This research presents a uniform approximation to the formulas of Benade and Keefe for the propagation constant of a cylindrical tube, valid for all tube radii and frequencies in the audio range. Based on this approximation, a simple expression is presented for a filter which closely matches the thermoviscous loss filter of a tube of specified length and radius at a given sampling rate. The form of this filter and the simplicity of coefficient calculation make it particularly suitable for real-time music applications where it may be desirable to have tube parameters such as length and radius vary during performance.

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When an unsmooth flexible tube rotates, rich tones are produced. We propose a physical model that simulates this behavior. The tube is modeled as an open-ended organ pipe blown by an air stream pumped by a rotationally induced pressure which follows Bernoulli’s principle.

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This paper describes an experimental research tool for block-based physical modeling and DSP computation. The goals of the development have been high abstraction level and flexibility in model specification without compromising computational efficiency in real-time simulation and application execution. To achieve both goals, the Lisp language is used for symbolic manipulation of computational block structures and C language for compilation of efficient executables. The primary motivation for this tool has been to enable flexible generation of physical models where twodirectional interaction between elements is needed. A particular feature of the system is support for mixed modeling by combining digital waveguides, finite difference schemes, wave digital filters, as well as traditional block-based DSP algorithms.

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The success recently encountered by physically-based modeling (or model-based approaches) for music should not mask the deep challenges that remain in this area. This article first proposes an overview of the various goals that researchers and musicians, respectively operating from scientific and end-user perspectives, may pursue. Among these goals, those recently proposed or particularly critical for the coming years of research are highlighted. The article then introduces ten criteria that summarize the main features an optimal physically-based modeling scheme or language should present. With respect to these, it proposes an evaluation of the major approaches to physically-based modeling. Key words: goals of the physically-based approach to sound synthesis and music creation, languages and schemes, enduser needs, perception, evaluation criteria, bibliographic overview.

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Main/Side (M/S) coding is a module deigned to reduce the channel irrelevancy and redundancies for multichannel coding in current audio standards. However, the success of this module relies on the careful design on four aspects: M/S decision, psychoacoustic model for the M/S channels, bits allocated between the M/S channels, and bit allocation in the M/S channels. This paper presents an efficient M/S method well designed from the four aspects. We conduct the experiments through the MP3 encoder to show the better quality and complexity than the Lame.

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The FIR-based reverberators, which convolve the input sequence with an impulse response modelling the concert hall, have better quality compared to the IIR-based approach. However, the high computational complexity of the FIR-based reverberators limits the applicability to most cost-oriented system. This paper introduces a method that uses perceptual criterion to reduce the complexity of convolution methods for reverberation. Also, an objective measurement criterion is introduced to check the perceptual difference from the reduction. The result has shown that the length of impulse response can be cut off by 60% without affecting the perceptual reverberation quality. The method is well integrated into the existing FFT-based approach is have around 30% speed-up.

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In this paper we compare different methods to compute music similarity between songs. The presented approaches have been reported by other authors in the field and we implemented minor improvements of them. We evaluated the different methods on a common database of MP3 encoded songs covering different genres, albums and artists. We used the best approach of the evaluation in a P2P scenario to compute song profiles and recommendations for similar songs. We will describe this integration in the second part of the paper.

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In the context of Computer Music, physical modeling is usually dedicated to the modeling of sound sources or physical instruments. This paper presents an innovative use of physical modeling in order to model and synthesize complex auditory effects such as collective acoustic phenomena producing metabolic emergent auditory organizations. As a case study, we chose the ‘dune effect’, which in open nature leads both to visual and auditory effects. The article introduces two particle physical models, able to collaborate. The first is dedicated the synthesis of spatial (or visual) dynamics effects of moving sand dunes. The second i s dedicated to the rendering of acoustical dynamics of “sounding sands”. Altogether, they lead to a multisensorial simulation of sand in dune. Keywords: auditory effects, metabolic effects, emergence, sonification process, sounding sands,

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The bandwidth-proportional noise-shaping criterion has been proposed to be the optimum bit allocation criterion in the sense of minimizing segmental masking-to-noise ratio [1]. A single loop bit allocation has been developed for ISO/IEC MPEG1layer 3 (MP3) [2] to reduce the complexity over the existing two nested loop method for MP3 encoder. Also, the allocation method also leads to the better compression quality. This paper extends the allocation method to the ISO/IEC MPEG-2/4 Advanced Audio Coding (AAC) [3] [4]. This paper derives the perceptual bandwidth for the scale factor bands, provides the bit consumption trade-off to transmit the bits for the scale factors, and illustrates an efficient sectioning method on the Huffman codebook. All these methods jointly integrate with the bit allocation method to improve the quality and the computing complexity of the widely used AAC reference encoder, FAAC [5].

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The internet Audio Rendering System (iARS) is an Internet browser extension extending the browser’s capabilities with real-time signal processing. The proposed system allows to receive audio streams from the Internet and apply various audio algorithms with no additional computational power needed from the server. iARS is part of the Internet Archive of Electronic Music (IAEM) project which is also presented in this paper.The IAEM is intended to be a platform to access an extensive and distributed archive of electronic music. It combines collaborative tools, real time signal processing on the client side and the content of the archive to a powerful teaching, research and publishing tool.

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