This paper concerns the implementation of a 3D transmission line matrix (TLM) algorithm based on a tetrahedral mesh structure and visualization of room acoustics simulation. Although a well known method, TLM algorithms implemented in 3D are less commonly found in the literature. We have implemented the TLM method using a tetrahedral mesh of pressure nodes with transmission lines lying superimposed on nearest neighbour bonds of a tetrahedral atomic lattice. Results of simulations are compared with those of a standard 3D cartesian mesh and a 2D mesh implementation of TLM. An important feature is a useful graphics interface designed for user-friendly control of room acoustics simulation and visualization in arbitrary shaped rooms containing objects of arbitrary size and number. The paper includes brief discussions of results of using different techniques for modeling totally absorptive or partially absorptive boundaries.

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RoomWeaver is a Digital Waveguide Mesh (DWM) based Integrated Development Environment (IDE) style research tool, similar in appearance and functionality to other current acoustics software. The premise of RoomWeaver is to ease the development and application of DWM models for virtual acoustic spaces. This paper demonstrates the basic functionality of RoomWeaver’s 3D modelling and Room Impulse Response (RIR) generation capabilities. A case study is presented to show how new DWM types can be quickly developed and easily tested using RoomWeaver’s built in plug-in architecture through the implementation of a hybrid-type mesh. This hybrid mesh is comprised of efficient, yet geometrically inflexible, finite difference DWM elements and the geometrically versatile, but slow, wave-based DWM elements. The two types of DWM are interfaced using a KW-pipe and this hybrid model exhibits a significant increase in execution speed and a smaller memory footprint than standard wave-based DWM models and allows nontrivial geometries to be successfully modelled.

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In order to achieve high-quality audio-realistic rendering in complex environments, we need to determine all the acoustic paths that go from sources to receivers, due to specular reflections as well as diffraction phenomena. In this paper we propose a novel method for computing and auralizing the reflected as well as the diffracted field in 2.5D environments. The method is based on a preliminary geometric analysis of the mutual visibility of the environment reflectors. This allows us to compute on the fly all possible acoustic paths, as the information on sources and receivers becomes available. The construction of a beam tree, in fact, is here performed through a look-up of visibility information and the determination of acoustic paths is based on a lookup on the computed beam tree. We also show how to model diffraction using the same beam tree structure used for modeling reflection and transmission. In order to validate the method we conducted an acquisition campaign over a real environment and compared the results obtained with our real-time simulation system.

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The aim of this paper is to give an overview of the techniques and principles for rendering the apparent source extent of sound sources in 3D audio displays. We mainly focus on techniques that use decorrelation as a mean to decrease the Interaural CrossCorrelation Coefficient (IACC) which has a direct impact on the perceived source extent. We then present techniques where decorrelation is varied in time and frequency, allowing to create tempo­ ral and spectral variations in the spatial extent of sound sources. Frequency dependant decorrelation can be employed to create an effect where a sound is spatially split in its different frequency bands, these having different positions and spatial extents. We fi­ nally present results of psychoacoustic experiments aimed at eval­ uating the effectiveness of decorrelation based techniques for the rendering of sound source extent. We found that the intended source extent matches well the mean perceived source extent by subjects.

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Analog compander systems have been used to suppress the perception of noise in low dynamic range analog signal storage (tape recording) and signal transmission (FM radio). Commercial compander systems have been analyzed with respect to their signal processing requirements. The general structures of single- and multiband compander systems have been implemented on a high performance audio PC workstation. Audio tests and measurements with the optimized compander algorithms and parameters show very good performance. Even for transmission channels with very low signal-to-noise ratio (SNR of only 40 dB) an optimized digital multi-band compander emulation removes the channel noise perceptively from the output signal of the transmission system.

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This paper introduces a method to transform voice based on modeling the radiated voice pulses in frequency domain. This approach tries to combine the strengths of classical time and frequency domain techniques into a single framework, providing both an independent control of each voice pulse and flexible timbre and phase modification capabilities.

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This paper takes the opportunity of presenting a set of new adaptive effects to propose a generic scheme for adaptive effects built upon a spectral source-filter decomposition and a Short-Time Fourier analysis-resynthesis. This allows for a better formalization of the involved signal processing algorithms and leads to a simple classification of adaptive effects already presented in the literature, that falls into this category. We discuss the motivation and the advantages of combining source-filter modeling and phase vocoder representation for the design of adaptive digital audio effects. Then we detail the general structure that includes STFT analysis and re-synthesis scheme, the source filter decomposition, and an adaptive control unit composed of a feature extraction system and a sound mapping unit that might be driven by a gestural control section.

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The main aim of this paper is to describe a new software program for esophageal speech treatment developed at the University of Deusto. The software tool, named “ESOIMPROVE”, allows both to characterize and to modify this speech, and provides the necessary framework to achieve a high quality and intelligible transformed esophageal speech by applying a complete range of sound effects and algorithms. In this field, this tool represents a considerable advance in the study of these voices. The final objective of the project is to obtain an esophageal speech with acceptable levels of quality and intelligibility, and some more works in this direction are being actually developed.

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In this paper it is described a new algorithm for esophageal speech regeneration, based on pitch and jitter modification. Traditional phase vocoder and resampling pitch scaling techniques have been used to develop a new adaptive method which scales the low esophageal speech pitch and applies a variable scaling factor significantly reducing its jitter. This method has shown to considerably improve esophageal speech quality, reducing its hoarseness and increasing its intelligibility. The presented algorithm pretends to be an important step forward in the regeneration of esophageal speech.

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