Detection and Processing Techniques of FECG Signal for Fetal Monitoring

Fetal electrocardiogram (FECG) signal contains potentially precise information that could assist clinicians in making more appropriate and timely decisions during labor. The ultimate reason for the interest in FECG signal analysis is in clinical diagnosis and biomedical applications. The extraction and detection of the FECG signal from composite abdominal signals with powerful and advance methodologies are becoming very important requirements in fetal monitoring. The purpose of this review paper is to illustrate the various methodologies and developed algorithms on FECG signal detection and analysis to provide efficient and effective ways of understanding the FECG signal and its nature for fetal monitoring. A comparative study has been carried out to show the performance and accuracy of various methods of FECG signal analysis for fetal monitoring. Finally, this paper further focused some of the hardware implementations using electrical signals for monitoring the fetal heart rate. This paper opens up a passage for researchers, physicians, and end users to advocate an excellent understanding of FECG signal and its analysis procedures for fetal heart rate monitoring system

IMPLEMENTACION DEL ALGORITMO INFOMAX PARA LA ATENUACION DE RUIDO EN LLAMADAS TELEFONICAS

Este trabajo se presenta la implementación de la arquitectura de un algoritmo de análisis de componentes independientes (ICA) en un dispositivo de arreglo de compuertas programable en campo (FPGA) utilizando el lenguaje de descripción de hardware (HDL) Verilog, que atenúa el ruido en las llamadas telefónicas. El algoritmo utilizado es el de maximización de la información llamado INFOMAX, el cual fue desarrollado por Te-Won Lee y mediante el cambio en las condiciones de operación evita la saturación de los pesos sinápticos. Los resultados arrojaron una atenuación del ruido de aproximadamente 30 dB

FPGA Implementation of Blind Source Separation using FastICA

Fast Independent Component Analysis (FastICA) is a statistical method used to separate signals from an unknown mixture without any prior knowledge about the signals. This method has been used in many applications like the separation of fetal and maternal Electrocardiogram (ECG) for pregnant women. This thesis presents an implementation of a fixed-point FastICA in field programmable gate array (FPGA). The proposed design can separate up to four signals using four sensors. QR decomposition is used to improve the speed of evaluation of the eigenvalues and eigenvectors of the covariance matrix. Moreover, a symmetric orthogonalization of the unit estimation algorithm is implemented using an iterative technique to speed up the search algorithm for higher order data input. The hardware is implemented using Xilinx virtex5-XC5VLX50t chip. The proposed design can process 128 samples for the four sensors in less than 63 ns when the design is simulated using 10 MHz clock

IMPLEMENTATION OF NOISE CANCELLATION WITH HARDWARE DESCRIPTION LANGUAGE

The objective of this project is to implement noise cancellation technique on an FPGA using Hardware Description Language. The performance of several adaptive algorithms is compared to determine the desirable algorithm used for adaptive noise cancellation system. The project will focus on the implementation of adaptive filter with least-meansquares (LMS) algorithm or normalized least-mean-squares (NLMS) algorithm to cancel acoustic noises. This noise consists of extraneous or unwanted waveforms that can interfere with communication. Due to the simplicity and effectiveness of adaptive noise cancellation technique, it is used to remove the noise component from the desired signal. The project is divided into four main parts: research, Matlab simulation, ModelSim simulation and hardware implementation. The project starts with research on several noise cancellation techniques, and then with Matlab code, Simulink and FDA tool, the adaptive noise cancellation system is designed with the implementation of the LMS algorithm, NLMS algorithm and recursive-least-square algorithm to remove the interference noise. By using the Matlab code and Simulink, the noise that interfered with a sinusoidal signal and a record of music can be removed. The original signal in turns can be retrieved from the noise corrupted signal by changing the coefficient of the filter. Since filter is the important component in adaptive filtering process, the filter is designed first before adding adaptive algorithm. A Finite Impulse Response (FIR) filter is designed and the desired result of functional simulation and timing simulation is obtained through ModelSim and Integrated Software Environment (ISE) software and FPGA implementation. Finally the adaptive algorithm is added to the filter, and implemented in the FPGA. The noise is greatly reduced in Matlab simulation, functional simulation and timing simulation. Hence the results of this project show that noise cancellation with adaptive filter is feasible

