subject:(speech processing)

Queens University

1. Zheng, Chenxi. Single-Microphone Speech Dereverberation: Modulation Domain Processing and Quality Assessment .

Degree: Electrical and Computer Engineering, 2011, Queens University

URL: http://hdl.handle.net/1974/6609

► In a reverberant enclosure, acoustic speech signals are degraded by reflections from walls, ceilings, and objects. Restoring speech quality and intelligibility from reverberated speech has… (more)

▼ In a reverberant enclosure, acoustic speech signals are degraded by reflections from walls, ceilings, and objects. Restoring speech quality and intelligibility from reverberated speech has received increasing interest over the past few years. Although multiple channel dereverberation methods provide some improvements in speech quality/ intelligibility, single-channel dereverberation remains an open challenge. Two types of advanced single-channel dereverberation methods, namely acoustic domain spectral subtraction and modulation domain filtering, provide small improvement in speech quality and intelligibility. In this thesis, we study single-channel dereverberation algorithms. Firstly, an upper bound of time-frequency masking (TFM) performance for dereverberation is obtained using ideal time-frequency masking (ITFM). ITFM has access to both the clean and reverberated speech signals in estimating the binary-mask matrix. ITFM implements binary masking in the short time Fourier transform (STFT) domain, preserving only those spectral components less corrupted by reverberation. The experiment results show that single-channel ITFM outperforms four existing multi-channel dereverberation methods and suggest that large potential improvements could be obtained using TFM for speech dereverberation. Secondly, a novel modulation domain spectral subtraction method is proposed for dereverberation. This method estimates modulation domain long reverberation spectral variance (LRSV) from time domain LRSV using a statistical room impulse response (RIR) model and implements spectral subtraction in the modulation domain. On one hand, different from acoustic domain spectral subtraction, our method implements spectral subtraction in the modulation domain, which has been shown to play an important role in speech perception. On the other hand, different from modulation domain filtering which uses a time-invariant filter, our method takes the changes of reverberated speech spectral variance along time into account and implements spectral subtraction adaptively. Objective and informal subjective tests show that our proposed method outperforms two existing state-of-the-art single-channel dereverberation algorithms.

Subjects/Keywords: Speech enhancement ; Dereverberation ; Speech signal processing

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Zheng, C. (2011). Single-Microphone Speech Dereverberation: Modulation Domain Processing and Quality Assessment . (Thesis). Queens University. Retrieved from http://hdl.handle.net/1974/6609

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

Zheng, Chenxi. “Single-Microphone Speech Dereverberation: Modulation Domain Processing and Quality Assessment .” 2011. Thesis, Queens University. Accessed January 17, 2021. http://hdl.handle.net/1974/6609.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

Zheng, Chenxi. “Single-Microphone Speech Dereverberation: Modulation Domain Processing and Quality Assessment .” 2011. Web. 17 Jan 2021.

Vancouver:

Zheng C. Single-Microphone Speech Dereverberation: Modulation Domain Processing and Quality Assessment . [Internet] [Thesis]. Queens University; 2011. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/1974/6609.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

Zheng C. Single-Microphone Speech Dereverberation: Modulation Domain Processing and Quality Assessment . [Thesis]. Queens University; 2011. Available from: http://hdl.handle.net/1974/6609

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

University of Wollongong

2. Cheng, Eva Chung-Wai. Spatial analysis of multiparty speech scenes.

Degree: PhD, 2012, University of Wollongong

URL: 090609 Signal Processing ; https://ro.uow.edu.au/theses/3713

► This thesis conducts a series of investigations into the estimation of speaker location cues from multi-party meeting speech recordings. As participants in meetings generally… (more)

