An Efficient DOCSIS Upstream Equalizer

The advancement in the CATV industry has been remarkable. In the beginning, CATV provided a few television channels. Now it provides a variety of advanced services such as video on demand (VOD), Internet access, Pay-Per-View on demand and interactive TV. These advances have increased the popularity of CATV manyfold. Current improvements focus on interactive services with high quality. These interactive services require more upstream (transmission from customer premises to cable operator premises) channel bandwidth. The flow of data through the CATV network in both the upstream and downstream directions is governed by a standard referred to as the Data Over Cable Service Interface Specification (DOCSIS) standard. The latest version is DOCSIS 3.1, which was released in January 2014. The previous version, DOCSIS 3.0, was released in 2006. One component of the upstream communication link is the QAM demodulator. An important component in the QAM demodulator is the equalizer, whose purpose is to remove distortion caused by the imperfect upstream channel as well as the residual timing offset and frequency offset. Most of the timing and frequency offset are corrected by timing and frequency recovery circuits; what remains is referred to as offset. A DOCSIS receiver, and hence the equalizer within, can be implemented with ASIC or FPGA technology. Implementing an equalizer in an ASIC has a large nonrecurring engineering cost, but relatively small per chip production cost. Implementing equalizer in an FPGA has very low non-recurring cost, but a relatively high per chip cost. If the choice technology was based on cost, one would think it would depends only on the volume, but in practice that is not the case. The dominant factor when it comes to profit, is the time-to-market, which makes FPGA technology the only choice. The goal of this thesis is to design a cost optimized equalizer for DOCSIS upstream demodulator and implement in an FPGA. With this in mind, an important objective is to establish a relationship between the equalizer’s critical parameters and its performance. The parameter-performance relationship that has been established in this study revealed that equalizer step size and length parameters should be 1/64 and approximately 20 to yield a near optimum equalizer when considering the MER-convergence time trade-off. In the pursuit of the objective another relationship was established that is useful in determining the accuracy of the timing recovery circuit. That relationship establishes the sensitivity both of the MER and convergence time to timing offset. The equalizer algorithm was implemented in a cost effective manner using DSP Builder. The effort to minimize cost was focused on minimizing the number of multipliers. It is shown that the equalizer can be constructed with 8 multipliers when the proposed time sharing algorithm is implemented

Fiber echo state network analogue for high-bandwidth dual-quadrature signal processing

All-optical platforms for recurrent neural networks can offer higher computational speed and energy efficiency. To produce a major advance in comparison with currently available digital signal processing methods, the new system would need to have high bandwidth and operate both signal quadratures (power and phase). Here we propose a fiber echo state network analogue (FESNA) — the first optical technology that provides both high (beyond previous limits) bandwidth and dual-quadrature signal processing. We demonstrate applicability of the designed system for prediction tasks and for the mitigation of distortions in optical communication systems with multilevel dual-quadrature encoded signals

Solutions for New Terrestrial Broadcasting Systems Offering Simultaneously Stationary and Mobile Services

