This collection includes both ASU Theses and Dissertations, submitted by graduate students, and the Barrett, Honors College theses submitted by undergraduate students. 

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Description
Speech is generated by articulators acting on

a phonatory source. Identification of this

phonatory source and articulatory geometry are

individually challenging and ill-posed

problems, called speech separation and

articulatory inversion, respectively.

There exists a trade-off

between decomposition and recovered

articulatory geometry due to multiple

possible mappings between an

articulatory configuration

and the speech produced. However, if measurements

are

Speech is generated by articulators acting on

a phonatory source. Identification of this

phonatory source and articulatory geometry are

individually challenging and ill-posed

problems, called speech separation and

articulatory inversion, respectively.

There exists a trade-off

between decomposition and recovered

articulatory geometry due to multiple

possible mappings between an

articulatory configuration

and the speech produced. However, if measurements

are obtained only from a microphone sensor,

they lack any invasive insight and add

additional challenge to an already difficult

problem.

A joint non-invasive estimation

strategy that couples articulatory and

phonatory knowledge would lead to better

articulatory speech synthesis. In this thesis,

a joint estimation strategy for speech

separation and articulatory geometry recovery

is studied. Unlike previous

periodic/aperiodic decomposition methods that

use stationary speech models within a

frame, the proposed model presents a

non-stationary speech decomposition method.

A parametric glottal source model and an

articulatory vocal tract response are

represented in a dynamic state space formulation.

The unknown parameters of the

speech generation components are estimated

using sequential Monte Carlo methods

under some specific assumptions.

The proposed approach is compared with other

glottal inverse filtering methods,

including iterative adaptive inverse filtering,

state-space inverse filtering, and

the quasi-closed phase method.
ContributorsVenkataramani, Adarsh Akkshai (Author) / Papandreou-Suppappola, Antonia (Thesis advisor) / Bliss, Daniel W (Committee member) / Turaga, Pavan (Committee member) / Arizona State University (Publisher)
Created2018
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Description
This work examines two main areas in model-based time-varying signal processing with emphasis in speech processing applications. The first area concentrates on improving speech intelligibility and on increasing the proposed methodologies application for clinical practice in speech-language pathology. The second area concentrates on signal expansions matched to physical-based models but

This work examines two main areas in model-based time-varying signal processing with emphasis in speech processing applications. The first area concentrates on improving speech intelligibility and on increasing the proposed methodologies application for clinical practice in speech-language pathology. The second area concentrates on signal expansions matched to physical-based models but without requiring independent basis functions; the significance of this work is demonstrated with speech vowels.

A fully automated Vowel Space Area (VSA) computation method is proposed that can be applied to any type of speech. It is shown that the VSA provides an efficient and reliable measure and is correlated to speech intelligibility. A clinical tool that incorporates the automated VSA was proposed for evaluation and treatment to be used by speech language pathologists. Two exploratory studies are performed using two databases by analyzing mean formant trajectories in healthy speech for a wide range of speakers, dialects, and coarticulation contexts. It is shown that phonemes crowded in formant space can often have distinct trajectories, possibly due to accurate perception.

A theory for analyzing time-varying signals models with amplitude modulation and frequency modulation is developed. Examples are provided that demonstrate other possible signal model decompositions with independent basis functions and corresponding physical interpretations. The Hilbert transform (HT) and the use of the analytic form of a signal are motivated, and a proof is provided to show that a signal can still preserve desirable mathematical properties without the use of the HT. A visualization of the Hilbert spectrum is proposed to aid in the interpretation. A signal demodulation is proposed and used to develop a modified Empirical Mode Decomposition (EMD) algorithm.
ContributorsSandoval, Steven, 1984- (Author) / Papandreou-Suppappola, Antonia (Thesis advisor) / Liss, Julie M (Committee member) / Turaga, Pavan (Committee member) / Kovvali, Narayan (Committee member) / Arizona State University (Publisher)
Created2016