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  • About
  • The Global ETD Search service is a free service for researchers to find electronic theses and dissertations. This service is provided by the Networked Digital Library of Theses and Dissertations.
    Our metadata is collected from universities around the world. If you manage a university/consortium/country archive and want to be added, details can be found on the NDLTD website.
151

Hardware implementation of an automatic speaker recognition system using artificial neural networks

Moonasar, Viresh January 2002 (has links)
Submitted in fulfillment of the academic requirements for the degree of Master of Technology in Electrical Engineering in the Department of Electronic Engineering, Faculty of Engineering, ML Sultan Technikon of Durban in South Africa, March 2002. / The use of speaker recognition technology in interactive voice response and electronic commerce systems has been limited. This is due to the lack of research attention and published results when compared to all the other areas of speech recognition technologies / M
152

Development of tests and preprocessing algorithms for evaluation and improvement of speech recognition units

Wasmeier, Hans January 1986 (has links)
This study considered the evaluation of commercially available isolated word, speaker dependent, speech recognition units, and preprocessing techniques that may be used for improving their performance. The problem was considered in three separate stages. A series of tests were designed to exercise an isolated word, speaker dependent, speech recognition unit. These tests provided a sound basis for determining a given unit's strengths and weaknesses. This knowledge permits a more informed decision on the best recognition device for a given price range. As well, this knowledge may be used in the design of a robust vocabulary, and creation of guidelines for best performance. The test vocabularies were based on the forty English phonemes identified by Rabiner and Schafer [28] and the test variations were representative of common variations which may be expected in normal use. A digital archive system was implemented for storing the voice input of test subjects. This facility provided a data base for an investigation of preprocessing techniques. As well, it permits the testing of different speech recognition units with the same voice input, providing a platform for device comparison. Several speech preprocessing and performance improvement techniques were then investigated. Specifically, two types of time normalization, the enhancement of low energy phonemes and a change in training technique were investigated. These techniques permit a more accurate analysis of the failure mechanism of the speech recognition unit. They may also provide the basis for a speech preprocessor design which could be placed in front of a commercial speech recognition unit. A commercially available speech recognition unit, the NEC SR100, was used as a measure of the effectiveness of the tests and of the improvements. Results of the study indicated that the designed tests and the preprocessing & performance improvement techniques investigated were useful in identifying the speech recognition unit's weaknesses. Also, depending on the economics of implementation, it was found that preprocessing may provide a cost effective solution to some of the recognition unit's shortcomings. / Applied Science, Faculty of / Electrical and Computer Engineering, Department of / Graduate
153

Speaker-independent access to a large lexicon

Mathan, Luc Stefan January 1987 (has links)
No description available.
154

Integration of multiple feature sets for reducing ambiguity in automatic speech recognition

MomayyezSiahkal, Parya. January 2008 (has links)
No description available.
155

Speaker normalizing transforms in speech recogniton by computer

Sejnoha, Vladimir. January 1982 (has links)
No description available.
156

Speaker recognition using digit utterances

Scrimgeour, J. Michael. January 1984 (has links)
No description available.
157

Experiments on automatic phonetic segmentation and transcription of speech

Lennig, Matthew. January 1983 (has links)
No description available.
158

Connectionist models applied to automatic speech recognition

Bengio, Yoshua January 1987 (has links)
No description available.
159

An approach to a robust speaker recognition system

Tran, Michael 06 June 2008 (has links)
This dissertation presents a design of a robust, automatic speaker recognition (ASR) system. The ASR system is designed to work with both text-independent and text-dependent speaker recognition. Several speaker spectral features are studied to determine their contribution in term of accuracy to the system. A new algorithm is designed to label a speaker voice as either male-type voice or female-type voice. Following this division, the processing time of the speaker identification for the ASR system will be reduced by about half. Rectangular window, Hamming window, first order preemphasis filter, and many proposed spectral distances are also investigated. The principal components analysis is used to achieve high degree of female-type and male-type separation as well as the speaker recognition accuracy. Spectral features are combined to improve the recognition performance of the system. In addition, many other system components such as speech endpoint detection, automatic noise thresholds, etc. are required to build correctly in order to achieve high speaker recognition accuracy. Multi-stage decision process is used both to improve and to speed up the decision if certain criteria are met. Finally, TIMIT acoustic continuous speech corpus is used to evaluate the speaker recognition performance and the robustness of the system. / Ph. D.
160

A feasibility study on the use of a voice recognition system for training delivery

Gibson, Marcia 25 August 2008 (has links)
This feasibility study examined the possibility of using an independent voice recognition system as the input device during a training delivery requirement. The intent was to determine whether the voice recognition system could be incorporated into a training delivery system designed to train students how to use the Communications Electronics Operating Instructions manual, a tool used for communicating over the radio network during military operations. This study showed how the voice recognition system worked in an integrated voice based delivery system for the purpose of delivering instruction. An added importance of the study was that the voice system was an independent speech recognition system. At the time this study was conducted, there did not exist a reasonably priced speech recognition system that interfaced with both graphics and authoring software which allowed any student to speak to the system without training the system to recognize the individual student's voice. This feature increased the usefulness and flexibility of the system. The methodology for this feasibility study was a development and evaluation model. This required a market analysis, development of the voice system and instructional course ware, testing the system using a sample population from the Armor School at Ft. Knox, Kentucky, and making required alterations. The data collection approach was multifaceted. There were surveys to be completed by each subject: a student profile survey, a pretest, a posttest, and an opinion survey about how well the instruction met expectations. Data was also collected concerning how often the recognition system recognized, did not recognize, or misrecognized the voice of each subject. The information gathered was analyzed to determine how well the voice recognition system performs in a training delivery application. The findings of this feasibility study indicated that an effective voice based training delivery system could be developed by integrating an IBM clone personal computer with a graphics board and supporting software, signal processing board and supporting software for audio output and input, and instructional authoring software. Training was delivered successfully since all students completed the course, 85% performed better on the posttest than on the pretest, and the mean gain scores more than satisfied the expected criterion for the training course. The misrecognition factor was 12%. An important finding of this study is that the misrecognition factor did not affect the students' opinion of how well the voice system operated or the students' learning gain. / Ed. D.

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