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

作者： Preihs, Stephan Ostermann, Jorn Leibniz Universität Hannover Institut für Informationsverarbeitung Appelstr. 9A Hannover Germany

ISBN: (纸本)9781467369978

This paper presents an extension and global optimization of a bit-allocation strategy for use in a low delay subband ADPCM-based audio coding scheme. The framework employed for the assignment of bits to subband quantizers is based on a subband level estimation of the signals used for the prediction error normalization. It is modified to reduce switching artifacts and extended to contain a mapping to predefined allocation presets. © 2015 IEEE.

关键词： dynamic bitallocation global optimization techniques low delay audio coding subband ADPCM

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ERROR ROBUST lowdelay audio coding USING SPHERICAL LOGARITHMIC QUANTIZATION 24

ERROR ROBUST LOWDELAY AUDIO CODING USING SPHERICAL LOGARITHM...

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24th European Signal Processing Conference (EUSIPCO)

作者： Preihs, Stephan Lamprecht, Timm Ostermann, Joern Leibniz Univ Hannover Inst Informat Verarbeitung Appelstr 9A D-30167 Hannover Germany

ISBN: (纸本)9780992862657

This paper reveals the potential gain in audio quality that can be achieved by combining Spherical Logarithmic Quantization (SLQ) with advanced broadband error robust low delay audio coding based on ADPCM. We briefly summarize the basic properties and mechanisms of SLQ and the employed ADPCM scheme and show how they can be combined in a freely parameterizable coding algorithm. The resulting codec includes techniques for error robustness and a shaping of the coding noise. We present results of optimizing the codec parameters in our framework for global optimization based on psychoacoustic measures. Our evaluation shows that by using SLQ instead of scalar quantization a PEAQ ODG-score improvement with a maximum of about 1 point and a mean of 0.2 can be achieved. An analysis of the bit-error behavior of the combined SLQ-ADPCM shows that a major improvement in bit-error performance results from our proposed efficient error detection and processing.

关键词： low delay audio coding Spherical Logarithmic Quantization ADPCM global optimization PEAQ

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Fast Algorithms for low-delay SBR Filterbanks in MPEG-4 AAC-ELD

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2012年第3期20卷 1022-1031页

作者： Chivukula, Ravi K. Reznik, Yuriy A. Devarajan, Venkat Jayendra-Lakshman, Mythreya Qualcomm Inc San Diego CA 92121 USA InterDigital Inc San Diego CA 92121 USA Univ Texas Arlington Dept Elect Engn Arlington TX 76019 USA

The MPEG committee has recently completed development of a new audio coding standard "MPEG-4 Advanced audio coding-Enhanced low delay" (AAC-ELD). AAC-ELD is targeted towards high-quality, full-duplex communication applications such as audio and video conferencing. AAC-ELD uses low delay spectral band replication (LD-SBR) technology together with a low delay AAC core encoder to achieve high coding efficiency and low algorithmic delays. In this paper, we present fast algorithms for computing LD-SBR filterbanks in AAC-ELD. The proposed algorithms map complex exponential modulation portion of the filterbanks to discrete cosine transforms of types IV and II. Our proposed mapping also allows to merge some multiplications with the windowing stage that precedes or succeeds the modulation step. This further reduces computational complexity. Our presentation includes detailed explanation and flow-graphs of the algorithms, complexity analysis, and comparisons with alternative implementations.

关键词： Advanced audio coding (AAC) discrete cosine transform (DCT) DCT-IV factorization fast algorithms filterbank low delay audio coding Moving Picture Experts Group (MPEG) spectral band replication (SBR)

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Spherical Logarithmic Quantization

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2010年第1期18卷 126-140页

作者： Matschkal, Bernd Huber, Johannes B. Inst Informat Transmiss D-91058 Erlangen Germany