221 p.[EN]Since the first broadcasted TV signal was transmitted in the early decades of the past century, the television broadcasting industry has experienced a series of dramatic changes. Most recently, following the evolution from analogue to digital systems, the digital dividend has become one of the main concerns of the broadcasting industry. In fact, there are many international spectrum authorities reclaiming part of the broadcasting spectrum to satisfy the growing demand of other services, such as broadband wireless services, arguing that the TV services are not very spectrum-efficient. Apart from that, it must be taken into account that, even if up to now the mobile broadcasting has not been considered a major requirement, this will probably change in the near future. In fact, it is expected that the global mobile data traffic will increase 11-fold between 2014 and 2018, and what is more, over two thirds of the data traffic will be video stream by the end of that period. Therefore, the capability to receive HD services anywhere with a mobile device is going to be a mandatory requirement for any new generation broadcasting system. The main objective of this work is to present several technical solutions that answer to these challenges. In particular, the main questions to be solved are the spectrum efficiency issue and the increasing user expectations of receiving high quality mobile services. In other words, the main objective is to provide technical solutions for an efficient and flexible usage of the terrestrial broadcasting spectrum for both stationary and mobile services. The first contributions of this scientific work are closely related to the study of the mobile broadcast reception. Firstly, a comprehensive mathematical analysis of the OFDM signal behaviour over time-varying channels is presented. In order to maximize the channel capacity in mobile environments, channel estimation and equalization are studied in depth. First, the most implemented equalization solutions in time-varying scenarios are analyzed, and then, based on these existing techniques, a new equalization algorithm is proposed for enhancing the receivers’ performance. An alternative solution for improving the efficiency under mobile channel conditions is treating the Inter Carrier Interference as another noise source. Specifically, after analyzing the ICI impact and the existing solutions for reducing the ICI penalty, a new approach based on the robustness of FEC codes is presented. This new approach employs one dimensional algorithms at the receiver and entrusts the ICI removing task to the robust forward error correction codes. Finally, another major contribution of this work is the presentation of the Layer Division Multiplexing (LDM) as a spectrum-efficient and flexible solution for offering stationary and mobile services simultaneously. The comprehensive theoretical study developed here verifies the improved spectrum efficiency, whereas the included practical validation confirms the feasibility of the system and presents it as a very promising multiplexing technique, which will surely be a strong candidate for the next generation broadcasting services.[ES]Desde el comienzo de la transmisión de las primeras señales de televisión a principios del siglo pasado, la radiodifusión digital ha evolucionado gracias a una serie de cambios relevantes. Recientemente, como consecuencia directa de la digitalización del servicio, el dividendo digital se ha convertido en uno de los caballos de batalla de la industria de la radiodifusión. De hecho, no son pocos los consorcios internacionales que abogan por asignar parte del espectro de radiodifusión a otros servicios como, por ejemplo, la telefonía móvil, argumentado la poca eficiencia espectral de la tecnología de radiodifusión actual. Asimismo, se debe tener en cuenta que a pesar de que los servicios móviles no se han considerado fundamentales en el pasado, esta tendencia probablemente variará en el futuro cercano. De hecho, se espera que el tráfico derivado de servicios móviles se multiplique por once entre los años 2014 y 2018; y lo que es más importante, se pronostica que dos tercios del tráfico móvil sea video streaming para finales de ese periodo. Por lo tanto, la posibilidad de ofrecer servicios de alta definición en dispositivos móviles es un requisito fundamental para los sistemas de radiodifusión de nueva generación. El principal objetivo de este trabajo es presentar soluciones técnicas que den respuesta a los retos planteados anteriormente. En particular, las principales cuestiones a resolver son la ineficiencia espectral y el incremento de usuarios que demandan mayor calidad en los contenidos para dispositivos móviles. En pocas palabras, el principal objetivo de este trabajo se basa en ofrecer una solución más eficiente y flexible para la transmisión simultánea de servicios fijos y móviles. La primera contribución relevante de este trabajo está relacionada con la recepción de la señal de televisión en movimiento. En primer lugar, se presenta un completo análisis matemático del comportamiento de la señal OFDM en canales variantes con el tiempo. A continuación, con la intención de maximizar la capacidad del canal, se estudian en profundidad los algoritmos de estimación y ecualización. Posteriormente, se analizan los algoritmos de ecualización más implementados, y por último, basándose en estas técnicas, se propone un nuevo algoritmo de ecualización para aumentar el rendimiento de los receptores en tales condiciones. Del mismo modo, se plantea un nuevo enfoque para mejorar la eficiencia de los servicios móviles basado en tratar la interferencia entre portadoras como una fuente de ruido. Concretamente, tras analizar el impacto del ICI en los receptores actuales, se sugiere delegar el trabajo de corrección de dichas distorsiones en códigos FEC muy robustos. Finalmente, la última contribución importante de este trabajo es la presentación de la tecnología LDM como una manera más eficiente y flexible para la transmisión simultánea de servicios fijos y móviles. El análisis teórico presentado confirma el incremento en la eficiencia espectral, mientras que el estudio práctico valida la posible implementación del sistema y presenta la tecnología LDM c

Tree-Structured Nonlinear Adaptive Signal Processing

In communication systems, nonlinear adaptive filtering has become increasingly popular in a variety of applications such as channel equalization, echo cancellation and speech coding. However, existing nonlinear adaptive filters such as polynomial (truncated Volterra series) filters and multilayer perceptrons suffer from a number of problems. First, although high Order polynomials can approximate complex nonlinearities, they also train very slowly. Second, there is no systematic and efficient way to select their structure. As for multilayer perceptrons, they have a very complicated structure and train extremely slowly Motivated by the success of classification and regression trees on difficult nonlinear and nonparametfic problems, we propose the idea of a tree-structured piecewise linear adaptive filter. In the proposed method each node in a tree is associated with a linear filter restricted to a polygonal domain, and this is done in such a way that each pruned subtree is associated with a piecewise linear filter. A training sequence is used to adaptively update the filter coefficients and domains at each node, and to select the best pruned subtree and the corresponding piecewise linear filter. The tree structured approach offers several advantages. First, it makes use of standard linear adaptive filtering techniques at each node to find the corresponding Conditional linear filter. Second, it allows for efficient selection of the subtree and the corresponding piecewise linear filter of appropriate complexity. Overall, the approach is computationally efficient and conceptually simple. The tree-structured piecewise linear adaptive filter bears some similarity to classification and regression trees. But it is actually quite different from a classification and regression tree. Here the terminal nodes are not just assigned a region and a class label or a regression value, but rather represent: a linear filter with restricted domain, It is also different in that classification and regression trees are determined in a batch mode offline, whereas the tree-structured adaptive filter is determined recursively in real-time. We first develop the specific structure of a tree-structured piecewise linear adaptive filter and derive a stochastic gradient-based training algorithm. We then carry out a rigorous convergence analysis of the proposed training algorithm for the tree-structured filter. Here we show the mean-square convergence of the adaptively trained tree-structured piecewise linear filter to the optimal tree-structured piecewise linear filter. Same new techniques are developed for analyzing stochastic gradient algorithms with fixed gains and (nonstandard) dependent data. Finally, numerical experiments are performed to show the computational and performance advantages of the tree-structured piecewise linear filter over linear and polynomial filters for equalization of high frequency channels with severe intersymbol interference, echo cancellation in telephone networks and predictive coding of speech signals

