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Loudspeaker System for Laboratory Measurements of Polar Patterns
Morkus, Filip ; Káňa, Ladislav (referee) ; Schimmel, Jiří (advisor)
In the first part the attention is aiming to the choice of suitable loudspeakers. For this choice it was necessary to set required criteria and parameters. The aim of the system was to have as wide reproducibility range of frequency as possible but on the other hand not to be too large. The other criterion was the quality of the loudspeakers and its price. In the second part there was frequency crossover suggested according to the parameters of the chosen loudspeakers. This was suggested for low frequency spectrum as well as for high frequency spectrum. When determining the dividing frequency there were thought over producer’s recommendations and frequency characteristics of the loudspeakers so that the dividing frequency was not set in the place where the loudspeaker does not have “constant” character. The third part concerns the construction of the loudspeaker case. There one can find more details about the construction of the loudspeaker system called D´Appolito and its constructional requirements. This part was kept to the previous parts because integrating “D´Appolito” needs setting the distance between the loudspeakers according to the dividing frequency. It was also necessary to count with their size. According to the loudspeakers parameters it was possible to calculate the capacity of the loudspeaker case, also its shape was said and geometric proportions were calculated. Then everything was simulated in suitable software. In the last part the complete loudspeaker system was measured in an anechoic room. There were the direction characteristics and frequency characteristics measured there. The results of the measuring were compared with the software simulations.
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Microphone arrays for spatial separation of acoustic signals
Grobelný, Petr ; Schimmel, Jiří (referee) ; Míča, Ivan (advisor)
The goal of this master’s thesis is to explore the possibilities of multichannel localization of acoustic signal sources and their following application on a real signal localization and separation, using Beamforming methods. During this thesis two beamforming methods were selected, namely Delay and Sum a Constant Directivity Beamforming - Circular Arrays, and were applicated on real environment signals using two microphone arrays’ geometries ULA (Uniform linear array) and UCA (Uniform Circular array).
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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.
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Sound level meter for embedded systems
Stejskal, Tomáš ; Schimmel, Jiří (referee) ; Botta, Miroslav (advisor)
The aim of this work is the design and implementation of a sound level meter for emdedded systems. It is designed sound level meter sensor. This sensor includes a microphone, microphone preamplifier and ADC. This sensor is connected to a development kit STM32F4 Discovery, where sound is processed. It is processed filtration, time weighting, calculation of sound leve and loudness weighting. The sound level is then sent via the serial communication USART. This thesis includes a theoretical analysis of noise and its human perceptions. It also describes the development platform used.
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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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Audio signal editing software for multi-speaker systems
Svěrák, Jan ; Schimmel, Jiří (referee) ; Sysel, Petr (advisor)
This master’s thesis focuses on creating audio editing software for multi-speaker systems which can be used for real-time signal processing, such as polarity inversion or delay ranging from one to several thousand samples. The thesis is split into three main chapters. The first chapter contains theoretical knowledge required for development of audio software, for example principles of digital signal processing. The second chapter comprises of an overview of the source files and descriptions of several significant classes and functions. The last chapter is intended as a user guide, therefore it contains instructions on how to operate the software via the user interface.
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Psychoacoustical Measurement of Binaural Hearing Characteristics
Novotný, Ota ; Trzos, Michal (referee) ; Schimmel, Jiří (advisor)
This diploma thesis deals with a binaural hearing issues (it means hearing by both of ears), a human hearing ability to locate position of a sound source at three-dimensional space and parameters that affect this ability. In the second part, it focuses on psychoacoustic experiment, its main features and errors that can occur and which affect a results credibility. Method of pair comparisons is described more closely here. The last part of this thesis describes a technical solution of experiment in Java environment. The application should have a graphical interface and should be able to register a new user and perform a psychoacoustical experiment. The process of experiment is following. The aplication selects a random position of defined virtual sound source on the defined range and it plays this sound into headphones on button click. The users task is to set the application controls, representing a virtual sound source position, that way, where the user hear the sound come from. On another button click the application plays the same sound, but this sound comes from application controls set position (set by user). User compares this pair of sounds and modifies the position of second sound source until these two positions are same. The application stores these results for later processing on another button click. Principles of generating testing sound sources (sine wave, narrowband noise and sound file with wav extension) and their 3D positioning by measured head model impulse responses correlation are described thereinafter. An ability of human hearing system to locate a virtual sound source in dependence on sound parameters is discussed in conclusion.
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Design and realization of the pedal effect on Arduino platform
Gajarský, Tomáš ; Schimmel, Jiří (referee) ; Krajsa, Ondřej (advisor)
The topic of my bachelor’s thesis is the design and the construction of the pedal effect used by the Arduino Due, which is supplemented by the possibility of controlling through the internal web interface, which is mediated by the modul Arduino Ethernet Shield. The following effects are included: echo, tremolo, vibrato, chorus and flanger. For editing the guitar signal in order to interact with AD microcontroller converter, is used the input filter preamplifier circuit. In order of editing the output signal from Arduino is used summing and filtring output circuit for the guitar amplifier. The basic component for these circuits is operating amplifier OPA1644. The goal of this thesis was to create the proposal of the connection schemes, printed circuit board, the program code of the microcontroller, simple web page, as well as the construction and the measurement of the pedal effect.
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