Preliminary candidate advanced avionics system for general aviation

An integrated avionics system design was carried out to the level which indicates subsystem function, and the methods of overall system integration. Sufficient detail was included to allow identification of possible system component technologies, and to perform reliability, modularity, maintainability, cost, and risk analysis upon the system design. Retrofit to older aircraft, availability of this system to the single engine two place aircraft, was considered

Engineering data compendium. Human perception and performance. User's guide

The concept underlying the Engineering Data Compendium was the product of a research and development program (Integrated Perceptual Information for Designers project) aimed at facilitating the application of basic research findings in human performance to the design and military crew systems. The principal objective was to develop a workable strategy for: (1) identifying and distilling information of potential value to system design from the existing research literature, and (2) presenting this technical information in a way that would aid its accessibility, interpretability, and applicability by systems designers. The present four volumes of the Engineering Data Compendium represent the first implementation of this strategy. This is the first volume, the User's Guide, containing a description of the program and instructions for its use

A Hybrid voice/text electronic mail system: an application of the integrated services digital network

The objective of this thesis is to present a useful application for the Integrated Services Digital Network (ISDN) that is expected to one day replace the analog phone system in use today. ISDN itself and its continuing evolution are detailed. The system developed as a part of this thesis involved the creation of an inexpensive phone terminal that can serve as an ISDN terminal and also as a bridge to a Local Area Network (LAN). The phone terminal provides a hybrid electronic mail system that allows the attachment of speech to text within a message. Messages created with this phone terminal could theoretically be sent locally using the LAN interface and globally using ISDN to other users with either phone terminals or multimedia personal computers. For this project, the two phone terminals created were interconnected via an Ethernet and using an 80486 PC to act as a Central Office System. This Central Office System provides speech/message storage for the phone terminals. It makes use of speech compression techniques to minimize the storage requirements. The speech compression techniques used as well as the field of speech coding in general are discussed

THE ROLE OF “FOCUS OF ATTENTION” ON THE LEARNING OF NON-NATIVE SPEECH SOUNDS: ENGLISH SPEAKERS LEARNING OF MANDARIN CHINESE TONES

Focus of attention (FOA) has been demonstrated to affect motor learning and performance of many motor skills. FOA refers to the performer’s focus while performing the task. The purpose of this dissertation was to assess the role of FOA in the speech domain. The research asked whether external or internal FOA would individually or differentially facilitate the learning of Mandarin Chinese tones by native English speakers. As a secondary question and experimental control, this study also examined whether the four tones were produced with the same accuracy. Forty-two females, between the ages of 18 and 24 were randomly assigned to one of three groups: external FOA (EFOA), internal FOA (IFOA) and control (C). During the acquisition phase, the groups were instructed to either focus on the sound produced (EFOA), the vibration in the voice box (IFOA), or no related FOA instructions (control). Participants were required to repeat the Mandarin words after an auditory model. To assess learning, the participants repeated the practiced words in a retention test, and repeated similar but unpracticed words during a transfer test. The data was collected in two sessions. The dependent variables were the root mean squared error (acoustic measure) and percentage of correctly perceived tones (perceptual measure). There was a significant difference among the four Mandarin Chinese tones for the three groups (Tones 1 and 4 were produced with significantly higher accuracy than Tones 2 and 3) before acquisition phase. There was, however, no significant difference among the three FOA groups on the dependent variables. The results contradict the FOA effects in the literature derived from limb motor learning and oral-nonspeech learning experiments. This study represents the first attempt to test the FOA in the speech domain. As such, it is premature to draw firm conclusions about the role of FOA in speech motor learning based on these results. The discussion focuses on factors that might have led to the current results. Because FOA represents a potential factor that might affect speech motor learning, future research is warranted to study the effect of FOA in the speech domain

Robust speaker identification against computer aided voice impersonation

Speaker Identification (SID) systems offer good performance in the case of noise free speech and most of the on-going research aims at improving their reliability in noisy environments. In ideal operating conditions very low identification error rates can be achieved. The low error rates suggest that SID systems can be used in real-life applications as an extra layer of security along with existing secure layers. They can, for instance, be used alongside a Personal Identification Number (PIN) or passwords. SID systems can also be used by law enforcements agencies as a detection system to track wanted people over voice communications networks. In this thesis, the performance of 'the existing SID systems against impersonation attacks is analysed and strategies to counteract them are discussed. A voice impersonation system is developed using Gaussian Mixture Modelling (GMM) utilizing Line Spectral Frequencies (LSF) as the features representing the spectral parameters of the source-target pair. Voice conversion systems based on probabilistic approaches suffer from the problem of over smoothing of the converted spectrum. A hybrid scheme using Linear Multivariate Regression and GMM, together with posterior probability smoothing is proposed to reduce over smoothing and alleviate the discontinuities in the converted speech. The converted voices are used to intrude a closed-set SID system in the scenarios of identity disguise and targeted speaker impersonation. The results of the intrusion suggest that in their present form the SID systems are vulnerable to deliberate voice conversion attacks. For impostors to transform their voices, a large volume of speech data is required, which may not be easily accessible. In the context of improving the performance of SID against deliberate impersonation attacks, the use of multiple classifiers is explored. Linear Prediction (LP) residual of the speech signal is also analysed for speaker-specific excitation information. A speaker identification system based on multiple classifier system, using features to describe the vocal tract and the LP residual is targeted by the impersonation system. The identification results provide an improvement in rejecting impostor claims when presented with converted voices. It is hoped that the findings in this thesis, can lead to the development of speaker identification systems which are better equipped to deal with the problem with deliberate voice impersonation.EThOS - Electronic Theses Online ServiceGBUnited Kingdo