<p>Cochlear implant listeners are presented with a time and frequency-quantized version of speech signals. In the frequency domain, resolution is limited by the number of electrodes in each listener's array. Current cochlear implant speech processing strategies implicitly assume that the information presented to each one of these electrodes is perceived as unique and independent. However, previous research suggests that stimuli presented on different electrodes can be indiscriminable (e.g. Zwolan et al., 1997; Throckmorton and Collins, 1999; Henry et al., 2000) . Additional studies suggest that stimuli presented on one electrode can influence the perception of stimuli on neighboring electrodes (e.g. Shannon, 1990; Chatterjee and Shannon, 1998; Boëx et al., 2003). Removing this redundant or occluded information could cause more distinct or perceivable information to be presented to the listener and possibly result in improved speech recognition.</p><p>Previous studies have used psychophysical data to identify the electrodes with the highest potential to confound speech recognition (Zwolan et al., 1997, Boëx et al., 2003, and Garadat et al., 2012). In order to minimize electrode interactions and maximize the amount of perceivable information, each of these studies used a single psychophysical metric to deactivate the electrodes across all time windows of the speech processing strategy. For some listeners, these reduced electrode sets resulted in improved speech recognition over using the of the electrodes in their array. These studies did not compare the results of using different psychophysical metrics to exclude electrodes for a group of listeners nor did they investigate speech recognition performance as a function of the number of electrodes excluded from the array.</p><p>In this work, three different psychophysical metrics were used to obtain a multidimensional estimate of the potential "usefulness'' of each electrode. These results were then used to inform two different methods of psychophysics-motivated electrode selection. The first method incorporated individual data into each listener's energy-driven speech processing strategy. For each time window, the electrodes with the highest energy that were also most likely to be perceived, according to the psychophysical data, were selected for stimulation. The second method sequentially excluded the electrodes with the highest potential to confound from the array across all time windows, resulting in a group of psychophysics-motivated electrode sets for each metric. Evaluating each of these electrode sets exhaustively would require a prohibitive amount of experimental time. To mitigate this problem, an adaptive procedure was developed to estimate performance as a function of cochlear implant parameters in a time-efficient manner. For each metric, the procedure estimated the set with the highest estimated probability of correct phoneme identification. Listeners' speech recognition performance using this electrode set was then compared to their performance using their full electrode array. For both electrode selection methods, listeners' speech recognition scores were generally comparable to those obtained in the clinical condition. This finding supports the hypothesis that listeners were not perceiving all the information presented to them using their clinical speech processing strategy and their complete set of electrodes. Additionally, these results suggest that improvements to the proposed electrode selection strategies should be in investigated in order to increase the amount of perceivable information presented to cochlear implant listeners.</p> / Dissertation

No description available.

Of the few existing models of temporal masking applicable to problems such as compression and enhancement, none are based on empirical data from the psychoacoustic literature, presumably because the multidimensional nature of the data makes the derivation of tractable functional models difficult. This thesis presents two new functional models of the temporal masking effect of the human auditory system, and their exploitation in audio compression and speech enhancement applications. Traditional audio compression algorithms do not completely utilise the temporal masking properties of the human auditory system, relying solely on simultaneous masking models. A perceptual wavelet packet-based audio coder has been devised that incorporates the first developed temporal masking model and combined with simultaneous masking models in a novel manner. An evaluation of the coder using both objective (PEAQ, ITU-R BS.1387) and extensive subjective tests (ITU-R BS.1116) revealed a bitrate reduction of more than 17% compared with existing simultaneous masking-based audio coders, while preserving transparent quality. In addition, the oversampled wavelet packet transform (ODWT) has been newly applied to obtain alias-free coefficients for more accurate masking threshold calculation. Finally, a low-complexity scalable audio coding algorithm using the ODWT-based thresholds and temporal masking has been investigated. Currently, there is a strong need for innovative speech enhancement algorithms exploiting the auditory masking effects of human auditory system that perform well at very low signal-to-noise ratio. Existing competitive noise suppression algorithms and those that incorporate simultaneous masking were examined and evaluated for their suitability as baseline algorithms. Objective measures using PESQ (ITU-T P.862) and subjective measures (ITU-T P.835) demonstrate that the proposed enhancement scheme, based on a second new masking model, outperformed the seven baseline speech enhancement methods by at least 6- 20% depending on the SNR. Hence, the proposed speech enhancement scheme exploiting temporal masking effects has good potential across many types and intensities of environmental noise. Keywords: human auditory system; temporal masking; simultaneous masking; audio compression; speech enhancement; subjective test; objective test.

The main findings of this research project were that under controlled acoustic conditions, a hearing aid can be trained to provide amplification settings that are closer to hypothetical preferred settings than were the initial untrained settings, and in everyday acoustic environments, hearing aid users can train an aid to provide an amplification settings that they prefer to the untrained settings on a significant majority of occasions.

Thesis (Ph. D.)--University of Missouri-Columbia, 2007. / The entire dissertation/thesis text is included in the research.pdf file; the official abstract appears in the short.pdf file (which also appears in the research.pdf); a non-technical general description, or public abstract, appears in the public.pdf file. Title from title screen of research.pdf file (viewed on March 4, 2008) Vita. Includes bibliographical references.

Thesis (M.S.)--Rochester Institute of Technology, 1989. / Includes bibliographical references (leaves 20-21).

Thesis (Ph. D.)--University of Missouri--Rolla, 2007. / Vita. The entire thesis text is included in file. Title from title screen of thesis/dissertation PDF file (viewed November 29, 2007) Includes bibliographical references (p. 66-67).

Thesis (Ph. D.)--University of Toronto, 2006. / Source: Dissertation Abstracts International, Volume: 67-06, Section: B, page: 3354. Advisor: Parham Aarabi. Includes bibliographical references.

Thesis (M.Sc.)--University of Melbourne, Dept. of Otolaryngology, 2000. / Typescript (photocopy). Includes bibliographical references (leaves 99-109).

Thesis (Ph. D.)--University of New South Wales, 2000. / Online copy varies slightly. Also available online.