This paper describes a versatile software toolbox, which has been developed for researching, teaching and developing in the field of multichannel sound signal processing. The software system runs on a PC and consists on 5 modules covering the main stages and aspects of multichannel sound reproduction using loudspeakers. A number of new and efficient algorithms have been specially implemented for this software.

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This papers deals with a 3-D audio system that has been developed for desktop multimedia environments. The system has the ability to place virtual sources at arbitrary azimuths and elevations around the listener’s head based on HRTF binaural synthesis. A listener seated in front of a computer and two loudspeakers placed at each side of the monitor have been considered. Transaural reproduction using loudspeakers has been used for rendering the sound field to listener ears. Furthermore the system can cope with slight movements of the listener head. Head position is monitored by means of a simple computer vision algorithm. Four head position coordinates (x,y,z,φ) in order to allow free movements of the listener are continuously estimated. Cross-talk cancellation filters and virtual sources locations are updated depending on these head coordinates.

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Wave Field Synthesis is a method for 3D sound reproduction, based on the precise construction of the desired wave field by using an array of loudspeakers. The main purpose of this work is to present a set of software tools that brings to the audio community a feasible an easy way to start working with wave field synthesis systems. First in the paper, an introduction to different 3D sound techniques and an overview of WFS theory and foundations are given. Next, a series of software tools specially developed to simulate, analyze and implement WFS systems are presented. The first software module helps the user in the design of the array of loudspeakers to be employed in the reproduction by computing the equations for each speaker signal excitation. Another tool simulates the wave field generated by the arrays and analyses both performance and quality of the acoustic field. Finally a user friendly tool for realtime convolution capable of producing the excitation signals for the array of loudspeakers is presented. Also, different experiments that have been carried out with this software in order to evaluate the precision and behaviour of different WFS configurations are presented and interpreted.

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Advanced multichannel sound systems such as Wave Field Synthesis (WFS) allow to recreate spatial wide sound scenes of sources. The recreation of the illusion of a 3D natural and realistic sound scene can be achieved by means of virtual rooms where the wave field is simulated. Such wave field is used as a source of information for the convolution of WFS sound sources with extrapolated impulsive responses in these virtual rooms. To obtain the needed plane waves for auralization, a complete description of the sound field is needed, including an accurate knowledge of the particle velocity. In this paper, virtual rooms are simulated by means of Finite-Differences Time Domain method. This method provides a complete solution of the sound field variables in a wide frequency band and can be used to produce both the impulsive responses of pressure and particle velocity for plane wave decomposition, prior to auralization. To illustrate its applicability, a set of rooms consisting of a typical auditorium room, a cinema and a perfect cube are shown and evaluated.

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In sound rendering systems using loudspeakers, the listening room adds echoes not considered by the reproduction system, thus deteriorating the rendered audio signal. Specifically, Wave Field Synthesis is a 3D audio reproduction system, which allows synthesizing a realistic sound field in a wide area by using arrays of loudspeakers. This paper proposes a room compensation approach based on a multichannel inverse filter bank calculated to compensate the room effects at selected points within the listening area. Time domain and frequency domain algorithms are proposed to accurately compute the bank of inverse filters. A comparative study between these algorithms by means of laboratory experiments is presented.

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