This study focused on the differences in the knowledge gained between instructionally designed mediated lecture instruction and traditional lecture instruction. The sample used in this study consisted of 107 new hospital employees required to attend an orientation. The experimental group consisted of 43 subjects and the control group consisted of 64 subjects. Each group was given a pre-, post- and post-delayed knowledge test and an attitude test. The following hypotheses were tested: (H₀1) There will be no differences in post-test scores between groups. (H₂) There will be no differences in post-delayed test scores between groups. (H₃) There will be no differences in attitudes between groups. An analysis of covariance was computed on the data to test hypothesis one and hypothesis two and a one-way analysis of variance to test hypothesis three. The results indicated no significant differences on post-test scores and Hypothesis One was retained. Hypothesis Two was rejected as a significant difference indicated an increase in knowledge retained over time by the experimental group. Hypothesis Three was rejected with the experimental group reacting much more favorably to the mediated lecture. The conclusions reached as a result of this study were that both groups were learning, regardless of the method of instruction. Also knowledge was retained over time much better by the experimental group when the information was presented in the instructionally designed mediated format. Finally, the favorable response by the experimental group to the mediated lecture indicated a preference for this teaching format. Recommendations for additional investigations were discussed.

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This thesis investigates whether certain practices improve the use of video-mediated communication; specifically video-mediated gazing (the act of looking directly into the camera) and face-to-face familiarisation prior to video-mediated meetings. This is done through comparisons of conditions where such practices are employed and control conditions. The successful adoption of these practices is assessed using a multi-level approach: investigating the communicative process, participant perceptions and task outcome. Participant perceptions are directed towards assessing the media, assessing other participants using the media, perceptions of task performance and communicative success, and perceptions of social co-presence. In cases where task outcome is assessed, an objective measurement of performance is taken. Communicative process is assessed through investigating participants use of gazing behaviour and verbal aspects of process: for example turn length, dialogue length and the numper of interruptions. Verbal aspects of process are also measured using Conversational Games analysis, where the functions of participants' utterances are assessed. The results show that participants who gaze at the camera are perceived more favourably. Accompanying speech with video-mediated gazing also results in improved recall of information. Face-to-face familiarisation alters participant perceptions of others using the media and feelings of social co-presence. It is concluded that for certain applications (specifically social tasks) and with an appropriate level of training (specifically with the use of video-mediated gazing) the use of such strategies benefits video-mediated communication.

The purpose of this study was to make an investigation of the use of audio-visual aids in the conduct of interscholastic athletics for boys in AA High Schools in Texas.

The aim of this thesis is to explore the possibility of integrating an AC3 audio de- coding module into the company’s current product. Due to limited left resources on the FPGA chip in the company’s current product, the focus of this thesis is to be resource efficient. In this thesis, a system for AC3 audio decoding is designed and implemented. In order to use less logic on FPGA, PicoBlaze soft processor is used to control the whole processing flow. The system is designed and synthe- sized for a Spartan-6 FPGA which can be easily ported to the company’s current platform.

Audio fingerprinting is a highly specific content-based audio retrieval technique. Given a short audio fragment as query, an audio fingerprinting system can identify the particular file that contains the fragment in a large library potentially consisting of millions of audio files. In this thesis, we investigate the possibility and feasibility of applying audio fingerprinting to do speech recognition in noisy environments based on speech reconstruction. To reconstruct noisy speech, the speech is divided into small segments of equal length at first. Then, audio fingerprinting is used to find the most similar segment in a large dataset consisting of clean speech files. If the similarity is above a threshold, the noisy segment is replaced with the clean segment. At last, all the segments, after conditional replacement, are concatenated to form the reconstructed speech, which is sent to a traditional speech recognition system. In the above procedure, a critical step is using audio fingerprinting to find the clean speech segment in a dataset. To test its performance, we build a landmark-based audio fingerprinting system. Experimental results show that this baseline system performs well in traditional applications, but its accuracy in this new application is not as good as we expected. Next, we propose three strategies to improve the system, resulting in better accuracy than the baseline system. Finally, we integrate the improved audio fingerprinting system into a traditional speech recognition system and evaluate the performance of the whole system. / Graduate

En traitement du signal audio, l’analyse est une étape essentielle permettant de comprendre et d’inter-agir avec les signaux existants. En effet, la qualité des signaux obtenus par transformation ou par synthèse des paramètres estimés dépend de la précision des estimateurs utilisés. Cependant, des limitations théoriques existent et démontrent que la qualité maximale pouvant être atteinte avec une approche classique peut s’avérer insuffisante dans les applications les plus exigeantes (e.g. écoute active de la musique). Le travail présenté dans cette thèse revisite certains problèmes d’analyse usuels tels que l’analyse spectrale, la transcription automatique et la séparation de sources en utilisant une approche dite “informée”. Cette nouvelle approche exploite la configuration des studios de musique actuels qui maitrisent la chaîne de traitement avant l’étape de création du mélange. Dans les solutions proposées, de l’information complémentaire minimale calculée est transmise en même temps que le signal de mélange afin de permettre certaines transformations sur celui-ci tout en garantissant le niveau de qualité. Lorsqu’une compatibilité avec les formats audio existants est nécessaire, cette information est cachée à l’intérieur du mélange lui-même de manière inaudible grâce au tatouage audionumérique. Ce travail de thèse présente de nombreux aspects théoriques et pratiques dans lesquels nous montrons que la combinaison d’un estimateur avec de l’information complémentaire permet d’améliorer les performances des approches usuelles telles que l’estimation non informée ou le codage pur. / In the field of audio signal processing, analysis is an essential step which allows interactions with existing signals. In fact, the quality of transformed or synthesized audio signals depends on the accuracy over the estimated model parameters. However, theoretical limits exist and show that the best accuracy which can be reached by a classic estimator can be insufficient for the most demanding applications (e.g. active listening of music). The work which is developed in this thesis revisits well known audio analysis problems like spectral analysis, automatic transcription of music and audio sources separation using the novel ``informed'' approach. This approach takes advantage of a specific configuration where the parameters of the elementary signals which compose a mixture are known before the mixing process. Using the tools which are proposed in this thesis, the minimal side information is computed and transmitted with the mixture signal. This allows any kind of transformation of the mixture signal with a constraint over the resulting quality. When the compatibility with existing audio formats is required, the side information is embedded directly into the analyzed audio signal using a watermarking technique. This work describes several theoretical and practical aspects of audio signal processing. We show that a classic estimator combined with the sufficient side information can obtain better performances than classic approaches (classic estimation or pure coding).