Speech modeling based on balanced reduction techniques

In this thesis an CELP algorithm based speech signal processing sys-tem is developed, in which, unlike the classical modeling where an Auto Regressive model is used, a stable Auto Regressive Moving Av-erage (ARM A) model employed. The basic idea is to first estimate the vocal tract filter with an Au...

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書目詳細資料
主要作者: Gao, Guangmin
其他作者: Li, Gang
格式: Theses and Dissertations
語言:English
出版: 2008
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在線閱讀:http://hdl.handle.net/10356/13366
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機構: Nanyang Technological University
語言: English
實物特徵
總結:In this thesis an CELP algorithm based speech signal processing sys-tem is developed, in which, unlike the classical modeling where an Auto Regressive model is used, a stable Auto Regressive Moving Av-erage (ARM A) model employed. The basic idea is to first estimate the vocal tract filter with an Auto Regressive(AR) model of very high order and then convert it into an ARMA model via the powerful Balanced Model Reduction(BMR) techniques. Thus, the difficulties in the direct estimation of ARMA parameters is avoided. Another advantage of this method is to estimate the vocal tract filter as one transfer function and hence no pitch detection is required, which may simplify the existing speech processing. It is believed that with the ARMA model obtained using this proposed mothed, the CELP algorithm can achieve a synthetic speech of high quality at very low bit rates.