Download Finite Difference Schemes on Hexagonal Grids for Thin Linear Plates with Finite Volume Boundaries The thin plate is a key structure in various musical instruments, including many percussion instruments and the soundboard of the piano, and also is the mechanism underlying electromechanical plate reverberation. As such, it is a suitable candidate for physical modelling approaches to audio effects and sound synthesis, such as finite difference methods—though great attention must be paid to the problem of numerical dispersion, in the interest of reducing perceptual artefacts. In this paper, we present two finite difference schemes on hexagonal grids for such a thin plate system. Numerical dispersion and computational costs are analysed and compared to the standard 13-point Cartesian scheme. An equivalent finite volume scheme can be related to the 13-point Cartesian scheme and a 19-point hexagonal scheme, allowing for fitted boundary conditions of the clamped type. Theoretical modes for a clamped circular plate are compared to simulations. It is shown that better agreement is obtained for the hexagonal scheme than the Cartesian scheme.
Download Prioritized Computation for Numerical Sound Propagation The finite difference time domain (FDTD) method is commonly used as a numerically accurate way of propagating sound. However, it requires extensive computation. We present a simple method for accelerating FDTD. Specifically, we modify the FDTD update loop to prioritize computation where it is needed most in order to faithfully propagate waves through the simulated space. We estimate for each potential cell update its importance to the simulation output and only update the N most important cells, where N is dependent on the time available for computation. In this paper, we explain the algorithm and discuss how it can bring enhanced accuracy and dynamism to real-time audio propagation.
Download Sinusoidal Synthesis Method using a Force-based Algorithmm In this paper we propose a synthesis method using a force-based algorithm to control frequencies of multiple sine waves. In order to implement this synthesis method, we analyze an existing sound source using a fast Fourier transform (FFT). Spectral peaks which have large magnitudes are regarded as heavy partials and assigned large attractive forces. A few hundred sine waves with stationary amplitudes are placed in a frequency space on which forces generated in the analysis phase are applied. The frequencies of the partials gravitate to the nearest peak of the reference spectrum from the source sound. As more sine waves are combined at the large peaks, the sound synthesized by the partials gradually transforms into the reference spectrum. In order to prevent the frequencies of the partials from gravitating onto localized peaks, each partial is assigned a repulsive force against all others. Through successful control of these attractive and repulsive forces, roughness and speed variation of the synthesis can be achieved. Moreover, by increasing or decreasing the number of partials according to the total amplitude of the source sound, amplitude envelope following is achieved.
Download A Method of Morphing Spectral Envelopes of the Singing Voice for Use with Backing Vocals The voice morphing process presented in this paper is based on the observation that, in many styles of music, it is often desirable for a backing vocalist to blend his or her timbre with that of the lead vocalist when the two voices are singing the same phonetic material concurrently. This paper proposes a novel application of recent morphing research for use with a source backing vocal and a target lead vocal. The function of the process is to alter the timbre of the backing vocal using spectral envelope information extracted from both vocal signals to achieve varying degrees of blending. Several original features are proposed for the unique usage context, including the use of LSFs as voice morphing parameters, and an original control algorithm that performs crossfades between synthesized and unsynthesized audio on the basis of voiced/unvoiced decisions.
Download Short-Time Time-Reversal on Audio Signals We present an analysis of short-time time-reversal on audio signals. Based on our analysis, we define parameters that can be used to control the digital effect and explain the effect each parameter has on the output. We further study the case of 50% overlap-add, then use this for a real-time implementation. Depending on the window length, the effect can modify the output sound variously, from adding overtones to adding reverse echoes. We suggest example use cases and digital effects setups for usage in sound design and recording.
Download A Statistical Approach to Automated Offline Dynamic Processing in the Audio Mastering Process Mastering audio is a complicated yet important step in music production. It is used for many purposes, an important one is to ensure a typical loudness for a piece of music within its genre. In order to automate this step we use a statistical model of the dynamic section. To allow a statistical approach we need to introduce some modifications to the compressor’s side-chain or more precisely to its ballistics. We then develop an offline framework to determine compressor parameters for the music at hand such that the signal’s statistic properties meet certain target properties, namely statistical central moments, which for example can be chosen genre specific. Finally the overall system is tested with songs which are available to us as unmastered, professionally mastered, and only compressed versions.
Download Revisiting Implicit Finite Difference Schemes for Three-Dimensional Room Acoustics Simulations on GPU Implicit finite difference schemes for the 3-D wave equation using a 27-point stencil on the cubic grid are presented, for use in room acoustics modelling and artificial reverberation. The system of equations that arises from the implicit formulation is solved using the Jacobi iterative method. Numerical dispersion is analysed and computational efficiency is compared to second-order accurate 27-point explicit schemes. Timing results from GPU implementations demonstrate that the proposed algorithms scale over their explicit counterparts as expected: by a factor of M + 2, where M is a fixed number of Jacobi iterations (eight can be sufficient in single precision). Thus, the accuracy of the approximation can be improved over explicit counterparts with only a linear increase in computational costs, rather than the quartic (in operations) and cubic (in memory) increases incurred when oversampling the grid. These implicit schemes are advantageous in situations where less than 1% dispersion error is desired.
Download A Preliminary Model for the Synthesis of Source Spaciousness We present here a basic model for the synthesis of source spaciousness over loudspeaker arrays. This model is based on two experiments carried out to quantify the contribution of early reflections and reverberation to the perception of source spaciousness.
Download Low Frequency Group Delay Equalization of Vented Boxes using Digital Correction Filters In this paper methods to determine the group delay of vented boxes and techniques for the design of filters for group delay equalization are presented. First the transfer function and the related group delay are explained. Then it is shown how the group delay can be computed or approximated for a certain alignment of the box. Furthermore it is shown how to derive the required parameters of the transfer function from a simple electrical measurement of the box, which allows the determination of the group delay without knowledge of the box design parameters. Two strategies for the design and implementation of digital correction filters are shown where one approach allows for a real-time adjustability of the delay. Finally, the performance with a real speaker is evaluated.
Download Exploring the Vectored Time Variant Comb Filter This paper presents the time variant vectored comb filter. It is an extension of the feedback delay network to time variant and nonlinear domains. Effects such as chorus and flanger, tap delay and pitch shifter are examined in the context of the feedback scheme. Efficient implementation of a stateless vectorizable LFO for modulation purposes is presented, along with a recursive formulation of the Hadamard matrix multiplication. The time variant comb filter is examined in various effect settings, and presented with source code and sound examples.