National Repository of Grey Literature 100 records found  beginprevious70 - 79nextend  jump to record: Search took 0.00 seconds. 
Application for Measurement of Threshold in Quiet and Masking Curves
Bednář, Jan ; Sysel, Petr (referee) ; Schimmel, Jiří (advisor)
Main aim of this thesis is to create an application for measuring threshold in quiet and masking curves in C++ language. First chapter describes basic teory of perceiving sound and phenomenon of masking. Second chapter focuses on description of different hearing measurement methods. Third chapter describes, how the classes and functions for frequency changing oscilator and evaluation were created and how the graph display function works. Next chapter describes, how to use the application properly and how to do the basic calibration, so the data will be displayed correctly. Last chapter shows the correct function, measured thresholds in quiet and masking curves for two subjects.
Secured audio codec for Asterisk PBX
Jakubíček, Michal ; Sysel, Petr (referee) ; Krajsa, Ondřej (advisor)
This thesis is focused on the design of secured audio codec for Asterisk PBX. The first chapter is focused on the basic division of traditional PBX producers and the open source PBX. The second chapter explains the structure of Asterisk PBX and its fundamental difference from a traditional PBX. Asterisk is based on components called modules, therefore the work also deals with the most important modules for operation of exchanges and their division of terms of support and dividing by the type of application and their properties. In this chapter there are described in more detail audio codec A-law and u-law. The third chapter contains simple instructions to get your orientation in the construction of the module for Asterisk PBX and this guide is accompanied by a simple example of creating a module demonstration of his method of translation, commissioning and loaded into Asterisk. Simulation of voice security is in the fourth chapter which provides a description of the proposed security solutions with subsequent implementation in Simulink. This simulation verifies the functionality of the solution proposed security phone call. In the fifth chapter outlines the historical use of encryption techniques primarily mirroring the spectrum and time division signal and comparing them with current modern digital technics. In the last sixth chapter is the actual implementation audio codec module with encryption.
Speech segmentation
Kašpar, Ladislav ; Galáž, Zoltán (referee) ; Sysel, Petr (advisor)
My diploma thesis is devoted to the problem of segmentation of speech. It includes the basic theory on this topic. The theory focuses on the calculation of parameters for seg- mentation of speech that are used in the practical part. An application for segmentation of speech has been written in Matlab. It uses techniques as segmentation of the signal, energy of the signal and zero crossing function. These parameters are used as input for the algorithm k–means.
Forward Error Correction in Digital Communication Systems
Kostrhoun, Jan ; Sysel, Petr (referee) ; Číž, Radim (advisor)
This work deals with forward error correction. In the work, basic methods and algorithms of error correction are described. For the presentation of encoding and decoding process of Hamming code, Reed-Müller code, Fire code, Reed-Solomon code and Trellis coded modulation programs in Matlab were created.
Protection of VoIP networks and their testing
Ulický, Ivan ; Sysel, Petr (referee) ; Krajsa, Ondřej (advisor)
Main goal of creating this diploma thesis is existence of increasingly amount of potential threats against IP voice networks (VoIP). The thesis is devoted to testing of various types of attacks and provides some possible solutions for this systems as well. The work points out to a various types of current attacks against either insecure or very little secure structures. The theoretical part is dedicated to analyse and description of wide spectrum of VoIP protocols including signaling protocols (SIP, IAX2), transport protocols (RTP, RTCP) and security protocols (SRTP, ZRTP, IPsec, SDES). Further attention is dedicated to the one of possible open source IP PBX solutions called Asterisk. There is shown a variety of possible attacks against this system due to its openness, because open systems always tend to be more susceptible for various attacks as they need an advanced administration and endless need for searching of new trends in area of security. The last block of the theoretical part is focused on common threats and types of attacks against VoIP networks. The practical part is about design and creation of web application called ,,VoIP Hacks using PHP” written in PHP scripting language and ist main task is to execute three basic attacks: eavesdropping, call drop and call flood. There is also a possibility of port scanning of selected network which is added as supplementary part of this application. The application can be comfortably managed from web browser user interface. All captured data can be displayed directly into the web browser. Tests of the application were performed on Google Chrome and Mozzila Firefox browsers. There is an accent placed on cooperation between the application and terminal linux programmes such as Tshark, BYE Teardown, INVITE flooder or Nmap, which all accept commands from web interface and interpret gained output values back to the web browser.
Identification of Speech Activity in Noisy Speech Signal
Pelikán, Martin ; Sysel, Petr (referee) ; Smékal, Zdeněk (advisor)
This paper is focused on identification of pauses in noisy speech signal and following filtering of the noise from the signal. Firstly the signal processing methods are theoretically described, then voice activity detectors and in the end noise filtering methods are described. Several voice activity detectors were created and their pause detection rate was compared.
Classification of transmission channels based on speech signal analysis
Báňa, Josef ; Sysel, Petr (referee) ; Atassi, Hicham (advisor)
The thesis examines the impact of VoIP transmission channel characteristics on speech parameters. It seeks ways of emulating properties of a selected VoIP transmission channel and creating a network emulation environment. Several scripts have been created in the Matlab development environment and used to modify and divide a continuous recording into parts identical to the original speeches, before passing through the transmission channel. Subsequently a database of speech recordings is created, as affected by selected characteristics - jitter, bandwidth, loss. Within these databases, symptoms are sought as the most evident characteristics of the transmission channel. Using correlation, symptoms are selected that are best suited for automated determination of the properties of transmission channel characteristics such as jitter, loss and bandwidth.
Comparison of voice and audio codecs
Lúdik, Michal ; Sysel, Petr (referee) ; Míča, Ivan (advisor)
This thesis deals with description of human hearing, audio and speech codecs, description of objective measure of quality and practical comparison of codecs. Chapter about audio codecs consists of description of lossless codec FLAC and lossy codecs MP3 and Ogg Vorbis. In chapter about speech codecs is description of linear predictive coding and G.729 and OPUS codecs. Evaluation of quality consists of description of segmental signal-to- noise ratio and perceptual evaluation of quality – WSS and PESQ. Last chapter deals with description od practical part of this thesis, that is comparison of memory and time consumption of audio codecs and perceptual evaluation of speech codecs quality.
Speech activity detector in digital signal processor
Kovařík, Jiří ; Mach, Václav (referee) ; Sysel, Petr (advisor)
In this diploma thesis were created voice activity detectors according to the standard ITU-T G.729 and G.723.1. The voice activity detectors were implements in the digital signal processor TMS320C6416 made by Texas Instruments. At the same time detectors were designed using by MATLAB programming language. The diploma thesis can be divided into two parts. In the theoretical section provides information on how to report detectors in the standard ITU-T G.729 and G.723.1. In the implementation part is described steps in the implementation of the detector in signal processor TMS320C6416 and there are discussed various differences compared to the documentation.
Digital filter for acoustic band, implemented by microcontroller
Hudec, Antonín ; Sysel, Petr (referee) ; Kubánek, David (advisor)
Microcontrollers of the ATmega family are not very suitable for real-time digital signal processing. Although they can reach 20 MIPS with maximum clocking of 20 MHz (most of the instructions are within one clock cycle), ATmega has just one 8bit multiplier unit, which is too little for demanding tasks like real-time digital signal filtration. My objective is to find out whether it is possible to filter digital signal with ATmega. If the answer is yes, what is the maximum sampling rate and what output quality can be achieved.

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