Colamus humanitatem : Nurturing human nature

In an essay on anger, the ancient philosopher Seneca warns of the futility of harboring negative emotions given the imminence of death—the ultimate human equalizer. Ancient philosophers like Seneca believed that emotions are based on cognitions (beliefs) and are therefore modifiable through spiritual exercises. Modern research shows that the emotional and cognitive aspects of human psychology are malleable (nurture), but also require gene expression (nature). A parallel between individual behavior and socio-political forces suggests a framework for the current environmental crisis— another human equalizer. Two critical questions are suggested: Is the amassed experience of the last few centuries sufficient to lead to corrective measures that would avoid environmental degradation? Or would a catastrophic event with significant longterm environmental degradation have to occur before corrective measures reach consensus at the socio-political level? 

Performance of GPRS coding scheme detection under severe multipath and co-channel interference as a function of soft-bit width

The General Packet Radio Service (GPRS) has been developed for the mobile radio environment to allow the migration from the traditional circuit switched connection to a more efficient packet based communication link particularly for data transfer. GPRS requires the addition of not only the GPRS software protocol stack, but also more baseband functionality for the mobile as new coding schemes have be en defined, uplink status flag detection, multislot operation and dynamic coding scheme detect. This paper concentrates on evaluating the performance of the GPRS coding scheme detection methods in the presence of a multipath fading channel with a single co-channel interferer as a function of various soft-bit data widths. It has been found that compressing the soft-bit data widths from the output of the equalizer to save memory can influence the likelihood decision of the coding scheme detect function and hence contribute to the overall performance loss of the system. Coding scheme detection errors can therefore force the channel decoder to either select the incorrect decoding scheme or have no clear decision which coding scheme to use resulting in the decoded radio block failing the block check sequence and contribute to the block error rate. For correct performance simulation, the performance of the full coding scheme detection must be taken into account.

Optimum decision delay of the finite-length DFE

This work investigates the optimum decision delay and tap-length of the finite-length decision feedback equalizer. First we show that, if the feedback filter (FBF) length Nb is equal to or larger than the channel memory v and the decision delay Δ is smaller than the feedforward filter (FFF) length Nf, then only the first Δ+1 elements of the FFF can be nonzero. Based on this result we prove that the maximum effective FBF length is equal to the channel memory v, and if Nb ≥ v and Nf is long enough, the optimum decision delay that minimizes the MMSE is Nf-1.

Flexible MMSE DFE using length and LAG adaptation

This paper investigates how to choose the optimum tap-length and decision delay for the decision feedback equalizer (DFE). Although the feedback filter length can be set as the channel memory, there is no closed-form expression for the feedforward filter length and decision delay. In this paper, first we analytically show that the two dimensional search for the optimum feedforward filter length and decision delay can be simplified to a one dimensional search, and then describe a new adaptive DFE where the optimum structural parameters can be self-adapted.

Adaptive B-spline neural network based nonlinear equalization for high-order QAM systems with nonlinear transmit high power amplifier

High bandwidth-efficiency quadrature amplitude modulation (QAM) signaling widely adopted in high-rate communication systems suffers from a drawback of high peak-toaverage power ratio, which may cause the nonlinear saturation of the high power amplifier (HPA) at transmitter. Thus, practical high-throughput QAM communication systems exhibit nonlinear and dispersive channel characteristics that must be modeled as a Hammerstein channel. Standard linear equalization becomes inadequate for such Hammerstein communication systems. In this paper, we advocate an adaptive B-Spline neural network based nonlinear equalizer. Specifically, during the training phase, an efficient alternating least squares (LS) scheme is employed to estimate the parameters of the Hammerstein channel, including both the channel impulse response (CIR) coefficients and the parameters of the B-spline neural network that models the HPA’s nonlinearity. In addition, another B-spline neural network is used to model the inversion of the nonlinear HPA, and the parameters of this inverting B-spline model can easily be estimated using the standard LS algorithm based on the pseudo training data obtained as a natural byproduct of the Hammerstein channel identification. Nonlinear equalisation of the Hammerstein channel is then accomplished by the linear equalization based on the estimated CIR as well as the inverse B-spline neural network model. Furthermore, during the data communication phase, the decision-directed LS channel estimation is adopted to track the time-varying CIR. Extensive simulation results demonstrate the effectiveness of our proposed B-Spline neural network based nonlinear equalization scheme.

