Orientador: Jaime Portugheis / Dissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica e de Computação / Made available in DSpace on 2018-08-11T03:16:26Z (GMT). No. of bitstreams: 1 Nascimento_VagnerValedo_M.pdf: 1654073 bytes, checksum: fe7023b9741222f4780c06c588baf3a4 (MD5) Previous issue date: 2007 / Resumo: Este trabalho apresenta um estudo sobre a decodificação turbo q-ária em sistemas FFH-CDMA. Para alcançar o principal objetivo do estudo, foi desenvolvido um algoritmo para implementar a decodificação suave q-ária. Este algoritmo se baseia em uma adaptação do método de decodificação de Chase para suportar alfabetos q-ários. A proposta de decodificação iterativa (turbo) define um procedimento para viabilizar a realimentação dos símbolos q-ários decodificados e de suas respectivas confiabilidades em cada iteração do decodificador. Simulações foram realizadas considerando canais gaussianos e FFH-CDMA. Os resultados obtidos nas simulações demonstram uma considerável melhoria no desempenho dos sistemas com a decodificação turbo q-ária, sem comprometer a complexidade. Entretanto, a utilização dos códigos produto reduzem a eficiência espectraf do sistema, sendo necessário, assim, compensá-Ia através do aumento do alfabeto do código q-ário. Palavras-chave: Códigos q-ários, Códigos Produto, Decodificação Suave, Decodificação TUrbo, SISO, Sistemas FFH-CDMA / Abstract: This work presents a study of q-ary turbo decoding applied to FFH-CDMA systems. To attain the main objective of this dissertation, an algorithm to implement the q-ary soft decoding was designed. Specifically, an adaptation of the Chase decoding method was proposed to comply with q-ary symbols requirements. The proposal of iterative (turbo) decoding makes possible the feedback of the decoded q-ary symbols and its reliabilities in each decoder iteration. Simulations were done considering Gaussian and FFH-CDMA channels. The results obtained in the simulations indicate a better system's performance when the q-ary turbo decoding is applied, without compromising complexity. However, the use of product codes decreases the system's spectral efficiency. Thus, it is necessary to increase the q-ary code's alphabet in order to mitigate this decréase. Keywords: Non-Binary Codes, Product Codes, Soft Decoding, Turbo Decoding, SISO, FFH-CDMA / Mestrado / Telecomunicações e Telemática / Mestre em Engenharia Elétrica

Contemporary and future speech telecommunication systems now utilise low bit rate (LBR) speech coding techniques in efforts to eliminate bandwidth expansion as a disadvantage of digital coding and transmission. These speech coders employ model-based approaches in compressing human speech into a number of parameters, using a well-known process known as linear predictive coding (LPC). However, a major side-effect observed in these coders is that errors in the model parameters have noticeable and undesirable consequences on the synthesised speech quality, and unless they are protected from such corruptions, the level of service quality will deteriorate rapidly. Traditionally, forward error correction (FEC) coding is used to remove these errors, but these require substantial redundancy. Therefore, a different perspective of the error control problems and solutions is necessary. In this thesis, emphasis is constantly placed on exploiting the constraints and residual redundancies present in the model parameters. It is also shown that with such source criteria in the LBR speech coders, varying degrees of error protection from channel corruptions are feasible. From these observations, error control requirements and methodologies, using both block- and parameter-orientated aspects, are analysed, devised and implemented. It is evident, that under the unusual circumstances which LBR speech coders have to operate in, the importance and significance of source reliant error control will continue to attract research and commercial interests. The work detailed in this thesis is focused on two LPC-based speech coders. One of the ideas developed for these two coders is an advanced zero redundancy scheme for the LPC parameters which is designed to operate at high channel error rates. Another concept proposed here is the use of source criteria to enhance the decoding capabilities of FEC codes to exceed that of maximum likelihood decoding performance. Lastly, for practical operation of LBR speech coders, lost frame recovery strategies are viewed to be an indispensable part of error control. This topic is scrutinised in this thesis by investigating the behaviour of a specific speech coder under irrecoverable error conditions. In all of the ideas pursued above, the effectiveness of the algorithms formulated here are quantified using both objective and subjective tests. Consequently, the capabilities of the techniques devised in this thesis can be demonstrated, examples of which are: (1) higher speech quality produced under noisy channels, using an improved zero-redundancy algorithm for the LPC filter coefficients; (2) as much as 50% improvement in the residual BER and decoding failures of FEC schemes, through the utilisation of source criteria in LBR speech coders; and (3) acceptable speech quality produced under high frame loss rates (14%), after formulating effective strategies for recovery of speech coder parameters. It is hoped that the material described here provide concepts which can help achieve the ideals of maximum efficiency and quality in LBR speech telecommunications.

