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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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Wavelet packet filterbanks for low time delay audio coding

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 1999年第3期7卷 310-322页

作者： Philippe, P de Saint-Martin, FM Lever, M CNET Rennes DIH HDM CIM F-35512 Cesson Sevigne France

We study the application of wavelet packet filterbanks to low bit-rate transparent audio coding, taking the audio coders' delay requirements into account, and propose low-delay coders based on wavelet packet filterbanks, We first develop a method of comparison between filterbanks for perceptual audio coding by estimating the necessary bit-rate for a transparent compression, We use this comparison method in order to select the best filters for our audio compression scheme, from a large set of orthogonal and biorthogonal wavelets. Different wavelet filters may be used at different stages of the tree-structured decomposition with a constraint on the overall delay taken into account. The optimization is carried out with a simulated annealing procedure, proposing two wavelet packet filterbanks, exhibiting average and low delays. They are inserted in a complete audio coder that employs vector quantization and considers psychoacoustic models. The use of the proposed filterbanks leads to the design of a new bit allocation procedure, taking into account the lack of selectivity of the equivalent synthesis filters in a wavelet packet filterbank, The resulting audio scheme is validated through listening tests. The wavelet packet filterbanks are shown to be a promising tool for audio coding, especially for low-delay coding: with average delay, the quality of the wavelet packet filterbanks is as good as with MPEG-1 Layer-2, both with 80 Kb/s, and when reducing the delay to 200 samples, 96 Kb/s are needed to achieve the same quality..

关键词： audio coding bit allocation optimal transformation wavelets

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coding bit rate scalability of MPEG audio data

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ELECTRONICS AND COMMUNICATIONS IN JAPAN PART III-FUNDAMENTAL ELECTRONIC SCIENCE 2001年第10期84卷 47-56页

作者： Nakajima, Y Yoneyama, A Sugano, M Yanagihara, H KDD R&D Labs Inc Multimedia Commun Lab Kamifukuoka Saitama 3568502 Japan

Users previously had to use transcoding methods in which a coded audio stream was decoded and the decoded audio then reencoded at the desired bit rate. Data processing of coded audio therefore usually involves complex and time-consuming tasks. In this paper, we describe three algorithms for bit rate scaling on coded MPEG data domains. One is a bandwidth limitation method to cut higher frequency components until the target data rate is satisfied. A second method uses a requantization process where a quantization step in each subband is modified, and a third method also utilizes a requantization process with a psychoacoustic model from bit allocation information obtained in the bitstream in order to improve bit rate scaling efficiency. The simulation results show that the requantization process provides very high conversion efficiency and that sound quality nearly equal to that of direct coding can be obtained by utilizing the psychoacoustic model. It was also shown that very fast scaling (factor of six) can be achieved compared with the conventional baseband transcoding method. (C) 2001 Scripta Technica.

关键词： MPEG audio coding audio coding scalability coding bit rate conversion requantization

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Optimum bit allocation in subband coding with nonideal reconstruction filters

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IEEE SIGNAL PROCESSING LETTERS 2001年第6期8卷 157-159页

作者： Caini, C Vanelli-Coralli, A Univ Bologna Dept Elect Comp Sci & Syst I-40126 Bologna Italy

Subband perceptual coding aims at providing the best subjective quality by decomposing the input signal into adjacent subbands, Every frequency component is then separately coded with a different number of bits in order to properly shape the quantization noise density power spectrum. In this letter, an optimum bit allocation algorithm is presented. As well as being applicable to either uniform or nonuniform frequency decompositions, it allows us to take into account the nonideal selectivity of the reconstruction filters. The effects of leakage for both short and long FIR filters have been investigated referring to a wavelet packet subband decomposition.

关键词： audio coding bit allocation filter banks multirate filters subband coding

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Modifying transients for efficient coding of audio

Modifying transients for efficient coding of audio

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

作者： R. Vafin R. Heusden W.B. Kleijn Department of Speech Music and Hearing KTH (Royal Institute of Technology) Stockholm Sweden

We propose a method for efficient representation of transients in audio signals. We estimate the transient component of an original audio signal and modify the locations of the transients in such a way that the transients can occur only at locations defined by a relatively coarse time grid. This procedure allows an efficient representation of transients with damped sinusoids. We also verify that the introduced modifications do not result in a perceptual difference between the original and the modified audio signals.

关键词： Signal analysis Signal processing Speech coding Multiple signal classification Auditory system Bit rate Damping Algorithm design and analysis audio coding Rate-distortion

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A comparison of warped and conventional linear predictive coding

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 2001年第5期9卷 579-588页

作者： Härmä, A Laine, UK Agere Syst Media Signal Proc Res Murray Hill NJ 07974 USA Aalto Univ Lab Acoust & Audio Signal Proc FIN-02015 Espoo Finland

Frequency-warped signal processing techniques are attractive to many wideband speech and audio applications since they have a clear connection to the frequency resolution of human hearing. A warped version of linear predictive coding (LPC) is studied in this paper, The performance of conventional and warped LPC algorithms are compared in a simulated coding system using listening tests and conventional technical measures. The results indicate that the use of warped techniques is beneficial especially in wideband coding and may result in savings of one bit per sample compared to the conventional algorithm while retaining the same subjective quality.

