Yuan-Fu Liao

Dr. Yuan-Fu Liao

Department of Electronic Engineering, National Taipei University of Technology, Taipei, Taiwan

Research Focus & Metrics

Academic Career Length

32Years (1995-Present)

(Calculated once Scilit publication data is synced: current year − year of first publication + 1)

Representative Publications

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1 January 1998 in IEEE Transactions on Speech and Audio Processing
Verified
Journal Article
Closed access
SCSin-Horng Chen
Sin-Horng Chen
Yuan-Fu LiaoYuan-Fu Liao
Yuan-Fu Liao
SCSong-Mao Chiang
Song-Mao Chiang
SCSaga Chang
Saga Chang
A novel recurrent neural network-based (RNN-based) front-end preclassification scheme for fast continuous Mandarin speech recognition is proposed. First, an RNN is employed to discriminate each input frame for the three broad classes of initial, final, and silence. A finite state machine (FSM) is then used to classify the input frame into four states including three stable states of initial (I), final (F), and silence (S), and a transient (T) state. The decision is made based on examining whether the RNN discriminates well between classes. We then restrict the search space for the three stable states in the following DP search to speed up the recognition process. The efficiency of the proposed scheme was examined by simulations in which we incorporate it with a hidden Markov model-based (HMM-based) continuous 411 Mandarin based-syllables recognizer. The experimental results showed that it can be used in conjunction with the beam search to greatly reduce the computational complexity of the HMM recognizer while keeping the recognition rate almost undegraded.

2 May 2014 in IEEE/ACM Transactions on Audio, Speech, and Language Processing
Verified
Journal Article
Closed access
SCSin-Horng Chen
Sin-Horng Chen
CHChiao-Hua Hsieh
Chiao-Hua Hsieh
CCChen-Yu Chiang
Chen-Yu Chiang
HHHsi-Chun Hsiao
Hsi-Chun Hsiao
YWYih-Ru Wang
Yih-Ru Wang
HYHsiu-Min Yu
Hsiu-Min Yu
A new data-driven approach to building a speaking rate-dependent hierarchical prosodic model (SR-HPM), directly from a large prosody-unlabeled speech database containing utterances of various speaking rates, to describe the influences of speaking rate on Mandarin speech prosody is proposed. It is an extended version of the existing HPM model which contains 12 sub-models to describe various relationships of prosodic-acoustic features of speech signal, linguistic features of the associated text, and prosodic tags representing the prosodic structure of speech. Two main modifications are suggested. One is designing proper normalization functions from the statistics of the whole database to compensate the influences of speaking rate on all prosodic-acoustic features. Another is modifying the HPM training to let its parameters be speaking-rate dependent. Experimental results on a large Mandarin read speech corpus showed that the parameters of the SR-HPM together with these feature normalization functions interpreted the effects of speaking rate on Mandarin speech prosody very well. An application of the SR-HPM to design and implement a speaking rate-controlled Mandarin TTS system is demonstrated. The system can generate natural synthetic speech for any given speaking rate in a wide range of 3.4-6.8 syllables/sec. Two subjective tests, MOS and preference test, were conducted to compare the proposed system with the popular HTS system. The MOS scores of the proposed system were in the range of 3.58-3.83 for eight different speaking rates, while they were in 3.09-3.43 for HTS. Besides, the proposed system had higher preference scores (49.8%-79.6%) than those (9.8%-30.7%) of HTS. This confirmed the effectiveness of the speaking rate control method of the proposed TTS system.

7 January 2002 in Speech Communication
Verified
Journal Article
Closed access
WWWern-Jun Wang
Wern-Jun Wang
Yuan-Fu LiaoYuan-Fu Liao
Yuan-Fu Liao
SCSin-Horng Chen
Sin-Horng Chen

1 June 1995 in Pattern Recognition
Verified
Journal Article
Closed access
WCWen-Yuan Chen
Wen-Yuan Chen
Yuan-Fu LiaoYuan-Fu Liao
Yuan-Fu Liao
SCSin-Horng Chen
Sin-Horng Chen

28 November 2019 in Journal of Signal Processing Systems
Verified
Journal Article
Closed access
PDF file
Yuan-Fu LiaoYuan-Fu Liao
Yuan-Fu Liao
YCYung-Hsiang Shawn Chang
Yung-Hsiang Shawn Chang
YLYu-Chen Lin
Yu-Chen Lin
WHWu-Hua Hsu
Wu-Hua Hsu
Matus PlevaMatus Pleva
Matus Pleva
Jozef JuhárJozef Juhár
Jozef Juhár

1 August 2007 in IEEE Transactions on Audio, Speech, and Language Processing
Verified
Journal Article
Closed access
Yuan-Fu LiaoYuan-Fu Liao
Yuan-Fu Liao
ZCZi-He Chen
Zi-He Chen
YJYau-Tarng Juang
Yau-Tarng Juang
Handsets that are not seen in the training phase (unseen handsets) are significant sources of performance degradation for speaker identification (SID) applications in the telecommunication environment. In this paper, a novel latent prosody analysis (LPA) approach to automatically extract the most discriminative prosodic cues for assisting in conventional spectral feature-based SID is proposed. The concept of the LPA approach is to transform the SID problem into a full-text document retrieval-like task via 1) prosodic contour tokenization, 2) latent prosody analysis, and 3) speaker retrieval. Experimental results of the phonetically balanced, read-speech, handset-TIMIT (HTIMIT) database demonstrated that the proposed method of fusing the LPA prosodic feature-based SID systems with maximum-likelihood a priori handset knowledge interpolation (ML-AKI) spectral feature-based SID outperformed both the pitch and energy Gaussian mixture model (Pitch-GMM) and the bigram of the prosodic state (Bigram) counterparts for both cases of counting all and only unseen handsets.