A robust sequential hypothesis testing method for brake squeal localisation

This contribution deals with the in situ detection and localisation of brake squeal in an automobile. As brake squeal is emitted from regions known a priori, i.e., near the wheels, the localisation is treated as a hypothesis testing problem. Distributed microphone arrays, situated under the automobile, are used to capture the directional properties of the sound field generated by a squealing brake. The spatial characteristics of the sampled sound field is then used to formulate the hypothesis tests. However, in contrast to standard hypothesis testing approaches of this kind, the propagation environment is complex and time-varying. Coupled with inaccuracies in the knowledge of the sensor and source positions as well as sensor gain mismatches, modelling the sound field is difficult and standard approaches fail in this case. A previously proposed approach implicitly tried to account for such incomplete system knowledge and was based on ad hoc likelihood formulations. The current paper builds upon this approach and proposes a second approach, based on more solid theoretical foundations, that can systematically account for the model uncertainties. Results from tests in a real setting show that the proposed approach is more consistent than the prior state-of-the-art. In both approaches, the tasks of detection and localisation are decoupled for complexity reasons. The localisation (hypothesis testing) is subject to a prior detection of brake squeal and identification of the squeal frequencies. The approaches used for the detection and identification of squeal frequencies are also presented. The paper, further, briefly addresses some practical issues related to array design and placement. (C) 2019 Author(s)

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

Implementation and evaluation of a low complexity microphone array for speaker recognition

Includes bibliographical references (leaves 83-86).This thesis discusses the application of a microphone array employing a noise canceling beamforming technique for improving the robustness of speaker recognition systems in a diffuse noise field

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

Prototype mixed-signal hardware for public safety radio interoperability

In performing their required duties public safety personnel from differing departments often need to communicate with one another using their in-car radios. However, in many cases, especially involving small departments, this interoperability doesn\u27t exist. A prototype circuit design has been developed and tested within the laboratory using two common radio systems: EFJohnson and Motorola. The preliminary results have shown successful operation as a system gateway between the two radio systems with good performance regarding audio signal latency and minimizing the push-to-talk signal generation delay

Software-Defined Radio Demonstrators: An Example and Future Trends

Software-defined radio requires the combination of software-based signal processing and the enabling hardware components. In this paper, we present an overview of the criteria for such platforms and the current state of development and future trends in this area. This paper will also provide details of a high-performance flexible radio platform called the maynooth adaptable radio system (MARS) that was developed to explore the use of software-defined radio concepts in the provision of infrastructure elements in a telecommunications application, such as mobile phone basestations or multimedia broadcasters

The Effects of Active Flow Control on High-Speed Jet Flow Physics and Noise

The work to be presented focuses on the noise generation of a fully turbulent, compressible jet flow within a large scale anechoic chamber. The investigations are aimed at understanding the complex nature of the jet flow field in an effort to reduce the far-field noise through active flow control and novel reduced-order modeling. The flow field of a highly subsonic, axisymmetric jet with a nozzle diameter of two inches (50.8 mm), is probed through the implementation of two-component particle image velocimetry (PIV) in the streamwise plane, along the jet\u27s centerline. These measurements are coupled with simultaneously sampled near and far-field pressure measurements, in an effort to understand the relationship between the complex flow field in the near region of the jet and large pressure fluctuation in the far-field, responsible for the noise. In order to reduce these large pressure fluctuations in the acoustic field, it is imperative to first understand the interaction of structures in the flow field and evaluate how this relates to the propagation of acoustic signatures to the far-field. We seek to establish a low-dimensional representation of the nonlinear, turbulent flow field through the implementation of reduced-order modeling in the form of proper orthogonal decomposition. In the first set of experiments conducted, active flow control is employed in the form of synthetic jet actuation at the nozzle lip, based on previous investigations. The effects of the flow control are observed using large-window PIV and far-field pressure measurements. The results suggest that an order epsilon input elicits an order one response, with both open and closed-loop flow control. While no noise reductions are seen in the far-field as compared to the uncontrolled jet, control authority over the jet is observed. The flow control greatly enhances mixing, thus reducing the length of the potential core and causing shear layer expansion. The second set of experiments involves the implementation of a time-resolved PIV system to effectively capture the temporal evolution of the flow physics in the streamwise plane. Low-dimensional velocity modes are directly correlated to low-dimensional acoustic modes in the far-field, using the observable inferred decomposition. Preliminary findings suggest that a small subset of low-dimensional velocity modes greatly contribute to the far-field acoustics. The spatiotemporal nature of these loud modes are investigated in the context of potential noise-producing events. It has been found that for the Mach 0.6 uncontrolled jet, focusing on the region near the collapse of the potential core, modes 6 and 14 appear to be the loud modes, contributing significantly to the far-field noise. Further exploration of mode 6 reveals a unique interaction of structures at very specific instances in time. Thus, it is concluded that from a low-dimensional viewpoint, we have identified the deterministic spatial structures in the velocity that most highly contributes to the noise in the far-field. It is possible from this analysis to begin to identify noise-producing events and examine these interactions in both time and space. Lastly, loud modes are identified for the controlled jet (using time-resolved PIV), however initial findings imply that the control greatly increases the complexity of the problem. Despite this fact, it is found that there may be similarities in the spatial structure of the loud modes for two different closed-loop control cases. In any case, through the use of active flow control and reduced-order modeling, preliminary steps have been taken to understand the sources of jet noise with respect to the flow physics, in an overall effort to efficiently achieve far-field noise reductions for practical applications