▼ This thesis conducts a series of investigations into the estimation of speaker location cues from multi-party meeting speech recordings. As participants in meetings generally remain stationary, speaker location information is fundamentally useful for higher-level tasks such as steering a microphone array beamformer towards an active speaker, or segmenting meeting speech into each speaker’s period of participation for ‘browsing’ of recordings and speech recognition. Whilst existing speaker location cues are typically Time-Delay Estimations (TDE) estimated from microphone array signals, this thesis proposes the use of level and time/phase-based spatial cues, motivated by the spatial cues utilised in recently standardised Spatial Audio Coding (SAC) paradigms. Implemented using leading and standardised SAC techniques, experiments compared the proposed SAC spatial cues with TDE and found the combination of TDE with SAC level-based cues to be the most accurate for speech segmentation. As meetings recordings predominantly contain speech content, front-end Linear- Prediction (LP) analysis using theoretical and standardised speech coders is then investigated with single and multichannel LP models. Whilst existing approaches estimate TDE from the Hilbert envelope of single-channel LP speech residuals, this thesis proposes the use of intra and interchannel multichannel prediction and found spatial cues estimated from the Hilbert envelope of LP residuals to be the most robust against reverberation. Further experiments investigating the effect of microphone array characteristics found the microphone directivity pattern to significantly influence spatial cue estimation: the omnidirectional and cardioid polar responses optimally suit time/phase and levelbased cues, respectively. In practice, however, switching microphone patterns or employing mixed pattern arrays is impractical. This thesis proposes the use of the Ambisonic B-format steerable ‘virtual microphone’ to enable the same physical microphones to be simultaneously used for optimal capture of both time/phase and level-based cues. Further, results indicate that steering the virtual microphone in real-time to an active speaker, localised using sound intensity techniques, also improves meeting speech capture. Thus, the work in this thesis has contributions in practical spatial meeting speech analysis, where investigations studied microphone array characteristics, spatial recording techniques, and algorithms in spatial cue estimation and speech processing as utilised in internationally standardised speech and audio coders.

Subjects/Keywords: speech signal processing; microphone arrays; meeting speech

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Cheng, E. C. (2012). Spatial analysis of multiparty speech scenes. (Doctoral Dissertation). University of Wollongong. Retrieved from 090609 Signal Processing ; https://ro.uow.edu.au/theses/3713

Chicago Manual of Style (16^th Edition):

Cheng, Eva Chung-Wai. “Spatial analysis of multiparty speech scenes.” 2012. Doctoral Dissertation, University of Wollongong. Accessed January 17, 2021. 090609 Signal Processing ; https://ro.uow.edu.au/theses/3713.

MLA Handbook (7^th Edition):

Cheng, Eva Chung-Wai. “Spatial analysis of multiparty speech scenes.” 2012. Web. 17 Jan 2021.

Vancouver:

Cheng EC. Spatial analysis of multiparty speech scenes. [Internet] [Doctoral dissertation]. University of Wollongong; 2012. [cited 2021 Jan 17]. Available from: 090609 Signal Processing ; https://ro.uow.edu.au/theses/3713.

Council of Science Editors:

Cheng EC. Spatial analysis of multiparty speech scenes. [Doctoral Dissertation]. University of Wollongong; 2012. Available from: 090609 Signal Processing ; https://ro.uow.edu.au/theses/3713

University of Texas – Austin

3. -5484-2532. Listening under pressure : the downside of motivation.

Degree: PhD, Communication Sciences & Disorders, 2017, University of Texas – Austin

URL: http://hdl.handle.net/2152/68142

► The desire for self-improvement is critical to human performance and learning outcomes. Paradoxically, however, being subjected to increased performance pressure can also result in “choking… (more)

▼ The desire for self-improvement is critical to human performance and learning outcomes. Paradoxically, however, being subjected to increased performance pressure can also result in “choking under pressure”. No studies have experimentally examined the extent to which motivation impacts native speech processing. This dissertation manipulated performance pressure in listeners, and systematically examined its impact on three speech-processing experiments. Sixty adult native English listeners and 45 non-native listeners with poorer English proficiency completed three speech processing experiments, twice – once to establish a baseline, and again to measure changes in performance. In these experiments using native English speech, listeners detected (illusionary) sound changes, categorized phonemes under lexical interference, and recognized words in noises. After baseline testing, half of the participants in each language group were instructed to work, with a fictitious partner, towards a performance-contingent monetary reward; the other half, as controls, simply performed the tasks a second time. This study demonstrated a negative impact of performance pressure on native listeners in all experiments. Relative to the controls, the motivation group were more susceptible to illusions, failed to ignore lexical interference despite prior exposure, and recognized fewer words in cognitively-demanding listening situations. Unexpectedly, relative to native listeners, non-native listeners perceived it as less important to perform well, and those who were in the high performance-pressure group requested significantly greater amount of money for improvement. These language-group differences in task-related attitudes might be a confounding factor that moderate the effect of motivation. By illustrating a complex interaction among motivation, listener status, and performance-induced demands, this dissertation highlights the importance of motivation in speech science. Advisors/Committee Members: Chandrasekaran, Bharath (advisor), Champlin, Craig A. (committee member), Henry, Maya L. (committee member), Peña, Elizabeth D. (committee member), Griffin, Zenzi M. (committee member).

Subjects/Keywords: Motivation; Speech processing; Speech and hearing science

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

-5484-2532. (2017). Listening under pressure : the downside of motivation. (Doctoral Dissertation). University of Texas – Austin. Retrieved from http://hdl.handle.net/2152/68142

Note: this citation may be lacking information needed for this citation format:
Author name may be incomplete

Chicago Manual of Style (16^th Edition):

-5484-2532. “Listening under pressure : the downside of motivation.” 2017. Doctoral Dissertation, University of Texas – Austin. Accessed January 17, 2021. http://hdl.handle.net/2152/68142.

