Todays’ systems for telecommunication are getting more and more complex. Automatic testing is required to guarantee quality of the systems produced. An actual example is the introduction of GPRS traffic in the GSM network nodes. This thesis investigates the need and demands for such an automatic testing of the traffic channels in the GSM system. A solution intended to be a part of the Ericsson TSS is proposed. One problem to be solved is that today’s tools for testing do not support testing of speech channels with the speech transcoder unit installed. As part of the investigation, a speech codec is implemented for execution on current hardware used in the test platform. The selected speech codec is the enhanced full rate codec, generating a bitstream of 12.2 kbit/s, and gives a good trade-off between compression and speech quality. The report covers the design of the test tool and the implementation of speech codec. Particularly performance problems in the implementation of the encoder will be addressed.

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.

There are different ways to store and process large amount of data. Hadoop iswidely used, one of the most popular platforms to store huge amount of dataand process them in parallel. While storing sensitive data, security plays animportant role to keep it safe. Security was not that much considered whenHadoop was initially designed. The initial use of Hadoop was managing largeamount of public web data so confidentiality of the stored data was not anissue. Initially users and services in Hadoop were not authenticated; Hadoop isdesigned to run code on a distributed cluster of machines so without properauthentication anyone could submit code and it would be executed. Differentprojects have started to improve the security of Hadoop. Two of these projectsare called project Rhino and Project Sentry [1].Project Rhino implements splittable crypto codec to provide encryptionfor the data that is stored in Hadoop distributed file system. It also developsthe centralized authentication by implementing Hadoop single sign on whichprevents repeated authentication of the users accessing the same services manytimes. From the authorization point of view Project Rhino provides cell-basedauthorization for Hbase [2].Project Sentry provides fine-grained access control by supporting role-basedauthorization which different services can be bound to it to provide authorizationfor their users [3].It is possible to combine security enhancements which have been done inthe Project Rhino and Project Sentry to further improve the performance andprovide better mechanisms to secure Hadoop.In this thesis, the security of the system in Hadoop version 1 and Hadoopversion 2 is evaluated and different security enhancements are proposed, consideringsecurity improvements made by the two aforementioned projects, ProjectRhino and Project Sentry, in terms of encryption, authentication, and authorization.This thesis suggests some high-level security improvements on theCentralized authentication system (Hadoop Single Sign on) implementationmade by Project Rhino.

Made available in DSpace on 2016-04-04T18:31:20Z (GMT). No. of bitstreams: 1 Alessandra B Bandeira.pdf: 1200038 bytes, checksum: 316684811248760f05faf3fdda1844f3 (MD5) Previous issue date: 2007-10-30 / This work investigates VoIP (Voice over IP) performance over WLAN (WVoIP) through qualitative and quantitative point of view, considering degradation in indoor environment. The results was obtained through a channel emulator to produce the Weibull distribution. Measurements results with a robot showed that this distribution is adequate to represent WLAN environment instability. The test utilized the CODECs (Coder Decoder) G711 and GSM (Global System for Mobile Communications). The work presents results that indicate the effect of the instability of wireless environment in the VoIP performance. The results indicate a significant variation of voice quality, considering the severity of the phenomena emulated. The behaviors of the CODECs are evaluated considering the different characteristics of bandwidth and robustness. / O trabalho investiga de forma quantitativa e qualitativa o desempenho de VoIP (Voz sobre IP) em uma rede WLAN (Wireless Local ?rea Network), denominado WVoIP, (Wireless VoIP) considerando degrada??o em ambiente indoor. Para atingir esse objetivo foram realizados testes utilizando um emulador de canal capaz de produzir fen?menos de propaga??o com distribui??o de Weibull. Medidas mostram que esta distribui??o ? adequada para avalia??o de instabilidade em redes WLANs. S?o utilizados dois codecs de voz: G711 e GSM (Global System for Mobile Communications). O trabalho apresenta resultados que indicam o efeito dos fen?menos de instabilidade das redes sem fio no desempenho de VoIP, que revelam varia??es significativas na qualidade de voz em fun??o da intensidade dos fen?menos emulados. S?o analisadas as quest?es relativas aos codecs de voz que possuem caracter?sticas de banda e robustez diferentes.

Made available in DSpace on 2016-04-04T18:31:26Z (GMT). No. of bitstreams: 1 Adenilson Boccato de Almeida.pdf: 2733171 bytes, checksum: 5b2d37bdad4b16ff26ccb172bbd6fbfc (MD5) Previous issue date: 2008-12-18 / This dissertation examines the quality of the transmission of VoIP (Voice over IP) in wireless LAN (Local Area Network Wireless). To achieve this objective tests were performed on the bench using voice codecs G729, G711, G723 and GSM analyzing characteristics of delay and loss of packets. The paper presents results that indicate the effect of the phenomena of instability of wireless networks in the performance of VoIP. These variations show significant changes in the quality of voice depending on the used environment and the distance between the transmitter and receiver. / Esta disserta??o analisa a qualidade da transmiss?o do VoIP (Voz sobre IP) em rede WLAN (Wireless Local ?rea Network). Para atingir esse objetivo foram realizados testes em bancada utilizando os codecs de voz G729, G711, G723 e GSM analisando suas caracter?sticas de atraso e perda de pacotes. O trabalho apresenta resultados que indicam o efeito dos fen?menos de instabilidade das redes sem fio no desempenho do VoIP. Essas varia??es revelam altera??es significativas na qualidade da voz em fun??o do ambiente utilizado e da dist?ncia entre o emissor e receptor.

In this thesis, the impact of packet losses on the quality of received videos sent across a network that exhibit normal network perturbations such as jitters, delays, packet drops etc has been examined. Dynamic behavior of a normal network has been simulated using Linux and the Network Emulator (NetEm). Peoples’ perceptions on the quality of the received video were used in rating the qualities of several videos with differing speeds. In accordance with ITU’s guideline of using Mean Opinion Scores (MOS), the effects of packet drops were analyzed. Excel and Matlab were used as tools in analyzing the peoples’ opinions which indicates the impacts that different loss rates has on the transmitted videos. Statistical methods used for evaluation of data are mean and variance. We conclude that people have convergence of opinions when losses become extremely high on videos with highly variable scene changes

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.

This project deals with possibilities of using the software defined radio conception for radio amateur use in a short wave band and its subsequent implementation into properly designed hardware. The aim of this work is to design a transceiver that would be capable of working in AM, FM, SSB, and CW modes. Within a theoretical part of the project the architectures of software defined radios and their components are discussed. This part was focused mainly on analog parts of the chain, such as amplifiers, filters and converters. Signal processing algorithms for both receiver and transmitter working in desired modes are studied subsequently and their computer models are built. Designed algorithms are implemented into FPGA structure (Virtex -5).