Space-time-frequency methods for interference-limited communication systems

textTraditionally, noise in communication systems has been modeled as an additive, white Gaussian noise process with independent, identically distributed samples. Although this model accurately reflects thermal noise present in communication system electronics, it fails to capture the statistics of interference and other sources of noise, e.g. in unlicensed communication bands. Modern communication system designers must take into account interference and non-Gaussian noise to maximize efficiencies and capacities of current and future communication networks. In this work, I develop new multi-dimensional signal processing methods to improve performance of communication systems in three applications areas: (i) underwater acoustic, (ii) powerline, and (iii) multi-antenna cellular. In underwater acoustic communications, I address impairments caused by strong, time-varying and Doppler-spread reverberations (self-interference) using adaptive space-time signal processing methods. I apply these methods to array receivers with a large number of elements. In powerline communications, I address impairments caused by non-Gaussian noise arising from devices sharing the powerline. I develop and apply a cyclic adaptive modulation and coding scheme and a factor-graph-based impulsive noise mitigation method to improve signal quality and boost link throughput and robustness. In cellular communications, I develop a low-latency, high-throughput space-time-frequency processing framework used for large scale (up to 128 antenna) MIMO. This framework is used in the world's first 100-antenna MIMO system and processes up to 492 Gbps raw baseband samples in the uplink and downlink directions. My methods prove that multi-dimensional processing methods can be applied to increase communication system performance without sacrificing real-time requirements.Electrical and Computer Engineerin

Performance enhancement for LTE and beyond systems

A thesis submitted to the University of Bedfordshire, in partial fulfilment of the requirements for the degree of Doctor of PhilosophyWireless communication systems have undergone fast development in recent years. Based on GSM/EDGE and UMTS/HSPA, the 3rd Generation Partnership Project (3GPP) specified the Long Term Evolution (LTE) standard to cope with rapidly increasing demands, including capacity, coverage, and data rate. To achieve this goal, several key techniques have been adopted by LTE, such as Multiple-Input and Multiple-Output (MIMO), Orthogonal Frequency-Division Multiplexing (OFDM), and heterogeneous network (HetNet). However, there are some inherent drawbacks regarding these techniques. Direct conversion architecture is adopted to provide a simple, low cost transmitter solution. The problem of I/Q imbalance arises due to the imperfection of circuit components; the orthogonality of OFDM is vulnerable to carrier frequency offset (CFO) and sampling frequency offset (SFO). The doubly selective channel can also severely deteriorate the receiver performance. In addition, the deployment of Heterogeneous Network (HetNet), which permits the co-existence of macro and pico cells, incurs inter-cell interference for cell edge users. The impact of these factors then results in significant degradation in relation to system performance. This dissertation aims to investigate the key techniques which can be used to mitigate the above problems. First, I/Q imbalance for the wideband transmitter is studied and a self-IQ-demodulation based compensation scheme for frequencydependent (FD) I/Q imbalance is proposed. This combats the FD I/Q imbalance by using the internal diode of the transmitter and a specially designed test signal without any external calibration instruments or internal low-IF feedback path. The instrument test results show that the proposed scheme can enhance signal quality by 10 dB in terms of image rejection ratio (IRR). In addition to the I/Q imbalance, the system suffers from CFO, SFO and frequency-time selective channel. To mitigate this, a hybrid optimum OFDM receiver with decision feedback equalizer (DFE) to cope with the CFO, SFO and doubly selective channel. The algorithm firstly estimates the CFO and channel frequency response (CFR) in the coarse estimation, with the help of hybrid classical timing and frequency synchronization algorithms. Afterwards, a pilot-aided polynomial interpolation channel estimation, combined with a low complexity DFE scheme, based on minimum mean squared error (MMSE) criteria, is developed to alleviate the impact of the residual SFO, CFO, and Doppler effect. A subspace-based signal-to-noise ratio (SNR) estimation algorithm is proposed to estimate the SNR in the doubly selective channel. This provides prior knowledge for MMSE-DFE and automatic modulation and coding (AMC). Simulation results show that this proposed estimation algorithm significantly improves the system performance. In order to speed up algorithm verification process, an FPGA based co-simulation is developed. Inter-cell interference caused by the co-existence of macro and pico cells has a big impact on system performance. Although an almost blank subframe (ABS) is proposed to mitigate this problem, the residual control signal in the ABS still inevitably causes interference. Hence, a cell-specific reference signal (CRS) interference cancellation algorithm, utilizing the information in the ABS, is proposed. First, the timing and carrier frequency offset of the interference signal is compensated by utilizing the cross-correlation properties of the synchronization signal. Afterwards, the reference signal is generated locally and channel response is estimated by making use of channel statistics. Then, the interference signal is reconstructed based on the previous estimate of the channel, timing and carrier frequency offset. The interference is mitigated by subtracting the estimation of the interference signal and LLR puncturing. The block error rate (BLER) performance of the signal is notably improved by this algorithm, according to the simulation results of different channel scenarios. The proposed techniques provide low cost, low complexity solutions for LTE and beyond systems. The simulation and measurements show good overall system performance can be achieved