Note: this citation may be lacking information needed for this citation format:
Author name may be incomplete

MLA Handbook (7^th Edition):

-5484-2532. “Listening under pressure : the downside of motivation.” 2017. Web. 17 Jan 2021.

Note: this citation may be lacking information needed for this citation format:
Author name may be incomplete

Vancouver:

-5484-2532. Listening under pressure : the downside of motivation. [Internet] [Doctoral dissertation]. University of Texas – Austin; 2017. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/2152/68142.

Note: this citation may be lacking information needed for this citation format:
Author name may be incomplete

Council of Science Editors:

-5484-2532. Listening under pressure : the downside of motivation. [Doctoral Dissertation]. University of Texas – Austin; 2017. Available from: http://hdl.handle.net/2152/68142

Note: this citation may be lacking information needed for this citation format:
Author name may be incomplete

4. Prajith, P. Investigations on the applications of dynamical instabilities and deterministic chaos for speech signal processing.

Degree: Physics, 2008, University of Calicut

URL: http://shodhganga.inflibnet.ac.in/handle/10603/3960

►

This study investigates the potential use of Reconstructed Phase Space (RPS) based parameters for Speech Signal Processing by utilizing nonlinear dynamical systems theory. In this… (more)

Subjects/Keywords: Physics; Speech signal processing; chaos

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Prajith, P. (2008). Investigations on the applications of dynamical instabilities and deterministic chaos for speech signal processing. (Thesis). University of Calicut. Retrieved from http://shodhganga.inflibnet.ac.in/handle/10603/3960

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

Prajith, P. “Investigations on the applications of dynamical instabilities and deterministic chaos for speech signal processing.” 2008. Thesis, University of Calicut. Accessed January 17, 2021. http://shodhganga.inflibnet.ac.in/handle/10603/3960.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

Prajith, P. “Investigations on the applications of dynamical instabilities and deterministic chaos for speech signal processing.” 2008. Web. 17 Jan 2021.

Vancouver:

Prajith P. Investigations on the applications of dynamical instabilities and deterministic chaos for speech signal processing. [Internet] [Thesis]. University of Calicut; 2008. [cited 2021 Jan 17]. Available from: http://shodhganga.inflibnet.ac.in/handle/10603/3960.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

Prajith P. Investigations on the applications of dynamical instabilities and deterministic chaos for speech signal processing. [Thesis]. University of Calicut; 2008. Available from: http://shodhganga.inflibnet.ac.in/handle/10603/3960

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

5. TEOH, EMILY SIEW. Investigating the Neural Correlates of Speech Processing & Selective Auditory Attention using EEG.

Degree: School of Engineering. Discipline of Electronic & Elect. Engineering, 2020, Trinity College Dublin

URL: http://hdl.handle.net/2262/92187

► Speech comprehension is a remarkable human ability. Most normal-hearing people are adept at attending to a speech stream even amidst a noisy multi-talker background and… (more)

▼ Speech comprehension is a remarkable human ability. Most normal-hearing people are adept at attending to a speech stream even amidst a noisy multi-talker background and parsing the layers of information it contains in real-time to uncover its meaning. The cortex is thought to be hierarchically organised to perform the latter feat, with higher levels processing increasingly abstract features. Nonetheless, the precise computations it undertakes and how these processes are modulated by attention remain incompletely understood. The development of novel encoding/decoding approaches for relating high temporal resolution neuroimaging modalities to representations of multivariate stimuli like continuous natural speech have enabled researchers to shed some light on this topic. An important recent discovery is that low-frequency, non-invasive EEG/MEG tracks the amplitude envelope of speech (a low-level acoustic measure conveying many cues important for speech comprehension) and that a robust reconstruction of the measure can be acquired from neural data. Moreover, during cocktail party listening tasks, this tracking and the accuracy of reconstruction have been shown to be modulated by attention. These findings have given rise to further research along several lines, including studies employing the framework to investigate how cortex encodes other speech processing stages along the hierarchy and how this encoding is affected by attention, as well as studies developing the idea of exploiting envelope reconstruction as a means of decoding attentional selection for the implementation of smart devices like steerable hearing aids. In this thesis, we employ a particular encoding/decoding approach ? the temporal response function (TRF) ? in conjunction with EEG to address several questions within these subareas. We first tested the efficacy of a state-of-the-art framework for utilising envelope reconstruction to decode auditory attention (O Sullivan et al., 2015) within the context of a cocktail party paradigm with moving talkers and showed that it is robust. This was motivated by the non-stationarity of real-world environments. We then considered if (1) the decoder weights themselves (i.e., the model weights mapping from EEG data to the acoustic envelope) and (2) alpha power might contain unique information that can be leveraged for the decoding of attention. We showed that incorporating a metric based on the consistency of model weights across subjects into the decoding framework yielded an improvement in performance above and beyond using envelope reconstruction alone. We then investigated the neural processing of prosody ? an aspect of spoken language that conveys another layer of meaning on top of linguistic units. In particular, we were interested to explore how prosodic pitch is encoded in low-frequency EEG during listening to continuous natural speech. We mapped two measures of prosodic pitch ? relative pitch and harmonic resolvability ? to concurrently-recorded EEG. These measures were inspired by an ECoG study showing… Advisors/Committee Members: Lalor, Edmund.

