No / Multi-antenna(MIMO) techniques are reported to improve the performance of radio communication systems in terms of their capacity and spectral efficiency. In combination with appropriate receiver technologies they can also provide savings in the required transmit power with respect to target bit error rate. Long Term Evolution(LTE), one of the candidates for fourth generation(4G) mobile communication systems has MIMO as one of its underlying technologies and ITU defined channel models for its propagating environment. This paper undertakes a comprehensive verification of the performance of transmit diversity MIMO in the downlink sector of LTE. It uses models built using MATLAB to carry out simulations. It is deduced that generally increasing transmit diversity configuration from 2x2 to 4x4 offers SNR savings in flat fading channels though with a user equipment moving at 30km/hr, deploying 2x2 offers higher SNR saving below 7dB. Furthermore bandwidth variation has minimal effect on the BER performance of transmit MIMO except at SNR values above 9dB while the gains of higher modulation schemes come with a transmit power penalty.

The DIRECT (DIviding RECTangles) algorithm JONESJOTi, a variant of Lipschitzian methods for bound constrained global optimization, has been applied to the optimal transmitter placement for indoor wireless systems. Power coverage and BER (bit error rate) are considered as two criteria for optimizing locations of a specified number of transmitters across the feasible region of the design space. The performance of a DIRECT implementation in such applications depends on the characteristics of the objective function, the problem dimension, and the desired solution accuracy. Implementations with static data structures often fail in practice because of unpredictable memory requirements. This is especially critical in S⁴W (Site-Specific System Simulator for Wireless communication systems), where the DIRECT optimization is just one small component connected to a parallel 3D propagation ray tracing modeler running on a 200-node Beowulf cluster of Linux workstations, and surrogate functions for a WCDMA (wideband code division multiple access) simulator are also used to estimate the channel performance. Any component failure of this large computation would abort the entire design process. To make the DIRECT global optimization algorithm efficient and robust, a set of dynamic data structures is proposed here to balance the memory requirements with execution time, while simultaneously adapting to arbitrary problem size. The focus is on design issues of the dynamic data structures, related memory management strategies, and application issues of the DIRECT algorithm to the transmitter placement optimization for wireless communication systems. Results for two indoor systems are presented to demonstrate the effectiveness of the present work. / Master of Science

The Motorola Inc. paging protocol FLEX is discussed. The design and construction of a FLEX paging protocol decoder is discussed in detail. It proposes a decoding solution that includes a radio frequency (RF) receiver and a decoder board. The RF receiver will be briefly discussed. The decoder design is the main focus of this thesis as it transforms the RF frequency modulated (FM) data from the receiver and converts it to FLEX data words. The decoder is designed to handle bit sampling, bit clock synchronization, FLEX packet detection, and FLEX data word collection. The FLEX data words are then sent by the decoder to an external computer through a serial link for bit processing and storage. A FLEX transmitter will send randomly generated data so that a bit error rate (BER) calculation can be made at a PC. Each receiver'9s noise power and noise bandwidth will be measured so that noise spectral density may be calculated. A complete measurement set-up will be shown on how these noise measurements are made. The BER at a known power level is recorded. This enables Eb/No curves to be generated so that results of the decoding algorithm may be compared. This is performed on two different receivers. / Master of Science

Radio frequency (RF) receivers are inherently non-linear due to non-linear components contained within the RF front-end such as the low noise amplifier (LNA) and mixer. When receivers operate in the non-linear region, this will affect the system performance due to intermodulation products, and cross-modulation, to name a few. Intermodulation products are the result of adjacent channel signals that combine and create intermodulation distortion of the received signal. We call these adjacent channel signals blockers. Receiving blockers are unavoidable in wideband receivers and their effect must be analyzed and properly addressed. This M.S. Thesis studies the effect of blockers on system performance, specifically the symbol error rate (SER), as a function of the receiver non-linearity figure and the blocking signal power and modulation format. There have been numerous studies on the effect of non-linearity in the probability of true and false detections in spectrum sensing when blockers are present. There has also been research showing the optimal modulation scheme for effective jamming. However, we are not aware of work analyzing the effect of modulated adjacent channel blockers on communication system performance. The approach taken in this paper is a theoretical derivation followed by numerical analysis aimed to quantify the effect of receiver nonlinearity on communication system performance as a function of (1) receiver characteristics, (2) blocking signal powers, (3) signal and blocker modulation format, and (4) phase-synchronized/non-synchronized blocker reception. The work focuses on M-PSK modulation schemes. For high blocker powers and non-linearity, the Es/No (Eb/No) performance loss can be as high as 4.7 dB for BPSK modulated signal and BPSK modulated blockers when received in sync with the desired signal. When blockers have a random phase offset with respect to the desired signal, the performance degradation is about 2 dB for BPSK modulated desired and blocker signals. It was found that for an BPSK transmitted signal with phase-synchronous blockers, the SER (BER) deteriorates the most when the blocking signals are of the same modulation. The effect is reduced, but still significant, as the modulation order of the signal of interest or the blockers, or both increases. / Master of Science

