Efficient Acquisition and Denoising of Full-Range Event-Related Potentials Following Transient Stimulation of the Auditory Pathway

This body of work relates to recent advances in the field of human auditory event-related potentials (ERP), specifically the fast, deconvolution-based ERP acquisition as well as single-response based preprocessing, denoising and subsequent analysis methods. Its goal is the contribution of a cohesive set of methods facilitating the fast, reliable acquisition of the whole electrophysiological response generated by the auditory pathway from the brainstem to the cortex following transient acoustical stimulation. The present manuscript is divided into three sequential areas of investigation : First, the general feasibility of simultaneously acquiring auditory brainstem, middle-latency and late ERP single responses is demonstrated using recordings from 15 normal hearing subjects. Favourable acquisition parameters (i.e., sampling rate, bandpass filter settings and interstimulus intervals) are established, followed by signal analysis of the resulting ERP in terms of their dominant intrinsic scales to determine the properties of an optimal signal representation with maximally reduced sample count by means of nonlinear resampling on a logarithmic timebase. This way, a compression ratio of 16.59 is achieved. Time-scale analysis of the linear-time and logarithmic-time ERP single responses is employed to demonstrate that no important information is lost during compressive resampling, which is additionally supported by a comparative evaluation of the resulting average waveforms - here, all prominent waves remain visible, with their characteristic latencies and amplitudes remaining essentially unaffected by the resampling process. The linear-time and resampled logarithmic-time signal representations are comparatively investigated regarding their susceptibility to the types of physiological and technical noise frequently contaminating ERP recordings. While in principle there already exists a plethora of well-investigated approaches towards the denoising of ERP single-response representations to improve signal quality and/or reduce necessary aquisition times, the substantially altered noise characteristics of the obtained, resampled logarithmic-time single response representations as opposed to their linear-time equivalent necessitates a reevaluation of the available methods on this type of data. Additionally, two novel, efficient denoising algorithms based on transform coefficient manipulation in the sinogram domain and on an analytic, discrete wavelet filterbank are proposed and subjected to a comparative performance evaluation together with two established denoising methods. To facilitate a thorough comparison, the real-world ERP dataset obtained in the first part of this work is employed alongside synthetic data generated using a phenomenological ERP model evaluated at different signal-to-noise ratios (SNR), with individual gains in multiple outcome metrics being used to objectively assess algorithm performances. Results suggest the proposed denoising algorithms to substantially outperform the state-of-the-art methods in terms of the employed outcome metrics as well as their respective processing times. Furthermore, an efficient stimulus sequence optimization method for use with deconvolution-based ERP acquisition methods is introduced, which achieves consistent noise attenuation within a broad designated frequency range. A novel stimulus presentation paradigm for the fast, interleaved acquisition of auditory brainstem, middle-latency and late responses featuring alternating periods of optimized, high-rate deconvolution sequences and subsequent low-rate stimulation is proposed and investigated in 20 normal hearing subjects. Deconvolved sequence responses containing early and middle-latency ERP components are fused with subsequent late responses using a time-frequency resolved weighted averaging method based on cross-trial regularity, yielding a uniform SNR of the full-range auditory ERP across investigated timescales. Obtained average ERP waveforms exhibit morphologies consistent with both literature values and the reference recordings obtained in the first part of this manuscript, with all prominent waves being visible in the grand average waveforms. The novel stimulation approach cuts acquisition time by a factor of 3.4 while at the same time yielding a substantial gain in the SNR of obtained ERP data. Results suggest the proposed interleaved stimulus presentation and associated postprocessing methodology to be suitable for the fast, reliable extraction of full-range neural correlates of auditory processing in future studies.Diese Arbeit steht im Zusammenhang mit aktuellen Entwicklungen auf dem Gebiet der ereigniskorrelierten Potentiale (EKP) des humanen auditorischen Systems, insbesondere der schnellen, entfaltungsbasierten EKP-Aufzeichnung sowie einzelantwortbasierten Vorverarbeitungs-, Entrauschungs- und nachgelagerten