Subjects/Keywords: EEG; Neural Signal Processing; Neural Speech Processing

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

TEOH, E. S. (2020). Investigating the Neural Correlates of Speech Processing & Selective Auditory Attention using EEG. (Thesis). Trinity College Dublin. Retrieved from http://hdl.handle.net/2262/92187

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

TEOH, EMILY SIEW. “Investigating the Neural Correlates of Speech Processing & Selective Auditory Attention using EEG.” 2020. Thesis, Trinity College Dublin. Accessed January 17, 2021. http://hdl.handle.net/2262/92187.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

TEOH, EMILY SIEW. “Investigating the Neural Correlates of Speech Processing & Selective Auditory Attention using EEG.” 2020. Web. 17 Jan 2021.

Vancouver:

TEOH ES. Investigating the Neural Correlates of Speech Processing & Selective Auditory Attention using EEG. [Internet] [Thesis]. Trinity College Dublin; 2020. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/2262/92187.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

TEOH ES. Investigating the Neural Correlates of Speech Processing & Selective Auditory Attention using EEG. [Thesis]. Trinity College Dublin; 2020. Available from: http://hdl.handle.net/2262/92187

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Univerzitet u Beogradu

6. Bilibajkić, Ružica, 1980-. Prepoznavanje artikulaciono-akustičnih odstupanja glasova u patološkom govoru.

Degree: Elektrotehnički fakultet, 2016, Univerzitet u Beogradu

URL: https://fedorabg.bg.ac.rs/fedora/get/o:12129/bdef:Content/get

►

Elektrotehnika - Akustika / Electrical engineering - Acoustics

U kliničkoj praksi procena artikulacionih poremećaja uglavnom se vrši pomoću subjektivnih mera i metoda, audio vizuelnom procenim… (more)

▼

Elektrotehnika - Akustika / Electrical engineering - Acoustics

U kliničkoj praksi procena artikulacionih poremećaja uglavnom se vrši pomoću subjektivnih mera i metoda, audio vizuelnom procenim od strane obučenih eksperata-logopeda. U tu svrhu postoji određeni broj standardnih testova i procedura na osnovu kojih se daje kvalitativna i kvantitativna ocena patološkog izgovora. Ovakav pristup podložan je uticaju faktora koji potiču od ispitivača i uslova ispitivanja, a koji nisu posledica patološkog izgovora. Postojanje automatskog sistema za procenu patologije u govoru, u prvom redu bi dovelo do toga da ovakvi faktori postanu zanemarljivi, čime bi se višestruko unapredila klinička praksa. Ovo istraživanje fokusirano je na formiranje jedinstvenog sistema za detekciju patologije govora koji se bazira na aktuelnim dostignućima na polju obrade govora i standardnim metodama koje se u te svrhe koriste u logopedskoj praksi. U radu je prikazan razvoj sistema za procenu artikulacionih odstupanja govora koji je osmišljen tako da prati metodologiju logopedskog pristupa. Predstavljeni su rezultati modelovanja auditornog sistema čoveka prilikom detekcije akustičkih obeležja u govornom signalu koja učestvuju u manifestaciji patologije. Data su i istraživanja na temu segmentacije glasova pogodnih za dalju obradu radi utvrđivanja postojanja i stepena patologije. Prikazan je i rad na izradi automatskog sistema za klasifikaciju govora u smislu postojanja ili odsustva patologije u govornom signalu. Poseban deo posvećen je istraživanjima na temu prepoznavanja specifičnih artikulacionih odstupanja pojedinačnog glasa. Rezultati istraživanja pokazali su da se upotrebom algoritama koji se koriste u obradi govora može izvesti sistem za automatsku procenu patološkog izgovora. Rezultati koji se dobijaju automatskom procenom u visokom procentu se podudaraju sa ocenama dobijenim od strane eksperata.

