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

作者： Rong Dong David Hermann Robert Brennan Edward Chau AMI Semiconductors Inc. Waterloo ONT Canada

To help develop ultra-low power wireless hearing aid products, we investigate the integration of subband audio coding with hearing aid applications. Both the audio coding and the hearing aid application use subband processing, but their requirements for the filterbanks are totally different. The hearing aid application typically uses an oversampled filterbank to reduce the aliasing in each subband, whereas the audio codec needs a critically sampled filterbank for maximal coding efficiency. A joint filterbank structure is proposed in this paper to satisfy these contradictive filterbank requirements. With this structure, the two filterbanks are combined into a single stereo filterbank operation, which can be efficiently implemented on a filterbank coprocessor. This structure substantially reduces the computation complexity, power consumption and memory usage.

关键词： Filter bank audio coding Auditory system Digital signal processing Codecs Coprocessors Signal processing algorithms Energy consumption Hearing aids Streaming media

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Using wavpack for real-time audio coding in interactive applications

Using wavpack for real-time audio coding in interactive appl...

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IEEE International Conference on Multimedia and Expo (ICME)

作者： Zefir Kurtisi Lars Wolf Institut für Betriebssysteme und Rechnerverbund Technische Universitäat Braunschweig Germany

Highly interactive real-time music applications like network music performance (NMP) currently lack appropriate audio codecs that meet both demands for low delay and high audio quality. Music audio codecs introduce unacceptable processing latencies, while low-delay speech codecs provide insufficient audio quality. In this article we describe how we integrated the WavPack audio coding scheme into our NMP system for interactive distributed network centric music performances. Our evaluation shows that WavPack performs very well in such types of delay-critical applications. In NMP it provides high compression ratios and good audio quality without additional buffering delays.

关键词： Codecs Delay Music Servers Real time systems audio coding Bit rate

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Model-based deadzone optimization for stack-run audio coding with uniform scalar quantization

Model-based deadzone optimization for stack-run audio coding...

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

作者： Marie Oger Stephane Ragot Marc Antonini France Télécom Research and Development TECH/SSTP/RVA Lannion France France Télécom R&D/TECH/SSTP Lannion Cedex Lab. I3S-UMR 6070 CNRS and Univ. of Nice Sophia Antipolis Sophia Antipolis France

In this paper we present a model-based method to optimize the dead- zone of uniform scalar quantization in transform audio coding. The input signal is coded in modified discrete cosine transform (MDCT) domain by uniform scalar quantization followed by context-based arithmetic coding. The optimal deadzone is derived using a non- asymptotic method, assuming that the distribution of MDCT coefficients is approximated by a generalized Gaussian model. We show that deadzone optimization improves slightly quality, especially at low bitrates.

关键词： audio coding Quantization Bit rate Shape Laplace equations Random variables Encoding Mean square error methods Lagrangian functions Research and development

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The Research and Implement of Mean-Quantization-Based Fragile Watermarking in AVS-S audio coding

The Research and Implement of Mean-Quantization-Based Fragil...

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Congress on Image and Signal Processing, CISP

作者： Bo Hang RuiMin Hu Wuhan University China

The AVS-S standard for security industry requires its audio information must have tamper-proof capability. We researched the fragile watermarking based on mean quantization principle and implement it in AVS-S audio coding. Firstly, a binary image was embedded into audio as the watermarking. The audio with watermarking should have no significant quality decrease. Then attack the audio sequence with a variety of methods. We calculate the watermarking similarity between the retrieved and the original to discover attack actions. The test showed this algorithm can achieve the desired results.

关键词： Watermarking audio coding Discrete cosine transforms Equations Code standards Data security Information security Quantization Standards development Video codecs

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Waveform approximating residual audio coding with perceptual pre- and post-filtering

Waveform approximating residual audio coding with perceptual...

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Asilomar Conference on Signals, Systems & Computers

作者： Jesper Kjoer Nielsen Jesper Rindom Jensen Mads Groesboll Christensen Soren Holdt Jensen Torben Larsen Department of Electronic Systems University of Aalborg Aalborg Denmark

We investigate waveform approximating residual coding for a sinusoidal parametric audio coder at low bit rates. The residual coding is based on the well-known pre- and post-filtering method with lossless coding which features perceptual weighting for short time segments. We compare the incurred perceptual distortion from joint quantization of the residual and the sinusoids for different bit rates. In addition to that, we develop a transform coding scheme for the coefficients in the pre- and post-filters which must be send as side information between the encoder and decoder. Our investigations show that the combination of the sinusoidal subcoder and the pre- and post-filtering entails an overall lower perceptual distortion for low as well as high bit rates. Also, the developed transform coding scheme enables efficient coding of the side information at a low bit rate.

关键词： audio coding Bit rate Transform coding Psychoacoustic models Decoding Signal processing Quantization MONOS devices Pulse modulation Councils

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Efficient implementation of a class of MDCT/IMDCT filterbanks for speech and audio coding applications

Efficient implementation of a class of MDCT/IMDCT filterbank...

