REVERBERATION: COMBOLUTION & ALGORITHMS
J. Dario R. Barbosa 311275087 (SID)
Digital Audio Systems, DESC9115, Semester 1 2012 Graduate Program in Audio and Acoustics Faculty of Architecture, Design and Planning, The University of Sydney
ABSTRACT One of the most interesting and fundamental phenomenon in acoustics is Reverberation. Reverberation is how sounds propagate from the source to the listener. Focusing in enclosure spaces, when a signal is emitted by a Figure 2. Simple comb filter. sound source a torrent of echoes from different surfaces in the room strikes the listener’s ear, producing an impression of 1.2. ARTIFICIAL REVERBERATORS space. During this trip the radiated sound is modified by the environment, interacting with objects, surfaces, physical Since the 1950’s engineers and acousticians have tried to parameters and geometric characteristics of the room. Such develop reverberation devices capable to simulate sound reflections give to the listener an impression of space, position, propagation in enclosures. The two most popular ways to create orientation, directional characteristics of the sound source, size artificial reverberation is by convolution and by algorithms. and shape of the room [1, 2, 3, 4]. ‘Magnetic tape reverberator’ introduced in 1961 by John Artificial reverberation can be created by convolution and Kellner is another way; please refer to reference [6] for further by algorithms. information.
1. INTRODUCTION Reverberation by convolution shown in Figure 3, is the mathematical process to create a third signal from two other signals, one signal is the sound and the other signal is the 1.1. ACOUSTIC BACKGROUND impulse response of the reverberation enclosure we want to simulate. Convolution is a FIR based system. Reverberation can be divided in the frequency spectrum in three sections: direct sound, early reflections and echo density also named late reverberation; Figure 1.
Figure 3. Convolution
Using algorithms is another method to create artificial reverberation. These algorithms are IIR based system. Figure 1. Direct sound, early reflections and echo density in the frequency spectrum.
The reverberation time is defined by W. C. Sabine as the 2. CONVOLUTION time interval in which reverberation level drops by 60dB [5]. Direct sound is the sound that travels from the source to the Reverberation can be classified as linear invariant system; listener. Early reflections are reflections that arrive between 50 therefore it can be described by its input response and can be and 80 ms after the direct signal. The time between direct sound implemented by means of a Finite Impulse Response (FIR) and early reflections is named Pre-Delay or Initial Time Gap. filter [7].
Considering sound pressure, the particle velocity of air is in There are two ways to create an impulse response signal. phase with pressure for long distances and in quadrature for One is by exiting the room with a pure tone signal which short distances. If the source is at long distance we consider the changes continuously from low to high frequencies creating a condition of propagation of plane waves. In short distances the full frequency spectrum and capturing its response by means of air velocity and pressure are in quadrature and therefore energy microphones refer to Figure 4, and the second one is by doesn’t radiate. Therefore, we will focus on plane waves. designing on a computer the impulse response of the room.
The harmonic series of plane waves can be reproduced by a comb filter refer to Figure 2, by means of a feedback delay line.
Figure 5. All pass filter
The z transform of the all pass filter is given by