Articulatory based speech models for blind speech dereverberation using sequential Monte Carlo methods
Articulatory based speech models for blind speech dereverberation using sequential Monte Carlo methods
Room reverberation leads to reduced intelligibility of audio signals. Enhancement is thus crucial for high-quality audio and scene analysis applications. This paper proposes to directly and optimally estimate the source signal and acoustic channel from the distorted observations. The remaining model parameters are sampled from a particle filter, facilitating real-time dereverberation. The approach was previously successfully applied to single- and multisensor blind dereverberation. Enhancement can be improved upon by accurately modelling the speech production system. This paper therefore extends the blind dereverberation approach to incorporate a novel source model based on parallel formant synthesis and compares the approach to one using a time-varying AR model, with parameters varying according to a random walk. Experimental data shows that dereverberation using the proposed model is improved for vowels, stop consonants, and fricatives.
2131-2135
European Signal Processing Conference
Evers, Christine
93090c84-e984-4cc3-9363-fbf3f3639c4b
Hopgood, James R.
ae180a4d-33bf-468d-ab66-5eeb152e7fc2
1 December 2010
Evers, Christine
93090c84-e984-4cc3-9363-fbf3f3639c4b
Hopgood, James R.
ae180a4d-33bf-468d-ab66-5eeb152e7fc2
Evers, Christine and Hopgood, James R.
(2010)
Articulatory based speech models for blind speech dereverberation using sequential Monte Carlo methods.
In European Signal Processing Conference.
European Signal Processing Conference.
.
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Conference or Workshop Item
(Paper)
Abstract
Room reverberation leads to reduced intelligibility of audio signals. Enhancement is thus crucial for high-quality audio and scene analysis applications. This paper proposes to directly and optimally estimate the source signal and acoustic channel from the distorted observations. The remaining model parameters are sampled from a particle filter, facilitating real-time dereverberation. The approach was previously successfully applied to single- and multisensor blind dereverberation. Enhancement can be improved upon by accurately modelling the speech production system. This paper therefore extends the blind dereverberation approach to incorporate a novel source model based on parallel formant synthesis and compares the approach to one using a time-varying AR model, with parameters varying according to a random walk. Experimental data shows that dereverberation using the proposed model is improved for vowels, stop consonants, and fricatives.
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Published date: 1 December 2010
Venue - Dates:
18th European Signal Processing Conference, EUSIPCO 2010, , Aalborg, Denmark, 2010-08-23 - 2010-08-27
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Local EPrints ID: 445886
URI: http://eprints.soton.ac.uk/id/eprint/445886
PURE UUID: 995cd4bc-c3ce-46d8-ae8a-27a5c5a89642
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Date deposited: 13 Jan 2021 17:31
Last modified: 23 Feb 2023 03:21
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Contributors
Author:
Christine Evers
Author:
James R. Hopgood
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