Analysemethoden. Ziel ist die Bereitstellung eines vollständigen Methodensatzes, der eine schnelle, zuverlässige Erfassung der gesamten elektrophysiologischen Aktivität entlang der Hörbahn vom Hirnstamm bis zum Cortex ermöglicht, die als Folge transienter akustischer Stimulation auftritt. Das vorliegende Manuskript gliedert sich in drei aufeinander aufbauende Untersuchungsbereiche : Zunächst wird die generelle Machbarkeit der gleichzeitigen Aufzeichnung von Einzelantworten der auditorischen Hirnstammpotentiale zusammen mit mittelspäten und späten EKP anhand von Referenzmessungen an 15 normalhörenden Probanden demonstriert. Es werden hierzu geeignete Erfassungsparameter (Abtastrate, Bandpassfiltereinstellungen und Interstimulusintervalle) ermittelt, gefolgt von einer Signalanalyse der resultierenden EKP im Hinblick auf deren dominante intrinsische Skalen, um auf dieser Grundlage die Eigenschaften einer optimalen Signaldarstellung mit maximal reduzierter Anzahl an Abtastpunkten zu bestimmen, die durch nichtlineare Neuabtastung auf eine logarithmische Zeitbasis realisiert wird. Hierbei wird ein Kompressionsverhältnis von 16.59 erzielt. Zeit-Skalen-Analysen der uniform und logarithmisch abgetasteten EKP-Einzelantworten zeigen, dass bei der kompressiven Neuabtastung keine relevante Information verloren geht, was durch eine vergleichende Auswertung der resultierenden, gemittelten Wellenformen zusätzlich gestützt wird - alle prominenten Wellen bleiben sichtbar und sind hinsichtlich ihrer charakteristischen Latenzen und Amplituden von der Neuabtastung weitgehend unbeeinflusst. Die uniforme und logarithmische Signalrepräsentation werden hinsichtlich ihrer Anfälligkeit für die üblicherweise bei der EKP-Aufzeichnung auftretenden physiologischen und technischen Störquellen vergleichend untersucht. Obwohl bereits eine Fülle von gut etablierten Ansätzen für die Entrauschung von EKP-Einzelantwortdarstellungen zur Verbesserung der Signalqualität und/oder zur Reduktion der benötigten Erfassungszeiten existiert, erfordern die wesentlich veränderten Störeigenschaften der vorliegenden, logarithmisch abgetasteten Einzelantwortdarstellungen im Gegensatz zu ihrem uniformen Äquivalent eine Neubewertung der verfügbaren Methoden für diese Art von Daten. Darüber hinaus werden zwei neuartige, effiziente Entrauschungsalgorithmen geboten, die auf der Koeffizientenmanipulation einer Sinogramm-Repräsentation bzw. einer analytischen, diskreten Wavelet-Zerlegung der Einzelantworten basieren und gemeinsam mit zwei etablierten Entrauschungsmethoden einer vergleichenden Leistungsbewertung unterzogen werden. Um einen umfassenden Vergleich zu ermöglichen, werden der im ersten Teil dieser Arbeit erhaltene EKP-Messdatensatz sowie synthetischen Daten eingesetzt, die mithilfe eines phänomenologischen EKP-Modells bei verschiedenen Signal-Rausch-Abständen (SRA) erzeugt wurden, wobei die individuellen Anstiege in mehreren Zielmetriken zur objektiven Bewertung der Performanz herangezogen werden. Die erhaltenen Ergebnisse deuten darauf hin, dass die vorgeschlagenen Entrauschungsalgorithmen die etablierten Methoden sowohl in den eingesetzten Zielmetriken als auch mit Blick auf die Laufzeiten deutlich übertreffen. Weiterhin wird ein effizientes Reizsequenzoptimierungsverfahren für den Einsatz mit entfaltungsbasierten EKP-Aufzeichnungsmethoden vorgestellt, das eine konsistente Rauschunterdrückung innerhalb eines breiten Frequenzbands erreicht. Ein neuartiges Stimulus-Präsentationsparadigma für die schnelle, verschachtelte Erfassung auditorischer Hirnstammpotentiale, mittlelspäter und später Antworten durch alternierende Darbietung von optimierten, dichter Stimulussequenzen und nachgelagerter, langsamer Einzelstimulation wird eingeführt und in 20 normalhörenden Probanden evaluiert. Entfaltete Sequenzantworten, die frühe und mittlere EKP enthalten, werden mit den nachfolgenden späten Antworten fusioniert, wobei eine Zeit-Frequenz-aufgelöste, gewichtete Mittelung unter Berücksichtigung von Regularität über Einzelantworten hinweg zum Einsatz kommt. Diese erreicht einheitliche SRA der resultierenden EKP-Signale über alle untersuchten Zeitskalen hinweg. Die erhaltenen, gemittelten EKP-Wellenformen weisen Morphologien auf, die sowohl mit einschlägigen Literaturwerten als auch mit den im ersten Teil dieses Manuskripts erhaltenen Referenzaufnahmen konsistent sind, wobei alle markanten Wellen deutlich in den Gesamtmittelwerten sichtbar sind. Das neuartige Stimulationsparadigma verkürzt die Erfassungszeit um den Faktor 3.4 und vergrößert gleichzeitig den erreichten SRA erheblich. Die Ergebnisse deuten darauf hin, dass die vorgeschlagene verschachtelte Stimuluspräsentation und die nachgelagerte EKP-Verarbeitungsmethodik zur schnellen, zuverlässigen Extraktion neuronaler Korrelate der gesamten auditorischen Verarbeitung im Rahmen zukünftiger Studien geeignet sind.Bundesministerium für Bildung und Forschung | Bimodal Fusion - Eine neurotechnologische Optimierungsarchitektur für integrierte bimodale Hörsysteme | 2016-201