Equalização neural aplicada a sistemas com modulação bidimensional em fibra óptica

Nowadays, optic fiber is one of the most used communication methods, mainly due to the fact that the data transmission rates of those systems exceed all of the other means of digital communication. Despite the great advantage, there are problems that prevent full utilization of the optical channel: by increasing the transmission speed and the distances involved, the data is subjected to non-linear inter symbolic interference caused by the dispersion phenomena in the fiber. Adaptive equalizers can be used to solve this problem, they compensate non-ideal responses of the channel in order to restore the signal that was transmitted. This work proposes an equalizer based on artificial neural networks and evaluates its performance in optical communication systems. The proposal is validated through a simulated optic channel and the comparison with other adaptive equalization techniques

On designing OTA-C graphic-equalizers with MOSFET-triode transconductors

A CMOS audio-equalizer based on a parallel-array of 2nd-order bandpass-sections is presented and realized with triode transconductors. It has a programmable 12db-boost/cut on each of its three decade-bands, easily achieved through the linear dependence of gm on VDS. In accordance with a 0.8μm n-well double-metal fabrication process, a range of simulations supports theoretical analysis and circuit performance at different boost/cut scenarios. For VDD=3.3V, fullboosting stand-by prover consumption is 1.05mW. THD=-42.61dB@1Vpp and may be improved by balanced structures. Thermal- and I/f-noise spectral densities are 3.2μV/Hz12 and 18.2μV/Hz12@20Hz, respectively, for a dynamic range of 52.3dB@1Vpp. The equalizer effective area is 2.4mm2. The drawback of the existing transmission-zero due to the feedthrough-capacitance of a triode input-device is also addressed. The proposed topology can be extended to the design of more complex graphic-equalizers and hearing-aids.

Deconvolução de processo sísmico não-estacionário

O presente trabalho trata da aplicação do filtro Kalman-Bucy (FKB), organizado como uma deconvolução (FKBD), para extração da função refletividade a partir de dados sísmicos. Isto significa que o processo é descrito como estocástico não-estacionário, e corresponde a uma generalização da teoria de Wiener-Kolmogorov. A descrição matemática do FKB conserva a relação com a do filtro Wiener-Hopf (FWH) que trata da contra-parte com um processo estocástico estacionário. A estratégia de ataque ao problema é estruturada em partes: (a) Critério de otimização; (b) Conhecimento a priori; (c) Algoritmo; e (d) Qualidade. O conhecimento a priori inclui o modelo convolucional, e estabelece estatísticas para as suas componentes do modelo (pulso-fonte efetivo, função refletividade, ruídos geológico e local). Para demostrar a versatilidade, a aplicabilidade e limitações do método, elaboramos experimentos sistemáticos de deconvolução sob várias situações de nível de ruídos aditivos e de pulso-fonte efetivo. Demonstramos, em primeiro lugar, a necessidade de filtros equalizadores e, em segundo lugar, que o fator de coerência espectral é uma boa medida numérica da qualidade do processo. Justificamos também o presente estudo para a aplicação em dados reais, como exemplificado.

