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International Conference on Multimedia and Expo

作者： Zhang, Shuhua Dou, Weibei Yang, Huazhong Tsinghua Univ Dept Elect Engn Tsinghua Natl Lab Informat Sci & Technol Beijing 100084 Peoples R China

ISBN: (纸本)9781424474936

This paper presents a linear operation called Maximal Coherence Rotation (MCR) on paired vectors for stereo audio coding. Intrigued by the idea that stronger coherence between paired channels will lead to higher stereo coding efficiency, we develop MCR to maximize the coherence restricted by being invertible and energy-conserving. It results in equal energy, minimized difference, and always non-negative coherence for the pair of channels processed. In binaural hearing, this can be viewed as turning a physical sound source at any azimuth to a virtual one on the median plane. A prototype MCR stereo coder shows significantly higher quality for some test sequences than that of AMR-WB+, one of the best low bitrate stereo coders in the public domain. And as a preprocessing tool for MPEG-4 Parametric Stereo (PS), MCR avoids out-of-phase inter-channel cancellation during 2-to-1 channel downmixing without any additional bandwidth requirement, thanks to the maximized coherence.

关键词： audio coding Binaural Hearing Inner Product Space Median Plane

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Energy-proportion based scheme for audio watermarking

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IET SIGNAL PROCESSING 2010年第5期4卷 576-587页

作者： Chen, S. -T. Huang, H. -N. Chen, C. -J. W, G. -D. Natl Chi Nan Univ Dept Elect Engn Puli 545 Nan Tou Taiwan Tunghai Univ Dept Math Taichung 40704 Taiwan

This study presents a audio watermarking that embeds information by energy-proportion scheme. By using normalised energy instead of probability, this study rewrites the entropy in information theory as an e function. In order to guarantee the robustness of watermarks, low-frequency coefficients of discrete wavelet transform (DWT) are mapped into the domain of this function. Then, the characteristic curve of energy-proportion function (CCEP) is obtained. Moreover, some characteristics and properties of this function are analysed and proved. Based on CCEP and the properties of energy-proportion function, this study proposes a novel audio watermarking scheme. In addition, the proposed energy-proportion scheme can extract the watermark without original audio signal. Finally, performance of the proposed scheme is assessed by signal-to-noise ratio (SNR), mean opinion score, embedding capacity and BER. Experimental results demonstrate that the embedded data are robust against most signal processing and attacks, such as re-sampling, MP3 compression, low-pass filtering and amplitude scaling.

关键词： energy proportion function signal sampling audio signal extraction error statistics audio watermarking signal- to-noise ratio amplitude scaling information theory signal resampling MP3 compression discrete wavelet transform Speech and audio coding DWT discrete wavelet transforms audio coding entropy Integral transforms signal processing BER SNR CCEP Other topics in statistics

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Representing Musical Sounds With an Interpolating State Model

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2010年第3期18卷 613-624页

作者： Klapuri, Anssi Virtanen, Tuomas Tampere Univ Dept Signal Proc FI-33720 Tampere Finland

A computationally efficient algorithm is proposed for modeling and representing time-varying musical sounds. The aim is to encode individual sounds and not the statistical properties of several sounds representing a certain class. A given sequence of acoustic feature vectors is modeled by finding such a set of "states" (anchor points in the feature space) that the input data can be efficiently represented by interpolating between them. The proposed interpolating state model is generic and can be used to represent any multidimensional data sequence. In this paper, it is applied to represent musical instrument sounds in a compact and accurate form. Simulation experiments were carried out which show that the proposed method clearly outperforms the conventional vector quantization approach where the acoustic feature data is k-means clustered and the feature vectors are replaced by the corresponding cluster centroids. The computational complexity of the proposed algorithm as a function of the input sequence length is O(T log T).