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

作者： Ravi K. Chivukula Yuriy A. Reznik Electrical Engineering Department University of Texas Arlington Arlington TX USA QUALCOMM Inc.orporated San Diego CA USA

In this paper, an efficient algorithm for implementing MDCT/IMDCT of lengths N = 5 ldr 2 m (m ges 2) is presented. Transforms for such lengths are of interest for speech and audio coding applications, such as recently issued and/or emerging standards G.729.1, ***-VBR, and EVRC-WB. In our design we utilize a mapping of MDCT of size N into JV/2-point DCT-IV and DCT-II with isolated pre- multiplications, which are subsequently moved in the windowing stage. We show that such a modified window is piece-wise symmetric, and can be stored using N/2 words. In our algorithm we also use an efficient factorization of 5- point DCT-II which requires only 4 multiplications by irrational factors. We compare our proposed algorithm with several alternative implementations and show that our design offers practically appreciable reduction in complexity and memory usage.

关键词： Speech coding audio coding Algorithm design and analysis Discrete cosine transforms Speech synthesis Speech analysis Strontium Speech codecs Sampling methods Delay

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A fully scalable audio coding structure with embedded psychoacoustic model

A fully scalable audio coding structure with embedded psycho...

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

作者： Te Li Susanto Rahardja Soo Ngee Koh Institute for Infocomm Research Singapore Nanyang Technological University Singapore

A fully scalable audio coding structure based on a novel combination of the non-core MPEG-4 scalable lossless audio coding (SLS), the state-of-the-art psychoacoustic model, joint stereo coding and the perceptually prioritized bit-plane coding is presented in this paper. The psychoacoustic information is implicitly embedded in the scalable bitstream with negligible amount of side information and trivial modification to the standardized SLS decoder. Results of extensive evaluation show that the subjective quality of scalable audio is improved significantly.

关键词： audio coding Psychoacoustic models Laser sintering Bit rate Decoding MPEG 4 Standard Psychology Scalability Phase change materials Tellurium

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Directional audio coding Using Planar Microphone Arrays

Directional Audio Coding Using Planar Microphone Arrays

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Hands-Free Speech Communication and Microphone Arrays, HSCMA

作者： Fabian Kuech Markus Kallinger Richard Schultz-Amling Giovanni del Galdo Jukka Ahonen Ville Pulkki Fraunhofer Institute of Integrated Circuits and Systems Erlangen Germany Laboratory of Acoustics and Audio Signal Processing Helsinki Finland

Multichannel sound systems become more and more established in modern audio applications. Consequently, the recording and the reproduction of spatial audio gains increasing attention. Directional audio coding (DirAC) represents an efficient approach to analyze spatial sound and to reproduce it using arbitrary loudspeaker configurations. In DirAC, the direction-of-arrival and the diffuseness of sound within frequency subbands is used to encode the spatial properties of the observed sound field. The estimation of these parameters is based on an energetic sound field analysis using three- dimensional microphone arrays. In practice, however, physical design constraints make three-dimensional microphone configurations often not acceptable. In this paper, we consider a new approach to microphone array processing that allows for an estimation of both direction-of-arrival of sound and diffuseness based on planar microphone configurations. The performance of the proposed method is evaluated via simulations and real measured data.

关键词： Microphone arrays audio coding Loudspeakers audio recording Frequency Parameter estimation Laboratories Acoustics audio systems Array signal processing

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2011年第1期19卷 24-33页

作者： den Brinker, Albertus C. Krishnamoorthi, Harish Verbitskiy, Evgeny A. Philips Res NL-5656 AE Eindhoven Netherlands Eindhoven Univ Technol Signal Proc Grp NL-5612 AZ Eindhoven Netherlands

Linear prediction has been successfully applied in many speech and audio processing systems. This paper presents the similarities and differences between two classes of linear prediction schemes, namely, Warped Linear Prediction (WLP) and Laguerre Linear Prediction (LLP). It is shown that both systems are closely related. In particular, we show that the LLP is in fact a WLP system where the optimization procedure is adapted such that the whitening property is automatically incorporated. The adaptation consists of a new linear constraint on the parameters. Furthermore, we show that an optimized WLP scheme where whitening is achieved by prefiltering before estimating the optimal coefficients results in a filter having all except the last reflection coefficient equal to those of the optimal LLP filter.

关键词： audio coding frequency warping linear prediction speech coding

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Auditory-inspired sparse representation of audio signals

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SPEECH COMMUNICATION 2011年第5期53卷 643-657页

作者： Pichevar, Ramin Najaf-Zadeh, Hossein Thibault, Louis Landili, Hassan Commun Res Ctr Ottawa ON K2H 8S2 Canada

This article deals with the generation of auditory-inspired spectro-temporal features aimed at audio coding. To do so, we first generate sparse audio representations we call spikegrams, using projections on gammatone/gammachirp kernels that generate neural spikes. Unlike Fourier-based representations, these representations are powerful at identifying auditory events, such as onsets, offsets, transients, and harmonic structures. We show that the introduction of adaptiveness in the selection of gammachirp kernels enhances the compression rate compared to the case where the kernels are non-adaptive. We also integrate a masking model that helps reduce bitrate without loss of perceptible audio quality. We finally propose a method to extract frequent audio objects (patterns) in the aforementioned sparse representations. The extracted frequency-domain patterns (audio objects) help us address spikes (audio events) collectively rather than individually. When audio compression is needed, the different patterns are stored in a small codebook that can be used to efficiently encode audio materials in a lossless way. The approach is applied to different audio signals and results are discussed and compared. This work is a first step towards the design of a high-quality auditory-inspired "object-based" audio coder. Crown Copyright (C) 2010 Published by Elsevier B.V. All rights reserved.

关键词： Sparse representations Masking Quantization Temporal data mining Episode discovery audio coding Matching pursuit Auditory pattern recognition

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