Spatial Acoustic Vector Based Sound Field Reproduction

Spatial sound field reproduction aims to recreate an immersive sound field over a spatial region. The existing sound pressure based approaches to spatial sound field reproduction focus on the accurate approximation of original sound pressure over space, which ignores the perceptual accuracy of the reproduced sound field. The acoustic vectors of particle velocity and sound intensity appear to be closely linked with human perception of sound localization in literature. Therefore, in this thesis, we explore the spatial distributions of the acoustic vectors, and seek to develop algorithms to perceptually reproduce the original sound field over a continuous spatial region based on the vectors. A theory of spatial acoustic vectors is first developed, where the spatial distributions of particle velocity and sound intensity are derived from sound pressure. To extract the desired sound pressure from a mixed sound field environment, a 3D sound field separation technique is also formulated. Based on this theory, a series of reproduction techniques are proposed to improve the perceptual performance. The outcomes resulting from this theory are: (i) derivation of a particle velocity assisted 3D sound field reproduction technique which allows for non-uniform loudspeaker geometry with a limited number of loudspeakers, (ii) design of particle velocity based mixed-source sound field translation technique for binaural reproduction that can provide sound field translation with good perceptual experience over a large space, (iii) derivation of an intensity matching technique that can reproduce the desired sound field in a spherical region by controlling the sound intensity on the surface of the region, and (iv) two intensity based multizone sound field reproduction algorithms that can reproduce the desired sound field over multiple spatial zones. Finally, these techniques are evaluated by comparing to the conventional approaches through numerical simulations and real-world experiments

Design and Evaluation of a Scalable and Reconfigurable Multi-Platform System for Acoustic Imaging

This paper proposes a scalable and multi-platform framework for signal acquisition and processing, which allows for the generation of acoustic images using planar arrays of MEMS (Micro-Electro-Mechanical Systems) microphones with low development and deployment costs. Acoustic characterization of MEMS sensors was performed, and the beam pattern of a module, based on an 8 × 8 planar array and of several clusters of modules, was obtained. A flexible framework, formed by an FPGA, an embedded processor, a computer desktop, and a graphic processing unit, was defined. The processing times of the algorithms used to obtain the acoustic images, including signal processing and wideband beamforming via FFT, were evaluated in each subsystem of the framework. Based on this analysis, three frameworks are proposed, defined by the specific subsystems used and the algorithms shared. Finally, a set of acoustic images obtained from sound reflected from a person are presented as a case study in the field of biometric identification. These results reveal the feasibility of the proposed systemSpanish research project SAM: TEC 2015-68170-R (MINECO/FEDER, UE

Active Noise Control Over Space: A Wave Domain Approach

Noise control and cancellation over a spatial region is a fundamental problem in acoustic signal processing. In this paper, we utilize wave-domain adaptive algorithms to iteratively calculate the secondary source driving signals and to cancel the primary noise field over the control region. We propose wave-domain active noise control algorithms based on two minimization problems: first, minimizing the wave-domain residual signal coefficients, and second, minimizing the acoustic potential energy over the region, and derive the update equations with respect to two variables, the loudspeaker weights and wave-domain secondary source coefficients. Simulation results demonstrate the effectiveness of the proposed algorithms, more specifically the convergence speed and the noise cancellation performance in terms of the noise reduction level and acoustic potential energy reduction level over the entire spatial region.DP14010341

Surround by Sound: A Review of Spatial Audio Recording and Reproduction

In this article, a systematic overview of various recording and reproduction techniques for spatial audio is presented. While binaural recording and rendering is designed to resemble the human two-ear auditory system and reproduce sounds specifically for a listener’s two ears, soundfield recording and reproduction using a large number of microphones and loudspeakers replicate an acoustic scene within a region. These two fundamentally different types of techniques are discussed in the paper. A recent popular area, multi-zone reproduction, is also briefly reviewed in the paper. The paper is concluded with a discussion of the current state of the field and open problemsThe authors acknowledge National Natural Science Foundation of China (NSFC) No. 61671380 and Australian Research Council Discovery Scheme DE 150100363

Development of a Method to Model an Enclosed, Coaxial Carbon Nanotube Speaker with Experimental Validation

Carbon nanotube (CNT) speakers operate on heat as compared to conventional loudspeakers that operate on vibration. CNT speakers are extremely lightweight, stretchable, flexible, and have high operating temperatures. Due to these advantages, CNT speakers are being considered as a viable replacement option for conventional loudspeakers. One such application is automotive exhaust noise control. The goal of this research is to design an enclosed, coaxial CNT speaker and to develop a modeling method to model this speaker using COMSOL Multiphysics. As part of this research, an enclosed, coaxial CNT speaker was designed and manufactured for automotive exhaust noise control. The first prototype was a proof of concept that the design is feasible, and the speaker works. Two additional prototypes have been developed to improve the manufacturing feasibility and performance. The first task undertaken during the modeling method development has been to create COMSOL models that simulated the CNT film temperature oscillation and the corresponding SPL. The simulation results have been compared with a MATLAB model for a planar CNT speaker. In addition, the SPL generated by the coaxial speaker has been compared with the simulated SPL generated by the CNT speaker. In addition, the performance of the coaxial speaker has been simulated in the presence of flow. Generally, a good correlation has been observed between the experimental SPL and simulated SPL. The models can be improved with the future development of improved material properties

