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24th International Congress on Acoustics, ICA 2022

作者： Jo, Byeongho Beack, Seungkwon Lee, Taejin Electronics and Telecommunications Research Institute Korea Republic of

In audio coding technology, the framework based on the modified discrete cosine transform (MDCT) has been mainly adopted in the literature and audio coding standards. The MDCT does not introduce blocking effects while allowing perfect reconstruction, and can evade the data rate increase with a 50% overlapped window. However, when using the temporal noise shaping for coding the transient signal, the undesired aliasing may occur, and the MDCT window should be designed under the time-domain aliasing cancellation constraint. These limitations can be simply avoided by using the discrete Fourier transform (DFT), but it increases the data rate by double because of the complex-valued representation of the DFT coefficients. In this study, we propose a scheme that effectively quantizes the complex-valued signal in the comparable data rate with the real-valued coding scheme. The proposed coding scheme is based on unrestricted polar quantization for complex variables combined with the psychoacoustic model. This study demonstrates the feasibility of the quantization scheme for the complex-valued coefficients of the same data rate with comparable perceptual distortion against the conventional MDCT-based audio coding. © ICA *** rights reserved

关键词： audio coding complex quantization discrete Fourier transform polar quantization psychoacoustics

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Modulation frequency and efficient audio coding

Modulation frequency and efficient audio coding

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Conference on Advanced Signal Processing Algorithms, Architectures, and Implementations XI

作者： Atlas, LE Vinton, MS Univ Washington Dept Elect Engn Seattle WA 98195 USA

ISBN: (纸本)0819441880

The concept of "modulation frequency" is shown to be a valuable insight into time-frequency transforms for audio coding. A two-dimensional transform, where the second dimension approximately decomposes the audio signal into modulation frequencies, is proposed. This transform, when applied to audio coding, provides high quality at low data rates and adapt gracefully to changes in available bandwidth. It is inherently scalable, meaning that channel conditions can be matched without the need for additional computation. Moreover, it is compact: in subjective tests our algorithm, coded at 32kilobits/seconds/channel, outperformed MPEG-l Layer 3 (MP3) coded at 56 kilobits/seconds/channel (both at 44.1 kHz). This potentially useful result motivates the need for further insight into the definition and analysis of modulation frequency. We thus define modulation frequency for a simple narrowband signal, propose a general bilinear framework for detection, and then propose a minimal set of conditions to extend this definition to broadband signals such as audio.

关键词： time-frequency analysis modulation frequency audio coding audio compression scalable coding

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audio coding via EMD

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DIGITAL SIGNAL PROCESSING 2020年 104卷 102770-102770页

作者： Boudraa, Abdel-Ouahab Khaldi, Kais Chonavel, Thierry Hadj-Alouane, Mounia Turki Komaty, Ali Ecole Navale Arts & Metiers Inst Technol IRENav CC 600 F-29240 Brest 9 France Jouf Univ Coll Sci & Arts Tabarjal POB 2014 Al Jouf 42421 Sakaka Saudi Arabia IMT Atlantique LabSTICC UMR 6285 BP 832 F-29285 Brest France El Manar Univ ENIT U2S BP 37 Tunis 1002 Tunisia Univ Sci & Arts USAL Lebanon Beirut Lebanon

In this paper an audio coding scheme based on the empirical mode decomposition in association with a psychoacoustic model is presented. The principle of the method consists in breaking down adaptively the audio signal into intrinsic oscillatory components, called Intrinsic Mode Functions (IMFs), that are fully described by their local extrema. These extrema are encoded. The coding is carried out frame by frame and no assumption is made upon the signal to be coded. The number of allocated bits varies from mode to mode and obeys to the coding error inaudibility constraint. Due to the symmetry of an IMF, only the extrema (maxima or minima) of one of its interpolating envelopes are perceptually coded. In addition, to deal with rapidly changing audio signals, a stationarity index is used and when a transient is detected, the frame is split into two overlapping sub-frames. At the decoder side, the IMFs are recovered using the associated coded maxima, and the original signal is reconstructed by IMFs summation. Performance of the proposed coding is analyzed and compared to that of MP3 and AAC codecs, and the wavelet-based coding approach. Based on the analyzed mono audio signals, the obtained results show that the proposed coding scheme outperforms the MP3 and the wavelet-based coding methods and performs slightly better than the AAC codec, showing thus the potential of the EMD for data-driven audio coding. (C) 2020 Elsevier Inc. All rights reserved.

