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Current sensor with digitized output signal
Hanuš, Ivan ; Vágner, Martin (referee) ; Havránek, Zdeněk (advisor)
The objective of this thesis is to describe principles of measuring of alternating current in low voltage power supply network with Rogowski coil and to study details of requirements for electronic current sensor with digital output. Furthermore to study the current solution of digitalization of output signal from Rogowski coil and to determine metrological characteristics of the digitalization module for processing the signal from the Rogowski coil as current sensor and to design a concept of an electronic current sensor unit with Rogowski coil and the digitalization module with the expected optical output. The theoretical part deals with the measurement of current using Rogowski coil, its characteristics and advantages compared to conventional current transformers. Further in the thesis there are specified the requirements for current sensor. The practical part deals with the design of digitalization module consisting of a pair of AD converters, a microcontroller and output circuits. Furthermore, the thesis deals with digital signal processing of signal from Rogowski coil. Furthermore, the thesis describes firmware for microcontroller and PC application. The result of thesis is functional circuit of electronic current sensor that processes signal from Rogowski coil and converts it to digital form.
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Digital signal processing from MEMS microphones
Koníček, Cyril ; Pristach, Marián (referee) ; Fujcik, Lukáš (advisor)
Proposed bachelor thesis is focused to the filter design for digital signal processing from digital microphone (MEMS type). The microphone types, structure of AMDP421 digital microphone and digital filter types are described in details. Selected filters are modelled, designed and simulated in MATLAB tool to meet the filter specification. The results are sufficiently commented and assessed. Afterwards the digital filters are described in VHDL language and implemented to the target FPGA circuit. In frame of the work the extended modules to SPARTAN 3 STARTER KIT are developed and designed to evaluate and demonstrate the proper function of the proposed filter design.
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Drum Machine for Musicians
Spáčil, Tomáš ; Fedra, Zbyněk (referee) ; Povalač, Aleš (advisor)
The aim of this work is to create a functional prototype of a drum machine with sound generating using a microcontroller. Introduction is devoted to general drum machine and an analysis of its parts, principle and basic modes of operation. The following chapter deals with the principle of digital signal processing and sound generation using a microcontroller. Part of this chapter is focused on mixing using compression methods. The core of this work is to design my own block structure and overall scheme of the drum machine with a description of the parts. Followed by analysis and implementation with theoretical and practical point of view. It is mainly about presentation and description of important parts of firmware codes, DPS design and another technical documentation useful for final prototype production. The conclusion contains a summary of the results with prezentation and discussion.
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Digital signal processing in real time
Zamazal, Zdeněk ; Mačák, Jaromír (referee) ; Rášo, Ondřej (advisor)
This work deals with digital signal processing in the field of adaptive filtering. Fundamental basics of adaptive filtering are described and primary aim is to create executable laboratory examples, using adaptive filtering, in LabView programming language. These laboratory examples are intended to be used by students fo studying and during laboratory lessons. Objective is to connect the examples with external devices, such as microphone. A microphone is used as an user's speech input acquiring interface. In the thesis is depicted Wiener's filter and problem of adaptive filtering is discussed. Contemporary adaptive algorithms are described and their applications as well. Most mentioned is the LMS algorithm and it's forms. Laboratory examples use following concepts: Adaptive Echo Cancellation, Active Noise Control and System Identification. Each of these examples is solely executable (need for LabView or Run-time engine), consisting also of theory with diagrams. Examples therefore are usable even without manual.
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Objective assessment and reduction of noise in musical signal
Rášo, Ondřej ; Makáň, Florian (referee) ; Krejčí, Jiří (referee) ; Balík, Miroslav (advisor)
The dissertation thesis focuses on objective assessment and reduction of disturbing background noise in a musical signal. In this work, a new algorithm for the assessment of background noise audibility is proposed. The listening tests performed show that this new algorithm better predicts the background noise audibility than the existing algorithms do. An advantage of this new algorithm is the fact that it can be used even in the case of a general audio signal and not only musical signal, i.e. in the case when the audibility of one sound on the background of another sound is assessed. The existing algorithms often fail in this case. The next part of the dissertation thesis deals with an adaptive segmentation scheme for the segmentation of long-term musical signals into short segments of different lengths. A new adaptive segmentation scheme is then introduced here. It has been shown that this new adaptive segmentation scheme significantly improves the subjectively perceived quality of the musical signal from the output of noise reduction systems which use this new adaptive segmentation scheme. The quality improvement is better than that achieved by other segmentation schemes tested.
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Advanced Algorithms for Satellite Communication Signal Processing
Záplata, Filip ; Bezoušek, Pavel (referee) ; Šimša, Jan (referee) ; Kasal, Miroslav (advisor)
Dizertační práce je zaměřena na softwarově definované přijímače určené k úzkopásmové družicové komunikaci. Komunikační kanály družicových spojů zahrnujících komunikaci s hlubokým vesmírem jsou zatíženy vysokými úrovněmi šumu, typicky modelovaného AWGN, a silným Dopplerovým posuvem signálu způsobeným mimořádnou rychlostí pohybu objektu. Dizertační práce představuje možné postupy řešení výpočetně efektivní digitální downkonverze úzkopásmových signálů a systému odhadu kmitočtu nosné úzkopásmových signálů zatížených Dopplerovým posuvem v řádu násobků šířky pásma signálu. Popis navrhovaných algoritmů zahrnuje analytický postup jejich vývoje a tam, kde je to možné, i analytické hodnocení jejich chování. Algoritmy jsou modelovány v prostředí MATLAB Simulink a tyto modely jsou využity pro ověření vlastností simulacemi. Modely byly také využity k experimentálním testům na reálném signálu přijatém z družice PSAT v laboratoři experimentálních družic na ústavu radioelektroniky.
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Homology search using digital signal processing methods
Kamar, Yana ; Jugas, Robin (referee) ; Maděránková, Denisa (advisor)
Thesis includes the theoretical introduction to molecular biology and genetics on the necessary level, including a description of the structure of DNA and the homologous gene. Described are fixed and physic-chemical kinds of nucleotide mapping, methods for processing digital signals. Numerical representations of genes that were programmed in MATLAB: unwrapped and accumulated phases, density vectors. Using the unwrapped phase and density vectors with windows of different lengths was performed CDS searching in the entire genome by calculation metric distances (euclidean and canberian) and correlation. Also, using the metric distances, a homologous gene was found in more or less similar bacterial genomes. The result is the approximate threshold of distance (euclidean and canberian) using to find homologous genes in genome.
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Analysis of Expressive Music Performance using Digital Signal Processing Methods
Fančal, Petr ; Mekyska, Jiří (referee) ; Smékal, Zdeněk (advisor)
The aim of this master's thesis is the analysis of musical compositions from the standpoint of time resources of music. The introduction briefly describes the basic musicological terms and variables that are in direct relationship to the time resources in expressive music performance. The following part of the work is devoted to the known methods of digital signal processing, suitable for music information retrieval from audio recordings. In the practical part these methods are demonstrated on three recordings in MATLAB environment and the results were compared in terms of used agogics.
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Advanced speech coding methods using digital signal processor
Zajíček, Marek ; Sysel, Petr (referee) ; Smékal, Zdeněk (advisor)
This master thesis describes the practical usage of AMR-WB (Adaptive Multi Rate - Wide Band) codec and its implementation on a digital signal processor which is integrated in functional voice communication system Siemens HiPath 4000. The first part is focused on the complete codec description, especially on an encoder and decoder. The second part partly describes signal processors and then is followed by the practical part of the implementation which is solved from the preliminary activities up to the optimalization of the final functional solution.
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