Ecualización y optimización de la codificación para la simulación de sistemas OFDM

Los sistemas basados en la técnica OFDM (Multiplexación por División de Frecuencias Ortogonales) son una evolución de los tradicionales sistemas FDM (Multiplexación por División de Frecuencia), gracias a la cual se consigue un mejor aprovechamiento del ancho de banda. En la actualidad los sistemas OFDM y sus variantes ocupan un lugar muy importante en las comunicaciones, estando implementados en diversos estándares como pueden ser: DVB-T (estándar de la TDT), ADSL, LTE, WIMAX, DAB (radio digital), etc. Debido a ello, en este proyecto se implementa un sistema OFDM en el que poder realizar diversas simulaciones para entender mejor su funcionamiento. Para ello nos vamos a valer de la herramienta Matlab. Los objetivos fundamentales dentro de la simulación del sistema es poner a prueba el empleo de turbo códigos (comparándolo con los códigos convolucionales tradicionales) y de un ecualizador. Todo ello con la intención de mejorar la calidad de nuestro sistema (recibir menos bits erróneos) en condiciones cada vez más adversas: relaciones señal a ruido bajas y multitrayectos. Para ello se han implementado las funciones necesarias en Matlab, así como una interfaz gráfica para que sea más sencillo de utilizar el programa y más didáctico. En los capítulos segundo y tercero de este proyecto se efectúa un estudio de las bases de los sistemas OFDM. En el segundo nos centramos más en un estudio teórico puro para después pasar en el tercero a centrarnos únicamente en la teoría de los bloques implementados en el sistema OFDM que se desarrolla en este proyecto. En el capítulo cuarto se explican las distintas opciones que se pueden llevar a cabo mediante la interfaz implementada, a la vez que se elabora un manual para el correcto uso de la misma. El quinto capítulo se divide en dos partes, en la primera se muestran las representaciones que puede realizar el programa, y en la segunda únicamente se realizan simulaciones para comprobar que tal responde nuestra sistema a distintas configuraciones de canal, y las a distintas configuraciones que hagamos nosotros de nuestro sistema (utilicemos una codificación u otra, utilicemos el ecualizador o el prefijo cíclico, etc…). Para finalizar, en el último capítulo se exponen las conclusiones obtenidas en este proyecto, así como posibles líneas de trabajo que seguir en próximas versiones del mismo. ABSTRACT. Systems based on OFDM (Orthogonal Frequency Division Multiplexing) technique are an evolution of traditional FDM (Frequency Division Multiplexing). Due to the use of OFDM systems are achieved by more efficient use of bandwidth. Nowadays, OFDM systems and variants of OFDM systems occupy a very important place in the world of communications, being implemented in standards such as DVB-T, ADSL, LTE, WiMAX, DAB (digital radio) and another more. For all these reasons, this project implements a OFDM system for performing various simulations for better understanding of OFDM system operation. The system has been simulated using Matlab. With system simulation we search to get two key objectives: to test the use of turbo codes (compared to traditional convolutional codes) and an equalizer. We do so with the intention of improving the quality of our system (receive fewer rates of bit error) in increasingly adverse conditions: lower signal-to-noise and multipath. For these reasons necessaries Matlab´s functions have been developed, and a GUI (User Graphical Interface) has been integrated so the program can be used in a easier and more didactic way. This project is divided into five chapters. In the second and third chapter of this project are developed the basis of OFDM systems. Being developed in the second one a pure theoretical study, while focusing only on block theory implemented in the OFDM system in the third one. The fourth chapter describes the options that can be carried out by the interface implemented. Furthermore the chapter is developed for the correct use of the interface. The fifth chapter is divided into two parts, the first part shows to us the representations that the program can perform, and the second one just makes simulations to check that our system responds to differents channel configurations (use of convolutional codes or turbo codes, the use of equalizer or cyclic prefix…). Finally, the last chapter presents the conclusions of this project and possible lines of work to follow in future versions.