关键词： Empirical mode decomposition Empirical mode compression audio coding Sub-band coding Stationarity index Psychoacoustic model

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A new error-mapping scheme for scalable audio coding 16

A new error-mapping scheme for scalable audio coding

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2014 16th IEEE International Workshop on Multimedia Signal Processing, MMSP 2014

作者： Huang, Haibin Rahardja, Susanto Institute for Infocomm Research 1 Fusionopolis Way Singapore Singapore Department of Electrical and Computer Engineering National University of Singapore 21 Lower Kent Ridge Road Singapore Singapore

ISBN: (纸本)9781479958962

In scalable audio coders, such as the MPEG-4 SLS, error-mapping is used to map quantization errors in the core coder to an error signal before passing through bit-plane coding. In this paper, we propose a new error-mapping scheme that is derived by observing statistical properties of the error signal. Compared with the error-mapping in SLS, the proposed scheme improves coding efficiency as well as computational complexity of the coder. An average improvement of 9 points in MUSHRA score has been achieved by the proposed scheme in subjective listening tests. The proposed error-mapping adds a useful new tool to the existing toolset for constructing next-generation scalable audio coders.

关键词： audio coding codecs computational complexity quantisation (signal) MPEG-4 SLS MUSHRA audio coders bit-plane coding error-mapping scheme quantization errors scalable audio coding audio coding Conferences Decoding Histograms Quantization (signal) Transform coding audio coding bit-plane coding complexity classes Quantization transform coding codecs histograms DEcoding ERROR SIGNAL ENCODER

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MDCT audio coding with pulse vector quantizers 40

MDCT audio coding with pulse vector quantizers

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40th IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2015

作者： Svedberg, Jonas Grancharov, Volodya Sverrisson, Sigurdur Norvell, Erik Toftgård, Tomas Pobloth, Harald Bruhn, Stefan SMN Ericsson Research Ericsson AB Stockholm Sweden

ISBN: (纸本)9781467369978

This paper describes a novel audio coding algorithm that is a building block in the recently standardized 3GPP EVS codec [1]. The presented scheme operates in the Modified Discrete Cosine Transform (MDCT) domain and deploys a Split-PVQ pulse coding quantizer, a noise-fill, and a gain control optimized for the quantizer's properties. A complexity analysis in terms of WMOPS is presented to illustrate that the proposed Split-PVQ concept and dynamic range optimized MPVQ-indexing are suitable for real-time audio coding. Test results from formal MOS subjective evaluations and objective performance figures are presented to illustrate the competitiveness of the proposed algorithm. © 2015 IEEE.

关键词： audio coding EVS MDCT Noise-Fill PVQ

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audio coding for TV and multimedia

Audio coding for TV and multimedia

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IBC 95 International Broadcasting Convention

作者： K. Brandenburg FhG-IIS Germany

Since 1988 the audio subgroup is ISO/IEC JTC1/SC29 WG11 (called MPEG) has been working on the standardization of high quality low bit-rate audio coding. Until now the standards 11172-3 (1) (MPEG-1 audio) and 13818-3 (2) (MPEG-2 audio) have been finished. The main focus of this paper is to give an overview on the ongoing work towards the NBC (Non Backwards-Compatible) extension to MPEG-2 audio and MPEG-4 audio coding.< >

关键词： audio coding Communication standards Codes Multimedia systems Radio broadcasting TV broadcasting

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audio coding and transmission for aeronautical broadcasts via satellite

Audio coding and transmission for aeronautical broadcasts vi...

