Analysis of a linear one-dimensional active noise control system by means of block diagrams and transfer functions

In arriving at the ideal filter transfer function for an active noise control system in a duct, the effect of the auxiliary sources (generally loudspeakers) on the waves generated by the primary source has invariably been neglected in the existing literature, implying a rigid wall or infinite impedance. The present paper presents a fairly general analysis of a linear one-dimensional noise control system by means of block diagrams and transfer functions. It takes into account the passive as well as active role of a terminal primary source, wall-mounted auxiliary source, open duct radiation impedance, and the effects of mean flow and damping. It is proved that the pressure generated by a source against a load impedance can be looked upon as a sum of two pressure waves, one generated by the source against an anechoic termination and the other by reflecting the rearward wave (incident on the source) off the passive source impedance. Application of this concept is illustrated for both the types of sources. A concise closed-form expression for the ideal filter transfer function is thus derived and discussed. Finally, the dynamics of an adaptive noise control system is discussed briefly, relating its standing-wave variables and transfer functions with those of the progressive-wave model presented here.

Dynamics of Cricket Sound Production

The clever designs of natural transducers are a great source of inspiration for man-made systems. At small length scales, there are many transducers in nature that we are now beginning to understand and learn from. Here, we present an example of such a transducer that is used by field crickets to produce their characteristic song. This transducer uses two distinct components-a file of discrete teeth and a plectrum that engages intermittently to produce a series of impulses forming the loading, and an approximately triangular membrane, called the harp, that acts as a resonator and vibrates in response to the impulse-train loading. The file-and-plectrum act as a frequency multiplier taking the low wing beat frequency as the input and converting it into an impulse-train of sufficiently high frequency close to the resonant frequency of the harp. The forced vibration response results in beats producing the characteristic sound of the cricket song. With careful measurements of the harp geometry and experimental measurements of its mechanical properties (Young's modulus determined from nanoindentation tests), we construct a finite element (FE) model of the harp and carry out modal analysis to determine its natural frequency. We fine tune the model with appropriate elastic boundary conditions to match the natural frequency of the harp of a particular species-Gryllus bimaculatus. We model impulsive loading based on a loading scheme reported in literature and predict the transient response of the harp. We show that the harp indeed produces beats and its frequency content matches closely that of the recorded song. Subsequently, we use our FE model to show that the natural design is quite robust to perturbations in the file. The characteristic song frequency produced is unaffected by variations in the spacing of file-teeth and even by larger gaps. Based on the understanding of how this natural transducer works, one can design and fabricate efficient microscale acoustic devices such as microelectromechanical systems (MEMS) loudspeakers.

Effect of polymeric properties on the operation of gel-type audio transducers.

A novel design of a moving-coil transducer coupled with a low-hardness elastomer called “the gel surround” is presented in this thesis. This device is termed a “gel-type audio transducer”. The gel-type audio transducer has been developed to overcome the problems that conventional loudspeakers have suffered - that is, the problem with size of the audio device against the quality of sound at low frequency range. Therefore the research work presented herein aims to develop the “gel-type audio transducer” as a next-generation audio transducer for miniaturized woofers. The gel-type audio transducer consists of the magnetic and coil-drive plate assembly, and these parts are coupled by the gel surround. The transducer is driven by the electromagnetic conversion mechanism (a moving-coil transducer) and its output driving force can be greatly enhanced by applying the novel mechanism of the gel surround especially at low frequency range, resulting in the enhanced acoustic efficiency. The transducer can be attached to a stiff and light panel with both the optimized impedance matching and minimised wave collisions. The performance of the gel-type audio transducer is greatly influenced by the mass of the magnetic assembly and compliance of the “gel surround”. But as the size of the magnet and its weight have to be kept minimal for a miniaturisation of the device, the focus of the research is on the effect of the of the gel surround. As a result, the effect of the gel surround, made of the RTV (room-temperature vulcanising) silicone elastomer, TPE (thermoplastic elastomer), and the silicone foam, on generation of the output driving force, the energy transfer from the transducer to a panel to which the transducer is attached, and sound radiation from the vibrating panel, was investigated. This effect was studied by COMSOL multiphysics (FE analysis) and thereby, the simulated results were verified by experiments such as the laser scanning measurement, DMA (dynamic mechanical analyzer), and the acoustic test. Successful development of prototypes of the gel-type audio transducers, with an enhanced acoustic efficiency at reduced size and weight, was achieved. Implementation of the transducers into consumer applications was also demonstrated with their commercial values.

