There has recently been an increasing interest in the generation of a sound field that is audible in one spatial region and inaudible in an adjacent region. The method proposed here ensures the control of the amplitude and phase of multiple acoustic sources in order to maximize the acoustic energy difference between two adjacent regions while also ensuring that evenly distributed source strengths are used. The performance of the method proposed is evaluated by computer simulations and experiments with real loudspeaker arrays in the shape of a circle and a sphere. The proposed method gives an improvement in the efficiency of radiation into the space in which the sound should be audible, while maintaining the acoustic pressure difference between two acoustic spaces. This is shown to give an improvement of performance compared to the contrast control method previously proposed.
Summary Excessive noise in hospitals adversely affects patients’ sleep and recovery, causes stress and fatigue in staff and hampers communication. The World Health Organization suggests sound levels should be limited to 35 decibels. This is probably unachievable in intensive care units, but some reduction from current levels should be possible. A preliminary step would be to identify principal sources of noise. As part of a larger project investigating techniques to reduce environmental noise, we installed a microphone array system in one with four beds in an adult general intensive care unit. This continuously measured locations and sound pressure levels of noise sources. This report summarises results recorded over one year. Data were collected between 7 April 2017 and 16 April 2018 inclusive. Data for a whole day were available for 248 days. The sound location system revealed that the majority of loud sounds originated from extremely limited areas, very close to patients’ ears. This proximity maximises the adverse effects of high environmental noise levels for patients. Some of this was likely to be appropriate communication between the patient, their caring staff and visitors. However, a significant proportion of loud sounds may originate from equipment alarms which are sited at the bedside. A redesign of the intensive care unit environment to move alarm sounds away from the bed‐side might significantly reduce the environmental noise burden to patients.
Controlled sound interference has been extensively investigated using a prototype dual layer loudspeaker array comprised of 16 loudspeakers. Results are presented for measures of array performance such as input signal power, directivity of sound radiation and accuracy of sound reproduction resulting from the application of conventional control methods such as minimization of error in mean squared pressure, maximization of energy difference and minimization of weighted pressure error and energy. Procedures for selecting the tuning parameters have also been introduced. With these conventional concepts aimed at the production of acoustically bright and dark zones, all the control methods used require a tradeoff between radiation directivity and reproduction accuracy in the bright zone. An alternative solution is proposed which can achieve better performance based on the measures presented simultaneously by inserting a low priority zone named as the -gray‖ zone. This involves the weighted minimization of mean-squared errors in both bright and dark zones together with the gray zone in which the minimization error is given less importance. This results in the production of directional bright zone in which the accuracy of sound reproduction is maintained with less required input power. The results of simulations and experiments are shown to be in excellent agreement.
Multi-zone sound control aims to reproduce multiple sound fields independently and simultaneously over different spatial regions within the same space. This paper investigates the multi-zone sound control problem formulated in the modal domain using the Lagrange cost function and provides a modal-domain analysis of the problem. The Lagrange cost function is formulated to represent a quadratic objective of reproducing a desired sound field within the bright zone and with constraints on sound energy in the dark zone and global region.
Object-based audio is an emerging representation for audio content, where content is represented in a reproductionformat-agnostic way and thus produced once for consumption on many different kinds of devices. This affords new opportunities for immersive, personalized, and interactive listening experiences. This article introduces an end-to-end object-based spatial audio pipeline, from sound recording to listening. A high-level system architecture is proposed, which includes novel audiovisual interfaces to support object-based capture and listenertracked rendering, and incorporates a proposed component for objectification, i.e., recording content directly into an object-based form. Text-based and extensible metadata enable communication between the system components. An open architecture for object rendering is also proposed. The system's capabilities are evaluated in two parts. First, listener-tracked reproduction of metadata automatically estimated from two moving talkers is evaluated using an objective binaural localization model. Second, object-based scene capture with audio extracted using blind source separation (to remix between two talkers) and beamforming (to remix a recording of a jazz group), is evaluated with perceptually-motivated objective and subjective experiments. These experiments demonstrate that the novel components of the system add capabilities beyond the state of the art. Finally, we discuss challenges and future perspectives for object-based audio workflows.
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