Advisors/Committee Members: Šumarac Pavlović, Dragana, 1967-.

Subjects/Keywords: digital speech processing; speech pathology detection; auditory model; speech segmentation

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Bilibajkić, Ružica, 1. (2016). Prepoznavanje artikulaciono-akustičnih odstupanja glasova u patološkom govoru. (Thesis). Univerzitet u Beogradu. Retrieved from https://fedorabg.bg.ac.rs/fedora/get/o:12129/bdef:Content/get

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

Bilibajkić, Ružica, 1980-. “Prepoznavanje artikulaciono-akustičnih odstupanja glasova u patološkom govoru.” 2016. Thesis, Univerzitet u Beogradu. Accessed January 17, 2021. https://fedorabg.bg.ac.rs/fedora/get/o:12129/bdef:Content/get.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

Bilibajkić, Ružica, 1980-. “Prepoznavanje artikulaciono-akustičnih odstupanja glasova u patološkom govoru.” 2016. Web. 17 Jan 2021.

Vancouver:

Bilibajkić, Ružica 1. Prepoznavanje artikulaciono-akustičnih odstupanja glasova u patološkom govoru. [Internet] [Thesis]. Univerzitet u Beogradu; 2016. [cited 2021 Jan 17]. Available from: https://fedorabg.bg.ac.rs/fedora/get/o:12129/bdef:Content/get.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

Bilibajkić, Ružica 1. Prepoznavanje artikulaciono-akustičnih odstupanja glasova u patološkom govoru. [Thesis]. Univerzitet u Beogradu; 2016. Available from: https://fedorabg.bg.ac.rs/fedora/get/o:12129/bdef:Content/get

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

University of North Carolina – Greensboro

7. Wendell, Tyler James. Feature extraction and feature reduction for spoken letter recognition.

Degree: 2016, University of North Carolina – Greensboro

URL: http://libres.uncg.edu/ir/listing.aspx?styp=ti&id=19822

► The complexity of finding the relevant features for the classification of spoken letters is due to the phonetic similarities between letters and their high dimensionality.… (more)

Subjects/Keywords: Automatic speech recognition – Data processing; Speech processing systems; Human-computer interaction

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Wendell, T. J. (2016). Feature extraction and feature reduction for spoken letter recognition. (Masters Thesis). University of North Carolina – Greensboro. Retrieved from http://libres.uncg.edu/ir/listing.aspx?styp=ti&id=19822

Chicago Manual of Style (16^th Edition):

Wendell, Tyler James. “Feature extraction and feature reduction for spoken letter recognition.” 2016. Masters Thesis, University of North Carolina – Greensboro. Accessed January 17, 2021. http://libres.uncg.edu/ir/listing.aspx?styp=ti&id=19822.

MLA Handbook (7^th Edition):

Wendell, Tyler James. “Feature extraction and feature reduction for spoken letter recognition.” 2016. Web. 17 Jan 2021.

Vancouver:

Wendell TJ. Feature extraction and feature reduction for spoken letter recognition. [Internet] [Masters thesis]. University of North Carolina – Greensboro; 2016. [cited 2021 Jan 17]. Available from: http://libres.uncg.edu/ir/listing.aspx?styp=ti&id=19822.

Council of Science Editors:

Wendell TJ. Feature extraction and feature reduction for spoken letter recognition. [Masters Thesis]. University of North Carolina – Greensboro; 2016. Available from: http://libres.uncg.edu/ir/listing.aspx?styp=ti&id=19822

Stellenbosch University

8. Rademan, Christiaan Frans. Improving visual speech synthesis using Decision Tree Models.

Degree: MEng, Electrical and Electronic Engineering, 2016, Stellenbosch University

URL: http://hdl.handle.net/10019.1/98728

►

ENGLISH ABSTRACT: Visual speech synthesis is essential for believable virtual character interaction. Traditionally, animation artists recreate the oral motions expected from speech utterances. In response,… (more)

