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A new bandwidth scalable wideband speech/audio coder

A new bandwidth scalable wideband speech/audio coder

引用

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

作者： Kyung Tae Kim Sung Kyo Jung Young Cheol Park Dae Hee Youn MCSP Laboratory Department of Electric & Electronic Engineering Yonsei University Seoul South Korea

ISBN: (纸本)0780374029

In this paper, we present a new bandwidth-scalable coder for wide band speech and audio signals. The proposed coder splits 8 kHz signal bandwidth into two narrow bands, and different coding schemes are applied to each band. The lower-band speech is coded with ITU-T G.729 Annex E, and the higher-band signal is compressed using a new algorithm based on the gammatone filter band with an invertible auditory model. Due to the split-band architecture and completely independent coding schemes for each band, the output speech of the decoder can be selected to be a narrowband or wideband coding according to the channel conditions. Subjective tests showed that, for wideband speech and audio signals, the proposed coder at 18 kbit/s produces superior quality to ITU-T 24 kbit/s G.722.1 with the shorter algorithmic delay.

关键词： Speech Narrowband Delay Speech coding Switches Decoding audio coding

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Observations on auditory excitation and masking patterns

Observations on auditory excitation and masking patterns

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： S. Voran U.S. Department of Commerce Institute of Telecommunication Sciences Boulder CO USA

Excitation patterns and masking patterns are used extensively in perceptual audio coders and quality assessment algorithms. Numerous algorithms for calculating these patterns have been proposed. This paper provides comparisons among the patterns generated by several of these algorithms. The comparisons are based on audio program material, rather than tones and noise. Explored areas include synthesis functions, spreading functions, masking indices, tonality measures, and the treatment of the absolute threshold of hearing. Several mathematical relations are provided to characterize observations in these areas. Patterns from simpler algorithms are considered as approximations to patterns from more complex algorithms, and the approximation error is characterized. Results may be useful to those who apply auditory excitation or masking patterns in their work.

关键词： Approximation algorithms Signal synthesis Frequency Approximation error Auditory system US Department of Commerce Tires audio coding Quality assessment Noise measurement

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Frequency selectivity via the SpEnt methodology for wideband speech compression

Frequency selectivity via the SpEnt methodology for wideband...

引用

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

作者： M.G. Kokes J.D. Gibson Multimedia Communications and Networking Laboratory Department of Electrical engineering Southern Methodist University Dallas TX USA

In speech and audio coding, frequency selectivity of the basis functions is an important property of the codec. The more precise the frequency selectivity, the less chance there is for audible coding effects due to uncanceled aliasing. We use Campbell's (1960) coefficient rate and the spectral entropy (SpEnt) of the source random process as a guide to formulate adaptive nonuniform modulated lapped biorthogonal transforms (NMLBT). The use of the NMLBT allows for efficient implementation of a time-varying transform which possesses both good frequency and time resolution at all instances, without the need for transitional filters. By coupling the SpEnt methodology with modulated lapped biorthogonal transforms (MLBT), we develop band combining strategies to produce an adaptive NMLBT. Due to the nature of the SpEnt methodology, the new frequency selection process comprises a non-linear approximation method to determine the best n basis functions to represent the current speech frame. We implement a wideband speech compression scheme based on this strategy and verify its improved performance in coding speech and audio signals at 16 and 24 kbps.

关键词： Frequency Wideband Speech coding Speech codecs audio coding Entropy Random processes Filters Approximation methods Speech processing

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Sound Field Model Violations in Parametric Spatial Sound Processing

Sound Field Model Violations in Parametric Spatial Sound Pro...

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IWAENC 2012;International Workshop on Acoustic Signal Enhancement

作者： Oliver Thiergart Emanuel A. P. Habets International Audio Laboratories Erlangen

Describing a sound field with a single audio signal and parametric side information offers an efficient way of storing, transmitting, or reproducing the recorded spatial sound. In many approaches, the parametric representation is based on a simple sound field model assuming for each time-frequency instance a single plane wave and diffuse sound. When multiple sources are active, we thus have to assume that the source signals do not overlap much in the time-frequency domain so that the single-wave model is still satisfied. In this paper, we show that a single-wave model is easily violated even for sparse signals such as speech. When the parametrized spatial sound is reproduced, these violations degrade the spatial quality. We further study whether the well-known measure of approximate W-disjoint orthogonality can be applied to detect the relevant model violations.

关键词： Speech Time frequency analysis Microphones audio coding Loudspeakers Indexes Acoustics

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Broadband direction estimation method utilizing combined pressure and energy gradients from optimized microphone array

Broadband direction estimation method utilizing combined pre...

引用

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

作者： Jukka Ahonen Ville Pulkki Department of Signal Processing and Acoustics Aalto University of Science and Technology Espoo Finland

This paper proposes a method for analyzing the direction of the arrival of sound by estimating the sound intensity vector from the pressure and energy gradients of closely-spaced omnidirectional microphones depending on frequency. Microphones with relatively large housing, which cause shadowing, are used here to provide inter microphone level differences in order to compute the energy gradients at high frequencies. The proposed method is evaluated in the direction analysis of a spatial-sound processing technique, Directional audio coding (DirAC). It is shown that the method provides a reliable direction estimation at the entire audio frequency range, whereas the traditional method employing the pressure gradients produces correct estimation in a limited frequency window only.

