Local Sound Field Reproduction using Two Closely Spaced Loudspeakers

When only two loudspeakers are used for the reproduction of sound for a single listener, time domain simulations show that it is advantageous that the two loudspeakers are very close together. The sound field reproduced by two loudspeakers that span 10 degrees as seen by the listener is simpler, and locally more similar to the sound field generated by a real sound source, than that reproduced by two loudspeakers that span 60 degrees. The basic physics of the problem is first explained by assuming that the sound propagates under free-field conditions. It is then demonstrated that when the influence of the listener on the incident sound waves is taken into account by modeling the listener's head as a rigid sphere, the results are qualitatively the same as in the free-field case. Consequently, two closely spaced loudspeakers are capable of accurately reproducing a desired sound field, not only at the ears of the listener but also in the, vicinity of the listener's head. This result, although counter-intuitive, is very encouraging. In particular, it suggests that many low-fidelity audio systems, such as those currently supplied with most multi-media computers, can be greatly improved.</p

The "stereo dipole" - A virtual source imaging system using two closely spaced loudspeakers

A system comprising only two closely spaced loudspeakers can create very convincing virtual images around a single listener. It is demonstrated that such a system is very robust with respect to head movement, and that the processing does not introduce any excessive artifacts. In practice, the loudspeakers ought to be "pair matched" in order to ensure accurate imaging.</p

Adaptive inverse filters for stereophonic sound reproduction

This paper describes a general theoretical basis for the design of adaptive digital filters used for the equalization of the response of multichannel sound reproduction systems. The work presented is applied first to the two channel case and then extended to deal with arbitrary numbers of channels. In the work described, the intention is not only to equalize both the response of the loudspeakers and the listening room but also the crosstalk transmission from right loudspeaker to left ear and vice versa. The formulation presented is thus a generalization of the Atal-Schroeder crosstalk canceller. However, the use of a least squares approach to the digital filter design, together with the use of appropriate modeling delays, also potentially enables the effective equalization of nonminimum phase components in the transmission path. A stochastic gradient algorithm is presented which facilitates the adaptation of the digital filters to the optimal solution, thereby providing the possibility of the design of the filters in situ in a given listening space. The work is illustrated using some experimental results for the two-channel case.</p

Inverse filter design and equalization zones in multichannel sound reproduction

A discussion is given of two techniques for designing inverse filters for use in multichannel sound reproduction systems. The first is the multiple-input/output inverse filtering theorem, which uses direct inversion of a matrix containing the coefficients of filters used to specify the electroacoustic transmission paths. The second is an adaptive technique based on the Multiple Error LMS algorithm. The theory presented reconciles the two approaches and furthermore, derives explicit conditions which must be fulfilled if an exact inverse is to exist. A formula is derived which gives the number of coefficients required in the inverse filters in terms of the number of coefficients used to represent the transmission paths. Some numerical examples are also presented which illustrate the dependence of the mean square error on both the choice of modeling delay and the number of coefficients in the inverse filters. Finally, the results of some simulations are given which demonstrate the acoustical possibilities associated with these filtering techniques.</p

Inverse filter of sound reproduction systems using regularization

We present a very fast method for calculating an inverse filter for audio reproduction system. The proposed method of FFT-based inverse filter design, which combines the well-known principles of least squares optimization and regularization, can be used for inverting systems comprising any number of inputs and outputs. The method was developed for the purpose of designing digital filters for multi-channel sound reproduction. It is typically several hundred times faster than a conventional steepest descent algorithm implemented in the time domain. A matrix of causal inverse FIR (finite impulse response) filters is calculated by optimizing the performance of the filters at a large number of discrete frequencies. Consequently, this deconvolution method is useful only when it is feasible in practice to use relatively long inverse filters. The circular convolution effect in the time domain is controlled by zeroth-order regularization of the inversion problem. It is necessary to set the regularization parameter β to an appropriate value, but the exact value of β is usually not critical. For single-channel systems, a reliable numerical method for determining β without the need for subjective assessment is given. The deconvolution method is based on the analysis of a matrix of exact least squares inverse filters. The positions of the poles of those filters are shown to be particularly important.</p

A multiple microphone recording technique for the generation of virtual acoustic images

A new recording technique based on multichannel digital signal processing is suggested. The system uses a dummy-head that is modeled as a rigid sphere with two pairs of microphones mounted on opposite sides of the sphere in the horizontal plane. Reversals - front back confusion, is a well- known phenomenon when localizing virtual acoustic images produced by either headphones or loudspeakers. Reproduction with two loudspeakers to the front of the listener causes rear virtual acoustic images to be perceived primarily at 'mirrored' angles in the frontal hemisphere. The problem is tackled here by using a multichannel signal processing technique rather than by mimicking accurately the acoustomechanical properties of a human head. The acoustic signals which are recorded at the microphones are filtered by a 4 x 4 matrix of digital filters before being transmitted via four loudspeakers. The performance of the system is investigated by means of computer simulations, objective measurements, and also by subjective experiments in an anechoic environment, where the listeners are asked to localize the perceived angle of the signals which were prerecorded with the sphere dummy-head. Successful discrimination of reversals is achieved primarily due to the dominant role of the interaural time delay (ITD) for localization at low frequencies, but the accuracy with which listeners can localize virtual acoustic images is reduced in comparison to a conventional two-ear dummy-head (e.g., KEMAR) with a two- loudspeaker arrangement. The system is robust with respect to head rotations - virtual acoustic images do not disappear and localization ability improves when listeners use small head rotations.</p

Fast deconvolution of multichannel systems using regularization

A very fast deconvoliition method, which is based on the fast Fourier transform (FFT), can be used to control the outputs from a multichannel plant comprising any number of control sources and error sensors. The result is a matrix of causal finite impulse response filters whose performance is optimized at a large number of discrete frequencies.</p

Local sound field reproduction using digital signal processing

This work shows how an acoustic wavefront can be reconstructed locally by using only a few loudspeakers. The loudspeaker inputs are calculated by passing a set of signals recorded by only a few microphones through a matrix of causal digital filters having finite impulse responses. These filters, referred to as the inverse filters, are calculated by inverting (in the least-squares sense) a matrix which contains the electroacoustic transfer functions from the loudspeakers to the microphones. In practice, it is crucial to use a modeling delay and a regularization factor in order to achieve an accurate inversion. The technique is illustrated with an example that shows how well four loudspeakers can reproduce a sound field that has been recorded with three microphones. When the recorded field does not contain energy at frequencies whose acoustical wavelengths are shorter than the distance between adjacent microphones, the original field is reproduced remarkably accurately in the vicinity of the microphones regardless of the positions of the loudspeakers.</p