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Control of Real-Time Digital Audio Signal Processing Algorithms using HTML Browser
Slováček, Martin ; Mačák, Jaromír (referee) ; Schimmel, Jiří (advisor)
VST (Virtual Studio Technology) is one of the most used for real-time audio processing. VST effects are inserted to host applications in the form of plug-ins. This thesis discusses a design of a VST host remote control using a Java applet displayed in a web browser. A graphical user interface of the applet consists of graphical controllers created in this thesis. The number of controlled parameters is passed on to the applet by server using an XML configuration file. Three types of controllers are available: a slider, a knob and a button. If a parameter value change occurs, the applet sends a message with the parameter values to the server. Subsequently, the server reacts to this change by setting the parameters of the loaded VST module. The proposed design was completely realized and practically tested.
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Acoustical measurent in real environment
Mach, Václav ; Mačák, Jaromír (referee) ; Míča, Ivan (advisor)
This bachelor thesis discusses the issue of acoustical measurement. There are dierent signals described, which are used for this purposes, both MLS and esweep, compared their advantages and disadvantages and made some practical demonstrations of impulse response measurement. Furthermore, the fundamentals of reberbation time measurement are described in theory. The practical part of this thesis consists of the measurement application, written in the C++ language and mainly targeted for Linux systems. Qt multiplatform toolkit has been utilized for the GUI part.
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Conversion of monophonic melody from the audio signal into the MIDI protocol stream
Krupička, Jan ; Rášo, Ondřej (referee) ; Mačák, Jaromír (advisor)
The aim of this thesis is to compare possibilities of the pitch extraction methods in the monophonic melody. There is presented the overview of the methods based on the speech pitch extraction techniques in the thesis. These methods uses frequency, time and „cepstral“ domain. They are compared in the term of success of the detection of various test signals. The part of the thesis specification is the implementation of these methods in Matlab. There are described basics of sound features at the beginning of this work. The overview of the musical tuning systems is mentioned and there is described a problem of the determination of the pitch from the detected frequency. There is considered an issue of MIDI protocol in the next part of the work. There are described the brief history and the essential structure of MIDI protocol. The last task of the work was the creation of the program in C language. The purpose of the program is to analyze the monophonic melody in audio signal form and assign note numbers to the detected sounds according to MIDI specification. After that the numbers are written into the standard MIDI file (SMF). There was implemented a correlation pitch detection algorithm in this program. It had the best results as compared to the others. There was used the fast correlation based on Fast Fourier transformation to accelerate computing of the correlation. The program was created in the form of MEX function, which provides various possibilities to be used in Matlab. There was also attached the description of the FFTW library, which was used to compute FFT.
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Implementation of digital sound synthesizer in VST technology
Paukeje, Ján ; Schimmel, Jiří (referee) ; Mačák, Jaromír (advisor)
This thesis is focused on synthesis of audio signals and algorithm implementation by the help of technology designed for real-time processing. Analyzing individual methods of sound synthesis, the MIDI communication protocol and procedure of real-time signal processing. It is developed an appropriate software simulation ,on ground of studying these aspects, implemented by using C++ language. A result of this work is sound generator supporting MIDI protocol as a plug-in module designed for Windows platforms.
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Application for measurement of audio effects
Frenštátský, Petr ; Schimmel, Jiří (referee) ; Mačák, Jaromír (advisor)
The term paper is focused on the implementation and description of the application for the measurement of sound effects, which are implemented as plug-ins. Application measures in the time domain impulse response, transfer characteristics of dynamic effects. Application also shows the waveforms of attack and release time. In frequency domain application measures the frequency spectrum of the signal and frequency response of the system. The application is accompanied by the measurement of external effects by ASIO interface.
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Reverb Digital audio effect based on convolution with impulse response of acoustic room
Tichý, Vladimír ; Rajmic, Pavel (referee) ; Mačák, Jaromír (advisor)
This work deals with a computer simulation of an acoustic room using its impulse response. Two different approaches to the simulation are described with their pros and cons and then the work is focused on the physical approach, which uses room’s impulse response during the simulation. Several methods for the extraction of the impulse response of the acoustic room are mentioned with their conditions of use. The detailed description of various algorithms for a real time convolution computing is followed by the cost analysis of frequency domain block convolution algorithms. Several algorithms are chosen, implemented and tested in Matlab environment. Then the most effective of them is chosen to be implemented in VST technology as the plug in module for real time room simulation.
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Digital signal processing in real time
Zamazal, Zdeněk ; Mačák, Jaromír (referee) ; Rášo, Ondřej (advisor)
This work deals with digital signal processing in the field of adaptive filtering. Fundamental basics of adaptive filtering are described and primary aim is to create executable laboratory examples, using adaptive filtering, in LabView programming language. These laboratory examples are intended to be used by students fo studying and during laboratory lessons. Objective is to connect the examples with external devices, such as microphone. A microphone is used as an user's speech input acquiring interface. In the thesis is depicted Wiener's filter and problem of adaptive filtering is discussed. Contemporary adaptive algorithms are described and their applications as well. Most mentioned is the LMS algorithm and it's forms. Laboratory examples use following concepts: Adaptive Echo Cancellation, Active Noise Control and System Identification. Each of these examples is solely executable (need for LabView or Run-time engine), consisting also of theory with diagrams. Examples therefore are usable even without manual.
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