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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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Cryptanalysis using neural networks
Budík, Lukáš ; Mačák, Jaromír (referee) ; Martinásek, Zdeněk (advisor)
This dissertation deals with analysis of current side canal by means of neural network. First part describes basis of cryptografy and dilemma of side canal. In the second part is theoretickly described neural network and correlative analysis. Third part describes practical analysis of calibres of current side canals by means of classifier which uses neural network in Matlab surrounding. This classifier is confronted with classifier which uses correlative analysis.
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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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Fluxmeter with graphical display of B-H curve
Ježek, Jaroslav ; Mačák, Jaromír (referee) ; Hanák, Pavel (advisor)
This work deals with simple fluxmeter which is able, together with other device, to show hysteresis loop. Hysteresis loop is a graphic expression of dependence of magnetic induction on intensity of magnetic field. Oscilloscope is used to display the hysteresis loop. This device is fully sufficient for the display. The measured objects are solenoids from various kinds of materials with the same shape. The main aim of this work is the design, realization and description of the fluxmeter. The fluxmeter consists of several partial blocks. The first one, on which this work is focused, is signal generator which is able to generace different kinds of signal. The generated signal comes on primary winding of solenoid where a magnetic field on a given intensity rises. An amplifier is used to obtain the sufficient intensity. Next thing this work is focused on is the design of the integrator which is necessary for the correct function of the fluxmeter. As suggested, there is shown the block diagram of linking of individual parts. There are described the measured results at the end of this work.
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Low bit rate voice encoders
Leitner, Jakub ; Mačák, Jaromír (referee) ; Pust, Radim (advisor)
The final thesis deals with coders and voice coders used in speech signal processing. The aim is to create an integral overview of coders and voice coders including a description of their properties, in the second part of the thesis a simulation of algorithms and methods of speech processing is performed in Matlab Simulink program.The basic methods of speech processing and a parametric LPC voice coder were simulated in time domain. In the LPC voice coder model there are implemented the algorithms for obtaining speech segment parameters. These are the algorithm for classification of voiced and unvoiced speech segment, LPC analysis and pitch detection. The output is a parametric signal that enables a receiver to synthesize a speech signal. The appendix 1 contains a list of names of coders or standard numbers of coders and their properties, the appendix 2 includes an overview of speech processing methods.
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Neural network utilization for etwork traffic predictions
Pavela, Radek ; Mačák, Jaromír (referee) ; Kacálek, Jan (advisor)
In this master’s thesis are discussed static properties of network traffic trace. There are also addressed the possibility of a predication with a focus on neural networks. Specifically, therefore recurrent neural networks. Training data were downloaded from freely accessible on the internet link. This is the captured packej of traffic of LAN network in 2001. They are not the most actual, but it is possible to use them to achieve the objective results of the work. Input data needed to be processed into acceptable form. In the Visual Studio 2005 was created program to aggregate the intensities of these data. The best combining appeared after 100 ms. This was achieved by the input vector, which was divided according to the needs of network training and testing part. The various types of networks operate with the same input data, thereby to make more objective results. In practical terms, it was necessary to verify the two principles. Principle of training and the principle of generalization. The first of the nominated designs require stoking training and verification training by using gradient and mean square error. The second one represents unknown designs application on neural network. It was monitored the response of network to these input data. It can be said that the best model seemed the Layer recurrent neural network (LRN). So, it was a solution developed in this direction, followed by searching the appropriate option of recurrent network and optimal configuration. Found a variant of topology is 10-10-1. It was used the Matlab 7.6, with an extension of Neural Network toolbox 6. The results are processed in the form of graphs and the final appreciation. All successful models and network topologies are on the enclosed CD. However, Neural Network toolbox reported some problems when importing networks. In creating this work wasn’t import of network functions practically used. The network can be imported, but the majority appear to be non-trannin. Unsuccessful models of networks are not presented in this master’s thesis, because it would be make a deterioration of clarity and orientation.
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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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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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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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