DocumentCode
323805
Title
Automatic estimation of formant and voice source parameters using a subspace based algorithm
Author
Yang, Chang-Sheng ; Kasuya, Hideki
Author_Institution
Fac. of Eng., Utsunomiya Univ., Japan
Volume
2
fYear
1998
fDate
12-15 May 1998
Firstpage
941
Abstract
An automatic method is proposed to estimate jointly formant and voice source parameters from a speech signal. A Rosenberg-Klatt (1990) model is used to approximate a voicing source waveform for voiced speech, whereas a white noise signal is assumed for the unvoiced speech. The vocal tract characteristic is represented by an IIR filter. The formant and anti-formant values are calculated from the IIR filter coefficients which are estimated by using the subspace-based system identification algorithm, while an exhaustive search procedure is applied to obtain the optimal source parameter values, where an error criterion is introduced in the frequency domain. An experiment has been performed to examine the performance of the proposed method with natural speech. The results show that the source parameters such as open and closure instants estimated by the method is in good agreement with those defined on the electro-glottograph signals and the formant values estimated are also accurate
Keywords
IIR filters; bioelectric potentials; electroencephalography; filtering theory; frequency-domain analysis; parameter estimation; search problems; speech processing; white noise; EEG waveform; IIR filter coefficients; Rosenberg-Klatt model; anti-formant values; automatic estimation; closure instant; electro-glottograph signals; error criterion; exhaustive search procedure; experiment; formant parameters; formant values; frequency domain; natural speech; open instant; performance; speech analysis; speech signal; subspace based algorithm; subspace-based system identification algorithm; vocal tract characteristic; voice source parameters; voiced speech; voicing source waveform; white noise signal; Filters; Frequency domain analysis; Natural languages; Parameter estimation; Signal analysis; Signal processing; Signal processing algorithms; Speech analysis; Speech synthesis; System identification;
fLanguage
English
Publisher
ieee
Conference_Titel
Acoustics, Speech and Signal Processing, 1998. Proceedings of the 1998 IEEE International Conference on
Conference_Location
Seattle, WA
ISSN
1520-6149
Print_ISBN
0-7803-4428-6
Type
conf
DOI
10.1109/ICASSP.1998.675421
Filename
675421
Link To Document