Modern devices provide a multitude of services that use radio frequencies in continual smaller packages. This size leads to an antenna used to transmit and receive information being usually very inefficient and a lot of power is wasted just to be able to transmit a signal. To mitigate this problem a new antenna was introduced by Dr. Manteghi that is capable of working efficiently across a large band. The antenna achieves this large band by doing quick frequency hopping across multiple channels. In order to test the performance of this antenna against more common antennas, a software radio was needed, such that tested antennas can be analyzed using multiple modulations. This paper presents a software defined radio system that was designed for the purpose of testing the bit-error rate of digital modulations schemes using described and other antennas. The designed system consists of a DSP, an FPGA, and commercially available modules. The combination allows the system to be flexible with high performance, while being affordable. Commercial modules are available for multiple frequency bands and capable of fast frequency switching required to test the antenna. The DSP board contains additional peripherals that allows for more complex projects in the future. The block structure of the system is also very educational as each stage of transmission and reception can be tested and observed. The full system has been constructed and tested using simulated and real signals. A code was developed for communication between commercial modules and the DSP, bit error rate testing, data transmission, signal generation, and signal reception. A graphical user interface (GUI) was developed to help user with information display and system control. This thesis describes the software-defined-radio design in detail and shows test results at the end. / Master of Science

Virginia Tech researchers have developed a broadband channel sounder that can measure channel quality while a wireless network is in operation. Channel measurements from the broadband sounder hold the promise of improving TCP/IP performance by trigging configuration changes in an adaptive data link layer protocol. We present an adaptive data link layer protocol that can use different levels of forward error correction (FEC) codes and link layer automatic retransmission request (ARQ) to improve network and transport layer performance. Using a simulation model developed in OPNET, we determine the effects of different data link layer protocol configurations on TCP/IP throughput and end-to-end delay using a Rayleigh fading channel model. Switching to higher levels of FEC encoding improves TCP/IP throughput for high bit error rates, but increases end-to-end delay of TCP/IP segments. Overall TCP/IP connections with link layer ARQ showed approximately 150 Kbps greater throughput than without ARQ, but lead to the highest end-to-end delay for high bit error rate channels. Based on the simulation results, we propose algorithms to maximize TCP/IP throughput and minimize end-to-end delay using the current bit error rate of the channel. We propose a metric, carrier-to-interference ratio (CIR) that is calculated from data retrieved from the broadband channel sounder. We propose algorithms using the carrier-to-interference ratio to control TCP/IP throughput and end-to-end delay. The thesis also describes a monitor program to use in the broadband wireless system. The monitor program displays data collected from the broadband sounder and controls the settings for the data link layer protocol and broadband sounder while the network is in operation. / Master of Science

Explosive growth in wireless technology caused by development in digital and RF circuit fabrications put some serious challenges on wireless system designers and link budget planning. Low transmit power, system coverage and capacity, high data rates, spatial diversity and quality of services (QOS) are the key factors in future wireless communication system that made it attractive. Dual-hop relaying is the promising underlying technique for future wireless communication to address such dilemmas. Based on dual-hop relaying this thesis addresses two scenarios. In the first case the system model employs dual-hop amplify and forward (AF) multiple input multiple output (MIMO) relay channels with transmit and receive antenna selection over independent Rayleigh fading channels where source and destination contain multiple antennas and communicate with each other with help of single antenna relay. It is assumed that the source and destination has perfect knowledge of channel state information (CSI). Our analysis shows that full spatial diversity order can be achieved with minimum number of antennas at source and destination i.e. min{N_s N_d }. In the second case the performance analysis of dual-hop amplify and forward (AF) multiple relay cooperative diversity network with best relay selection schemes over Rayleigh fading channels is investigated where the source and destination communicate with each other through direct and indirect links. Only the performance of best relay is investigated which participates in the transmission alone. The relay node that achieves highest SNR at the destination is selected as a best relay. Once again our analysis shows that full diversity order can be achieved with single relay with fewer resources compare to the regular cooperative diversity system. / Muhammad Adeel Islam 0046 700 412 343

