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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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Implementation of digital metronome with oversampling in VST technology
Křupka, Aleš ; Přikryl, Lubor (referee) ; Mačák, Jaromír (advisor)
This bachelor’s thesis presents solution to requirement of implementation of digital metronome in VST technology. In the beginning, there is analysis of basic characteristics of digital metronome, that helps to suggest algorithm realizing digital metronome. Algorithm is encapsulated into C++ class with appropriate interface, which is suitable for usage in different technologies of digital sound signal processing. This way encapsulated metronome is implemented in VST technology as plug-in. For this reason, thesis contains a brief introduction to VST software development kit, definitions of its basic methods and way how to do graphical user interface of VST plug-in. For realizing of function enabling choice of any sound loaded from wav file, there is a description of RIFF format, that specifies the format of saving multimedia data into drive. Using this knowledge, the function of loading from wav file is implemented. In order to adjust sampling frequency of loaded sound, thesis deals with problems of sample rate conversion and implements resampling function.
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Real-Time Generation of Band-Limited Digital Audio Signals
Maule, Petr ; Rajmic, Pavel (referee) ; Schimmel, Jiří (advisor)
Master’s thesis deals with the generation of digital audio signals with band-limited frequency spectrum, i.e. without the aliasing distortion. Various methods of generating band-limited rectangular, triangular, and sawtooth waveforms are described in the theoretical part. The described methods are programmed in the Matlab programming environment and compared in terms of real-time parameter changes, such as duty cycle change of rectangular waveform or continuous change of frequency. The main part of the thesis describes implementation of methods of successive integration of band-limited impulse train and method of differentiated parabolic waveforms in C++ language. The implemented methods were integrated into a plug-in of VST technology that generates an audio signal in real time. The implemented methods are compared in terms of computational complexity and distortion of the generated signal.
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Simulation of analogue audio effecs using the nonlinear filters
Otoupalík, Petr ; Káňa, Ladislav (referee) ; Mačák, Jaromír (advisor)
This thesis deals with a simulation of analogue audio effects using the nonlinear models that replace the analogue nonlinear devices in discrete domain. The thesis describes Volterra system model and simplified Volterra system model that can be realized in two ways, either Wiener model, or Hammerstein model. The method for the analysis and modeling of audio and acoustic nonlinear systems is presented in this thesis. This method allows through knowledge of the input swept-sine signal and the response of the analogue nonlinear system to the input signal to determine the coefficients of the discrete nonlinear system. This allows simulating the analogue nonlinear system in discrete domain. The method was first tested and then used successfully for simulation of the analogue nonlinear system in discrete domain. Concretely, it was simulated a musical guitar effect of the type of distortion. Last part of this thesis is devoted a description of VST technology and an implementation of VST plug-in module, which realizations Hammerstein model.
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Implementation of digital metronome with oversampling in VST technology
Křupka, Aleš ; Přikryl, Lubor (referee) ; Mačák, Jaromír (advisor)
This bachelor’s thesis presents solution to requirement of implementation of digital metronome in VST technology. In the beginning, there is analysis of basic characteristics of digital metronome, that helps to suggest algorithm realizing digital metronome. Algorithm is encapsulated into C++ class with appropriate interface, which is suitable for usage in different technologies of digital sound signal processing. This way encapsulated metronome is implemented in VST technology as plug-in. For this reason, thesis contains a brief introduction to VST software development kit, definitions of its basic methods and way how to do graphical user interface of VST plug-in. For realizing of function enabling choice of any sound loaded from wav file, there is a description of RIFF format, that specifies the format of saving multimedia data into drive. Using this knowledge, the function of loading from wav file is implemented. In order to adjust sampling frequency of loaded sound, thesis deals with problems of sample rate conversion and implements resampling function.
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Simulation of analogue audio effecs using the nonlinear filters
Otoupalík, Petr ; Káňa, Ladislav (referee) ; Mačák, Jaromír (advisor)
This thesis deals with a simulation of analogue audio effects using the nonlinear models that replace the analogue nonlinear devices in discrete domain. The thesis describes Volterra system model and simplified Volterra system model that can be realized in two ways, either Wiener model, or Hammerstein model. The method for the analysis and modeling of audio and acoustic nonlinear systems is presented in this thesis. This method allows through knowledge of the input swept-sine signal and the response of the analogue nonlinear system to the input signal to determine the coefficients of the discrete nonlinear system. This allows simulating the analogue nonlinear system in discrete domain. The method was first tested and then used successfully for simulation of the analogue nonlinear system in discrete domain. Concretely, it was simulated a musical guitar effect of the type of distortion. Last part of this thesis is devoted a description of VST technology and an implementation of VST plug-in module, which realizations Hammerstein model.
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|
Real-Time Generation of Band-Limited Digital Audio Signals
Maule, Petr ; Rajmic, Pavel (referee) ; Schimmel, Jiří (advisor)
Master’s thesis deals with the generation of digital audio signals with band-limited frequency spectrum, i.e. without the aliasing distortion. Various methods of generating band-limited rectangular, triangular, and sawtooth waveforms are described in the theoretical part. The described methods are programmed in the Matlab programming environment and compared in terms of real-time parameter changes, such as duty cycle change of rectangular waveform or continuous change of frequency. The main part of the thesis describes implementation of methods of successive integration of band-limited impulse train and method of differentiated parabolic waveforms in C++ language. The implemented methods were integrated into a plug-in of VST technology that generates an audio signal in real time. The implemented methods are compared in terms of computational complexity and distortion of the generated signal.
|
|
|
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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