Jordan Cheer

Robustness and Regularization of Personal Audio Systems

As well as being able to reproduce sound in one region of space, it would be useful to reduce the level of reproduced sound in other spatial regions, with a “personal audio” system. For mobile devices this is motivated by issues of privacy for the user and the need to reduce annoyance for other people nearby. Such personal audio systems can be realized with arrays of loudspeakers that become superdirectional at low frequencies, when the array dimensions are small compared with the acoustic wavelength. The design of the array then becomes a compromise between performance and array effort, defined as the sum of mean squared driving signals. Various methods of formulating this tradeoff as a regularization problem have been suggested and the connection between these formulations is discussed. Large array efforts are due to strongly self-cancelling multipole arrays. A concern is then the robustness of such an array to variations in the acoustic environment and driver sensitivity and position. The design of an array that is robust to these uncertainties then leads to a generalization of regularization.

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.

Minimally radiating sources for personal audio

In order to reduce annoyance from the audio output of personal devices, it is necessary to maintain the sound level at the user position while minimizing the levels elsewhere. If the dark zone, within which the sound is to be minimized, extends over the whole far field of the source, the problem reduces to that of minimizing the radiated sound power while maintaining the pressure level at the user position. It is shown analytically that the optimum two-source array then has a hypercardioid directivity and gives about 7 dB reduction in radiated sound power, compared with a monopole producing the same on-axis pressure. The performance of other linear arrays is studied using monopole simulations for the motivating example of a mobile phone. The trade-off is investigated between the performance in reducing radiated noise, and the electrical power required to drive the array for different numbers of elements. It is shown for both simulations and experiments conducted on a small array of loudspeakers under anechoic conditions, that both two and three element arrays provide a reasonable compromise between these competing requirements. The implementation of the two-source array in a coupled enclosure is also shown to reduce the electrical power requirements.

Modeling local active sound control with remote sensors in spatially random pressure fields

A general formulation is presented for the optimum controller in an active system for local sound control in a spatially random primary field. The sound field in a control region is selectively attenuated using secondary sources, driven by reference sensors, all of which are potentially remote from this control region. It is shown that the optimal controller is formed of the combination of a least-squares estimation of the primary source signals from the reference signals, and a least-squares controller driven by the primary source signals themselves. The optimum controller is also calculated using the remote microphone technique, in both the frequency and the time domains. The sound field under control is assumed to be stationary and generated by an array of primary sources, whose source strengths are specified using a spectral density matrix. This can easily be used to synthesize a diffuse primary field, if the primary sources are uncorrelated and far from the control region, but can also generate primary fields dominated by contributions from a particular direction, for example, which is shown to significantly affect the shape of the resulting zone of quiet.

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.

The effect of reverberation on personal audio devices

Personal audio refers to the creation of a listening zone within which a person, or a group of people, hears a given sound program, without being annoyed by other sound programs being reproduced in the same space. Generally, these different sound zones are created by arrays of loudspeakers. Although these devices have the capacity to achieve different sound zones in an anechoic environment, they are ultimately used in normal rooms, which are reverberant environments. At high frequencies, reflections from the room surfaces create a diffuse pressure component which is uniform throughout the room volume and thus decreases the directional characteristics of the device. This paper shows how the reverberant performance of an array can be modeled, knowing the anechoic performance of the radiator and the acoustic characteristics of the room. A formulation is presented whose results are compared to practical measurements in reverberant environments. Due to reflections from the room surfaces, pressure variations are introduced in the transfer responses of the array. This aspect is assessed by means of simulations where random noise is added to create uncertainties, and by performing measurements in a real environment. These results show how the robustness of an array is increased when it is designed for use in a reverberant environment.

Comments on "Complete parallel narrowband active noise control systems"

In the above paper Chang and Kuo show that the convergence of parallel narrowband active noise control systems is limited by the interaction between the multiple narrowband adaptive filters. To overcome this problem they propose a new algorithm that provides separate error signals to each of the narrowband adaptive filters, generated by filtering the error signal using bandpass filters tuned to the control frequency of each of the adaptive filters. Although these bandpass filters do not introduce an additional phase shift at the control frequency they do introduce an additional group delay into the secondary path. It is shown in this correspondence that although introducing these bandpass filters allows an increase in the step size it may also limit the convergence speed compared to the conventional algorithm.

Multichannel feedback control of interior road noise

Active noise control systems offer a potential method of reducing the weight of passive acoustic treatments and, therefore, increasing a vehicles’ fuel efficiency. The active control of engine noise can be implemented cost-effectively by using the car audio loudspeakers as control sources and an array of low-cost microphones as error sensors. Such systems have been commercially implemented, but without also controlling road noise their subjective benefits may be limited. The active control of road noise using a feedforward control strategy has also been practically demonstrated, but these systems require a number of accelerometers to be mounted to the vehicle’s structure to obtain a coherent reference signal and, therefore, lead to a significant implementation cost. This paper proposes a multichannel feedback system for the active control of road noise, which uses an array of microphones and car audio loudspeakers, which is common to a feedforward engine noise control system. The design of the multichanne...

Combining the remote microphone technique with head-tracking for local active sound control

This paper describes practical integration of the remote microphone technique with a head-tracking device in a local active noise control system. The formulation is first reviewed for the optimized observation filter and nearfield pressure estimation. The attenuation performance and stability of an adaptive active headrest system combined with the remote microphone technique are then studied. The accuracy of the nearfield estimation and the effect of the head-tracking on the control performance are investigated in real-time experiments. The regularization factor of the observation filter is selected as a trade-off between its accuracy and its robustness. The integrated active headrest system is used to estimate and attenuate disturbance signals at a listeners ears from a single tonal primary source, while a commercial head-tracking device detects and provides the real-time head position to the active headrest system whose responses are updated accordingly.

An Investigation of Delayless Subband Adaptive Filtering for Multi-Input Multi-Output Active Noise Control Applications

The broadband control of noise and vibration using multi-input, multi-output (MIMO) active control systems has a potentially wide variety of applications. However, the performance of MIMO systems is often limited in practice by high computational demand and slow convergence speeds. In the somewhat simpler context of single-input, single-output broadband control, these problems have been overcome through a variety of methods including subband adaptive filtering. This paper presents an extension of the subband adaptive filtering technique to the MIMO active control problem and presents a comprehensive study of both the computational requirements and control performance. The implementation of the MIMO filtered-x LMS algorithm using subband adaptive filtering is described and the details of two specific implementations are presented. The computational demands of the two MIMO subband active control algorithms are then compared to that of the standard full-band algorithm. This comparison shows that as the number of subbands employed in the subband algorithms is increased, the computational demand is significantly reduced compared to the full-band implementation provided that a restructured analysis filter-bank is employed. An analysis of the convergence of the MIMO subband adaptive algorithm is then presented and this demonstrates that although the convergence of the control filter coefficients is dependent on the eigenvalue spread of the subband Hessian matrix, which reduces as the number of subbands is increased, the convergence of the cost function is limited for large numbers of subbands due to the simultaneous increase in the weight stacking distortion. The performance of the two MIMO subband algorithms and the standard full-band algorithm has then been assessed through a series of time-domain simulations of a practical active control system and it has been shown that the subband algorithms are able to achieve a significant increase in the convergence speed compared to the full-band implementation.