Complex-valued Adaptive Digital Signal Enhancement For Applications In Wireless Communication Systems

In recent decades, the wireless communication industry has attracted a great deal of research efforts to satisfy rigorous performance requirements and preserve high spectral efficiency. Along with this trend, I/Q modulation is frequently applied in modern wireless communications to develop high performance and high data rate systems. This has necessitated the need for applying efficient complex-valued signal processing techniques to highly-integrated, multi-standard receiver devices. In this dissertation, novel techniques for complex-valued digital signal enhancement are presented and analyzed for various applications in wireless communications. The first technique is a unified block processing approach to generate the complex-valued conjugate gradient Least Mean Square (LMS) techniques with optimal adaptations. The proposed algorithms exploit the concept of the complex conjugate gradients to find the orthogonal directions for updating the adaptive filter coefficients at each iteration. Along each orthogonal direction, the presented algorithms employ the complex Taylor series expansion to calculate time-varying convergence factors tailored for the adaptive filter coefficients. The performance of the developed technique is tested in the applications of channel estimation, channel equalization, and adaptive array beamforming. Comparing with the state of the art methods, the proposed techniques demonstrate improved performance and exhibit desirable characteristics for practical use. The second complex-valued signal processing technique is a novel Optimal Block Adaptive algorithm based on Circularity, OBA-C. The proposed OBA-C method compensates for a complex imbalanced signal by restoring its circularity. In addition, by utilizing the complex iv Taylor series expansion, the OBA-C method optimally updates the adaptive filter coefficients at each iteration. This algorithm can be applied to mitigate the frequency-dependent I/Q mismatch effects in analog front-end. Simulation results indicate that comparing with the existing methods, OBA-C exhibits superior convergence speed while maintaining excellent accuracy. The third technique is regarding interference rejection in communication systems. The research on both LMS and Independent Component Analysis (ICA) based techniques continues to receive significant attention in the area of interference cancellation. The performance of the LMS and ICA based approaches is studied for signals with different probabilistic distributions. Our research indicates that the ICA-based approach works better for super-Gaussian signals, while the LMS-based method is preferable for sub-Gaussian signals. Therefore, an appropriate choice of interference suppression algorithms can be made to satisfy the ever-increasing demand for better performance in modern receiver design

FGPA Implementation of Low-Complexity ICA Based Blind Multiple-Input-Multiple-Output OFDM Receivers

In this thesis Independent Component Analysis (ICA) based methods are used for blind detection in MIMO systems. ICA relies on higher order statistics (HOS) to recover the transmitted streams from the received mixture. Blind separation of the mixture is achieved based on the assumption of mutual statistical independence of the source streams. The use of HOS makes ICA methods less sensitive to Gaussian noise. ICA increase the spectral efficiency compared to conventional systems, without any training/pilot data required. ICA is usually used for blind source separation (BSS) from their mixtures by measuring non-Gaussianity using Kurtosis. Many scientific problems require FP arithmetic with high precision in their calculations. Moreover a large dynamic range of numbers is necessary for signal processing. FP arithmetic has the ability to automatically scale numbers and allows numbers to be represented in a wider range than fixed-point arithmetic. Nevertheless, FP algorithm is difficult to implement on the FPGA, because the algorithm is so complex that the area (logic elements) of FPGA leads to excessive consumption when implemented. A simplified 32-bit FP implementation includes adder, Subtractor, multiplier, divider, and square rooter The FPGA design is based on a hierarchical concept, and the experimental results of the design are presented

Implementación en FPGA del algoritmo ICA para Cancelación de ruido en dispositivos móviles

