National Repository of Grey Literature 202 records found  beginprevious182 - 191nextend  jump to record: Search took 0.00 seconds. 
Ultra low noise switching power supply
Raba, Jaroslav ; Leuchter,, Jan (referee) ; Šrámek, Petr (advisor)
This thesis deals with the issue of switching power supply in terms of noise ratios in the output voltage. In the theoretical part analyzes the basic principle of classical and resonant topology switching power converters in terms of output voltage ripple and switching noise. There are also some possibilities of compensation to minimize switching noise and output ripple. The second part describes the custom design a low noise switching power supply, which is composed of an active power factor corrector, power converters and ripple limiter. The main power converter is designed as an LLC resonant converter. The last part deals with the actual construction of the proposed source, its measurement and presentation of the results.
Design of 10-channel audio equalizer with optimization of frequency characteristic and spectral audio analyzer
Štěrba, Václav ; Dvořák, Radek (referee) ; Šotner, Roman (advisor)
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.
LPC Speech Coding
Zapletal, Ondřej ; Kyselý, František (referee) ; Rajmic, Pavel (advisor)
The contents of the thesis "LPC speech coding" are studies of this method of a parametric source coding, explanation of mathematical procedures that are used in it (linear prediction, autocorrelation, Levinson-Durbin algorithm, transfer to a form suitable for transmission, Chebyshev root searching polynomial method) and acquaintance with the signification and application of that method in real speech encoders. The task of the original project of this thesis is a description and simulation of a simple speech encoder based on LPC, which transforms a real speech signal into a bit flow, which contains all of the significant parameters for its backward reconstruction (LSF coefficients, pitch period, excitation level, voice detection - AMDF method). One part of this thesis is a discussion about currently used speech encoders.
Spectral impuls signal analyzis
Měcháček, Radek ; Kubásek, Radek (referee) ; Fiala, Pavel (advisor)
Analysis signals, especially then pulsed signals and their next processing, today alreadyquaere to many different technological line. From communication engineering over telecommunication and acoustics, after as much as field - club application or multimedia. This work is introduction to the problems methods processing pulsed signals. Is distributive how from the standpoint of theory, so from look future application usage, by the help of programmatic analyses basic signals. Work bears basic description, notions, routes and method for work and analysis with widthspectral and ultraswidthspectral signals. Further be created tool for generation, filtration and near processing of several basic pseudo - random signals.
Adaptive filtering of ECG Signals
Nejezchleba, Zdeněk ; Kozumplík, Jiří (referee) ; Smital, Lukáš (advisor)
The aim was to test the methods for suppression 50 Hz noise with adaptive filtering. When using the general scheme of adaptive and deterministic scheme to suppress hum. The work is a theoretical derivation of adaptive algorithms and some examples of modeling in MATLAB.
Multimedia support of the course BSIS
Pasečný, Jan ; Šebesta, Vladimír (referee) ; Sigmund, Milan (advisor)
This paper takes aim at creating a consistent form of study materials, supplemented with illustrative examples, for Signals and systems subject. The thesis starts with basic characteristics of acoustic, image, biological and communication signals. Characteristics of linear signals and AD&DA conversion has been added to the next part and to complete the submission, discrete signals follow. Diploma thesis as a whole contains basic theoretical description of problematics, which it tries to supplement with interesting examples, connections, graphs and matlab scripts for illustrative presentation of mentioned problematics.
Segmentation techniques for image processing
Bajgar, Aleš ; Číka, Petr (referee) ; Říha, Kamil (advisor)
For image segmentation are used several methods. Good results can be achieved by the correct choice of method. Best results are using more methods together. Methods such as thresholding, the search area or methods founded on edge detection in an image. We can use different filters for noise removal. For example Gaussian, median, etc. Lowpass filter can also be used. Image segmentation Mean Shift, K-mean, watershed and pyramid segmentation performed by using methods from the OpenCV library.
Identification of pauses in noisy speech signal
Kepák, Petr ; Míča, Ivan (referee) ; Smékal, Zdeněk (advisor)
The basic problem of speech is a complete separation of the natural noise which arise from correct articulation of voiced and unvoiced consonants from noise and disturbance environment. Objective of this master’s thesis is to find an effective method that could identify the pauses without speech activity, which can identify the properties of noise and disturbance. Once the noise is correctly identified, it is already possible to use different methods for its removal. The master’s thesis describes two methods of pauses identification. These methods are programmed in Matlab and tested on nine speech recordings. Methods analysis of the results was performed using the ROC (Receiver Operating Characteristic) curves. In the end are summarized results analysis of created methods.
Pure tone audiometer
Blahák, Petr ; Rampl, Ivan (referee) ; Chmelař, Milan (advisor)
Human hearing is to collect information from the outside world and is one of the basic human senses. Part of this thesis is devoted to acoustics, properties of the human ear in terms of perception of sounds and methods, which human ears are investigating. Audiometer is an instrument which is most often used in healthcare. The main content of this thesis is to design pure tone audiometer, which is important for the subjective tests of human hearing non-invasive method.
Modern coding of speech signals using overcomplete models
Zapletal, Ondřej ; Průša, Zdeněk (referee) ; Rajmic, Pavel (advisor)
The theoretical contents of this thesis are studies of overcomplete models. Those are the models of signals, on which is set for their parametrization more variables, than it's necessary and consequently there's computed so-called sparse solution via iteration algorithms. A goal of this analysis is a selection just of the considerable (sparse) parameters. The theory is based on a linear algebra, vector spaces, bases and so-called frames. The task of the individual project of this thesis is a description and simulation of two speech coders: a classical coder based on linear predictive speech coding and a coder, that's making use of overcomplete stochastic ARMA processes models. A part of their realization is to simulate their decoders and a analyze their reconstruction quality. For their realization there is used MATLAB and an overcomplete models' library (toolbox frames).

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