Marcos F. Simón Gálvez

A superdirective array of phase shift sources

A superdirective array of audio drivers is described, which is compact compared with the acoustic wavelength over some of its frequency range. In order to minimize the overall sound power output, and hence reduce the excitation of the reverberant field when used in an enclosed space, the individual drivers are made directional by using phase shift enclosures. The motivating application for the array is the enhancement of sound from a television, in a particular region of space, to aid hearing impaired listeners. The design is initially investigated, using free-field simulations, by comparing the performance of 8 monopoles, 8 phase shift loudspeakers, and a double array of 16 monopoles, with a contrast maximization formulation. The construction and testing of an array of 8 drivers is then discussed, together with its measured response in an anechoic environment. The result of using acoustic contrast maximization is then compared with a least squares formulation, which demonstrates that the performance of the least squares solution can be made similar to that given by acoustic contrast maximization in this application, with a suitable choice of the target field.

Time domain optimization of filters used in a loudspeaker array for personal audio

This paper describes a time domain design method for calculating the coefficients of FIR filters used to drive a loudspeaker array for personal audio. A motivating application is to boost the television audio in a certain spatial region with the aim of increasing the speech intelligibility of the hearing impaired. As the array is of small size, superdirective beamforming is applied to increase the low and mid frequency directional performance of the radiator. The filters for such arrays have previously been designed one frequency at a time, leading to non-causal filters that require a modeling delay for real time implementation. By posing the filter optimization in the time domain, the filter responses can be causally constrained, and the optimization is performed once for all frequencies. This paper considers the performance of such filters by carrying out off-line simulations, firstly using the impulse responses of point sources in the free field, and then with the measured responses of a loudspeaker array in an anechoic chamber. The simulation results show how the time domain optimization allows the creation of filters with either a low order or a low modeling delay.

A Low-Frequency Panning Method With Compensation for Head Rotation

Amplitude panning produces interaural time difference (ITD) cues that help localize images in directions between loudspeakers. However, if the panning gains are static, then the ITD cues produced in this way vary inconsistently as the listeners head rotates, compared with a real source, and so the dynamic ITD cues are inaccurate. This effect destabilizes the perception of the image and overall scene, and is worse for loudspeakers that are more widely spaced relative to the listener. Based on a simple head model that is accurate in the low-frequency ITD regime, the ITD is calculated for a general field, including those produced by panning. A simple formula is derived relating head orientation, image direction, and a field description vector. Panning functions are then found that compensate for head orientation and are valid for any image direction. For the special case when the listener is facing the image, the functions are equivalent to vector base amplitude panning. The performance is first assessed objectively using measured binaural responses, rather than the simple head model. Subjective comparison is then made with pre-existing listening tests and new listening tests in which the listeners head is tracked to control the panning gains in real-time. These show that images can be stabilized as predicted, and, furthermore, that with the same panning functions, images can be produced in all directions using two loudspeakers placed in front.

Low-complexity, listener’s position-adaptive binaural reproduction over a loudspeaker array

This work presents a method for binaural reproduction over a loudspeaker array that adapts to the listener’s position by using a cross-talk cancellation approach that is updated in real-time. This is obtained by combining the audio signal processing system together with a computer vision mechanism that estimates listener’s position. A novel approach to adapt the cross-talk cancellation filters is introduced here, which employs filters created for a central listening position using a free-field propagation model and far field beamforming techniques, hence requiring little memory and processing. The paper introduces simulations to show the effectiveness and robustness of the formulation together with free-field measurements of performance using a 16 loudspeaker compact array.

Hearing impaired cochlear response simulation

A model is introduced which allows the vibration of the basilar membrane to be estimated for different degrees of hearing loss. The model is based on a discrete lumped parameter model of the human cochlea, which uses a three dimensional description of the fluid coupling. The hearing losses are assumed to be caused by the combined malfunction of the outer hair cells (OHCs), the inner hair cells (IHCs), and the endocochlear potential driving the system. OHC loss and damage to endochoclear potential are modeled by a reduction of the cochlear amplifier gain, which is obtained by reducing the feedback gain of the OHCs. IHC loss is modeled as an overall reduction in basilar membrane response. The distribution of OHC and IHC loss along the cochlea are derived using an iterative method, which matches the output vibration amplitude of the model to that assumed to generate the hearing impaired audiogram.