Application of Carbon Nanotube Thermophones as Duct Noise Cancelling Speakers: Using New Technology with Old Theories

Certain materials can produce sound through heat fluctuations. In recent times, the material capable of this was discovered to be Carbon Nanotubes (CNT). The idea is to create a flexible speaker that can be made into any shape or form to create sound. Once a speaker has been created, an active noise control system can be designed to cancel loud sounds in building and pipe ducts. This technology, similar to noise cancelling headphones, allows a user to create a modular system to mitigate noise in everyday life. Thus, the purpose of this research is to illustrate the abilities of a CNT speaker system to perform active noise control on an HVAC duct application. This paper will start with a brief introduction into CNT technology and thermophone acoustics before diverging into active noise control technology. It will illustrate planar and non –planar wave cancellation and showcase a small scale accelerated life test

A room acoustics measurement system using non-invasive microphone arrays

This thesis summarises research into adaptive room correction for small rooms and pre-recorded material, for example music of films. A measurement system to predict the sound at a remote location within a room, without a microphone at that location was investigated. This would allow the sound within a room to be adaptively manipulated to ensure that all listeners received optimum sound, therefore increasing their enjoyment. The solution presented used small microphone arrays, mounted on the room's walls. A unique geometry and processing system was designed, incorporating three processing stages, temporal, spatial and spectral. The temporal processing identifies individual reflection arrival times from the recorded data. Spatial processing estimates the angles of arrival of the reflections so that the three-dimensional coordinates of the reflections' origin can be calculated. The spectral processing then estimates the frequency response of the reflection. These estimates allow a mathematical model of the room to be calculated, based on the acoustic measurements made in the actual room. The model can then be used to predict the sound at different locations within the room. A simulated model of a room was produced to allow fast development of algorithms. Measurements in real rooms were then conducted and analysed to verify the theoretical models developed and to aid further development of the system. Results from these measurements and simulations, for each processing stage are presented

Theory and Design of Spatial Active Noise Control Systems

The concept of spatial active noise control is to use a number of loudspeakers to generate anti-noise sound waves, which would cancel the undesired acoustic noise over a spatial region. The acoustic noise hazards that exist in a variety of situations provide many potential applications for spatial ANC. However, using existing ANC techniques, it is difficult to achieve satisfying noise reduction for a spatial area, especially using a practical hardware setup. Therefore, this thesis explores various aspects of spatial ANC, and seeks to develop algorithms and techniques to promote the performance and feasibility of spatial ANC in real-life applications. We use the spherical harmonic analysis technique as the basis for our research in this work. This technique provides an accurate representation of the spatial noise field, and enables in-depth analysis of the characteristics of the noise field. Incorporating this technique into the design of spatial ANC systems, we developed a series of algorithms and methods that optimizes the spatial ANC systems, towards both improving noise reduction performance and reducing system complexity. Several contributions of this work are: (i) design of compact planar microphone array structures capable of recording 3D spatial sound fields, so that the noise field can be monitored with minimum physical intrusion to the quiet zone, (ii) derivation of a Direct-to-Reverberant Energy Ratio (DRR) estimation algorithm which can be used for evaluating reverberant characteristics of a noisy environment, (iii) propose a few methods to estimate and optimize spatial noise reduction of an ANC system, including a new metric for measuring spatial noise energy level, and (iv) design of an adaptive spatial ANC algorithm incorporating the spherical harmonic analysis technique. The combination of these contributions enables the design of compact, high performing spatial ANC systems for various applications

Spatial Noise-Field Control With Online Secondary Path Modeling: A Wave-Domain Approach

Due to strong interchannel interference in multichannel active noise control (ANC), there are fundamental problems associated with the filter adaptation and online secondary path modeling remains a major challenge. This paper proposes a wave-domain adaptation algorithm for multichannel ANC with online secondary path modelling to cancel tonal noise over an extended region of two-dimensional plane in a reverberant room. The design is based on exploiting the diagonal-dominance property of the secondary path in the wave domain. The proposed wave-domain secondary path model is applicable to both concentric and nonconcentric circular loudspeakers and microphone array placement, and is also robust against array positioning errors. Normalized least mean squares-type algorithms are adopted for adaptive feedback control. Computational complexity is analyzed and compared with the conventional time-domain and frequency-domain multichannel ANCs. Through simulation-based verification in comparison with existing methods, the proposed algorithm demonstrates more efficient adaptation with low-level auxiliary noise.DP14010341