Denna undersökning söker ett svar på hur den relativt vana lyssnarens bedömning av ljudkvalitet påverkas av ett så kallat öppet test, där det som bedöms är känd för lyssnaren, jämfört med ett blindtest, där detta objekt är okänt. Frågan appliceras på kvalitetsbedömningen av digitala kodningstekniker, d.v.s. hur lyssnaren påverkas av att valet av kodningsteknik som avlyssnas är känd eller inte. För att ta reda på detta genomfördes ett lyssningstest med nio deltagare. Deltagarna fick betygssätta perceptuellt kodade ljudfiler mot en känd referens, både som ett blindtest samt i ett öppet test. Resultatet är mångtydigt och inga generella slutsatser för hur lyssnaren påverkas av ett öppet test jämfört med ett blindtest går att uppfatta. Resultatet visar dock att påverkan ett öppet test har på lyssnarens bedömning är högst individuell. Lyssningstest i form av blindtest bör därför användas för att uppnå pålitligast resultat.

The topic of the master's thesis is about transmission of multimedia via network with help of compression’s algorithm used in codec MPEG-4 part 10 and Real Time Protocol (RTP). First part of this master thesis will be familiarization with basic terms about multimedia. It will be queried about techniques of compression axioms for audio and video, justification for necessary use of compression in transmitting multimedia via network. Also, here will be subscribed principles and properties of Real Time Protocol, which was designed for multimedia stream. From the start, competent codec for implementation of two applications (client, server) with read raw video and audio data, will choose the following : compression with help of codec Theora (Vorbis), transfer via network in RTP packets, decode and play at visual form for end user. Complete subscription of implementation for both applications written in programming languages C/C++ and Java, will be certain. The basic overview about this problematic experience with implementation, which were here concluded, are the reasons for reading this master thesis.

Diploma thesis focuses on the selection of suitable HW solution for parallell processing of multiple audio sources. Compares several different platforms based on architectures ARM, x86 and Epiphany, compares their performance in serial and parallel data processing, their energy consumption and price.

This thesis deals with the issues of detecting the codec used for the encoding of voice data carried by the RTP protocol without having access to the information carried by signalisation protocols in VoIP applications. Its main goal is to create and implement a fast algorithm for detecting the codec used for voice transfer via the RTP protocol. This algorithm should be fast enough to be used for offline analysis of captured data as well as for real-time online analysis. For research of possibilities were compared two approaches of detection. Detection by the characteristics of the codecs was chosen to solve the problem itself. Within the solution was performed data analysis and implementation of the application followed by testing on data.

This master’s thesis deals with a modification of existing project which is used as a helpful tool for tracking and measuring digital audio interface I2S. The original design was created by Ing. Martin Stejskal, Polymorphic USB – I2S Interface. Modifications are based on practical one year experience when the device was tested and deals with new requirements for extension. This work describes and justify individual changes of hardware and software part of project.

Issue of this diploma thesis is focused on analysis of audio codecs used within IP telephony. Attention of teoretical part is given mostly to audio codecs according to ITU-T recommendations, but also to signaling protocols used here. For practical part of analysis is chosen router Cisco 2821 and IP phones Cisco 7975G. Configuration is done over operating system Cisco IOS. Chosen signaling protocol is SCCP. For analysis itself are chosen 2 analysers – L-580FX and Fluke NetTool. These are used in combination with program Wireshark. Analysed parameters are latency, packet lost, bandwidth, jitter and mean opinion score. Measured values are presented in graphs and tables and they are discussed. Next output of the thesis is laboratory excercise, which deals with analysis of audio codecs.

This work deals with the creation of electronically controlled audio system for the language classroom. Possible solutions compares analog and digital form. Furthermore, describes the final system that was designed. The resulting system is implemented by analog amplifiers and analog crosspoint matrix. Hardware control is realized by microcontroller and that is controlled from the computer.

This paper deals with development and construction of an electronic drum module. The purpose of the device is to capture signals from an electronic drumkit and to produce sound accordingly. Firstly, a protoype with no sound output is constructed to demonstrate an ability to capture input signals. Based on its function, parameters for a final device are determined. Then, electronic component selection and design of printed circuit boards is described with an aim to maximalize polyphony and minimize latency of the sound generator. After that, firmware with software mixing algorithm is designed. Lastly, testing and measurement of real device parameters is performed.

This thesis deals with description of lossy video compression. Theoretical part of the work describes the fundamentals of the video compression and standarts for lossy as well lossless video and still image compression. The practical part follows up with design of Java program for simulation of MPEG codec.

The thesis Data transmission error-protection with BCH codes deals with a large class of random-error correcting cyclic codes which are able to protect binary data and can be used for example in data storages, high speed modems. Bose, Chaudhuri and Hocquenghem (BCH) codes operate over algebraic structures called Galois fields. The BCH encoding is the same as cyclic encoding and can be done with linear feedback shift register but decoding is more complex and can be done with different algorithms - in this thesis there are two algorithms for decoding Peterson and Berlekam-Massey mentioned. The aim of this thesis is to find BCH code which is able to correct t = 6 independent errors in up to data sequence n = 150 bits, then peruse possible realizations of the codecs and set criteria for the best realization, then design and test this realization. This thesis is split into three main parts. In the first part there are encoding and decoding methods of the BCH code generally described. The second part deals with selecting of the right code and realization. There was chosen BCH (63,30) code and realization with FPGA chip. In the last part is described design of BCH encoder and decoder and compilation in the Altera design software.