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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Francisco Pinto Martin Vetterli EPFL-IC-LCAV Ecole Polytechnique Fédérale de Lausanne Lausanne Switzerland

We address the problem of integrating directional analysis of sound into the filterbank of a spatial audio coder, with the purpose of processing and coding with some degree of independence the plane waves traveling in different directions. A plane wave represents an elementary waveform in the spatio-temporal analysis of the sound field, the same way a complex exponential is an elementary waveform in the time domain analysis of signals. Since a two-dimensional separable filterbank is not flexible enough for this purpose, we propose a non-separable approach based on the quincunx filterbank with diamond-shaped filters, cascaded with a base transform filterbank. This solution provides an invertible and critically sampled decomposition of the spatio-temporal spectra into subbands representing the different directions of wave propagation.

关键词： Signal analysis audio coding Signal processing Conferences Acoustic signal processing Acoustic applications Spatiotemporal phenomena Acoustic propagation Frequency Multidimensional signal processing

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Rate-distortion optimized quantization in multistage audio coding

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2006年第1期14卷 311-320页

作者： Vafin, R Kleijn, WB Royal Inst Technol KTH Dept Signals Sensors & Syst Tallinn Estonia Royal Inst Technol Dept Signals Sensors & Syst S-10044 Stockholm Sweden

In this work, we develop a new method for quantization in multistage audio coding. Given a (perceptual) distortion measure and a bit-rate constraint, we analytically derive the optimal rate distribution between subcoders (stages) and the corresponding optimal quantizers using high-rate theory. The analytical solutions for optimal quantizers allow a coder to easily adapt to changes in bit-rate requirements. As an illustration of the new method, we consider quantization in a two-stage sinusoidal/wave form coder that is a widely used combination in audio coding. We show that at low total rates most of the rate should be assigned to the sinusoidal (model-based, subspace) subcoder, while at high total rates most of the rate should be assigned to the waveform (full-space) subcoder. We compare the new method to a reference quantization method that does not use rate-distortion optimization. A significantly higher performance of the new method is shown by means of a listening test.

关键词： audio coding high-rate theory modified discrete cosine transform (MDCT) multistage coding quantization rate-distortion optimization sinusoidal coding waveform coding

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SCALABLE SUPERWIDEBAND EXTENSION FOR WIDEBAND coding

SCALABLE SUPERWIDEBAND EXTENSION FOR WIDEBAND CODING

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Tammi, Mikko Laaksonen, Lasse Ramo, Anssi Toukomaa, Henri Nokia Res Ctr Tampere Finland

ISBN: (纸本)9781424423538

Recent trends in speech and audio codec standardization include scalability and extending the signal bandwidth beyond wideband (WB) to superwideband (SWB). In this paper we introduce a SWB extension for the ITU-T G.718 WB codec. In the SWB extension the high frequency content is generated utilizing the quantized MDCT domain coefficients of the WB core, which enables low additional delay. The proposed implementation is scalable with 4 kbps layers. In the first layer two different coding modes are used depending on the input signal type. The proposed SWB extension is evaluated with listening tests and complexity analysis.

关键词： audio coding superwideband extension scalability

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Fast Algorithm for Modulated Complex Lapped Transform

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IEEE SIGNAL PROCESSING LETTERS 2009年第1-3期16卷 30-33页

作者： Dai, Xingdong Wagh, Meghanad D. LSI Corp Allentown PA 18109 USA Lehigh Univ Dept Elect & Comp Engn Bethlehem PA 18015 USA

A new algorithm for the modulated complex lapped transform (MCLT) with a sine windowing function is presented. It is shown that by merging the windowing operation with the main computation, both the real and the imaginary parts of the MCLT with 2N inputs can be obtained from two N-point discrete cosine transforms of type II (DCTs-II) of appropriate inputs. The resulting algorithm is computationally very efficient. In general, the value of N is an even number. When N is a power of 2, the proposed algorithm uses only N log N + 2 real multiplications (including the scaling factors in the DCT computation), with none of those being outside the DCT blocks.

关键词： audio coding fast algorithm modified discrete cosine transform modified discrete sine transform modulated complex lapped transform

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Development of a re-configurable ambisonic decoder for irregular loudspeaker configuration

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IET CIRCUITS DEVICES & SYSTEMS 2009年第4期3卷 197-203页

作者： Tsang, P. W. M. Cheung, K. W. K. City Univ Hong Kong Dept Elect Engn Hong Kong Hong Kong Peoples R China

This study reports a heuristic genetic algorithm to determine the decoding parameters in a first-order ambisonic system for reconstructing a three-dimensional sound field with an arbitrary quad speaker configuration. On this basis, a hardware prototype has been developed using a field programmable gate array (FPGA) to decode ambisonic signals that are encoded in the standard B-format. To allow direct coupling with digital audio sources, the input and output channels of the decoder are implemented with the 12S interface. Evaluations reveal that the decoding parameters derived by this method are superior to existing approaches in terms of flexibility in loudspeaker configuration and optimisation of some of the essential factors in surround sound reconstruction.