▼

ENGLISH ABSTRACT: Visual speech synthesis is essential for believable virtual character interaction. Traditionally, animation artists recreate the oral motions expected from speech utterances. In response, we present decision tree-based clustering techniques which are employed in automating visual speech animation. This is achieved using a small dataset of phoneticallyannotated audiovisual speech. Our work focuses on extending existing tree-based clustering algorithms by improving on the modelling of coarticulation effects. This is accomplished by capturing the motion of natural speech segments, referred to as dynamic visemes, and conserving their parameters during clustering and speech synthesis. Dynamic visemes are defined as the trajectories of oral features segmented by triphone boundaries. By applying simple search and concatenation criteria, our visual speech synthesis system uses decision trees to better predict which dynamic visemes to use. Experimentation guided all design decisions, suggesting which oral features were of greatest importance, identifying an appropriate dynamic viseme length and finding an effective interpolation method for conserving coarticulation. We evaluate the performance of our visual speech synthesis models by computing squared error differences between synthesised and measured feature trajectories. Perceptual tests also asked participants to compare virtual characters animated by model outputs. Both measured and perceptual tests show that our approaches lead to a clear improvement over a comparable baseline. Through our research, we intended on making speech synthesis more accessible. Therefore, the conversational agents are based on the freely available MakeHuman and Blender software components. The customised oral feature motion capture system is also easily reproduced and requires only consumer grade recording equipment.

AFRIKAANSE OPSOMMING: Visuele spraaksintese is noodsaaklik om geloofwaardige interaksie met virtuele karakters moontlik te maak. In die verlede het animasiekunstenaars mondbewegings vanaf werklike spraak nageboots. In hierdie studie bied ons tegnieke aan wat gebaseer is op saambondeling met behulp van besluitnemingsbome. Hierdie tegnieke word gebruik om die animasie van visuele spraak te outomatiseer, en maak gebruik van ’n klein datastel van foneties geannoteerde oudiovisuele spraak. Ons werk fokus op die uitbrei van bestaande besluitnemingsboom-saambondelingsalgoritmes, deur die modellering van koartikulasie-effekte te verbeter. Dit word moontlik gemaak deur eers die beweging van natuurlike spraaksegmente (viseme) vas te vang, en dan hul parameters te bewaar tydens die saambondeling en spraaksintese. Dinamiese viseme word gedefinieer as die trajekte van mondeienskappe, gesegmenteer deur trifoongrense. Deur eenvoudige soek- en saamvoegingskriteria toe te pas, kan ons visuele spraaksintesestelsel van besluitnemingsbome gebruik maak om beter te voorspel watter viseme aangewend moet word. Alle ontwerpsbesluite is deur ekspermintering gelei, om bv.…

Advisors/Committee Members: Niesler, T. R., Stellenbosch University. Faculty of Engineering. Dept. of Electrical and Electronic Engineering..

Subjects/Keywords: UCTD; Speech synthesis; Speech processing systems; Computer animation; Speech processing systems – Digital techniques; Decision trees

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Rademan, C. F. (2016). Improving visual speech synthesis using Decision Tree Models. (Thesis). Stellenbosch University. Retrieved from http://hdl.handle.net/10019.1/98728

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

Rademan, Christiaan Frans. “Improving visual speech synthesis using Decision Tree Models.” 2016. Thesis, Stellenbosch University. Accessed January 17, 2021. http://hdl.handle.net/10019.1/98728.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

Rademan, Christiaan Frans. “Improving visual speech synthesis using Decision Tree Models.” 2016. Web. 17 Jan 2021.

Vancouver:

Rademan CF. Improving visual speech synthesis using Decision Tree Models. [Internet] [Thesis]. Stellenbosch University; 2016. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/10019.1/98728.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

Rademan CF. Improving visual speech synthesis using Decision Tree Models. [Thesis]. Stellenbosch University; 2016. Available from: http://hdl.handle.net/10019.1/98728

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

University of Illinois – Urbana-Champaign

9. Bryan, Jacob. A sensorimotor basis of speech communication.

Degree: PhD, Electrical & Computer Engr, 2019, University of Illinois – Urbana-Champaign

URL: http://hdl.handle.net/2142/104777

► This dissertation presents the development of sensorimotor primitives as a means of constructing a language-agnostic model of speech communication. Insights from major theories in speech… (more)

Subjects/Keywords: speech; langauge; artificial intelligence; speech signal processing; speech communication; speech articulation; sensorimotor primitives; speech primitives; variational autoencoder; inverse channel encoder

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Bryan, J. (2019). A sensorimotor basis of speech communication. (Doctoral Dissertation). University of Illinois – Urbana-Champaign. Retrieved from http://hdl.handle.net/2142/104777

Chicago Manual of Style (16^th Edition):

Bryan, Jacob. “A sensorimotor basis of speech communication.” 2019. Doctoral Dissertation, University of Illinois – Urbana-Champaign. Accessed January 17, 2021. http://hdl.handle.net/2142/104777.

MLA Handbook (7^th Edition):

Bryan, Jacob. “A sensorimotor basis of speech communication.” 2019. Web. 17 Jan 2021.

Vancouver:

Bryan J. A sensorimotor basis of speech communication. [Internet] [Doctoral dissertation]. University of Illinois – Urbana-Champaign; 2019. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/2142/104777.

