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Robust Single-Channel Speech Enhancement and Speaker Localization in Adverse Environments

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dc.contributor.author Mosayyebpour, Saeed
dc.date.accessioned 2014-04-30T22:40:58Z
dc.date.available 2015-04-26T11:22:05Z
dc.date.copyright 2014 en_US
dc.date.issued 2014-04-30
dc.identifier.uri http://hdl.handle.net/1828/5342
dc.description.abstract In speech communication systems such as voice-controlled systems, hands-free mobile telephones and hearing aids, the received signals are degraded by room reverberation and background noise. This degradation can reduce the perceived quality and intelligibility of the speech, and decrease the performance of speech enhancement and source localization. These problems are difficult to solve due to the colored and nonstationary nature of the speech signals, and features of the Room Impulse Response (RIR) such as its long duration and non-minimum phase. In this dissertation, we focus on two topics of speech enhancement and speaker localization in noisy reverberant environments. A two-stage speech enhancement method is presented to suppress both early and late reverberation in noisy speech using only one microphone. It is shown that this method works well even in highly reverberant rooms. Experiments under different acoustic conditions confirm that the proposed blind method is superior in terms of reducing early and late reverberation effects and noise compared to other well known single-microphone techniques in the literature. Time Difference Of Arrival (TDOA)-based methods usually provide the most accurate source localization in adverse conditions. The key issue for these methods is to accurately estimate the TDOA using the smallest number of microphones. Two robust Time Delay Estimation (TDE) methods are proposed which use the information from only two microphones. One method is based on adaptive inverse filtering which provides superior performance even in highly reverberant and moderately noisy conditions. It also has negligible failure estimation which makes it a reliable method in realistic environments. This method has high computational complexity due to the estimation in the first stage for the first microphone. As a result, it can not be applied in time-varying environments and real-time applications. Our second method improves this problem by introducing two effective preprocessing stages for the conventional Cross Correlation (CC)-based methods. The results obtained in different noisy reverberant conditions including a real and time-varying environment demonstrate that the proposed methods are superior compared to the conventional TDE methods. en_US
dc.language English eng
dc.language.iso en en_US
dc.rights.uri http://creativecommons.org/publicdomain/zero/1.0/ *
dc.subject skewness en_US
dc.subject early and late reverberation en_US
dc.subject noise en_US
dc.subject single-microphone en_US
dc.subject spectral subtraction en_US
dc.subject Time Delay Estimation (TDE) en_US
dc.subject Time Difference of Arrival (TDOA) en_US
dc.subject Adaptive Inverse Filtering (AIF) en_US
dc.subject Generalized Cross-Correlation (GCC) en_US
dc.subject room impulse response (RIR) en_US
dc.title Robust Single-Channel Speech Enhancement and Speaker Localization in Adverse Environments en_US
dc.type Thesis en_US
dc.contributor.supervisor Gulliver, T. Aaron
dc.contributor.supervisor Esmaeili, Morteza
dc.degree.department Department of Electrical and Computer Engineering en_US
dc.degree.level Doctor of Philosophy Ph.D. en_US
dc.rights.temp Available to the World Wide Web en_US
dc.identifier.bibliographicCitation S. Mosayyebpour, M. Esmaeili, and T. A. Gulliver, "Single-Microphone Early and Late Reverberation Suppression in Noisy Speech," IEEE Trans. Audio, Speech, Lang. Process.,vol. 21, no. 2, pp. 322-335, Feb. 2013. en_US
dc.identifier.bibliographicCitation S. Mosayyebpour, H. Sheikhzadeh, T. A. Gulliver, and M. Esmaeili, "Single- Microphone LP Residual Skewness-based Approach for Inverse Filtering of Room Impulse Response," IEEE Trans. Audio, Speech and Lang. Process., vol. 20, pp. 1617-1632, July 2012. en_US
dc.identifier.bibliographicCitation S. Mosayyebpour, A. Keshavarz, M. Biguesh, T. A. Gulliver, and M. Esmaeili "Speech-Model based Accurate Blind Reverberation Time Estimation Using an LPC Filter," IEEE Trans. Audio, Speech, Lang. Process., vol. 20, no. 6, pp. 1884-1893, Aug. 2012. en_US
dc.identifier.bibliographicCitation S. Mosayyebpour, H. Lohrasbipeydeh, M. Esmaeili, and T. A. Gulliver, "Time Delay Estimation via Minimum-Phase and All-Pass Component Processing," in Proc. IEEE Int. Conf. Acoustics, Speech and Signal Process. (ICASSP), Vancouver, BC, pp. 4285-4289, May 2013. en_US
dc.identifier.bibliographicCitation S. Mosayyebpour, T. A. Gulliver, and M. Esmaeili, "Single-Microphone Speech Enhancement by Skewness Maximization and Spectral Subtraction," International Workshop on Acoustic Signal Enhancement (IWAENC), pp. 1-4, Sep. 2012. en_US
dc.identifier.bibliographicCitation S. Mosayyebpour, A. Sayyadiyan, M. Zareian, and A. Shahbazi, "Single Channel Inverse Filtering of Room Impulse Response by Maximizing Skewness of LP Residual," IEEE Int. Conf. on Signal Acquisition and Process. (ICSAP), pp. 130-134, Feb. 2010. en_US
dc.identifier.bibliographicCitation S. Mosayyebpour, A. Sayyadiyan, E. Soltan Mohammadi, A. Shahbazi, and A. Keshavarz, "Time Delay Estimation using One Microphone Inverse Filtering in a Highly Reverberant Room," Proc. IEEE Int. Conf. on Signal Acquisition and Process. (ICSAP), pp. 140-144, Feb. 2010. en_US
dc.description.scholarlevel Graduate en_US
dc.description.proquestcode 0544 en_US
dc.description.proquestcode 0984 en_US


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