This thesis describes and models wirelesss transmission intereferences in real communication channels. A signal received on a fading channel is subjected to a multiplicative distortion and to the usual additive noise. Real channel adds to the signal noise and fadings. Gaussian noise is a result of channel awgn. Fadings is implicated by multipath propagation of signal in Rayleigh and Rician channels. Main goal of this project is to program BERsolve, which is created in tool GUIDE in Matlab. User program BERsolve makes it possible to analyse bit error rate and symbol error rate for some types of channel models, which are defined by standard COST207. BERsolve offers some other functions i.e. display time behaviour, constellation diagram and spectrum. This program provides us with an overview representation of problem by multipath interferences in communication channels.

This dissertation thesis deals with new progressive channel coding methods for data transmission using satellite transponder. The design of the system for SMS transmission, in which novel turbo coding methods are applied, is discussed too. An achievement of the lowest output power and the smallest user aperture is the principal aim of the new method applications. Design of system that would be able to the error free SMS transmission by very low signal to noise ratio is analyzed in this dissertation thesis. The work is focused on energy budget, modification and implementation of new turbo code types and using unique properties, development of new bit error rate estimation methods and methods for determination of final SMS form. The main contribution is the new type of turbo code development that have optimal properties for this usage, development of new bit error rate estimation method and development of method that is able to determine final form of SMS on the basis SMS frame structure and turbo decoding theory when the bit error rate is not zero.

Det manuella arbetssättet för undertextframtagning är en tidskrävande och kostsam process. Arbetet undersöker AI-verktyget Whisper och dess potential att ersätta processen som används idag. Processen innefattar både transkribering och översättning.  För att verktyget ska kunna göra denna transkribering och översättning behöver den i första hand kunna omvandla tal till text. Detta kallas för taligenkänning och är baserat på upptränade språkmodeller. Precisionen för transkriberingen kan mätas med ordfelfrekvens (Word Error Rate – WER) och för översättningen med COMET-22.  Resultaten visade sig klara av Microsofts krav för maximalt tillåten WER och anses därför vara tillräckligt bra för användning. Resultaten indikerade även att de maskinproducerade översättningarna uppnår tillfredställande kvalitet. Undertextframtagning, som är det andra steget i processen, visade sig Whisper ha svårare för när det gäller skapandet av undertexter. Detta gällde både för transkriberingen i originalspråk samt den engelsköversatta versionen. Kvaliteten på undertexternas formatering, som mäts med SubER-metoden, kan tolkas som för låga för att anses vara användbara. Resultaten låg i intervallet 59 till 96% vilket innebär hur stor del av den automatiskt tillverkade undertexten behöver korrigeras för att matcha referensen.  Den övergripande slutsatsen man kan dra är att Whisper eventuellt kan ersätta den faktiska transkriberings -och översättningsprocessen, då den både är snabbare och kostar mindre resurser än det manuella tillvägagångssättet. Den är dock inte i skrivande stund tillräcklig för att ersätta undertextframtagningen. / The manual process of subtitling creation is a time consuming and costly process. This study examines the AI-tool Whisper and its potential of substituting the process used today. The process consists of both speech recognition and speech translation.  For the tool to accomplish the transcription and translation, it first needs to be able to convert speech-to-text. This is called speech recognition and is based on trained speech models. The precision for the transcription can be measured using the Word Error Rate (WER), while the translation uses COMET-22 for measuring precision.  The results met the requirements for maximal allowed WER-value and were therefore considered to be usable. The results also indicated that the machine produced translations reached satisfactory quality. Subtitle creation, which is the second part of the process, turned out to be more of a challenge for Whisper. This applied to both the transcription in the original language and the English translated version.  The quality of the subtitling format, measured using the SubER-method, can be interpreted as too low to be considered useful. The results were in the interval of 59 to 96% which informs how large part of the automatically created subtitle need to be corrected to match the reference.  The conclusion one can draw is that Whisper could eventually substitute the actual transcription and translation process, since it is both faster and costs less resources than the manual process. Though it is not good enough, in the moment of writing, to substitute the subtitling creation.