Implementación de efectos de audio en la tarjeta Raspberry Pi B+

Este proyecto consiste en el diseño e implementación de un procesador digital de efectos de audio en tiempo real orientado a instrumentos eléctricos tales como guitarras, bajos, teclados, etc. El procesador está basado en la tarjeta Raspberry Pi B+, ordenador de placa reducida de bajo coste, desarrollado en Reino unido y cuyo lanzamiento tuvo lugar en el año 2012. En primer lugar, ha sido necesario lograr que la tarjeta asuma la funcionalidad de un procesador de audio en tiempo real. Para ello se ha instalado un sistema operativo Linux orientado a Raspberry (Raspbian) y se ha hecho uso de Pure Data (Pd): lenguaje de programación gráfico que fue desarrollado en los años 90 por Miller Puckette con intención de ser enfocado a la creación de eventos multimedia y de música por computador. El papel que desempeña Pd es de capa intermedia entre el hardware y el software ya que se encarga de tomar bloques de N muestras del convertidor analógico/digital y encaminarlas a través del flujo de señal diseñado gráficamente. En segundo lugar, se han implementado diferentes efectos de audio de distintas características. Así pues, se encuentran efectos basados en retardos, filtros digitales y procesadores de dinámica. Concretamente, los efectos implementados son los siguientes: delay, flanger, vibrato, reverberador de Schroeder, filtros (paso bajo, paso alto y paso banda), ecualizador paramétrico y compresor y expansor de dinámica. Estos efectos han sido implementados en lenguaje C de acuerdo con la API de Pd. Con esto se ha conseguido obtener un objeto por cada efecto, el cual es “instanciado” en Pd pudiendo ejecutarlo en tiempo real. En este proyecto se expone la problemática que supone cada paso del diseño proponiendo soluciones válidas. Además se incluye una guía paso a paso para configurar la tarjeta y lograr realizar un bypass de señal y un efecto simple partiendo desde cero. ABSTRACT. This project involves the design and implementation of a digital real-time audio processor for electrical instruments (guitars, basses, keyboards, etc.). The processor is based on the Raspberry Pi B + card: low cost computer, developed in UK in 2012. First, it was necessary to make the cards assume the functionality of a real time audio processor. A Linux operating system called Raspberry (Raspbian) was installed. In this Project is used Pure Data (Pd): a graphical programming language developed in the 90s by Miller Puckette intending to be focused on creating multimedia and computer music events. The role of Pd is an intermediate layer between the hardware and the software. It is responsible for taking blocks of N samples of the analog/digital converter and route it through the signal flow. Secondly, it is necessary to implemented the different audio effects. There are delays based effects, digital filter and dynamics effects. Specifically, the implemented effects are: delay, flanger, vibrato, Schroeder reverb, filters (lowpass, highpass and bandpass), parametric equalizer and compressor and expander dynamics. These effects have been implemented in C language according to the Pd API. As a result, it has been obtained an object for each effect, which is instantiated in Pd. In this Project, the problems of every step are exposed with his corresponding solution. It is inlcuded a step-by-step guide to configure the card and achieve perform a bypass signal process and a simple effect.

Changes in global gene expression patterns during development and maturation of the rat kidney

We set out to define patterns of gene expression during kidney organogenesis by using high-density DNA array technology. Expression analysis of 8,740 rat genes revealed five discrete patterns or groups of gene expression during nephrogenesis. Group 1 consisted of genes with very high expression in the early embryonic kidney, many with roles in protein translation and DNA replication. Group 2 consisted of genes that peaked in midembryogenesis and contained many transcripts specifying proteins of the extracellular matrix. Many additional transcripts allied with groups 1 and 2 had known or proposed roles in kidney development and included LIM1, POD1, GFRA1, WT1, BCL2, Homeobox protein A11, timeless, pleiotrophin, HGF, HNF3, BMP4, TGF-α, TGF-β2, IGF-II, met, FGF7, BMP4, and ganglioside-GD3. Group 3 consisted of transcripts that peaked in the neonatal period and contained a number of retrotransposon RNAs. Group 4 contained genes that steadily increased in relative expression levels throughout development, including many genes involved in energy metabolism and transport. Group 5 consisted of genes with relatively low levels of expression throughout embryogenesis but with markedly higher levels in the adult kidney; this group included a heterogeneous mix of transporters, detoxification enzymes, and oxidative stress genes. The data suggest that the embryonic kidney is committed to cellular proliferation and morphogenesis early on, followed sequentially by extracellular matrix deposition and acquisition of markers of terminal differentiation. The neonatal burst of retrotransposon mRNA was unexpected and may play a role in a stress response associated with birth. Custom analytical tools were developed including “The Equalizer” and “eBlot,” which contain improved methods for data normalization, significance testing, and data mining.