Over the past two decades there has been substantial growth in speech communications and new speech related applications. Bandwidth constraints led researchers to investigate ways of compressing speech signals whilst maintaining speech quality and intelligibility so as to increase the possible number of customers for the given bandwidth. Because of this a variety of speech coding techniques have been proposed over this period. At the heart of any proposed speech coding method is quantisation of the speech production model parameters that need to be transmitted to the decoder. Quantisation is a controlling factor for the targeted bit rates and for meeting quality requirements. The objectives of the research presented in this thesis are twofold. The first enabling the development of a very low bit rate speech coder which maintains quality and intelligibility. This includes increasing the robustness to various operating conditions as well as enhancing the estimation and improving the quantisation of speech model parameters. The second objective is to provide a method for enhancing the performance of an existing speech related application. The first objective is tackled with the aid of three techniques. Firstly, various novel estimation techniques are proposed which are such that the resultant estimated speech production model parameters have less redundant information and are highly correlated. This leads to easier quantisation (due to higher correlation) and therefore to bit saving. The second approach is to make use of the joint effect of the quantisation of spectral parameters (i.e. LSF and spectral amplitudes) for their big impact on the overall bit allocation required. Work towards the first objective also includes a third technique which enhances the estimation of a speech model parameter (i.e. the pitch) through a robust statistics-based post-processing (or tracking) method which operates in noise contaminated environments. Work towards the second objective focuses on an application where speech plays an important role, namely echo-canceller and noise-suppressor systems. A novel echo-canceller method is proposed which resolves most of the weaknesses present in existing echo-canceller systems and improves the system performance.

M.Ing. (Electrical & Electronic Engineering) / Coding is indispensable in modem communications and storage systems. For instance, a Reed-Solomon error-correction code ensures higher data integrity for the Compact Disc (CD) system. Modulation codes, such as the( d, k) codes, can furthermore be employed to enable synchronization between transmitter and receiver (or between the read and write processes in storage systems), and also to achieve compliance with bandwidth restrictions. In some cases, a 'combined' code is designed to function both as an error control code and a (d, k) modulation code. In this study, we consider such an existing class of error control (d, k) block codes. Of particular interest is the performance of these codes, determined mostly in terms of the probability of block error over certain selected channels. It is important to be able to judge or predict the performance of a communication system in terms of the probability of receiving incorrect information, and this probability depends not only on the specific error control code used, but also on the statistical structure of the channel error processes. The main contributions of this thesis are seen to be the following: 1) New codes and improvements on previous codes, originating from the generalization of existing theory. 2) Mathematical bounds and approximations on block error rates, compared with measured results from computer simulations.

Ph.D. (Electrical & Electronic Engineering) / In the ideal communication system no noise is present and error free communication is possible. In practice, however, several factors influence the correctness of· the communication. One of the most important of these factors is the synchronization of the message. Synchronization techniques form an integral part of data communication systems and without synchronization no comprehensible message can be received. An example of a communication system in which synchronization errors occur is the plesiochronous communication network which is used in many telephone networks [49,50]. A common problem with the use of the multiplexers in such a network is that output pulses may occur that do not contain valid data, due to minor discrepancies in the clock frequencies of the incoming signals. These inserted bits are termed ''justification bits" and their presence is signaled over the link by the justification control bits or stuffing control bits, which are included in the frame [49 and 50]. Synchronization of the network is dependent, among other factors, on the correct decoding of the stuffing control bits. Synchronization at the receiver can also be lost when the frame markers are not recognizable due to errors on the channel...

A performance evaluation of CSTC (Cooperative Space Time Coding) with spatially cor-related fading and imperfect channel estimation in Gaussian as well as impulsive noise is presented. Closed form expressions for the pairwise error probability conditioned on the estimated channel gains are derived by assuming the components of the received vector are independent given the estimated channel gains. An expurgated union bound using the limiting before averaging technique given the estimated channel gains is then obtained. Although this assumption is not strictly valid, simulation results show that the bound is accurate in estimating the diversity order as long the channel estimation is not very poor. It is found that CSTC with block fading channels can reduce the frame error rate (FER) relative to SUSTC (Single User Space Time Coding) with quasi-static fading channels, even when the channel gains for each user are strongly correlated and when the channel estimations are very poor. A decision metric for CSTC with spatially correlated fading, imperfect channel estimation, and impulsive mixture Gaussian noise is derived which yields lower FERs than the Gaussian noise decision metric. Simulation results show that the FER performance of CSTC with mixture Gaussian noise outperforms CSTC with Gaussian noise at low SNR. At high SNR, the FER performance of CSTC with Gaussian noise is better than the FER performance of CSTC with mixture Gaussian noise due to the heavy tail of the mixture Gaussian noise. / Applied Science, Faculty of / Electrical and Computer Engineering, Department of / Graduate