关键词： first-order ambisonic system loudspeakers heuristic programming heuristic genetic algorithm standard B-format acoustic signal processing irregular loudspeaker configuration decoding arbitrary quad speaker configuration field programmable gate array digital audio sources reconfigurable architectures re-configurable ambisonic decoder encoding decoding parameters field programmable gate arrays signal reconstruction three-dimensional sound field genetic algorithms direct coupling hardware prototype acoustic field audio coding surround sound reconstruction

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A Frequency/Detector Pruning Approach for Loudness Estimation

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IEEE SIGNAL PROCESSING LETTERS 2009年第11期16卷 997-1000页

作者： Krishnamoorthi, Harish Spanias, Andreas Berisha, Visar Arizona State Univ Dept Elect Engn Tempe AZ 85287 USA

In this letter, we propose a frequency and detector pruning approach for reducing the computational complexity associated with loudness estimation. The frequency pruning approach exploits the principles of psychoacoustics such that the total neural activity is preserved. The detector pruning approach evaluates the excitation/loudness patterns at nonuniform sample locations and employs signal interpolation techniques to obtain their corresponding high resolution estimates. Comparative results with the Moore and Glasberg loudness estimation process reveal that the proposed pruning approach for loudness estimation performs consistently well for different types of audio signals with a significant reduction in the computational complexity.

关键词： audio coding loudness psychoacoustics speech processing

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A Novel Low Bit Rate audio Bandwidth Extension Method

A Novel Low Bit Rate Audio Bandwidth Extension Method

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2nd International Symposium on Knowledge Acquisition and Modeling

作者： Hang, Bo Hu, Ruimin Ma, Ye Xiangfan Univ Math & Comp Sci Coll Xiangfan 441053 Peoples R China Wuhan Univ Natl Engn Res Ctr Multimedia Software Wuhan 430072 Peoples R China

ISBN: (纸本)9780769538884

In present communication system, high quality audio signal is supposed to be provided with low bit rate and low computational complexity. This paper proposed a novel audio coding bandwidth extension method, which can improve decoded audio quality with increasing only a few coding bits per frame and a little computational complexity. This method calculate high-frequency synthesis filter by using codebook mapping method, and transmit only quantified gain corrections in high-frequency part of multiplexing coding bit stream. The preliminary test show that this method can provide comparable audio quality with lower bit consumption and computational complexity compared to the high frequency regeneration of AMR-WB+.

关键词： bandwidth extension audio coding codebook mapping

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ENcoding THE SINUSOIDAL MODEL OF AN audio SIGNAL USING COMPRESSED SENSING

ENCODING THE SINUSOIDAL MODEL OF AN AUDIO SIGNAL USING COMPR...

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IEEE International Conference on Multimedia and Expo

作者： Griffin, Anthony Hirvonen, Toni Mouchtaris, Athanasios Tsakalides, Panagiotis Univ Crete Inst Comp Sci Fdn Res & Technol Hellas FORTH ICS Iraklion Crete Greece Univ Crete Dept Comp Sci Iraklion Crete Greece

ISBN: (纸本)9781424442904

In this paper, the compressed sensing (CS) methodology is applied to the harmonic part of sinusoidally-modeled audio signals. As this part of the model is sparse by definition in the frequency domain, we investigate how CS can be used to encode this signal at low bitrates, instead of encoding the sinusoidal parameters (amplitude, frequency, phase) as current state-of-the-art methods do. We extend our previous work by considering an improved system model, by comparing our model to other schemes, and exploring the effect of incorrectly reconstructed frames. We show that encouraging results can be obtained by our approach, although inferior at this point compared to state-of-the-art. Good performance is obtained using 24 bits per sinusoid as indicated by our listening tests.

关键词： audio coding compressed sensing sinusoidal model signal reconstruction signal sampling

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RATE DISTRIBUTION BETWEEN MODEL AND SIGNAL FOR MULTIPLE DESCRIPTIONS

RATE DISTRIBUTION BETWEEN MODEL AND SIGNAL FOR MULTIPLE DESC...

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Klejsa, Janusz Kleijn, W. Bastiaan Royal Inst Technol ACCESS Linnaeus Ctr S-10044 Stockholm Sweden

ISBN: (纸本)9781424423538

We consider the rate allocation problem for multiple-description quantization of the signal described by an adaptive model with a fixed structure. The source modeling in coding generally results in a two-stage description of the data, where one of the stages describes the model parameters, and the other describes the signal. Such a setup implies the existence of a trade-off between the rate spent on the parameters and the rate spent on the signal. We optimize this trade-off analytically for the multiple-description case using a method inspired by Minimum Description Length principle. We also provide an algorithm for optimizing the rate allocation between the components of the model-based multiple description coder. Finally we experimentally confirm our results. Our method facilitates the rate-adaptive multiple-description coding.

关键词： source modeling multiple description coding (MDC) audio coding

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BANDWIDTH EXTENSION FOR CHINA AVS-M STANDARD

BANDWIDTH EXTENSION FOR CHINA AVS-M STANDARD

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Zhan, Jie Choo, Kihyun Oh, Eunmi Samsung Elect Co Ltd Yongin Gyeonggi Do South Korea

ISBN: (纸本)9781424423538

We proposed a new frequency domain BandWidth Extension (BWE) technology. In the new technology, FFT based frequency domain gain shaping combined with Linear Prediction coding (LPC) based spectral envelope shaping is used for generating high frequency signals. To preserve the amount of noise component in the reconstructed band, gain reduction controlled by Spectrum Flatness Measurement (SFM) is employed. Subjective testing results show that the presented technology exhibits a comparable performance compared to 3GPP AMR-WB+ with the same bit-rate in the framework of audio Video coding of China Standard (AVS) Part 10 - Mobile Speech and audio Codec. This technology has been formally adopted as the artificial high band coding module in AVS P10.

关键词： audio coding LPC Speech coding Standardization

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