Investigation on localisation accuracy for first and higher order ambisonics reproduced sound sources

Ambisonics and higher order ambisonics (HOA) technologies aim at reproducing sound field either synthesised or previously recorded with dedicated microphones. Based on a spherical harmonic decomposition, the sound field is more precisely described when higher-order components are used. The presented study evaluated the perceptual and objective localisation accuracy of the sound field encoded with four microphones of order one to four and decoded over a ring of loudspeakers. A perceptual test showed an improvement of the localisation with higher order ambisonic microphones. Reproduced localisation indices were estimated for the four microphones and the respective synthetic systems of order one to four. The perceptual and objective analysis revealed the same conclusions. The localisation accuracy depends on the ambisonic order as well as the source incidence. Furthermore, impairments linked to the microphones were highlighted.

Spectral impairment of enclosed higher order ambisonics

Ambisonics is spatial audio technique that attempts to recreate a physical sound field over as large an area as possible. Higher Order Ambisonic systems modelled with near field loudspeakers in free field as well as in a simulated room are investigated. The influence of reflections on the image quality is analysed objectively for both a studio-sized and large reproduction environment using the relative intensity of the reproduced sound field. The results of a simulated enclosed HOA system in the studio-sized room are compared to sound field measurements in the reproduced area.

Off-Centre Localisation Performance of Ambisonics and HOA For Large and Small Loudspeaker Array Radii

Ambisonics and Higher Order Ambisonics (HOA) are scalable spatial audio techniques that attempt to present a sound scene to listeners over as large an area as possible. A localisation experiment was carried out to investigate the performance of a first and third order system at three listening positions - one in the centre and two off-centre - using a 5 m radius loudspeaker array. The results are briefly presented and compared to those of an earlier experiment on a 2.2 m loudspeaker array. In both experiments the off-centre listeners were placed such that the ratio of distance from the centre to the array radius was constant in both experiments. The test used a reverse target-pointer adjustment method to determine the error, both signed and absolute, for each combination of listening position and system. The results for both arrays were found to be very similar, suggesting that the relative amplitude of the loudspeakers, which were the same in both cases, was more dominant for localisation than the arrival time differences, which differed between array sizes.

Investigation on the influence of the order on ambisonics reproduced sound scene over headphones

To investigate the performance of ambisonics systems reproduced over headphones, a pairwise comparison test was carried out. Binaurally reproduced sound scenes for 2D ambisonic orders 1 to 4 decoded on 2M + 2 virtual loudspeakers using two decoder options, basic and mixed basic and maxrE were used. Similarity ratings are obtained from pairwise comparisons between all of the combinations of systemsa and a binaural reference.

Proxemics: The World is a Deaf Machine

A multi-channel sound installation involving fixed loudspeakers and multiple mp3 players, commissioned by the Sainsbury Centre of Visual Arts, Norwich as a response to the sculpture of Ian Tyson and the architecture of Denys Lasdun.

Spatial hearing and sound perception in musical composition

This thesis explores the possibilities of spatial hearing in relation to sound perception, and presents three acousmatic compositions based on a musical aesthetic that emphasizes this relation in musical discourse. The first important characteristic of these compositions is the exclusive use of sine waves and other time invariant sound signals. Even though these types of sound signals present no variations in time, it is possible to perceive pitch, loudness, and tone color variations as soon as they move in space due to acoustic processes involved in spatial hearing. To emphasize the perception of such variations, this thesis proposes to divide a tone in multiple sound units and spread them in space using several loudspeakers arranged around the listener. In addition to the perception of sound attribute variations, it is also possible to create rhythm and texture variations that depend on how sound units are arranged in space. This strategy permits to overcome the so called "sound surrogacy" implicit in acousmatic music, as it is possible to establish cause-effect relations between sound movement and the perception of sound attribute, rhythm, and texture variations. Another important consequence of using sound fragmentation together with sound spatialization is the possibility to produce diffuse sound fields independently from the levels of reverberation of the room, and to create sound spaces with a certain spatial depth without using any kind of artificial sound delay or reverberation.

Adaptive speech enhancement with varying noise backgrounds

We present a new approach for speech enhancement in the presence of non-stationary and rapidly changing background noise. A distributed microphone system is used to capture the acoustic characteristics of the environment. The input of each microphone is then classified either as speech or one of the predetermined noise types. Further enhancement of speech in respective microphones is carried out using a modified spectral subtraction algorithm that incorporates multiple noise models to quickly adapt to rapid background noise changes. Tests on real world speech captured under diverse conditions demonstrate the effectiveness of this method.