Empirical models based on flow resistivity are commonly used to determine the bulk properties of porous sound absorbing materials. The bulk properties include the complex wavenumber and complex characteristic impedance which can be used directly in simulation models. Moreover, the bulk properties can also be utilized to determine the normal incidence sound absorption and specific acoustic impedance for sound absorbing materials of any thickness and for design of layered materials. The sound absorption coefficient of sound absorbing materials is measured in an impedance tube using wave decomposition and the measured data is used to determine the flow resistivity of the materials by least squares curve fitting to empirical equations. Results for several commonly used foams and fibers are tabulated to form a rudimentary materials database. The same approach is then used to determine the flow resistivity of compressed sound absorbing materials. The flow resistivities of the compressed materials are determined as a function of the compression ratio. Results are then used in conjunction with transfer matrix theory to predict the sound absorptive performance of layered compressed absorbers with good agreement to measurement.

Partial enclosures are commonly used to reduce machinery noise. However, it is well known in industry that enclosures sometimes amplify the sound at low frequencies due to strong acoustic resonances compromising the performance. These noise issues are preventable if predicted prior to prototyping and production. Though boundary and finite element approaches can be used to accurately predict partial enclosure insertion loss, modifications to the model require time for remeshing and solving. In this work, partial enclosure performance at low frequencies is simulated using a plane wave transfer matrix approach. Models can be constructed and the effect of design modifications can be predicted rapidly. Results are compared to finite element analysis and measurement with good agreement. The approach is then used to design and place resonators into a sample enclosure. Improvements in enclosure performance are predicted using plane wave simulation, compared with acoustic finite element analysis, and then validated via measurement.

Bibliography: p. 199-207. Electronic publication; full text available in PDF format; abstract in HTML format. Describes the basis of a theoretical and experimental project, directed at the design and evaluation of a practical active noise control system suitable for a single light engine aircraft. The performance of virtual sensors were evaluated both analytically and experimentally in progressively more complex environments to identify their capabilities and limitations. Electronic reproduction.[Australia] :Australian Digital Theses Program,2001.

Bibliography : leaves 237-251. / x, 251 leaves : ill. ; 30 cm. / Title page, contents and abstract only. The complete thesis in print form is available from the University Library. / Thesis (Ph.D.)--University of Adelaide, Dept. of Mechanical Engineering, 1994

In this thesis, we are interested in processing noisy speech signals that are meant to be heard by humans, and hence we approach the noise-suppression problem from a perceptual perspective. We develop a noise-suppression paradigm that is based on a model of the human auditory system, where we process signals in a way that is natural to the human ear. Under this paradigm, we transform an audio signal in to a perceptual domain, and processes the signal in this perceptual domain. This approach allows us to reduce the background noise and the audible artifacts that are seen in traditional noise-suppression algorithms, while preserving the quality of the processed speech. We develop a single- and dual-microphone algorithm based on this perceptual paradigm, and conduct subjecting tests to show that this approach outperforms traditional noise-suppression techniques. Moreover, we investigate the cause of audible artifacts that are generated as a result of suppressing the noise in noisy signals, and introduce constraints on the noise-suppression gain such that these artifacts are reduced.

Achieving a low noise level is of great interest to the forest machine industry. Traditionally this is obtained by using passive noise reduction, i.e. by using materials for sound isolation and sound absorption. Especially designs to attenuate low frequency noise tend to be bulky and impractical from an installation point of view. An alternative solution to the problem is to use active noise control (ANC). The basic principle of ANC is to generate an anti-noise signal designed to destructively interfere with the unwanted noise. In this thesis two algorithms (Feedback FxLMS and Feedforward FxLMS) are implemented and evaluated for use in the ANC-system. The ANC-system is tuned to the specific environment in the driver’s cabin of a Komatsu forest machine. The algorithms are first tested in a simulated environment and then in real-time inside a forest machine. Simulations are made both in Matlab and in C using both generated signals and recorded signals. The C code is implemented on the Analog Devices Blackfin DSP card BF526. The result showed a significantly reduction of the sound pressure level (SPL) in the driver’s cabin. The noise attenuation obtained using the Feedback FxLMS was approximately 14 dB for a tonal 100 Hz signal and 11 dB using recorded engine noise from a forest machine at 850 rpm.

Thesis--Technische Universität Darmstadt, 2008. / Includes bibliographical references.

Thesis (D.Tech.: Electronic Engineering)-Dept. of Electronic Engineering, Durban University of Technology, 2007 vii, 150 leaves / Detecting and minimising distortion in audio signals is an important aspect of sound engineering. Distortion of a signal passing through an audio system may be caused by a number of factors and it is necessary to detect these effects for optimal sound. The problem is of interest to users and operators of high quality audio equipment and transmission facilities. The objective of this thesis was the development of techniques for the blind identification of distortion in a high quality audio signal using digital signal processing techniques. The techniques developed are based on digital signal processing techniques and statistical analysis of a recorded audio signal, which is treated as a random, non-stationary signal.

The design of a battery-powered, portable headphone active noise control system with TI TMS320C5416 DSP is described. The preliminary implementation of the system on a C5416 DSK is also explained. The problems of fixed-point implementation are described and solutions are proposed. Sign-sign Fx-LMS algorithm with a dead-zone is introduced and used as the adaptation algorithm. Effective use of dynamic range to improve the accuracy in filtering operations is discussed. Details of the designed battery-powered DSP board are given and board software development process is explained. The DSK system and designed portable system is compared against two commercially available analog systems under three different types of noises / composition of tones, drill noise and propeller plane cabin noise. The results reveal that adaptive system has better overall performance.

The objective of our research is to develop system architectures and CMOS circuits that interface with high-Q silicon microgyroscopes to implement navigation-grade angular rate sensors. The MEMS sensor used in this work is an in-plane bulk-micromachined mode-matched tuning fork gyroscope (M² – TFG ), fabricated on silicon-on-insulator substrate. The use of CMOS transimpedance amplifiers (TIA) as front-ends in high-Q MEMS resonant sensors is explored. A T-network TIA is proposed as the front-end for resonant capacitive detection. The T-TIA provides on-chip transimpedance gains of 25MΩ, has a measured capacitive resolution of 0.02aF /√Hz at 15kHz, a dynamic range of 104dB in a bandwidth of 10Hz and consumes 400μW of power. A second contribution is the development of an automated scheme to adaptively bias the mechanical structure, such that the sensor is operated in the mode-matched condition. Mode-matching leverages the inherently high quality factors of the microgyroscope, resulting in significant improvement in the Brownian noise floor, electronic noise, sensitivity and bias drift of the microsensor. We developed a novel architecture that utilizes the often ignored residual quadrature error in a gyroscope to achieve and maintain perfect mode-matching (i.e.0Hz split between the drive and sense mode frequencies), as well as electronically control the sensor bandwidth. A CMOS implementation is developed that allows mode-matching of the drive and sense frequencies of a gyroscope at a fraction of the time taken by current state of-the-art techniques. Further, this mode-matching technique allows for maintaining a controlled separation between the drive and sense resonant frequencies, providing a means of increasing sensor bandwidth and dynamic range. The mode-matching CMOS IC, implemented in a 0.5μm 2P3M process, and control algorithm have been interfaced with a 60μm thick M2−TFG to implement an angular rate sensor with bias drift as low as 0.1°/hr ℃ the lowest recorded to date for a silicon MEMS gyro.