Implementation of a wireless underwater video link

Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2008.This electronic version was submitted by the student author. The certified thesis is available in the Institute Archives and Special Collections.Includes bibliographical references (p. 73-75).Autonomous underwater vehicles (AUVs) are increasingly being considered for remotely supervised missions, primarily for routine subsea inspection tasks currently performed by tethered remotely operated vehicles (ROVs). This project is a step in the development of a high speed, networked wireless communication capability for AUVs using MIT Sea Grant's software defined Reconfigurable Modem (R-Modem) acoustic communications platform. We have demonstrated a test implementation of live streaming video with a digital camera connected to an R-Modem transceiver; and explored a range of tuning parameters for the video link.by James Paul Morash.M.Eng

Comparison of proposals for the future aeronautical communication system LDACS

Um zukünftigen Kapazitätsbedarf in aeronautischer Navigation abzudecken, werden neue Bord und Boden Kommunikationsdienste gebraucht. Die europäische Organisation für Sicherheit und Luftnavigation, Eurocontrol, unterstützte die Entwicklung zweier Vorschläge für ein solches System. Der erste Vorschlag, genannt LDACS1, ist ein digitales Breitband OFDM basiertes System, welches vom Institut für Kommunikation und Navigation, DLR entwickelt wurde. Der zweite Vorschlag, LDACS2 wird von einem Projektteam bestehend aus EGIS ASVIA, Helios SWEDAVIA und anderen entwickelt. LDACS2 folgt einem single carrier Steuerung mit einer GMSK Modulation. Beide Systeme sind für das Bedienen des aeronautischen Teils des L-Band (960-1164 MHz) gedacht. Diese Frequenz wird jedoch bereits von verschiedenen aeronautischen alte Systemen wie z.B. zivile Luftfahrtnavigation DME oder militärische Kommunikationssystemen (vereinigtes taktisches Informationsverteilungssystem JTIDS) eingesetzt. Darüber hinaus, LDACS ist offen für in der Luft befindlich Empfangsstörungen. Ein entscheidender Punkt im Auswahlprozess für eine der LDACS Systeme ist die Gewährleistung für das Nebeneinander von LDACS und des legacy Systems. Einerseits muss bewiesen werden, dass LDACS nur einen geringen Einfluss auf das legacy System hat. Andererseits muss eine verlässliche Funktion trotz Empfangsstörung (Beeinträchtigung) gewährleistet werden. In dieser Masterarbeit ist die Leistung von LDACS2 analysiert. Die Aufgabe umfasst einige theoretische Überlegungen für Ermittlungen von Kapazität, spektrale Leistungsfähigkeit, Skalierbarkeit und die mögliche Zahl gleichzeitiger Nutzer. Das Ergebnis zeigt die Beschränkung der angebotenen bit rates pro Nutzer gemäß der limitierten Bandbreite. Jedoch für gering bis mittelmäßigen Inanspruchnahme von Anwendern, die angebotenen bit rates sind innerhalb einer akzeptablen Reichweite. Der Hauptteil dieser Arbeit befasst sich mit der Anwendung des LDACS2 Systems gemäß der Simulations-Software. Das umfasst die gesamte physikalische Schichtung und die grundlegenden Teile der höheren Schichtung. Besonderer Schwerpunkt ist auf die Anwendung und Beurteilung von wirksamen Kanal Entzerrung Algorithms, Analyse und Auswertung. Neben AWGN Kanälen wurden auch praxisbezogenen Luftfahrtfrequenzen angewandt. Es stellte sich heraus, dass das Kanalkodierung in dieser Ausführung nicht genügend.Ilmenau, Techn. Univ., Masterarbeit, 201