Spherical logarithmic quantization (SLQ) is a vector quantization method for efficiently digitizing analog signals at a high dynamic range and with very low distortion while preserving the original waveform as closely as possible. SLQ is able to operate at a low data rate of, e.g., 2 bits per sample and at a very low signal delay of about ten samples, this corresponds to approximately 200 mu s for high-quality audio signals. The technique of SLQ is universally applicable (i.e., not restricted to, e.g., audio signals) and achieves an efficient digital representation of waveforms with high longterm as well as high segmental signal-to-noise ratios. The aim of this paper is to give a detailed description of the SLQ algorithm and to present simulation results on the performance of this new quantization scheme that combines several advantages. After a review of some important basic principles concerning quantization, linear prediction and multidimensional spheres, the SLQ encoder is described. To short vectors of signal samples which are represented in sphere coordinates, logarithmic quantization is applied to the radius and uniform quantization is applied to the angles. This results in the advantage of a constant signal-to-noise ratio over a very high dynamic range at a small loss with respect to the rate-distortion theory. In order to increase the signal-to-noise ratio by exploitation of correlations within the source signal, a solution for the problem of combining this vector quantization scheme with scalar adaptive differential pulse code modulation (ADPCM), i.e., ADPCM with sample by sample backward recursion is presented. Furthermore, an indexing scheme for the quantization cells covering the surface of a multidimensional unit sphere is presented and simulation results using different source signals are given.

关键词： Dynamic range logarithmic quantization low delay audio coding vector quantization

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AN SBR TOOL FOR VERY low delay APPLICATIONS WITH FLEXIBLE CROSSOVER FREQUENCY

AN SBR TOOL FOR VERY LOW DELAY APPLICATIONS WITH FLEXIBLE CR...

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2010 IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Werner, Michael Schuller, Gerald Ilmenau Univ Technol Helmholtzpl 2 D-98693 Ilmenau Germany Fraunhofer IDMT D-98693 Ilmenau Germany

ISBN: (纸本)9781424442966

In the field of audio coding a technique called Spectral Band Replication (SBR) is used in several codecs to reduce the data rate. We already developed an SBR tool with very little algorithmic delay for use in low delay applications like teleconferencing or live music performance but it produces a relatively high amount of side information. Further it only supports an SBR crossover frequency at half the signal bandwidth. This paper presents an enhanced SBR tool for low delay applications which uses a method known from speech coding called Codebook Mapping (CBM) to get a reduced side information data rate and an optional 3 band decomposition to reach low crossover frequencies. This leads to a low delay SBR tool with a reduced amount of side information and flexible crossover frequencies.

关键词： low delay audio coding bandwidth extension spectral band replication codebook mapping

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AN ERROR ROBUST ULTRA low delay audio CODER USING AN MA PREDICTION MODEL

AN ERROR ROBUST ULTRA LOW DELAY AUDIO CODER USING AN MA PRED...

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Wabnik, Stefan Schuller, Gerald Kraemer, Ferenc Fraunhofer IDMT Inst Digital Media Technol Ehrenbergstr 31 D-98693 Ilmenau Germany Tech Univ Ilmenau Inst Media Technol D- 98693 Ilmenau Germany

ISBN: (纸本)9781424423538

This paper compares two prediction structures for predictive perceptual audio coding in the context of the Ultra low delay (ULD) coding scheme. One structure is based on the commonly used AR signal model, leading to an IIR predictor in the decoder. The other structure is based on an MA signal model, leading to an FIR predictor in the decoder. We find that the AR-based predictor has a slightly better performance in case of an undisturbed transmission channel, but the MA-based predictor has a much better performance in case of transmission errors. For a Bit Error Rate (BER) of 1.0e-5, the perceptual quality of the proposed MA model predictor achieves a mean Objective Difference Grade (ODG) of -0.66 ODG whereas the AR. model predictor only reaches -3.42 ODG.

关键词： low delay audio coding Linear predictive coding Moving average processes Autoregressive processes Robustness

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Adaptive time-frequency resolution in modulated transform at reduced delay

Adaptive time-frequency resolution in modulated transform at...