关键词： audio coding frequency-warping listening test speech coding warped linear prediction

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audio coding using steady state harmonics and residuals

Audio coding using steady state harmonics and residuals

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IEEE Workshop on Multimedia Signal Processing

作者： K.N. Hamdy A.H. Tewfik Department of Electrical and Computer Engineering University of Minnesota Minneapolis MN USA

In this paper, we solve the problem of inter and intra-frame frequency leakage in harmonic based audio coding approaches. Specifically, we first use an FFT to estimate the harmonic parameters. Because of the limited frequency resolution of the windowed FFT, there is an uncertainty in the measured frequency within one frequency bin. To minimize this uncertainty, we use these estimates as seeds to a minimum least squares problem and solve for the optimal harmonic parameters. The resulting residual should be essentially free of harmonics and consist of only attacks and noise. We are preserving the attacks and coding them separately. The resulting residual noise is modeled and coded using wavelet transforms and noise modeling. We also present a solution to the problem of inter-frame frequency fading and highlight the benefit of using a rectangular coordinate representation of the harmonic data. The improved coder that we propose can be viewed as an object based approach to audio coding and leads to perceptually pleasant progressive transmission and representation schemes for multimedia applications.

关键词： audio coding Steady-state Frequency Speech synthesis Signal synthesis Harmonic analysis Speech analysis Discrete wavelet transforms Wavelet transforms Signal analysis

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audio coding using parametric piecewise modeling

Audio coding using parametric piecewise modeling

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IEEE International Conference on Personal Wireless Communication

作者： M. Imine A.R. Prasad Lucent Technol. Netherlands Wireless Communications & Networking Division Lucent Technologies Inc. Nieuwegein Netherlands

We study the application of parametric piecewise modeling (PPM), a novel coding technique, for audio coding. PPM is based on either the polynomial, Bezier or Chebyshev parametric representation. It has already been proposed for curve and surface modeling use in computer graphics and has been recently proposed for source and channel coding. PPM speeds up the process of generating graphics, has increased compression capability and to a certain extent can predict missing parts of the signal. Results show that high compression of audio signals is achieved using PPM. Further we can see that error correction, to a certain extent, can be achieved using PPM due to its prediction capabilities.

关键词： audio coding Source coding Channel coding Polynomials Computer graphics Signal processing Application software Chebyshev approximation Signal generators Error correction

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An efficient lossless coding algorithm and its application to audio coding

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ELECTRONICS AND COMMUNICATIONS IN JAPAN PART III-FUNDAMENTAL ELECTRONIC SCIENCE 1999年第4期82卷 71-78页

作者： Takamizawa, Y Iwadare, M Sugiyama, A NEC Corp Ltd Informat Technol Res Labs Kawasaki Kanagawa 216 Japan

This paper proposes a tonal component coding algorithm for a codec that employs a transform followed by Huffman coding, such as MPEG-2 audio AAC (advanced audio coding) [1]. After the input audio signal is mapped onto the frequency domain, the proposed algorithm removes local maxima components that degrade the coding efficiency. As a result of this withdrawal, the flatness of the spectrum increases and the efficiency of Huffman coding is improved. The removed components are encoded separately as side information. When the frequency resolution of the time/frequency mapping is high, this algorithm works more effectively since local maximum samples appear more frequently with such a mapping. Simulation results show that this algorithm achieves as much as an 11% bit reduction per frame and improves the coding efficiency in 41% of all audio frames. (C) 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(4): 71-78, 1999.

关键词： audio coding MPEG AAC low bit-rate coding lossless coding Huffman coding

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Digital watermarking: Algorithms and applications

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IEEE SIGNAL PROCESSING MAGAZINE 2001年第4期18卷 33-46页

作者： Podilchuk, CI Delp, EJ Bell Labs Murray Hill NJ 07974 USA Purdue Univ W Lafayette IN 47907 USA

Digital watermarking of multimedia content has become a veryactive research area over the last several years. A general frameworkfor watermark embedding and detection/decoding is presented here alongwith a review of some of the algorithms for different media typesdescribed in the literature. We highlight some of the differences basedon application such as copyright protection, authentication, tamperdetection, and data hiding as well as differences in technology andsystem requirements for different media types such as digital images,video, audio and text

关键词： audio coding copy protection copyright data compression data encapsulation decoding image coding message authentication multimedia systems reviews video coding word processing authentication copyright protection data hiding digital images digital waterma Video coding Hidden data message authentication Digital watermarking DEcoding MULTI-MEDIA SYSTEM Digital images copyrights audio coding image coding Word Processing Data compression data encapsulation copy protection Authentication

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