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

Noise cancelling in acoustic voice signals with spectral subtraction

The main purpose of study throughout this entire End of Degree Project would be the noise removal within speech signals, focusing on the diverse amount of algorithms using the spectral subtraction method. A Matlab application has been designed and created. The application main goal is to remove any meaningless thing considered as a disturb element when trying to perceive a voice; that is, anything considered as a noise. Noise removal is the basis for any voice processing that the user wants to do later, as speech recognition, save the clean audio, voice analysis, etc. A studio on four algorithms has been executed, in order to perform the spectral subtraction: Boll, Berouti, Lockwood & Boudy, and Multiband. This document presents a theoretical study and its implementation. Moreover, in order to have ready for the user a suitable implementation of an application, an intuitive and simple interface has been designed. This document shows how the different algorithms work in some voices and with various types of noise. A few amounts of noises are ideal, used by its mathematical characteristics, while others, are quite common and presented in daily routine, it is presented as for example, the noise of a bus. To apply the method of spectral subtraction is necessary the implementation of a Vocal Activity Detector, able to recognize in which precise moments of the audio there is voice or not. Two types have been studied and implemented: the first one establishes the meaning of voice according to a threshold which is adequate to this record, while the second one is the combination of Zero Crossing Rate and energy. In the end, once the application is implemented, evaluating its performances was the next process, either in an objective and a subjective form. People stand point was considered and asked, in order to obtain the proper functioning of the application along different types of noise, voice, variables, algorithm, etc.Este Trabajo de Fin de Grado, consiste en el estudio de la eliminación de ruido en voces; en concreto en el estudio de distintos algoritmos para el método de la resta espectral. Se ha creado una aplicación en el programa de cálculo Matlab cuyo uso es la eliminación de todo aquello que nos pueda molestar a la hora de escuchar una voz, es decir, lo que se considera ruido. La eliminación de ruido es la base de cualquier tratamiento de voz que se quiera aplicar posteriormente; desde reconocimiento de voz, el análisis de la misma, la conservación de la grabación limpia. etc. Se ha hecho un estudio de cuatro algoritmos para llevar a cabo esta resta espectral: Boll, Berouti, Lockwood & Boudy y Multibanda. En este documento se encuentra tanto un estudio teórico, así como su implementación. Para la implementación de una aplicación que pueda ser usada por un usuario, se ha diseñado una interfaz fácil e intuitiva de usar, en ésta se muestra cómo funcionan los distintos algoritmos en distintas voces y con distintos tipos de ruido, algunos ideales, usados en las medidas oficiales de ruido por sus concretas características matemáticas, y otros, los de la vida cotidiana como el ruido de un autobús. Para aplicar el método de la resta espectral es necesario la implementación de un Detector de Actividad Vocal (VAD) que reconozca en qué momentos del audio hay voz o no. Se han estudiado e implementado dos: Uno de ellos establece qué es voz según un límite adecuado a esa grabación y el otro es la combinación de la Tasa de Cruces por Cero (ZCR) y la energía. Por último, una vez implementada esta aplicación se ha procedido a evaluar su funcionamiento, tanto de una forma objetiva como subjetiva, a través de la escucha de distintas personas, las cuales dan su opinión, para poder obtener el comportamiento de la aplicación con distintos tipos de ruidos, voces, variables, algoritmos, etc.Ingeniería de Sistemas Audiovisuale