Conventional studies in communication networks mostly focus on securely and reliably transmitting  data from a source node (or multiple source nodes) to multiple destinations. A more general problem appears when the destination nodes are interested in obtaining  functions of the data available in distributed source nodes. For obtaining a function, transmitting all the data to a destination node and then computing the function might be inefficient. In order to exploit the network resources efficiently, the general problem offers distributed computing in combination with coding and communication. This problem has applications in distributed systems, e.g., in wireless sensor networks, in distributed storage systems, and in distributed computing systems. Following this general problem formulation, we study the optimal and secure recovery of the lost data in storage nodes and in reconstructing a version of a file in distributed storage systems.   The significance of this study is due to the fact that the new trends in communications including big data, Internet of things, low latency, and high reliability communications challenge the existing centralized data storage systems. Distributed storage systems can rectify those issues by  distributing  thousands of storage nodes (possibly around the globe), and then benefiting users by bringing data to their proximity.  Yet, distributing the storage nodes brings new challenges. In these distributed systems, where storage nodes  are connected through links and servers, communication plays a main role in their performance. In addition,  a part of network may fail or due to communication failure or delay there might exist multi versions of a file. Moreover, an intruder can overhear the communications between storage nodes and obtain some information about the stored data. Therefore, there are challenges on  reliability, security, availability, and consistency.   To increase reliability, systems need to store redundant data in storage nodes and employ error control codes. To maintain the  reliability  in a dynamic environment where storage nodes can fail, the system should have an autonomous repair process. Namely, it should regenerate the failed nodes by the help of other storage nodes. The repair process demands bandwidth, energy, or in general transmission costs.  We propose novel techniques to reduce the repair cost in distributed storage systems.   First, we propose {surviving nodes cooperation} in repair, meaning that surviving nodes can combine their received data with their own stored data and then transmit toward the new node. In addition, we study the repair problem in multi-hop networks and consider the cost of transmitting data between storage nodes.  While classical repair model assumes the availability of direct links between the new node and surviving nodes, we consider that such links may not be available either due to failure or their costs.  We formulate an optimization problem to minimize the repair cost and compare two systems, namely with and without surviving nodes cooperation.   Second, we study the repair problem where the links between storage nodes are lossy e.g., due to server congestion, load balancing, or unreliable physical layer (wireless links).  We model the lossy links by packet erasure channels and then derive the fundamental bandwidth-storage tradeoff in packet erasure networks. In addition, we propose dedicated-for-repair storage nodes to reduce the repair-bandwidth.   Third, we generalize the repair model by proposing the concept of partial repair. That is, storage nodes may lose parts of their stored data. Then in partial repair, the lost data is recovered by exchanging data between storage nodes and using the available data in storage nodes as side information. For efficient partial-repair,  we propose two-layer coding in distributed storage systems and then we derive the optimal bandwidth in partial repair.   Fourth, we study security in distributed storage systems.  We investigate security in partial repair. In particular, we propose codes that make the partial repair secure in the senses of strong and weak information-theoretic security definitions.   Finally, we study consistency in distributed storage systems. Consistency means that distinct users obtain the latest version of a file in a system that stores multi versions of a file. Given the probability of receiving a version by a storage node and the constraint on the node storage space, we aim to find the optimal encoding of multi versions of a file that maximizes the probability of obtaining the latest version of a file or a version close to the latest version by a read client that connects to a number of storage nodes. / <p>Pages 153-168 are removed due to copyright reasons.</p><p>QC 20161012</p><p></p>

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The purpose of this study was to investigate perceptions of high school students who made music with code in Sonic Pi. This qualitative multiple case study focused on individuals in an extracurricular club at a public charter high school who volunteered to participate on-site and remotely asynchronously via Canvas learning management system. This study was guided by five research questions, including: (1) What musical ideas, if any, do participants report learning or demonstrate through making music with code in Sonic Pi? (2) How does making music with code impact participants’ perceptions of their music making? (3) How does making music with code impact participants’ perceptions of their ability to learn to make music? (4) How does making music with code impact participants’ interest in music courses? (5) How does making music with code impact participants’ interest in computer science courses? Participants completed research study materials, including a series of tutorials for Sonic Pi. Data included answers to questionnaires and surveys, multimedia artifacts including the source code and exported audio of participants’ music making, and interviews of participants that were codified and analyzed in two cycles, utilizing descriptive coding, values coding, and longitudinal coding. Participants’ code and multimedia artifacts revealed a close alignment to the four properties of sound, including: pitch, duration, intensity/amplitude, and timbre. Participants’ artifacts revealed themes and demonstrated ideas extending beyond the four properties, including: form, non-traditional music notation, and randomization. Participants all agreed their coded artifacts are music. Additionally, participants’ varied responses about musicianship and composers suggests that making music is something anyone can engage in, regardless of how one identifies themself. All participants agreed that Sonic Pi is a useful tool for learning and understanding musical concepts and that Western staff notation is not required knowledge for making music. Participants’ interests in music or computer science courses were impacted by their prior experiences in music and/or coding. This study concludes with a discussion of themes based on the findings.