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Software Analyzer for Audio Effects
Frenštátský, Petr ; Sysel, Petr (referee) ; Schimmel, Jiří (advisor)
The utilisation of personal computers for a conditioning of audio devices has shown a significant increase, since the digital signal processing (DSP) was introduced. The expansion of the DSP has allowed implementing analyses to obtain frequency and linear characteristics, distortion parameters (THD, THD+N, WHD, SINAD), a rate of crosstalk or a signal-to-noise ratio. In this work a software analyser is developed, which is able to obtain qualitative parameters of hardware audio devices that are connected with a sound card. For an efficient communication between the sound card and the personal computer the ASIO driver is used. The application is capable to measure audio effects that are implemented in VST plug-ins. The software is developed in C++ language and the implemented analyses are based on the AES17 recommendation.
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Multiplatform audio player
Henzely, Július ; Schimmel, Jiří (referee) ; Sysel, Petr (advisor)
This master's thesis focuses on problematic of creating applications based on multiplatform framework. This thesis also includes fundamental theoretical knowledge about relational database system SQLite, which has been used in a practical part of the thesis. Programing of multiplatform audio player which could be used in speech-language pathology clinic is essential portion of the practical part. Player was enhanced with the database of patients and ability to generate XML files.
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Multiplatform audio player
Malár, Ladislav ; Mach, Václav (referee) ; Sysel, Petr (advisor)
This master thesis deals with creating of the cross-platform audio player with the possibility of its use in the speech-language pathology clinic. The theoretical part is focused on the comparison of two multi-platform libraries, of which was one chosen for creating of the player. Subsequently on functionality SQLite database, which was also used as part of the final application. The basis of the practical part is the creation of the cross-platform audio player which is extended about database of patients and also extended about speech signal analysis with the help of VST plugin.
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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.
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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.
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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.
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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.
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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.
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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.
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