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Application for Demonstrating Principles of Hybrid Sound Synthesis
Prokop, Šimon ; Indrák, Michal (referee) ; Schimmel, Jiří (advisor)
The master thesis deals with the description and application of the main principles of sound signal generators contained in hybrid synthesizers. The individual methods and ways of generating audio signals were first studied and based on the theory, scripts were then created in Matlab to demonstrate the principle functions of the individual generators. Based on the logic of these scripts, a polyphonic hybrid synthesizer was created in VST3 format in the C++ language with the JUCE framework. The synthesizer allows the user to choose between basic readout methods Wavecycle and Wavetable, audio signal filtering or oversampling of the overall signal process.
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Experimental Audio Effect Based on Dynamic Signal Filtering
Leitgeb, David ; Miklánek, Štěpán (referee) ; Schimmel, Jiří (advisor)
The aim of this thesis is the implementation of an experimental audio effect based on dynamic signal filtering. Two implementations of the audio effect were made, the first one using the Matlab environment and its Audio Toolbox extension, and the second one in the C++ programming language using the JUCE framework. The main digital signal processing types present in the proposed audio effect include: frequency filtering, signal delay and modulation of parameters using low-frequency oscillators. The initial prototype of the audio effect mainly serves to verify correct functionality of the proposed algorithms. The result of the second implementation is a fully functional plug-in module usable, for example, for music production. Both versions are available in the VST format, which can be used with numerous digital audio workstations. The thesis is divided into three main parts: theoretical introduction, implementation of the audio effect, and demonstration of its functionality. Theoretical introduction describes several areas of digital signal processing which are crucial for the correct implementation of the individual components of the audio effect. Implementation of the proposed audio effect is described in the following part of this thesis. Description of the implementation in Matlab is mostly done by showcasing parts of the source code. The C++ version of the audio effect is for the most part described verbally. The last part of this thesis focuses on the demonstration of both versions of the proposed effect. This part features several graphical examples demonstrating various use cases of the proposed effect. Audio files corresponding to these examples are included in the appendix.
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Audio effect based on signal spectrum modification
Vejvoda, Zdeněk ; Sysel, Petr (referee) ; Schimmel, Jiří (advisor)
The thesis describes the design and the engineering solution of a hardware electronic sound effect that is a mixture of a frequency shifter and a delay effect. Electronic music intruments, sound effects and particularly modular synthesizer of the Eurorack standard are introduced. The firmware, developed utilizing the Max Gen environment and running on an ARM chip, prompted an examination of the Max platform. Finally, the thesis explores the calculations and processes required to construct a fully functional effect module. The device was assembled, tested and measured.
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Automation of measurement of directional characteristics of sound sources
Nimmerrichter, Benjamin ; Liska, Matej (referee) ; Schimmel, Jiří (advisor)
This bachelors thesis deals with the topic of automating the measuring of directional sound radiation characteristics of sound sources with the use of one to sixteen, possibly thirty-two microphones connected to an audio device, which supports the ASIO driver technology, which is connected to a computer with an installation of the MATLAB software. In the scope of this thesis a program was created, which is capable of measuring the 2D and 3D directional characteristics of sound sources and it can show the frequency spectrum of the signal. This work addresses the problematics of the effective value of discrete signals, calibration of microphones for measuring sound pressure level, the calculation and measuring of the level of acoustic pressure in dB (SPL), the autocorrelation of a discrete signal, plotting of the measurement in a polar or spherical plot and the calculation of the signal to noise ratio. This thesis contains a theoretical introduction to the measurement and to the propagation of sound, then the composition of the program and the measurement and an assessment of the measured results.
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Audio codecs and their perceptual quality evaluation
Hudec, Andrej ; Schimmel, Jiří (referee) ; Rajmic, Pavel (advisor)
The bachelor thesis deals with audio codecs and tests to evaluate their listening quality. The thesis offers a description of the properties of human hearing, a brief introduction to the issues of audio coding, types of audio data compression and listening tests. The bachelor thesis also contains a theoretical description of the coding processes of selected audio codecs, evaluation of measuring their computational complexity and evaluation of listening quality of each of them using listening tests. HTML, CSS, Javascript and Python programming languages were used to implement the listening test environment. The audio files examined in the listening tests were obtained using the FFmpeg library.
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Audio visualizer
Jelínková, Jana ; Schimmel, Jiří (referee) ; Říha, Kamil (advisor)
The aim of this thesis is to create an audio visualizer. That means an object, whose parameters will be changed in real time based on chosen parameters of an audio. The first part of this thesis deals with different kinds of audio visualizers through history till today and also deals with some artistic theories about visualization. The second part deals with the main solution of the visualizer in Pure Data and principles used for audio processing and also describes development of an external for Pure Data.
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Microphone Array for Estimation of Direction of Arrival of Sound
Kubišta, Ladislav ; Balík, Miroslav (referee) ; Schimmel, Jiří (advisor)
This bachelor's work describe detection of direction receiving sound by analizing of the sound. Work is based on methods of time delay estimation. Programmed algorithm of estimation direction works on cross–correlation and some selected cross–correlation methods. Results of measuring, as programming sound, so sound recorded in laboratory conditions and real enviroment, are mentioned in conclusion. All calculations were done by platform Matlab
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Design and Realization of Input and Output Circuits of AD and DA Converters for Audio Signal Processing
Stejskal, Tomáš ; Koton, Jaroslav (referee) ; Schimmel, Jiří (advisor)
This bachelor's thesis aims at designing the circuit diagram of input and output circuits of analog to digital and, simultaneously, digital to analog convertors intended to process audio signals and their realization. The introductory theoretical part deals with the qualitative requirements concerning the way low-frequency audio amplifiers process audio signal. The following part focuses on the active component base selection with respect to the achievement of the required quality and parameters as defined in the task description. The next part comprises the design of the input and output amplifier circuit diagram itself and then the realization of their functional samples. The input amplifier provides some of the functions which are common with high quality sound production; such as attenuator activation, input signal phase inversion, and MUTE function. This amplifier consists of three logically sequenced parts. A preamplifier with approximately 60 dB amplification and with high separation between a useful signal and noise stands for the first part; a high-pass filter that suppresses interfering signals stands for the second part; and an output amplifier ensuring the balanced output for the analog to digital convertor stands for the last part. The output amplifier, which is connected to the digital to analog convertor, is also embraced of three logically sequenced parts. The first one includes the input amplifier which transforms the input balanced signal to the unbalanced one. The second part is created of an output amplifier that has the line level balanced output.
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