17 research outputs found

    Analysis of Root Displacement Interpolation Method for Tunable Allpass Fractional-Delay Filters

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    Efficient Synthesis of Room Acoustics via Scattering Delay Networks

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    An acoustic reverberator consisting of a network of delay lines connected via scattering junctions is proposed. All parameters of the reverberator are derived from physical properties of the enclosure it simulates. It allows for simulation of unequal and frequency-dependent wall absorption, as well as directional sources and microphones. The reverberator renders the first-order reflections exactly, while making progressively coarser approximations of higher-order reflections. The rate of energy decay is close to that obtained with the image method (IM) and consistent with the predictions of Sabine and Eyring equations. The time evolution of the normalized echo density, which was previously shown to be correlated with the perceived texture of reverberation, is also close to that of IM. However, its computational complexity is one to two orders of magnitude lower, comparable to the computational complexity of a feedback delay network (FDN), and its memory requirements are negligible

    Interpolation of Low Order HRTF Filters using a Zero Displacement Measure

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    Head-related transfer functions (HRTF) are generally measured at discrete azimuth and elevation values. Continuous motion of virtual sound images can only be obtained by interpolation of HRTFs for intermediate azimuth and elevation values in an immersive audio system. HRTFs can be modelled as a combination of low-order minimum-phase finite impulse response (FIR) filters and a constant delay line. A new method for interpolation of FIR filter zeros using a proximity-based vectorial displacement measure is proposed in this work

    Multichannel Dereverberation Theorems and Robustness Issues

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    Multichannel dereverberation amounts to the inversion of a multiple-input/multiple-output linear time-invariant system. In this paper, necessary and sufficient conditions for perfect dereverberation using stable and finite impulse response (FIR) filters are established. It is then shown that the inverse system given by the pseudoinverse of the original transfer function matrix exhibits a noise reduction property. A necessary and sufficient condition under which this pseudoinverse system is FIR is also given. Further, an FIR approximation to the pseudoinverse system is considered and the effects of the length of this approximation on the dereverberation accuracy are investigated. Finally, an analytical and numerical assessment of the dependence of the dereverberation accuracy on the accuracy of the acquisition of room impulse responses is provided

    Gunshot Sounds

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    Recordings of the sounds of four different pistols (Browning BDA 380, Glock 19c, Glock 21, Rossi Magnum 357) in an indoor firing range. The room impulse response of the firing range is also included

    Morphing of Transient Sounds Based on Shift-Invariant Discrete Wavelet Transform and Singular Value Decomposition

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    In this paper, a new morphing algorithm for transient sounds is introduced. Input sounds are first projected onto orthogonal bases from which intermediate or novel sounds can be generated. The proposed algorithm uses a shift invariant version of discrete wavelet transform and the singular value decomposition (SVD) to represent the input sound signals over a set of orthogonal bases. Interpolation is carried out between the weight vectors from the SVD to produce a new weight vector used for synthesising a new set of wavelet coefficients. The morphed sound is generated by taking the inverse discrete wavelet transform of the combined weighted bases. The proposed algorithm not only generates a range of new sounds, but also represents the input sounds in a more compact fashion
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