National Repository of Grey Literature 348 records found  1 - 10nextend  jump to record: Search took 0.01 seconds. 
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.
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.
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
Perceptual Audio Coding
Novák, Vladimír ; Rajmic, Pavel (referee) ; Schimmel, Jiří (advisor)
his thesis describes Perceptual Audio Coding of MPEG1 Layer 3 format (ISO/IEC 11172-3), principles and algorithms of psychoacoustic model. MATLAB application for modeling of Psychoacoustic model 2 of this audio format is developed.
Application for Real-Time Dual-Channel Analysis of Electroacoustic System
Bača, Petr ; Balík, Miroslav (referee) ; Schimmel, Jiří (advisor)
Master’s thesis contains the theory for realization of the researched software. Besides other, it describes Fourier transform, frequency response function, coherence, impulse response and group delay. Real life application of the software is discussed. Software is invented in the MatLab environment. Further, thesis provides testing of the software and shows its commented outcomes.
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.
Implementation of the Parametric Equalizer into VST3 Plug-in
Pevný, Jindřich ; Schimmel, Jiří (referee) ; Frenštátský, Petr (advisor)
This thesis deals with the field of DSP (Digital Signal Processing), specifically with structures of digital parametric shelving filters, which can be used as building blocks of parametric equalizer. The objective of this thesis is to learn the options of indirect parametric filter design using decomposition of transfer functions into the function of an all-pass filter and implement these using Steinberg’s VST3 SDK.
Audio Interface for Signal Procesor Using IEEE 1394 Bus
Staroň, Martin ; Přinosil, Jiří (referee) ; Schimmel, Jiří (advisor)
The scope of bachelor’s thesis is a design of the connection between the Freescale digital signal processor DSP56300 family and the DFM Audio DICE II iKit interface by way of the TDM bus. The DICE II iKit interface uses serial IEEE1394 FireWire bus that transmits an audio stream from a host computer. This signal receives the DICE II iKit interface that converts audio data from IEEE1394 FireWire to the TDM bus. The signal processor that process the sended audio data in a real time is connected to this TDM bus. The theoretical part of this work describes the theory of the digital signal processor, in detail DSP56300 Freescale family. After that there is a basic specification of the IEEE1394 FireWire bus. The theoretical part of this bachelor‘s thesis is ended by a description the TDM bus, the I2S interface and the AudifiedDSP Solution system. A connection of the signal processor and the DICE II interface was solved in the practical part, included the debbuging and the operating of the hadle subroutine for the duplex communication. The results of this thesis is might utilise for the duplex communication between the signal processor and aplication software working on the host computer.
Tube headphone amplifier with USB DAC as an input
Rozkopal, Tomáš ; Schimmel, Jiří (referee) ; Vlček, Lukáš (advisor)
This thesis focuses on transferring of sound through USB bus. There is analysed a principal of the bus, it´s physical properties and a description of information transmission. Behaviour of the end point is described on an example of an audio device. The second part of the thesis describes demands on digital/analogue converters, their attributes such as resolution, speed of conversion and signal to noise ratio. Furthermore there are described two basic conversions between digital and analogue conversion. The third part of the thesis analyses a principal of tubes and explains a principal of vacuum diode, triode, tetrode and pentode. The individual types of tubes are supplemented by currently produced types and brief descriptions of their use in a circuit. Possible solutions of designing data transfer between a computer and an end point are discussed in the fourth part of the thesis. Different available microcontrollers are compared. The design continues with solving a problem with connecting a DA converter using I2S bus and there is also designed DA converter with voltage output. Demands for analogue filter and numerically derived main assumptions for end amplifier are explained in the final part of the thesis. Tube filter low - pass type is designed and possible solution of an end amplifier is discussed.
Implementation of Waveshaper Audio Effect
Leitgeb, David ; Miklánek, Štěpán (referee) ; Schimmel, Jiří (advisor)
The aim of this thesis is the implementation of a non-linear audio effect called waveshaper. This type of distortion effect contains the following building blocks: user defined transfer function, several types of filters and an oversampling processor with multiple stages of oversampling. The first prototype of this audio effect was implemented using Matlab and its Audio Toolbox extension. Due to certain limitations of this prototype, the whole audio effect was later completely rewritten in C++. This new implementation uses the JUCE framework which is mainly used for audio application development. The transition to this framework allowed real time editing of the transfer function and a VST3 build of the effect. In addition to a brief introduction of the used system types, motivation for oversampling and the description of the implementation for both prototypes, this thesis also includes graphical examples demonstrating their correct functionality. Audio files related to these examples are included in the electronic attachment.

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