|
|
Microcontroller based system for voice transfer
Kapitančik, Maroš ; Kaczmarczyk, Václav (referee) ; Bradáč, Zdeněk (advisor)
The aim of this thesis have to introduce us with digitalized transmission of speech with limited data flow and construct device witch can comunicate on this level. Theoretical part is describing base creation of sound human preception and issues with digitizing. Proposed device uses hardware codec VS1063a. Coding is applied with IMA ADPCM. Scale of incoming analog audio signal can be up to 4kHz. Inside system is controled by microcontroler, witch take care about others parts. For comunication with other moduls is used wireless transceivers module. Software part describes confuguration and selected options of every individual part. Next is explained principle of entire system. For comunication was implemented packet data structure. This implementation raised data flow but saved error state in transmit.
|
|
|
Noise map
Lokaj, Jiří ; Kubičková, Alena (referee) ; Rozman, Jiří (advisor)
The aim of this paper is to introduce the measurement of noise and create a noise map in a geographic information system. The first part is focused on describing the physical properties of sound in space, atmospheric and physiological acoustics. It also deals with the physiological effects of noise on the human body and technology needed for measure and process noise. Other part describes the structure of a geographic information system and noise map. The last part is about the practical creation of noise maps, includes a description of the real measurement results and evaluation
|
|
|
Microcontroller based system for voice transfer
Kapitančik, Maroš ; Kaczmarczyk, Václav (referee) ; Bradáč, Zdeněk (advisor)
The aim of this thesis have to introduce us with digitalized transmission of speech with limited data flow and construct device witch can comunicate on this level. Theoretical part is describing base creation of sound human preception and issues with digitizing. Proposed device uses hardware codec VS1063a. Coding is applied with IMA ADPCM. Scale of incoming analog audio signal can be up to 4kHz. Inside system is controled by microcontroler, witch take care about others parts. For comunication with other moduls is used wireless transceivers module. Software part describes confuguration and selected options of every individual part. Next is explained principle of entire system. For comunication was implemented packet data structure. This implementation raised data flow but saved error state in transmit.
|
|
|
Application for Measurement of Musical Instruments Directional Characteristics
Ustohalová, Iveta ; Orlovský, Kristián (referee) ; Mach, Václav (advisor)
This bachelor thesis is focused on the directional charateristics of acoustics music instruments measurements. Compilalation of measuring apparatus including microphone array and the individual steps of measurement and post-processing of audio recordings in *.wav format using program created in MATLAB and ment for this purpose is described. The output are graphs showing the directional charakteristics of selected instruments - violin, clarinet, guitar and accordion in selected one-third octave bands. The measurement results are interpreted and if available, compared with results given in literature.
|
|
|
Adjusting Acoustic of Listening Room for Subjective Assessment of Quality of Audiovisual Systems
Bijota, Jan ; Kurc, David (referee) ; Schimmel, Jiří (advisor)
The thesis deals with acoustics of the listening room, which has to be modified to fulfil requirements in reccomendation ITU-R BS.1116. It analyses electro-acoustic reproduction chain, geometric and acoustic properties of the listening room and it also deals with conditions, which the reccomendation imposes on reference listening area. Thesis proposes a solution of listening arrangements and listening room model design according to reccomendation. This model design will be used for acoustic simulation program purposes. Resources of this thesis were speaker data sheet and my own measuring of reference monitor loudspeakers, reference listening room and reference sound field. Listening arangement was made in Microsoft Visio 2013 software. SketchUp Make software was used for model design of reference room. This model design was imported to EASE simulating software and then I started reference sound field simulation. Results of measurements and simulation don't fulfil requirements in reccomendation ITU-R BS.1116. Room can be modified by bass traps, but this modification will only reduce reverbation time. This thesis offers an insight of reproduction chain analysis and also may serve for future modification purposes of reference listening room.
|
|
|
Tube Microphone
Škoda, Adam ; Šotner, Roman (referee) ; Staněk, Miroslav (advisor)
The presented thesis deals with a tube microphone using a condenser microphone capsule for sound capturing. Moreover, two essential aspects of a tube microphone – the basis of its operation and its design – are also discussed in this thesis. In the designed microphone, there is a vacuum tube known as ECC81. This vacuum tube is used as an amplifying element of an input sound signal. This paper also includes the choice of the most suitable microphone circuit, its modification and the design of a voltage source used for feeding and heating the vacuum tube. Some appropriate simulations were applied to find out the functionality of all designed parts of the microphone system. Following step was the practical realization of all microphone parts as well as the activation and the complete parameters and characteristics measurements of realized microphone system. After the provided measurements, the microphone was compared to equivalent commercial products. Evidently, the realized microphone is qualitatively similar to other commercial products. Presented thesis also contains the list of components and PCBs.
|
|
|
Software for simulation of acoustic pressure distribution
Hampl, Filip ; Schimmel, Jiří (referee) ; Kurc, David (advisor)
This semestral thesis deals with the creation of software to simulate the distribution of sound pressure. Basic principles of sound propagation are explained which are necessary for an understanding of the issue. There is a description of the velocity potential and its application in creating simulations. Another issue is the introduction of wavefield synthesis, which is essentially a technology for simulation of primary sources with secondary sources. Advantages of wavefiled synthesis are discussed as well as disadvantageous aspects. The practical part of this work is creating a user interface in Matlab, which allows user to plot the distribution of sound pressure of one or interaction of more emitters. Another possible outcome is to display the directional characteristics of real resources. At the very end of the work the direction of further work is outlined.
|
|
|
Generator of Harmonic Sounds and Chords
Bednář, Jan ; Mašek, Jan (referee) ; Schimmel, Jiří (advisor)
Major aim of this bachelor’s thesis is creation of VST plug-in module, that is able to generate sound for more tones in the same time. This thesis describes a brief theory of music acoustics which aims to show basic relations between tones, their higher harmonics and impact on the sound color. The function of temperament system of tuning, relations between intervals and way of making chords is described. Next things mentioned in this thesis are basics of VST technology, how to programme with VST SDK and basic MIDI messages for implementation in VST plug-in.
|
|
|
Simple synthesizer with eight-bit controller
Kuczman, Petr ; Frýza, Tomáš (referee) ; Fedra, Zbyněk (advisor)
This work contains information about physical properties of sound and briefly defines basic types of synthesis of audio signals. The more detailed view was putted on pulse weight modulation that has been used. Furthermore, it deals with possibilities provided by connected peripheries. There was created a list of controlling components that summarizes their properties and the real chance of implementation with eight-bit microprocessor. Second part of the work contains description of methods and structure of program that simulates synthesizer itself as well as methods for uploading values in to registers.
|
|
|
Dictionary learning for sparse signal reconstruction
Ozdobinski, Roman ; Rajmic, Pavel (referee) ; Mach, Václav (advisor)
This bachelor thesis discusses the dictionary learning for the reconstruction of signal based on sparse representations. There are methods of static and optimized dictionary of matrices described that are used with approximative Orthogonal Matching Pursuit algorithm to reconstruct the missing groups of samples in the audio signal. There is theoretically analyzed algorithm for learning K-SVD dictionary together with its implementation in Matlab. Furthermore, the selected dictionaries are compared to the various types of audio signals.
|
guest ::