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

作者： Virette, David Koevesi, Balazs Philippe, Pierrick France Telecom R&D Orange Labs F-22307 Lannion France

ISBN: (纸本)9781424414833

Cosine-modulated transforms such as the Modified Discrete Cosine Transform (MDCT) are key elements in audio coding. They allow efficient energy compaction and perceptual irrelevancy reduction. The frequency localization can be adapted to the signal characteristics and fast implementations exist. All of this has made MIXT the most popular transform in audio coding. In this paper we solve a problem never addressed in the literature: in order to enable variable MDCT sizes in communication codecs, we demonstrate how frequency resolution can be adapted on the fly without using transition windows hence decreasing coding delay. A low complexity implementation of the method is also proposed.

关键词： Cosine transforms transform coding low delay audio coding time-varying filters

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ADAPTIVE PRE- AND POST-FILTERING FOR A SUBBAND ADPCM-BASED low delay audio CODEC

ADAPTIVE PRE- AND POST-FILTERING FOR A SUBBAND ADPCM-BASED L...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics (WASPAA)

作者： Preihs, Stephan Wacker, Christoph Ostermann, Jorn Leibniz Univ Hannover Inst Informationsverarbeitung Appelstr 9A D-30167 Hannover Germany

ISBN: (纸本)9781479974504

This paper addresses the combination of a low delay subband ADPCM-based audio codec with adaptive pre- and post-filtering for psychoacoustic noise shaping. We present how our basic scheme for error robust subband coding can be combined with two cascades of band shelving filters. The gain parameters of these filters are adapted by an algorithm that is based on power estimates which are obtained from the bandpass filtered subband signals of the codec. Since the subband coding as well as the adaptive pre- and post-filtering include several partially interacting parameters that are hard to adjust manually, a framework for their psychoacoustic controlled global optimization is presented. Experiments and their results show that the combination of our low delay subband ADPCM-based audio codec with adaptive pre- and post-filtering allows for a significant improvement of audio quality without any additional signaling overhead. Furthermore, the global optimization enables finding a psychoacoustical meaningful operating point.

关键词： Adaptive pre- and post-filtering low delay audio coding subband ADPCM global parameter optimization

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AN SBR TOOL FOR VERY low delay APPLICATIONS WITH FLEXIBLE CROSSOVER FREQUENCY

AN SBR TOOL FOR VERY LOW DELAY APPLICATIONS WITH FLEXIBLE CR...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Michael Werner Gerald Schuller Ilmenau University of Technology Helmholtzplatz 2 98693 Ilmenau Germany Fraunhofer IDMT Ehrenbergstrasse 29 98693 Ilmenau Germany

ISBN: (纸本)9781424442959

关键词： low delay audio coding bandwidth extension spectral band replication codebook mapping

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ADAPTIVE PRE- AND POST-FILTERING FOR A SUBBAND ADPCM-BASED low delay audio CODEC

ADAPTIVE PRE- AND POST-FILTERING FOR A SUBBAND ADPCM-BASED L...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Stephan Preihs Christoph Wacker Jorn Ostermann Leibniz Universitat Hannover Institut fur Informationsverarbeitung

ISBN: (纸本)9781479974511

This paper addresses the combination of a low delay subband ADPCM-based audio codec with adaptive pre- and post-filtering for psychoacoustic noise shaping. We present how our basic scheme for error robust subband coding can be combined with two cascades of band shelving filters. The gain parameters of these filters are adapted by an algorithm that is based on power estimates which are obtained from the bandpass filtered subband signals of the codec. Since the subband coding as well as the adaptive pre- and post-filtering include several partially interacting parameters that are hard to adjust manually, a framework for their psychoacoustic controlled global optimization is presented. Experiments and their results show that the combination of our low delay subband ADPCM-based audio codec with adaptive pre-and post-filtering allows for a significant improvement of audio quality without any additional signaling overhead. Furthermore, the global optimization enables finding a psychoacoustical meaningful operating point.

关键词： Adaptive pre- and post-filtering low delay audio coding Subband ADPCM Global parameter optimization

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