Detection of schizophrenia: A machine learning algorithm for potential early detection and prevention based on event-related potentials

We show that event-related potentials can be used to detect schizophrenia with a high degree of precision. With our machine learning algorithm we achieve a balanced accuracy of 96.4 , which exceeds all results with comparable approaches. For this we use additional sensors on the left and right hemisphere in addition to the common central sensors. The experimental design when recording the data takes into account the dysfunction of the schizophrenic efference copy. Due to its serious consequences, schizophrenia is a social issue in which early detection and prevention plays a central role. In the future, machine learning could be used to support early interventions. When the first symptoms appear, potential patients could be tested for the dysfunction typical for schizophrenia. In this way, risk groups and potential patients could be adequately treated before the onset of psychosis

Biomedical signal filtering for noisy environments

 Luke\u27s work addresses issue of robustly attenuating multi-source noise from surface EEG signals using a novel Adaptive-Multiple-Reference Least-Means-Squares filter (AMR-LMS). In practice, the filter successfully removes electrical interference and muscle noise generated during movement which contaminates EEG, allowing subjects to maintain maximum mobility throughout signal acquisition and during the use of a Brain Computer Interface

Brain-wave measures of workload in advanced cockpits: The transition of technology from laboratory to cockpit simulator, phase 2

The present Phase 2 small business innovation research study was designed to address issues related to scalp-recorded event-related potential (ERP) indices of mental workload and to transition this technology from the laboratory to cockpit simulator environments for use as a systems engineering tool. The project involved five main tasks: (1) Two laboratory studies confirmed the generality of the ERP indices of workload obtained in the Phase 1 study and revealed two additional ERP components related to workload. (2) A task analysis' of flight scenarios and pilot tasks in the Advanced Concepts Flight Simulator (ACFS) defined cockpit events (i.e., displays, messages, alarms) that would be expected to elicit ERPs related to workload. (3) Software was developed to support ERP data analysis. An existing ARD-proprietary package of ERP data analysis routines was upgraded, new graphics routines were developed to enhance interactive data analysis, and routines were developed to compare alternative single-trial analysis techniques using simulated ERP data. (4) Working in conjunction with NASA Langley research scientists and simulator engineers, preparations were made for an ACFS validation study of ERP measures of workload. (5) A design specification was developed for a general purpose, computerized, workload assessment system that can function in simulators such as the ACFS

