Background noise is particularly damaging to speech intelligibility for people with hearing loss. The problem of reducing noise in hearing aids is one of great importance and great difficulty. Over the years, many solutions and different algorithms have been implemented in order to provide the optimal solution to the problem. Beamforming has been used for a long time and has therefore been extensively researched. Studying the performance of Minimum Variance Distortionless Response (MVDR) beamforming with a three- and four- microphone array compared to the conventional two-microphone array, the aim is to implement a speech signal enhancement and a noise reduction algorithm. By using multiple microphones, it is possible to achieve spatial selectivity, which is the ability to select certain signals based on the angle of incidence, and improve the performance of noise reduction beamformers. This thesis proposes the use of beamforming, an existing technique in order to create a new way to reduce noise transmitted by hearing aids. In order to reduce the complexity of that system, we use hybrid cascades, which are simpler beamformers of two inputs each and connected in series. The configurations that we consider are a three-microphone linear array (monaural beamformer), a three-microphone configuration with a two-microphone linear array and the 3rd microphone in the ear (monaural beamformer), a three-microphone configuration with a two-microphone linear array and the 3rd microphone on contra-lateral ear (binaural beamformer), and finally four-microphone configurations. We also investigate the performance improvement of the beamformer with more than two microphones for the different configurations, against the two-microphone beamformer reference. This can be measured by using objective measurements, such as the amount of noise suppression, target energy loss, output SNR, speech intelligibility index and speech quality evaluation. These objective measurements are good indicators of subjective performance. In this project, we prove that most hybrid structures can perform satisfyingly well compared to the full complexity beamformer. The low complexity beamformer is designed with a fixed target location (azimuth), where its weights are calibrated with respect to a target signal located in front of the listener and for a diffuse noise field. Both second- and third- order beamformers are tested in different acoustic scenarios, such as a car environment, a meeting room, a party occasion and a restaurant place. In those scenarios, the target signal is not arriving at the hearing aid directly from the front side of the listener and the noise field is not always diffuse. We thoroughly investigate what are the performance limitations in that case and how well the different cascades can perform. It is proven that there are some very critical factors, which can affect the performance of the fixed beamformer, concerning all the hybrid structures that were examined. Finally, we show that lower complexity cascades for both second- and third- order beamformers can perform similarly well as the full complexity beamformers when tested for a set of multiple Head Related Transfer Functions (HRTFs) that correspond to a real head shape.

The blind source separation problem has received a great deal of attention in previous years. The aim of this problem is to estimate a set of original source signals from a set of linearly mixed signals through any number of signal processing techniques. While many methods exist that attempt to solve the blind source separation problem, a new technique is being used that uniquely separates audio sources as they are received from a microphone array. In this thesis a new algorithm is proposed that that utilizes the ICA algorithm in conjunction with a filtering technique that separates source signals and then removes sources of interference so that a signal of interest can be accurately tracked. Experimental results will compare a common blind source separation technique to the new algorithm and show that the new algorithm can detect a signal of interest and accurately track it as it moves through an anechoic environment. / Master of Science

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Popularity of speech communication is rapidly increasing in various contexts such as conferencing systems, mobile/fixed electronic devices and laptops thus leading to a heightened demand for new services and improved speech quality. Dictaphones used for dictations usually have one microphone. Single microphone does not give enough degree of freedom to allow estimation of location of the source. Microphone array makes use of multiple microphones for spatial filtering suppressing the background noise. This report aims for speech enhancement utilizing the benefits inherited with microphone arrays to find direction of desired speaker and focus the listening beam in that direction. A comparison is made between Generalized Cross Correlation (GCC) methods for locating the source in real office environment. Beamforming is implemented to make the microphone array listen in the desired direction thus reducing the interference from other sources. Minimum Variance Distortion-less Response (MVDR) approach is shown to give better results compared to more simplistic techniques. Perceptual based Eigen filter incorporating human hearing models in subspace incorporated in the suppressor eliminates the residual noise. Objective system performance is evaluated by estimating Signal-to-Noise-Ratio improvement (SNRI), segmental SNR, signal degradation and noise suppression. Perpetual Evaluation of Speech Quality (PESQ) gives Mean Opinion Score for subjective evaluation. / asim_zolo@yahoo.com, akbarali45@gmail.com

Ultrasonic guided wave imaging methods offer a cost-effective mechanism to perform in situ structural health monitoring (SHM) of large plate-like structures, such as commercial aircraft skins, ship hulls, storage tanks, and civil structures. However, current limits in imaging quality, environmental sensitivities, and implementation costs, among other things, are preventing widespread commercial adoption. The research presented here significantly advances state of the art guided wave imaging techniques using inexpensive, spatially distributed arrays of piezoelectric transducers. Novel adaptive imaging techniques are combined with in situ estimation and compensation of propagation parameters; e.g., dispersion curves and transducer transfer functions, to reduce sensitivity to unavoidable measurement inaccuracies and significantly improve resolution and reduce artifacts in guided wave images. The techniques can be used not only to detect and locate defects or damage, but also to characterize the type of damage. The improved ability to detect, locate, and now characterize defects or damage using a sparse array of ultrasonic transducers is intended to assist in the establishment of in situ guided wave imaging as a technically and economically viable tool for long-term monitoring of plate-like engineering structures.

The main body of this work concerns cultural as in Vlašim after the second world war from 1945 to 1989. There are four specific groups that can be categorized as cultural associations; a nature group Mr. Vlasak group, Blaník Vlašim Choir and an amateur theatre group and friends of fine arts Vlašim. The first three clubs date comfortably within our choose era and they are important to deepen our knowledge of cultural societies of that time. To understand the history of these various cultural societies in Vlašim, it is necessary to broader our perspective. When we see these cultural groups in the context of the development of federal activities, we can see how the town was guided in its development. The majority of the research has been done using archival material from Museum Podblanicka, the Prague Regional Archive and also the Benešov State Archive. Added to this were other written historical accounts of the time as well as interviews conducted with knowledgable first hand sources. Powered by TCPDF (www.tcpdf.org)