Spatial dissection of a soundfield using spherical harmonic decomposition

A real-world soundfield is often contributed by multiple desired and undesired sound sources. The performance of many acoustic systems such as automatic speech recognition, audio surveillance, and teleconference relies on its ability to extract the desired sound components in such a mixed environment. The existing solutions to the above problem are constrained by various fundamental limitations and require to enforce different priors depending on the acoustic condition such as reverberation and spatial distribution of sound sources. With the growing emphasis and integration of audio applications in diverse technologies such as smart home and virtual reality appliances, it is imperative to advance the source separation technology in order to overcome the limitations of the traditional approaches. To that end, we exploit the harmonic decomposition model to dissect a mixed soundfield into its underlying desired and undesired components based on source and signal characteristics. By analysing the spatial projection of a soundfield, we achieve multiple outcomes such as (i) soundfield separation with respect to distinct source regions, (ii) source separation in a mixed soundfield using modal coherence model, and (iii) direction of arrival (DOA) estimation of multiple overlapping sound sources through pattern recognition of the modal coherence of a soundfield. We first employ an array of higher order microphones for soundfield separation in order to reduce hardware requirement and implementation complexity. Subsequently, we develop novel mathematical models for modal coherence of noisy and reverberant soundfields that facilitate convenient ways for estimating DOA and power spectral densities leading to robust source separation algorithms. The modal domain approach to the soundfield/source separation allows us to circumvent several practical limitations of the existing techniques and enhance the performance and robustness of the system. The proposed methods are presented with several practical applications and performance evaluations using simulated and real-life dataset

Accurate aeroacoustic measurements in closed-section hard-walled wind tunnels

Noise emissions from aircraft are of major concern to aircraft manufacturers. There are various analytical, empirical and numerical tools to help in the design of quieter aircraft, however aeroacoustic measurements in wind tunnels are still required. There is a growing interest in simultaneous aerodynamic and aeroacoustic measurements in hard-walled closed-section wind tunnels. The research hypothesis of this work is whether accurate aeroacoustic measurements are possible in this type of wind tunnel. Two issues are of particular concern: the reverberation sound field and high background noise levels. De-reverberation, based on an Image Source Model (ISM), is proposed to tackle the first issue by incorporating the reflections in the focused beamformer. This technique is computationally fast and easy to implement. Source Power Integration and deconvolution techniques are shown to be still valid in de-reverberation. Measurements in a closed section wind tunnel have shown that an ISM gives a better estimate of the Green's functions, when compared to free-space Green's functions. Furthermore de-reverberation yielded more accurate source strength estimates from the beamformer. Qualitatively, de-convolved results were no different than when using free-space Green's functions. Simulations have shown that the ISM can become unstable at high frequencies if position errors are present. It is therefore recommended to limit the application of the ISM to frequencies below 10 kHz. At low frequencies the accuracy of beamforming levels is highly dependent on the level of noise contamination of the input data. Removing the diagonal of the cross spectral matrix might not be sufficient to eliminate this noise

A study into the design of steerable microphones arrays

Beamforming, being a multi-channel signal processing technique, can offer both spatial and temporal selective filtering. It has much more potential than single channel signal processing in various commercial applications. This thesis presents a study on steerable robust broadband beamformers together with a number of their design formulations. The design formulations allow a simple steering mechanism and yet maintain a frequency invariant property as well as achieve robustness against practical imperfectio

Sparse Array Design for Wideband Beamforming with Reduced Complexity in Tapped Delay-lines

Sparse wideband array design for sensor location optimization is highly nonlinear and it is traditionally solved by genetic algorithms (GAs) or other similar optimization methods. This is an extremely time-consuming process and an optimum solution is not always guaranteed. In this work, this problem is studied from the viewpoint of compressive sensing (CS). Although there have been CS-based methods proposed for the design of sparse narrowband arrays, its extension to the wideband case is not straightforward, as there are multiple coefficients associated with each sensor and they have to be simultaneously minimized in order to discard the corresponding sensor locations. At first, sensor location optimization for both general wideband beamforming and frequency invariant beamforming is considered. Then, sparsity in the tapped delay-line (TDL) coefficients associated with each sensor is considered in order to reduce the implementation complexity of each TDL. Finally, design of robust wideband arrays against norm-bounded steering vector errors is addressed. Design examples are provided to verify the effectiveness of the proposed methods, with comparisons drawn with a GA-based design method

Speech Enhancement using Multiple Transducers

In this thesis, three methods of speech enhancement techniques are investigated with applications in extreme noise environments. Various beamforming techniques are evaluated for their performance characteristics in terms of signal to (distant) noise ratio and tolerance to design imperfections. Two suitable designs are identified with contrasting performance characteristics — the second order differential array, with excellent noise rejection but poor robustness; and a least squares design, with adequate noise rejection and good robustness. Adaptive filters are introduced in the context of a simple noise canceller and later a post-processor for a dual beamformer system. Modifications to the least mean squares (LMS) filter are introduced to tolerate cross-talk between microphones or beamformer outputs. An adaptive filter based post-processor beamforming system is designed and evaluated using a simulation involving speech in noisy environments. The beamforming methods developed are combined with the modified LMS adaptive filter to further reduce noise (if possible) based on correlations between noise signals in a beamformer directed to the talker and a complementary beamformer (nullformer) directed away from the talker. This system shows small, but not insignificant, improvements in noise reduction over purely beamforming based methods. Blind source separation is introduced briefly as a potential future method for enhancing speech in noisy environments. The FastICA algorithm is evaluated on existing data sets and found to perform similarly to the post-processing system developed in this thesis. Future avenues of research in this field are highlighted