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Objective assessment and reduction of noise in musical signal
Rášo, Ondřej ; Makáň, Florian (referee) ; Krejčí, Jiří (referee) ; Balík, Miroslav (advisor)
The dissertation thesis focuses on objective assessment and reduction of disturbing background noise in a musical signal. In this work, a new algorithm for the assessment of background noise audibility is proposed. The listening tests performed show that this new algorithm better predicts the background noise audibility than the existing algorithms do. An advantage of this new algorithm is the fact that it can be used even in the case of a general audio signal and not only musical signal, i.e. in the case when the audibility of one sound on the background of another sound is assessed. The existing algorithms often fail in this case. The next part of the dissertation thesis deals with an adaptive segmentation scheme for the segmentation of long-term musical signals into short segments of different lengths. A new adaptive segmentation scheme is then introduced here. It has been shown that this new adaptive segmentation scheme significantly improves the subjectively perceived quality of the musical signal from the output of noise reduction systems which use this new adaptive segmentation scheme. The quality improvement is better than that achieved by other segmentation schemes tested.
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Real-Time Data Exchange technology and its applications
Meluzín, Ivo ; Šilhavý, Pavel (referee) ; Krajsa, Ondřej (advisor)
Bachelor ‘s thesis target the analysis of the Real-Time Data eXchange technology. There are mentioned some preferences for using kit development of TMS320C6416 DSK invented by Texas Instruments Inc., in the way of signal porcessing with help of some software applications. On these basis, there is built a model in Simulink. Sequentially, we activate whole system, which is running on developer‘s kit for methods of counting, processing and reading data from interface RTDX.
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Algorithm Database of Digital Audio Signal Processing with Automatic Classification and Searching using Web Interface
Kouba, Petr ; Sysel, Petr (referee) ; Schimmel, Jiří (advisor)
The aim of the bachelor was to design a database structure for classification source code and implementations of digital audio effects algorithms. Classification criteria for these algorithms are described in the thesis. A new project can be added to the database using web form or via of the automatic classification. The automatic classification is called a function for automatic recognition information in source files. For this purpose a format of saving catalog information into the source file was designed. These saved data can be further recognized by the suitable algorithm which utilizes PHP standard functions and then they inserted into the database. The saved projects can be searched by the project name, author’s name, the project description, or the criteria described in the introduction of this thesis.
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Software Analyzer for Audio Effects
Frenštátský, Petr ; Sysel, Petr (referee) ; Schimmel, Jiří (advisor)
The utilisation of personal computers for a conditioning of audio devices has shown a significant increase, since the digital signal processing (DSP) was introduced. The expansion of the DSP has allowed implementing analyses to obtain frequency and linear characteristics, distortion parameters (THD, THD+N, WHD, SINAD), a rate of crosstalk or a signal-to-noise ratio. In this work a software analyser is developed, which is able to obtain qualitative parameters of hardware audio devices that are connected with a sound card. For an efficient communication between the sound card and the personal computer the ASIO driver is used. The application is capable to measure audio effects that are implemented in VST plug-ins. The software is developed in C++ language and the implemented analyses are based on the AES17 recommendation.
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Drum Machine for Musicians
Spáčil, Tomáš ; Fedra, Zbyněk (referee) ; Povalač, Aleš (advisor)
The aim of this work is to create a functional prototype of a drum machine with sound generating using a microcontroller. Introduction is devoted to general drum machine and an analysis of its parts, principle and basic modes of operation. The following chapter deals with the principle of digital signal processing and sound generation using a microcontroller. Part of this chapter is focused on mixing using compression methods. The core of this work is to design my own block structure and overall scheme of the drum machine with a description of the parts. Followed by analysis and implementation with theoretical and practical point of view. It is mainly about presentation and description of important parts of firmware codes, DPS design and another technical documentation useful for final prototype production. The conclusion contains a summary of the results with prezentation and discussion.
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FFT implementation in FPGA and ASIC
Dvořák, Vojtěch ; Bohrn, Marek (referee) ; Fujcik, Lukáš (advisor)
The aim of this thesis is to design the implementation of fast Fourier transform algorithm, which can be used in FPGA or ASIC circuits. Implementation will be done in Matlab and then this form of implementation will be used as a reference model for implementation of fast Fourier transform algorithm in VHDL. To verify the correctness ofdesign verification enviroment will be created and verification process wil be done. Program that will generate source code for various parameters of the module performing a fast Fourier transform will be created in the last part of this thesis.
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Design of sigma-delta digital-to-analog converter in CMOS technology
Soukup, Luděk ; Pristach, Marián (referee) ; Fujcik, Lukáš (advisor)
This master’s thesis deals with the issue of digital to analog conversion and possibility of its realization in digital circuits. Goal of this project is to design sigma-delta digital to analog converter with resolution of 14 bits and frequency band (0 ÷ 20) kHz. Main functional blocks: interpolator and modulator sigma-delta will be realized like digital structures. Reconstruction filter will be realized like an analog structure. For design a check of parameters of designed converter programs MATLAB and Simulink are used. Designed digital structures will be described by VHDL language.
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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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Advanced speech coding methods using digital signal processor
Zajíček, Marek ; Sysel, Petr (referee) ; Smékal, Zdeněk (advisor)
This master thesis describes the practical usage of AMR-WB (Adaptive Multi Rate - Wide Band) codec and its implementation on a digital signal processor which is integrated in functional voice communication system Siemens HiPath 4000. The first part is focused on the complete codec description, especially on an encoder and decoder. The second part partly describes signal processors and then is followed by the practical part of the implementation which is solved from the preliminary activities up to the optimalization of the final functional solution.
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