Actualmente se ha incrementado el uso de dispositivos móviles como celulares y tabletas electrónicas, las cuales manipulan muchas aplicaciones, y requieren de procesadores con mayor potencial de procesamiento. En este artículo se propone el uso de una arquitectura paralela en un FPGA (del inglés Field Programmable Gate Array), que realice la cancelación del ruido en tiempo real e independiente de las tareas del procesador de un equipo. La técnica utilizada es el Análisis de Componentes Independientes (ICA del inglés Independent Component Analysis), que se basa en las propiedades estadísticas de las señales en espacios multidimensionales, calcula los pesos sinápticos dato a dato y originalmente está inspirada en la arquitectura de las Redes Neuronales Artificiales. En el desarrollo se realizó la arquitectura con Simulink mediante el software de Matlab y posteriormente con un lenguaje de Descripción de Hardware HDL-Verilog (HDL del inglés hardware description language), se realizó la implementación en un FPGA; La conversión de las señales analógicas a digitales y digitales a analógicas, se realizó mediante un ADC y un DAC de 12 bits a una frecuencia de muestreo de 44Khz; y se obtuvieron resultados de la simulación y de losexperimentales

Time-shared channel identification for adaptive noise cancellation in breath sound extraction

Abstract: Noise artifacts are one of the key obstacles in applying continuous monitoring and conrputer-assisted analysis of lung sounds. Traditional adaptive noise cancellation (ANC) methodologies work reasonably well when signal and noise are stationary and independent. Clinical lung sound auscultation encounters an acoustic environment in which breath sounds are not stationary and often correlate with noise. Consequently, capability of ANC becomes significantly compromised. This paper introduces a new methodology for extracting authentic lung sounds from noise-corrupted measurements. Unlike traditional noise cancellation methods that rely on either frequency band separation or sig3M/noise independence to achieve noise reduction, this methodology combines the traditional noise canceling n{ethods with the unique feature of time-split stages in breathing sounds. By employing a multi-sensor system, the method first employs a high-pass filter to elhninate the off-hand noise, and then performs time-shared blind identification and noise cancellation with recursion from breathing cycle to cycle. Since no frequency separation or signal/noise independence is required, this method potentially has a robust and reliable capability of noise reduction, complementing the traditional methods

Efficient Blind Source Separation Algorithms with Applications in Speech and Biomedical Signal Processing

Blind source separation/extraction (BSS/BSE) is a powerful signal processing method and has been applied extensively in many fields such as biomedical sciences and speech signal processing, to extract a set of unknown input sources from a set of observations. Different algorithms of BSS were proposed in the literature, that need more investigations, related to the extraction approach, computational complexity, convergence speed, type of domain (time or frequency), mixture properties, and extraction performances. This work presents a three new BSS/BSE algorithms based on computing new transformation matrices used to extract the unknown signals. Type of signals considered in this dissertation are speech, Gaussian, and ECG signals. The first algorithm, named as the BSE-parallel linear predictor filter (BSE-PLP), computes a transformation matrix from the the covariance matrix of the whitened data. Then, use the matrix as an input to linear predictor filters whose coefficients being the unknown sources. The algorithm has very fast convergence in two iterations. Simulation results, using speech, Gaussian, and ECG signals, show that the model is capable of extracting the unknown source signals and removing noise when the input signal to noise ratio is varied from -20 dB to 80 dB. The second algorithm, named as the BSE-idempotent transformation matrix (BSE-ITM), computes its transformation matrix in iterative form, with less computational complexity. The proposed method is tested using speech, Gaussian, and ECG signals. Simulation results show that the proposed algorithm significantly separate the source signals with better performance measures as compared with other approaches used in the dissertation. The third algorithm, named null space idempotent transformation matrix (NSITM) has been designed using the principle of null space of the ITM, to separate the unknown sources. Simulation results show that the method is successfully separating speech, Gaussian, and ECG signals from their mixture. The algorithm has been used also to estimate average FECG heart rate. Results indicated considerable improvement in estimating the peaks over other algorithms used in this work