关键词： Microphones Arrays Estimation Frequency estimation Indexes Shadow mapping audio coding

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AVS Digital audio Processing Technology

AVS Digital Audio Processing Technology

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International Conference on Innovative Computing, Information and Control (ICICIC)

作者： Cong Zhang RuiMin Hu HaoJun Ai Key Laboratory of Multimedia & Network Communications Engineering Wuhan University of China Wuhan China Wuhan University Wuhan Hubei CN

During the last decade, new digital audio applications have emerged for network multimedia, wireless multimedia communication, and multimedia computing. In response of this need, considerable research has been devoted to the development of high-fidelity digital audio. Several international or commercial digital audio standards are also presented, such as the ISO/MPEG family and the Dolby AC-3. Although these standards are publicly available, it does not mean they are free. Therefore, China own audio and video compression standard (AVS) is presented. In this paper, AVS audio coding and decoding technologies are discussed in some detail and concludes with a brief discussion of future research directions

关键词： Decoding audio coding Multimedia computing Switches Multimedia communication Psychoacoustic models Phase change materials ISO standards Video compression Standards development

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The measurement of audio-codec sound quality

The measurement of audio-codec sound quality

引用

IEEE Instrumentation and Measurement Technology Conference

作者： G. Bucci F. Franciosi P. Valocchi Dipartimento di Ingegneria Elettrica L'Aquila Italy Scuola Superiore Guglielmo Reiss Romoli S.p.A Coppito Italy

One of the most difficult issues about audio-codec design and evaluation is the analysis of the reproduced sound quality. Standard measurements (such as the signal-to-noise ratio) aren't adequate to make an accurate comparison between different codecs due to the masking phenomena of the human ear. The aim of this paper is to propose a simple objective test for the measurement of the coding distortion in analysis-by-synthesis codecs.

关键词： Codecs Psychoacoustic models World Wide Web Measurement standards Humans Ear Testing Computational complexity audio coding Frequency

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Benchmarking flexible adaptive time-frequency transforms for underdetermined audio source separation

Benchmarking flexible adaptive time-frequency transforms for...

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

作者： Andrew Nesbit Emmanuel Vincent Mark D. Plumbley Electronic Engineering & Computer Science Queen Mary College University of London London UK METISS Group IRISA/INRIA Rennes France

ISBN: (纸本)9781424423538

We have implemented several fast and flexible adaptive lapped orthogonal transform (LOT) schemes for underdetermined audio source separation. This is generally addressed by time-frequency masking, requiring the sources to be disjoint in the time-frequency domain. We have already shown that disjointness can be increased via adaptive dyadic LOTs. By taking inspiration from the windowing schemes used in many audio coding frameworks, we improve on earlier results in two ways. Firstly, we consider non-dyadic LOTs which match the time-varying signal structures better. Secondly, we allow for a greater range of overlapping window profiles to decrease window boundary artifacts. This new scheme is benchmarked through oracle evaluations, and is shown to decrease computation time by over an order of magnitude compared to using very general schemes, whilst maintaining high separation performance and flexible signal adaptivity. As the results demonstrate, this work may find practical applications in high fidelity audio source separation.

关键词： Time frequency analysis Source separation audio coding Fourier transforms Computer science Local activities Equations High performance computing Discrete cosine transforms Yield estimation

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Overview of MPEG-4 audio and its applications in mobile communications

Overview of MPEG-4 audio and its applications in mobile comm...

引用

International Conference on Communication Technology (ICCT)

作者： J. Herre B. Grill Audio Department Fraunhofer Institute of Integrated Circuits and Systems Erlangen Germany

MPEG-4 is the most recent member of the well-known family of ISO/MPEG coding standards, and provides an integrated set of audio coders with attractive novel capabilities, including bit rate and bandwidth scalability. The paper gives an introduction into the underlying design concepts of MPEG-4 and provides an overview of MPEG-4 audio coding technology and its features. In particular, it discusses the technical provisions which make MPEG-4 audio very well suited for applications in the domain of mobile communications.

关键词： MPEG 4 Standard Mobile communication Bit rate audio coding Channel capacity ISO standards Code standards IEC standards Standards development Bandwidth

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Tutorial on Critical Listening of Multi-Channel audio Codec Performance

Tutorial on Critical Listening of Multi-Channel Audio Codec ...

引用

SMPTE Technical Conference

作者： Sunil Bharitkar Grant Davidson Louis Fielder Poppy Crum Dolby Laboratories Inc. 100 Potrero Avenue San Francisco CA 94103

ISBN: (纸本)9781622765386

Listening for impairments introduced by multichannel audio codecs is an important task. Classical objective methods are not adequate in assessing audio coding schemes. Accordingly, the ITU-R BS.1116 & 1534 recommendations provide guidelines for subjective evaluation of codecs. This paper provides a tutorial on the proper conditions to do reliable codec testing. Several key components covered are, proper experimental design, selection of listening panel and training of listeners, developing the test methodology, selecting balanced program material, loudspeaker/room and sound-field requirements, listening for artifacts, and statistical analysis. This paper addresses these various components including the sound-field requirements since, as per the ITU: “The characteristics of the reference sound field at the listening area are most important for the subjective perception of, or the quality assessment of, auditory events and their reproducibility at other listening places or rooms. These characteristics result from the interaction of the loudspeaker(s) and the listening room”.

关键词： audio codec evaluation ITU-R Listening tests BS.1116 Critical listening setup Subjective testing Auscultation Listening Tutorials audio coding acoustic field Quality assessment codecs Loudspeakers International Telecommunication Union Intensive Care Units subjective evaluation Listeners design and selection physical impairment

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