Council of Science Editors:

Bryan J. A sensorimotor basis of speech communication. [Doctoral Dissertation]. University of Illinois – Urbana-Champaign; 2019. Available from: http://hdl.handle.net/2142/104777

10. Hosom, John-Paul. Automatic time alignment of phonemes using acoustic-phonetic information.

Degree: PhD, 2000, Oregon Health Sciences University

URL: doi:10.6083/M4CF9N1F ; http://digitalcommons.ohsu.edu/etd/175

Subjects/Keywords: Speech processing systems

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Hosom, J. (2000). Automatic time alignment of phonemes using acoustic-phonetic information. (Doctoral Dissertation). Oregon Health Sciences University. Retrieved from doi:10.6083/M4CF9N1F ; http://digitalcommons.ohsu.edu/etd/175

Chicago Manual of Style (16^th Edition):

Hosom, John-Paul. “Automatic time alignment of phonemes using acoustic-phonetic information.” 2000. Doctoral Dissertation, Oregon Health Sciences University. Accessed January 17, 2021. doi:10.6083/M4CF9N1F ; http://digitalcommons.ohsu.edu/etd/175.

MLA Handbook (7^th Edition):

Hosom, John-Paul. “Automatic time alignment of phonemes using acoustic-phonetic information.” 2000. Web. 17 Jan 2021.

Vancouver:

Hosom J. Automatic time alignment of phonemes using acoustic-phonetic information. [Internet] [Doctoral dissertation]. Oregon Health Sciences University; 2000. [cited 2021 Jan 17]. Available from: doi:10.6083/M4CF9N1F ; http://digitalcommons.ohsu.edu/etd/175.

Council of Science Editors:

Hosom J. Automatic time alignment of phonemes using acoustic-phonetic information. [Doctoral Dissertation]. Oregon Health Sciences University; 2000. Available from: doi:10.6083/M4CF9N1F ; http://digitalcommons.ohsu.edu/etd/175

University of Alberta

11. Porretta, Vincent J. Perception and lexical processing of gradient foreign accentedness.

Degree: PhD, Department of Linguistics, 2015, University of Alberta

URL: https://era.library.ualberta.ca/files/4f16c5678

► This dissertation examines native speaker perception and processing of the variability inherent to non-native speech, specifically, Mandarin-accented English. In order to accomplish this, subjective ratings… (more)

▼ This dissertation examines native speaker perception and processing of the variability inherent to non-native speech, specifically, Mandarin-accented English. In order to accomplish this, subjective ratings were first collected as a measure of perceived foreign accentedness. The influence of linguistic variables (both acoustic and lexical) was investigated with regard to the perception of gradient foreign accentedness. The results of the ratings study indicate that the perception of gradient accentedness is influenced by measures of acoustic distance (i.e., magnitude of difference between a given production and an average native production) as well as properties of the lexical items themselves (e.g., neighborhood density and phonotactic probability). These ratings were then utilized to investigate how the gradient nature of accentedness influences lexical processing across behavioral, (visual word) eye-tracking, and electrophysiological methods. Additionally, it was also possible to examine how self-reported listener experience with Chinese-accented speakers influences the processing of gradient accentedness. These studies provide converging and complementary evidence that processing of these tokens varies non-linearly along the accentedness continuum and by level of listener experience. The electrophysiological results indicate that the pattern of processing, including the allocation of perceptual and attentional resources, changes as a result of increased exposure to Chinese-accented English. The effect of experience is also seen in both reaction time and visual world eye-tracking data. The results of a cross-modal priming study indicate that degree of foreign accent modulates the strength with which lexical representations are primed and that listener experience with the accent in question mitigates this effect. Visual world eye-tracking presents similar results, showing that the time-course of word recognition is slowed as accentedness increases, though the ability to decode the signal is enhanced for listeners with greater experience. Taken together, the results come to bear on our understanding of how gradient foreign-accented speech maps onto linguistic representations and how those representations may change and adapt over time to accommodate the variability inherent to foreign-accented speech.

Subjects/Keywords: Perception; Lexical processing; Foreign-accented speech

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Porretta, V. J. (2015). Perception and lexical processing of gradient foreign accentedness. (Doctoral Dissertation). University of Alberta. Retrieved from https://era.library.ualberta.ca/files/4f16c5678

Chicago Manual of Style (16^th Edition):

Porretta, Vincent J. “Perception and lexical processing of gradient foreign accentedness.” 2015. Doctoral Dissertation, University of Alberta. Accessed January 17, 2021. https://era.library.ualberta.ca/files/4f16c5678.