Recent Advances in Signal Processing

The signal processing task is a very critical issue in the majority of new technological inventions and challenges in a variety of applications in both science and engineering fields. Classical signal processing techniques have largely worked with mathematical models that are linear, local, stationary, and Gaussian. They have always favored closed-form tractability over real-world accuracy. These constraints were imposed by the lack of powerful computing tools. During the last few decades, signal processing theories, developments, and applications have matured rapidly and now include tools from many areas of mathematics, computer science, physics, and engineering. This book is targeted primarily toward both students and researchers who want to be exposed to a wide variety of signal processing techniques and algorithms. It includes 27 chapters that can be categorized into five different areas depending on the application at hand. These five categories are ordered to address image processing, speech processing, communication systems, time-series analysis, and educational packages respectively. The book has the advantage of providing a collection of applications that are completely independent and self-contained; thus, the interested reader can choose any chapter and skip to another without losing continuity

Decoding speech comprehension from continuous EEG recordings

Human language is a remarkable manifestation of our cognitive abilities which is unique to our species. It is key to communication, but also to our faculty of generating complex thoughts. We organise, conceptualise, and share ideas through language. Neuroscience has shed insightful lights on our understanding of how language is processed by the brain although the exact neural organisation, structural or functional, underpinning this processing remains poorly known. This project aims to employ new methodology to understand speech comprehension during naturalistic listening condition. One achievement of this thesis lies in bringing evidence towards putative predictive processing mechanisms for language comprehension and confront those with rule-based grammar processing. Namely, we looked on the one hand at cortical responses to information-theoretic measures that are relevant for predictive coding in the context of language processing and on the other hand to the response to syntactic tree structures. We successfully recorded responses to linguistic features from continuous EEG recordings during naturalistic speech listening. The use of ecologically valid stimuli allowed us to embed neural response in the context in which they naturally occur when hearing speech. This fostered the development of new analysis tools adapted for such experimental designs. Finally, we demonstrate the ability to decode comprehension from the EEG signals of participants with above-chance accuracy. This could be used as a better indicator of the severity and specificity of language disorders, and also to assess if a patient in a vegetative state understands speech without the need for any behavioural response. Hence a primary outcome is our contribution to the neurobiology of language comprehension. Furthermore, our results pave the way to the development of a new range of diagnostic tools to measure speech comprehension of patients with language impairment.Open Acces

Sleep Stage Classification: A Deep Learning Approach

Sleep occupies significant part of human life. The diagnoses of sleep related disorders are of great importance. To record specific physical and electrical activities of the brain and body, a multi-parameter test, called polysomnography (PSG), is normally used. The visual process of sleep stage classification is time consuming, subjective and costly. To improve the accuracy and efficiency of the sleep stage classification, automatic classification algorithms were developed. In this research work, we focused on pre-processing (filtering boundaries and de-noising algorithms) and classification steps of automatic sleep stage classification. The main motivation for this work was to develop a pre-processing and classification framework to clean the input EEG signal without manipulating the original data thus enhancing the learning stage of deep learning classifiers. For pre-processing EEG signals, a lossless adaptive artefact removal method was proposed. Rather than other works that used artificial noise, we used real EEG data contaminated with EOG and EMG for evaluating the proposed method. The proposed adaptive algorithm led to a significant enhancement in the overall classification accuracy. In the classification area, we evaluated the performance of the most common sleep stage classifiers using a comprehensive set of features extracted from PSG signals. Considering the challenges and limitations of conventional methods, we proposed two deep learning-based methods for classification of sleep stages based on Stacked Sparse AutoEncoder (SSAE) and Convolutional Neural Network (CNN). The proposed methods performed more efficiently by eliminating the need for conventional feature selection and feature extraction steps respectively. Moreover, although our systems were trained with lower number of samples compared to the similar studies, they were able to achieve state of art accuracy and higher overall sensitivity

Models and analysis of vocal emissions for biomedical applications

This book of Proceedings collects the papers presented at the 3rd International Workshop on Models and Analysis of Vocal Emissions for Biomedical Applications, MAVEBA 2003, held 10-12 December 2003, Firenze, Italy. The workshop is organised every two years, and aims to stimulate contacts between specialists active in research and industrial developments, in the area of voice analysis for biomedical applications. The scope of the Workshop includes all aspects of voice modelling and analysis, ranging from fundamental research to all kinds of biomedical applications and related established and advanced technologies