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Audio Interface for Embedded PC
Staroň, Martin ; Hanák, Pavel (referee) ; Schimmel, Jiří (advisor)
The scope of my master thesis is a designing computer sound interface including measurement of audio performance. This work is concerning both design analog front - ends and digital support circuits. The sigma delta Analog to Digital (ADC) and Digital to Analog (DAC) converters is included in this conception. Those converters has been made into two separate printed circuit boards. All signal paths in this circuitry are utilizing differential mode that are quoted as balanced among audio engineers. Modern circuit components are used in this design, such as fully differential operational amplifiers, electronically controlled gain preamplifiers, low drop linear stabilizers with low noise level, DC component suppression circuits and low jitter active components. Theoretical part of this thesis contains specification of choosed sound defitions, questioning audio program loudness leveling. Further criteria of suitable active and passive components are included. In this thesis the simulations of fundamental circuits block are meant likewise. Practical part involve complete layout of printed circuit boards of and prototyping. Designed prototype device has wide application usage. It is intended to use not only as industrial computers, but also as dedicated sound converters, measurement cards, mixing consoles, switching matrixes, active loudspeakers, embedded systems.
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Noise Source Identification Using Nearfield Acoustical Holography
Nevole, Tomáš ; Míča, Ivan (referee) ; Schimmel, Jiří (advisor)
This master’s thesis deals with problems of noise source identification using nearfield acoustical holography (NAH). In the beginning there is the summary of basic terms and values of a sound pressure field, which is unnecessary for understanding of the theme. In the next part the thesis continues with more detailed description of the NAH technology and the historical context of its emergence. Measurement equipment which is used for scanning of sound pressure fields is also introduced. In addition, the kinds of NAH (according the shape of the wave front) are showed and the planar NAH is descripted most closely. Because of the NAH algorithms are implemented in the wave number domain (k-space), there is also a chapter focused to this problem in the thesis. There are briefly descripted some similar methods in next chapter, like statistically optimized NAH, (SONAH) and iterative NAH with recursive filtration. The main product of the thesis is the practical part represented by testing application. That is created in the Matlab environment and is able to calculate and display hologram of the scanned array by the planar NAH method using the “k-space” filter. The application supposes a planar sound source and in other cases the accuracy of the reconstruction is not guaranteed. There are also given some holograms calculated with the application.
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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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Noise Source Identification Using Beamforming
Kurc, David ; Míča, Ivan (referee) ; Schimmel, Jiří (advisor)
This master's thesis is focused on the noise source identification using microphone arrays and beamforming as the signal processing method. It describes parts of such a system and provides a comparison with other systems that serve a similar purpose (eg. NAH). Various types of microphone arrays are mentioned with their influence on the resulting ability to identify the noise source. We are further focusing on Delay-And-Sum technology, on which we are explaining the basic principles and constraints of beamforming. The practical part describes the implementation of the DAS method in MATLAB and C language, the specific structures of built microphone arrays and assembly of complete systems capable of identifying sources of noise. These systems were tested by performing a practical experiment. Achievements in the form of distribution maps of acoustic energy in the focused space are interpreted in the last chapter.
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Digital Audio Signal Feature Extraction and Modification in Dynamic Plane
Kramoliš, Ondřej ; Sysel, Petr (referee) ; Schimmel, Jiří (advisor)
This thesis deals with basic methods of root mean square and peak value measurement of a digital acoustic signal, algotithms to measure audio programme loudness and true-peak audio level according to recommendation ITU-R BS.1770-1 and digital systems for control of signal dynamic range. It shows achieved results of root mean square and peak value measurement and results of implementation of dynamic processor with general piecewise linear non-decreasing static curve and algorithms according to recommendation ITU-R BS.1770-1.
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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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Utilizing psychoacoustic model and Wavelet Packet Transform for purposes of audio signal watermarking
Heitel, Tomáš ; Schimmel, Jiří (referee) ; Rajmic, Pavel (advisor)
This Thesis deals with a method to enforce the intellectual property rights and protect digital media from tampering – Digital Audio Watermarking. The main aim of this work is implement an audio watermarking algorithm. The theoretical part defined basic terms, methods and processes, which are used in this area. The practical part shows a process of embedding the digital signature into a host signal and her backward extraction. The embedding rule used spread spectrum technique and a psychoacoustic model. The implemented psychoacoustic model involves two properties of the human auditory system which are frequency masking and representation the frequency scale on limited bands called critical bands. The model is relatively new and based on the DWPT. In terms of above model is then the digital watermark embedded in the wavelet domain. This algorithm is implemented in technical software MATLAB. One part of this work focuses on robustness tests of the algorithm. Common signal processing modifications are applied to the watermarked audio as follows: Cutting of the audio, re-sampling, lossy compression, filtering, equalization, modulation effects, noise addition. The last part of the thesis presents subjective and objective methods usable in order to judge the influence of watermarking embedding on the quality of audio tracks called transparency.
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Beamforming using microphone arrays
Bartoň, Zdeněk ; Schimmel, Jiří (referee) ; Míča, Ivan (advisor)
The aim of the master thesis is to sum up theoretical information about beamforming methods of microphone arrays and to verify their functionality. At the beginning of this work there are simulated different varietes of linear uniform and nonuniform microphone arrays and circular arrays. The results are verificated by a practical measurement in ideal conditions. Then I will focuse on implementation of the DAS(Delay And Sum), SAB(Sub Array Beamforming), CDB(Constant Directivity Beamforming), CDB-CA(CDB-Circular Arrays) beamformer including theoretical and practical verification of the functionality in ideal conditions. At the end of this thesis are all beamforming methods compared with each other at SNR(signal to Noise Ratio) and directivity parameters.
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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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