This paper deals with study of human movement and using that in identification. In the first part of my work are explained characteristics of human movement and factors which take effect on these characteristics. Practical part is dealing with design of program which is solving mentioned problems. The input of program is created by video sequence with lateral movement of human. The program is finding coordinates of lower limbs joints. From this coordinates are created locomotion characteristics used for human identification. Matching of time behaviors is based on correlation.

The aim of the master thesis is to sum up theoretical information about beamforming methods of microphone arrays and to verify their functionality. At the beginning of this work there are simulated different varietes of linear uniform and nonuniform microphone arrays and circular arrays. The results are verificated by a practical measurement in ideal conditions. Then I will focuse on implementation of the DAS(Delay And Sum), SAB(Sub Array Beamforming), CDB(Constant Directivity Beamforming), CDB-CA(CDB-Circular Arrays) beamformer including theoretical and practical verification of the functionality in ideal conditions. At the end of this thesis are all beamforming methods compared with each other at SNR(signal to Noise Ratio) and directivity parameters.

This master's thesis is focused on localization static (entering semester project) and moving acoustic sources (entering master's thesis) by the help of microphonic arrays. In the first part deal with common problems of localization. Further are here described types of microphonic arrays, simplifying possibilities which delimited this problems and general information about room acoustics. In the next part of this master's thesis are step by step mentioned methods localization of acoustic sources. In practical part were used algorithms: Steered-Beamformer-Based Locators and TDOA-Based Locators. Last part of this master's work includes results of these algorithms.

The main objective of this study is to learn about the audiovisual stimulator and to create hardware resolution of stimulation LED glasses and in environment of the program LabView application, which operate this stimulation LED glasses and in the same time create sound of stimulation. Use environment of the program LabView. Application, which is create in environment of the program LabView, enable operate stimulation LED glasses and arrange sound from three source with two different method, which use modern AVS. Application contains a lot of security, informative and agreement components.

The goal of this master’s thesis is to explore the possibilities of multichannel localization of acoustic signal sources and their following application on a real signal localization and separation, using Beamforming methods. During this thesis two beamforming methods were selected, namely Delay and Sum a Constant Directivity Beamforming - Circular Arrays, and were applicated on real environment signals using two microphone arrays’ geometries ULA (Uniform linear array) and UCA (Uniform Circular array).

The aim of this work is to create an automated workplace for measuring the basic parameters of audio amplifiers using a graphical programming environment LabVIEW. Subsequently, the functionality will be verified in practical measurements on the real amplifier.

In mentioned theses there are described the basic tenets of the photovoltaic panels operation. There is described the progress of the photovoltaic panels. Theses is focused on the theoretical knowledges about the equivalent circuits and their aplication in the proposed simulation system. The goal of theses is to create the surroundings in the program Agilent VEE Pro 8.0 and to create the program for the simulation of the serial and paralel conjunction of the photovoltaic panels. The resultant program enables the simulation of the I – V and P – V characteristics of the real photovoltaic panels.

This thesis describes the design of the active load for adjusting the maximum power point of PV module and the module loaded with the defined parameters for measuring the effectiveness and identifying the characteristics of the PV module.

This work deals with the design of a 10-zones equalizer with optimized frequency characteristics with a spectrum audio analyzer. In this work the problem of processing audio signals using equalization for filtering the interference frequencies, correction of frequency cover signal boost or suppression of the required zones of the audible spectrum are also analyzed. The influence of subjective perception of sound intensity of the audio signals reproduction and its use in working with equalizer is discussed too. The work describes the principles and usage of the audio-analyzer as a tool for the optimization of the audio equalization setting when ensuring the appropriate listening conditions of music reproduction, spoken word, sounds, etc. It also focuses on the signal source for testing audio-chains, their generation and measurement using the audio analyzer. The equalizer equipment, audio-analyzer generator of reference signals equipment and power supply are designed as a single unit.

My diploma thesis describes upgrade of flowbench serving as test rig for regulation valves characteristic measurement. At the beggining, there is an introduction to the water regulation valves theory followed by current machine condition description. Next chapter defines needed improvements and at the end follows proposition of improvements and necessary component definition and final realization. Final chapter summarizes complete project.