关键词： Acoustic signal processing audio coding discrete cosine transforms (DCTs) interpolation vector quantization

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A High-Quality Speech and audio Codec With Less Than 10-ms Delay

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2010年第1期18卷 58-67页

作者： Valin, Jean-Marc Terriberry, Timothy B. Montgomery, Christopher Maxwell, Gregory Octasic Inc Montreal PQ H1Y 3L1 Canada Xiph Org Fdn Somerville MA 02144 USA RedHat Inc Westford MA 01886 USA Juniper Networks Inc Herndon VA 20171 USA

With increasing quality requirements for multimedia communications, audio codecs must maintain both high quality and low delay. Typically, audio codecs offer either low delay or high quality, but rarely both. We propose a codec that simultaneously addresses both these requirements, with a delay of only 8.7 ms at 44.1 kHz. It uses gain-shape algebraic vector quantization in the frequency domain with time-domain pitch prediction. We demonstrate that the proposed codec operating at 48 kb/s and 64 kb/s out-performs both G. 722.1C and MP3 and has quality comparable to AAC-LD, despite having less than one fourth of the algorithmic delay of these codecs.

关键词： audio coding low-delay speech coding super-wideband transform coding

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audio coding with Signal Adaptive Block based Filter Bank Switching

Audio Coding with Signal Adaptive Block based Filter Bank Sw...

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International Conference on Electrical Engineering (ICEE)

作者： M. Saleem M.T. Ali Department of Electrical Engineering University of Engineering and Technology Lahore Pakistan Communication Systems Lab Research Center University of Engineering and Technology Lahore Pakistan

An improved efficiency perceptual audio codec is presented which analyzes each block of input signal and selects a suitable time/frequency mapping transform for it. The selection is based on statistics of the input signal vis-a-vis energy compaction and resolution power properties of the transforms employed which include the DFT (uniform subbands), DFT (critical subbands), DCT and CELP( for speech only blocks). The performance of the codec is compared with the widely famous MPEG-1 Layer-III algorithm. Efficiency enhancement is indicated by improved grades of subjective listening test for the proposed codec compared to those for MPEG-1 Layer-III at similar bit rates.. The paper concludes with a discussion of future research implications of the work.

关键词： audio coding Filter bank Codecs Signal analysis Frequency Signal mapping Statistics Compaction Energy resolution Signal resolution

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A Comparative Evaluation of Techniques for Single-Frame Discrimination of Nonstationary Sinusoids

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2010年第3期18卷 498-508页

作者： Wells, Jeremy J. Murphy, Damian T. Univ York Dept Elect Intelligent Syst Grp Audio Lab York YO10 5DD N Yorkshire England

Many spectral analysis and modification techniques require the separation of sinusoidal from nonsinusoidal signal components of a Fourier spectrum. Techniques exist for the estimation of the parameters of nonstationary sinusoids, and for discriminating these from other components, within a single Fourier frame. We present a comparative study of five methods for sinusoidal discrimination, considering their effectiveness and their computational cost.

关键词： audio coding music spectral analysis

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audio Signal Representations for Indexing in the Transform Domain

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2010年第3期18卷 434-446页

作者： Ravelli, Emmanuel Richard, Gael Daudet, Laurent Univ Paris 06 Inst Jean Rond Alembert LAM F-75015 Paris France Telecom ParisTech Inst Telecom CNRS LTCI F-75014 Paris France

Indexing audio signals directly in the transform domain can potentially save a significant amount of computation when working on a large database of signals stored in a lossy compression format, without having to fully decode the signals. Here, we show that the representations used in standard transform-based audio codecs (e.g., MDCT for AAC, or hybrid PQF/MDCT for MP3) have a sufficient time resolution for some rhythmic features, but a poor frequency resolution, which prevents their use in tonality-related applications. Alternatively, a recently developed audio codec based on a sparse multi-scale MDCT transform has a good resolution both for time- and frequency-domain features. We show that this new audio codec allows efficient transform-domain audio indexing for three different applications, namely beat tracking, chord recognition, and musical genre classification. We compare results obtained with this new audio codec and the two standard MP3 and AAC codecs, in terms of performance and computation time.