Chromatic dispersion compensation using full-field maximum-likelihood sequence estimation

We investigate full-field detection-based maximum-likelihood sequence estimation (MLSE) for chromatic dispersion compensation in 10 Gbit/s OOK optical communication systems. Important design criteria are identified to optimize the system performance. It is confirmed that approximately 50% improvement in transmission reach can be achieved compared to conventional direct-detection MLSE at both 4 and 16 states. It is also shown that full-field MLSE is more robust to the noise and the associated noise amplifications in full-field reconstruction, and consequently exhibits better tolerance to nonoptimized system parameters than full-field feedforward equalizer. Experiments over 124 km spans of field-installed single-mode fiber without optical dispersion compensation using full-field MLSE verify the theoretically predicted performance benefits.

A 10 Mb/s visible light communication system using a low bandwidth polymer light-emitting diode

In this paper we experimentally demonstrate a 10 Mb/s error free visible light communications (VLC) system using polymer light-emitting diodes (PLEDs) for the first time. The PLED under test is a blue emitter with ∼600 kHz bandwidth. Having such a low bandwidth means the introduction of an intersymbol interference (ISI) induced penalty at higher transmission speeds and thus the requirement for an equalizer. In this work we improve on previous literature by implementing a decision feedback equalizer, rather than a linear equalizer. Considering 7% and 20% forward error correction codes, transmission speeds up to ∼12 Mb/s can be supported.

Fiber nonlinearity-induced penalty reduction in CO-OFDM by ANN-based nonlinear equalization

We experimentally demonstrate ∼2 dB quality (Q)-factor enhancement in terms of fiber nonlinearity compensation of 40 Gb/s 16 quadrature amplitude modulation coherent optical orthogonal frequency-division multiplexing at 2000 km, using a nonlinear equalizer (NLE) based on artificial neural networks (ANN). Nonlinearity alleviation depends on escalation of the ANN training overhead and the signal bit rate, reporting ∼4 dB Q-factor enhancement at 70 Gb/s, whereas a reduction of the number of ANN neurons annihilates the NLE performance. An enhanced performance by up to ∼2 dB in Q-factor compared to the inverse Volterra-series transfer function NLE leads to a breakthrough in the efficiency of ANN.

Comparison of DSP-based nonlinear equalizers for intra-channel nonlinearity compensation in coherent optical OFDM

A novel versatile digital signal processing (DSP)-based equalizer using support vector machine regression (SVR) is proposed for 16-quadrature amplitude modulated (16-QAM) coherent optical orthogonal frequency-division multiplexing (CO-OFDM) and experimentally compared to traditional DSP-based deterministic fiber-induced nonlinearity equalizers (NLEs), namely the full-field digital back-propagation (DBP) and the inverse Volterra series transfer function-based NLE (V-NLE). For a 40 Gb/s 16-QAM CO-OFDM at 2000 km, SVR-NLE extends the optimum launched optical power (LOP) by 4 dB compared to V-NLE by means of reduction of fiber nonlinearity. In comparison to full-field DBP at a LOP of 6 dBm, SVR-NLE outperforms by ∼1 dB in Q-factor. In addition, SVR-NLE is the most computational efficient DSP-NLE.