (SQVH) is proposed in this paper. In this codec the signal sampled at 32 kHz can be coded in terms of scalable bit rates. The core codec is International Telecommunication Union Telecommunication Standardization Sector (ITU-T) G.729.1 which can process signal with 7 kHz bandwidth. Besides there are five extra bit-rates added and the bandwidth is extended to 14 kHz. The additional bit-rates include 36, 40, 48, 56, and 64 kb/s. Some new methods used in the additional layers are proposed in this paper. The objective and subjective listening tests show that this codec has good performance compared with reference codec.

关键词： Speech coding audio coding Codecs Speech processing Signal processing Bandwidth Wideband Signal processing algorithms Standardization Mobile communication

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A modified distortion metric for audio coding

A modified distortion metric for audio coding

引用

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

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

ISBN: (纸本)9781424423538

Current audio coding standards employ the modified discrete cosine transform (MDCT) where overlapped frames of audio are windowed and transformed to the frequency domain. Encoding parameters are chosen so as to minimize a distortion measure subject to a rate constraint. At the decoder, inverse transformation involves additional windowing and overlap-add of frames. An analysis of the time domain error in the reconstructed frame reveals that distortion metrics based solely on the MDCT domain error are in fact unable to capture the effects of windowing and overlap-add at the decoder. The main contribution of this paper is a modified distortion metric that does capture these effects via modified discrete sine transform analysis. When incorporated into an Advanced audio Coder the proposed distortion metric significantly improves subjective quality of reconstructed audio.

关键词： audio coding Decoding Discrete transforms Quantization Distortion measurement Frequency domain analysis Discrete Fourier transforms Discrete cosine transforms Time domain analysis Entropy

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Estimating spatial cues for audio coding in MDCT domain

Estimating spatial cues for audio coding in MDCT domain

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： Shuixian Chen Ruimin Hu Shuhua Zhang Computer School Wuhan University of China Wuhan China National Engineering Research Center of Multimedia Software Wuhan University of China Wuhan China Department of Electronic Engineering Tsinghua University Beijing China

Although widely used otherwise, MDCT is excluded in the current scheme for spatial cues representation, due to its lacking of phase information and energy conservation. But combining MDCT with MDST overcomes the difficulties. Moreover, MDST spectra can be built perfectly from neighboring MDCT spectra. The MDCT-MDST conversion, in matrix form, is approximating to a banded sparse matrix. When applied to spatial audio coding using MDCT based core coders, this method avoids separate transforming for cues representation and saves significant computation. Listening tests also show that it has same audio quality as other complex transform based methods.

关键词： audio coding Discrete transforms Power generation economics Delay Testing Energy conservation Sparse matrices Computational complexity Mirrors Filter bank

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