关键词： audio coding audio indexing sparse representations time-frequency representations

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audio Sparse Decompositions in Parallel

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IEEE SIGNAL PROCESSING MAGAZINE 2010年第2期27卷 90-96页

作者： Daudet, Laurent Ecole Normale Super F-75231 Paris France Univ Paris 06 LAM F-75252 Paris 05 France Univ Paris 07 F-75221 Paris 05 France

Greedy methods are often the only practical way of solving very large sparse approximation problems. Among such methods, Matching Pursuit (MP) is undoubtedly one of the most widely used, due to its simplicity and relatively low overhead. Since MP works sequentially, however, it is not straightforward to formulate it as a parallel algorithm, to take advantage of multi-core platforms for real-time processing. In this paper, we investigate how a slight modification of MP makes it possible to break down the decomposition into multiple local tasks, while avoiding blocking effects. Our simulations on audio signals indicate that this Parallel Local Matching Pursuit (PLoMP) gives results comparable to the original MP algorithm, but could potentially run in a fraction of the time on-the-fly sparse approximations of high-dimensional signals should soon become a reality.

关键词： Matching pursuit algorithms Signal processing algorithms Discrete transforms Discrete wavelet transforms Image coding audio coding Transform coding Wavelet transforms Power harmonic filters Signal analysis

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audio coding Improvement Using Evolutionary Speech/Music Discrimination

Audio Coding Improvement Using Evolutionary Speech/Music Dis...

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IEEE International Conference on Fuzzy Systems (FUZZ-IEEE)

作者： J.E. Munoz Exposito S. Garcia Galan N. Ruiz Reyes P. Vera Candeas Telecommunication Engineering Department Polytechnic School University of Jaén Jaen Spain

Automatic speech/music discrimination is an important tool used in many multimedia applications, becoming a research topic of interest in the last years. This paper presents our last works in the speech/music discrimination field, aiming to improve the coding efficiency of standard audio coders (i.e. MP3, AAC) when speech and music signals are involved. In order to discriminate between speech and music, a fuzzy rules-based expert system is incorporated into the decision-taking stage of traditional speech/music discrimination systems. The knowledge base of the fuzzy expert system has been obtained by means of a typical genetic learning algorithm (the Pittsburgh algorithm). The proposed speech/music discrimination scheme manages the operation of an intelligent audio coder, which selects a GSM coder for speech frames and an AAC coder for music ones, resulting in a lower bit rate regarding the case of using a standardized audio coder (AAC in this work). Further, the intelligent audio coder has been designed aiming to obtain a similar subjective audio quality than AAC. GSM operates at 13 kbits/s, while in the experiments the bit rate specification for AAC has been 32 kbits/s for one-channel audio signals.

关键词： audio coding Speech coding Hybrid intelligent systems GSM Bit rate Code standards Digital audio players Multiple signal classification Fuzzy systems Expert systems

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Quantized Sparse Approximation with Iterative Thresholding for audio coding

Quantized Sparse Approximation with Iterative Thresholding f...

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

作者： M. Yaghoobi T. Blumensath M. Davies Institute for Digital Communications Joint Research Institute for Signal and Image Processing University of Edinburgh UK

Sparse coding is a new field in signal processing with possible applications to source coding. In this paper we present a new method that combines the problems of sparse signal approximation with coefficient quantization. This method uses overcomplete dictionaries and exploits signal redundancy. The proposed method will be derived as an extension of a recently presented method (iterative thresholding) to find sparse representations of signals. Because in digital communication and storage we need a quantized representation of the signal, instead of quantization of sparse representations a posteriori, we propose a refined method that combines sparse approximation and quantization. To compare the proposed method to a posteriori quantization, we present an audio example.

关键词： audio coding Quantization Dictionaries Source coding Signal representations Transform coding Matching pursuit algorithms Digital communication Image coding Decoding

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