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Proceedings of GLOBECOM '93. IEEE Global Telecommunications Conference

作者： F.F. Tzeng B.R.U. Bhaskar COMSAT Laboratories Clarksburg MD USA

A system design for aeronautical audio broadcasts, with a C-band uplink and an L-band downlink, via Inmarsat space segments is presented. Near-transparent-quality compression of 5-kHz bandwidth audio at 17 kbit/s is achieved based on a hybrid technique that employs linear predictive modeling and transform-domain residual quantization. High-resolution spectral modeling, block-size adaptation, and adaptive bit allocation are key components of the design. Concatenated Reed-Solomon/convolutional codes with quadrature phase shift keying are selected for bandwidth and power efficiency. An RF bandwidth of 20 kHz per channel, and a decoded bit error rate of 10/sup -6/ with an E/sub b//N/sub 0/ of 4.5 dB are obtained. Frequency-division multiple access was selected over code-division multiple access. A link budget computation based on a worst-case scenario indicates sufficient system power margins. The transponder occupancy analysis for 72 audio channels reveals ample remaining capacity for emerging aeronautical services.< >

关键词： audio coding Broadcasting Bandwidth Power system modeling L-band Downlink Predictive models Quantization Adaptation model Bit rate

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audio coding using variable-depth multistage quantizers

Audio coding using variable-depth multistage quantizers

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Data Compression Conference (DCC)

作者： F. Kossentini M.W. Macon M.J.T. Smith Department of EE University of British Columbia Vancouver BC Canada School of ECE Georgia Tech Atlanta GA USA

Digital coding of high-fidelity audio signals has become a key technology in the development of cost-effective multimedia systems. Most audio coding algorithms rely on (i) removal of statistical redundancies in the signal and (ii) exploitation of masking properties of the human auditory system to hide distortions in the coded signal. Transform and subband coders provide a convenient framework for time-frequency domain signal analysis and coding based on these two principles. We present a subband coding algorithm for high-quality coding of 48 kHz sampled audio signals that is based on perceptual transformation, variable-depth multistage uniform quantization, and complexity-constrained higher-order entropy coding. The construction does not require any training and avoids completely the need to send side information.

关键词： audio coding Quantization Multi-stage noise shaping Humans Multimedia systems Auditory system Distortion Time frequency analysis Signal analysis Entropy coding

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audio coding with signal adaptive filterbanks

Audio coding with signal adaptive filterbanks

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

作者： J. Princen J.D. Johnston AT and T Bell Laboratories Inc. Murray Hill NJ USA

In this paper we present a high quality audio coding system based on a novel nonuniform modulated filterbank coupled with time-varying cosine modulated filterbanks in a cascade architecture. The system makes use of psychoacoustic thresholds in a natural way to adapt the resolution of the filterbank to achieve high coding gain on a wide range of signal types. Results show that the system provides excellent quality at 64 kb/s and good quality at 48 kb/s for monophonic coding.

关键词： audio coding Adaptive filters Filter bank Signal resolution Frequency Psychoacoustic models Psychology Quantization Noise level Modulation coding

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audio coding using the best level wavelet packet transform and auditory masking

Audio coding using the best level wavelet packet transform a...

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International Conference on Signal Processing Proceedings (ICSP)

作者： S. Hosoi J. Toyama M. Shimbo Division of Systems & Information Engineering Graduate School of Engineering Hokkaido University Sapporo Japan

In this paper, we propose an audio coder that aims at high quality even in nonstationary segments of music signals. First, we divide input signals into subbands by using wavelet packet decomposition. We use a critical band approximate wavelet packet decomposition tree for efficient auditory masking. Dynamic wavelet packet decomposition is then used for effective signal representation. By choosing the basis from a wavelet packet tree, the best level method is used. The bases of the best level method has more entropy than that of the best basis method, but by choosing the appropriate frame length, it has similar entropy to that of the bases of the best basis method. Thus we can reduce side information. We propose an adaptive frame method whose criteria is the entropy of best level basis.

关键词： audio coding Wavelet packets Wavelet transforms Entropy Bit rate Quantization Frequency Decoding Signal resolution Tree data structures

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