Algorithms and structures for long adaptive echo cancellers

The main theme of this thesis is adaptive echo cancellation. Two novel independent approaches are proposed for the design of long echo cancellers with improved performance. In the first approach, we present a novel structure for bulk delay estimation in long echo cancellers which considerably reduces the amount of excess error. The miscalculation of the delay between the near-end and the far-end sections is one of the main causes of this excess error. Two analyses, based on the Least Mean Squares (LMS) algorithm, are presented where certain shapes for the transitions between the end of the near-end section and the beginning of the far-end one are considered. Transient and steady-state behaviours and convergence conditions for the proposed algorithm are studied. Comparisons between the algorithms developed for each transition are presented, and the simulation results agree well with the theoretical derivations. In the second approach, a generalised performance index is proposed for the design of the echo canceller. The proposed algorithm consists of simultaneously applying the LMS algorithm to the near-end section and the Least Mean Fourth (LMF) algorithm to the far-end section of the echo canceller. This combination results in a substantial improvement of the performance of the proposed scheme over both the LMS and other algorithms proposed for comparison. In this approach, the proposed algorithm will be henceforth called the Least Mean Mixed-Norm (LMMN) algorithm. The advantages of the LMMN algorithm over previously reported ones are two folds: it leads to a faster convergence and results in a smaller misadjustment error. Finally, the convergence properties of the LMMN algorithm are derived and the simulation results confirm the superior performance of this proposed algorithm over other well known algorithms

Performance of SC-FDMA with diversity techniques over land mobile satellite channel

La demanda de la alta velocidad de datos resulta en una importante interferencia entre símbolos para los sistemas monoportadora en canales de ancho de banda y potencia limitada. Superar la selectividad en el tiempo y la frecuencia del canal de propagación requiere el uso de potentes técnicas de procesamiento de señales. Ejemplos recientes incluyen el uso de múltiples antenas en el transmisor / receptor, en la técnica conocida como Multiple-Input Multiple-Output (MIMO). En ciertos entornos (tales como el enlace ascendente de un enlace móvil) por lo general sólo una antena está disponible en la transmisión. Por lo tanto, sólo esquemas con entrada individual y salida única (Single Input Single Output, SISO) o transmisiones con entrada única y múltiples salidas (Single Input Multiple Output, SIMO) son factibles. La multiplexación por división ortogonal en frecuencia (Orthogonal Frequency-Division Multiplexing, OFDM) es una técnica de modulación ampliamente utilizada por su robustez frente a la selectividad en frecuencia de los canales, su escalabilidad y su compatibilidad con MIMO. Sin embargo, sufre de una alta relación de potencia de pico a promedio (Peak-to-Average Power Ratio, PAPR) que necesita amplificadores de alta potencia muy lineales, lo que resulta costoso energéticamente para la transmisión. La técnica monoportadora con acceso múltiple por división de frecuencia (Single Carrier Frequency-Division Multiple Access , SC-FDMA) se ha convertido en una alternativa a la técnica de OFDM que se utiliza específicamente en el enlace ascendente de LTE. SC-FDMA es capaz de reducir la PAPR en la transmisión, dando lugar a una relajación de las limitaciones en cuanto a la eficiencia de potencia necesaria en los terminales de usuario y las unidades satélite. SC-FDMA puede ser descrito como una versión de OFDMA en el que se incluyen una etapa de pre-codificación y de pre-codificación inversa en el transmisor y el receptor respectivamente. Así, los símbolos se transmiten en tiempo, pero después de ser procesados en la frecuencia. Incluso con el uso de OFDMA o SC-FDMA, la ISI tiene que ser compensada por la igualación, que normalmente se realiza en el dominio de frecuencia. El objetivo de esta tesis es proporcionar un análisis matemático del comportamiento de SC-FDMA en un canal móvil terrestre por satélite (Land Mobile Satellite, LMS). Para este propósito, el canal se modela como un canal Rice sombreado tal que la línea de visión (Line of Sight, LOS) sigue la distribución de Nakagami. En primer lugar, se describen las técnicas de modulación multiportadora OFDMA y SC-FDMA. A continuación, se lleva a cabo un análisis de OFDMA y SC-FDMA basado en el ruido complejo recibido a la entrada del detector. Se evalúa la probabilidad de error de bit (Bit Error Rate, BER) de SC-FDMA para diferentes profundidades del desvanecimiento y de la diversidad de antena en el receptor. También se evalúa la eficiencia espectral de SC-FDMA para el canal LMS. Por último, se abordan las técnicas de diversidad y se evalúan las técnicas conocidas como Maximal Ratio Combining (MRC) y Equal Gain Combining (EGC)