MLA Handbook (7^th Edition):

Porretta, Vincent J. “Perception and lexical processing of gradient foreign accentedness.” 2015. Web. 17 Jan 2021.

Vancouver:

Porretta VJ. Perception and lexical processing of gradient foreign accentedness. [Internet] [Doctoral dissertation]. University of Alberta; 2015. [cited 2021 Jan 17]. Available from: https://era.library.ualberta.ca/files/4f16c5678.

Council of Science Editors:

Porretta VJ. Perception and lexical processing of gradient foreign accentedness. [Doctoral Dissertation]. University of Alberta; 2015. Available from: https://era.library.ualberta.ca/files/4f16c5678

Georgia Tech

12. Li, Kehuang. Some new applications of phase information to speech processing.

Degree: PhD, Electrical and Computer Engineering, 2019, Georgia Tech

URL: http://hdl.handle.net/1853/62636

► With the fast growing of deep neural network models, more and more tasks have been boosted when move on to deep models. Speech processing applications,… (more)

Subjects/Keywords: Speech processing; Phase; Deep neural network

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Li, K. (2019). Some new applications of phase information to speech processing. (Doctoral Dissertation). Georgia Tech. Retrieved from http://hdl.handle.net/1853/62636

Chicago Manual of Style (16^th Edition):

Li, Kehuang. “Some new applications of phase information to speech processing.” 2019. Doctoral Dissertation, Georgia Tech. Accessed January 17, 2021. http://hdl.handle.net/1853/62636.

MLA Handbook (7^th Edition):

Li, Kehuang. “Some new applications of phase information to speech processing.” 2019. Web. 17 Jan 2021.

Vancouver:

Li K. Some new applications of phase information to speech processing. [Internet] [Doctoral dissertation]. Georgia Tech; 2019. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/1853/62636.

Council of Science Editors:

Li K. Some new applications of phase information to speech processing. [Doctoral Dissertation]. Georgia Tech; 2019. Available from: http://hdl.handle.net/1853/62636

Georgia Tech

13. Gimarc, Charles Edward. Application of an optimum quantizer algorithm to PCM and ADPCM speech coders.

Degree: MS, Electric engineering, 1980, Georgia Tech

URL: http://hdl.handle.net/1853/15738

Subjects/Keywords: Speech processing systems

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Gimarc, C. E. (1980). Application of an optimum quantizer algorithm to PCM and ADPCM speech coders. (Masters Thesis). Georgia Tech. Retrieved from http://hdl.handle.net/1853/15738

Chicago Manual of Style (16^th Edition):

Gimarc, Charles Edward. “Application of an optimum quantizer algorithm to PCM and ADPCM speech coders.” 1980. Masters Thesis, Georgia Tech. Accessed January 17, 2021. http://hdl.handle.net/1853/15738.

MLA Handbook (7^th Edition):

Gimarc, Charles Edward. “Application of an optimum quantizer algorithm to PCM and ADPCM speech coders.” 1980. Web. 17 Jan 2021.

Vancouver:

Gimarc CE. Application of an optimum quantizer algorithm to PCM and ADPCM speech coders. [Internet] [Masters thesis]. Georgia Tech; 1980. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/1853/15738.

Council of Science Editors:

Gimarc CE. Application of an optimum quantizer algorithm to PCM and ADPCM speech coders. [Masters Thesis]. Georgia Tech; 1980. Available from: http://hdl.handle.net/1853/15738

Oregon State University

14. Herr, Alan Jack. Pattern recognition of speech using hybrid computer feature extraction.

Degree: PhD, Electrical and Computer Engineering, 1974, Oregon State University

URL: http://hdl.handle.net/1957/45574

► Most technological problems associated with man-machine conversations have been solved and are well documented in both the contemporary technical and lay literature. The major remaining… (more)

Subjects/Keywords: Speech processing systems

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Herr, A. J. (1974). Pattern recognition of speech using hybrid computer feature extraction. (Doctoral Dissertation). Oregon State University. Retrieved from http://hdl.handle.net/1957/45574

Chicago Manual of Style (16^th Edition):

Herr, Alan Jack. “Pattern recognition of speech using hybrid computer feature extraction.” 1974. Doctoral Dissertation, Oregon State University. Accessed January 17, 2021. http://hdl.handle.net/1957/45574.

MLA Handbook (7^th Edition):

Herr, Alan Jack. “Pattern recognition of speech using hybrid computer feature extraction.” 1974. Web. 17 Jan 2021.

Vancouver:

Herr AJ. Pattern recognition of speech using hybrid computer feature extraction. [Internet] [Doctoral dissertation]. Oregon State University; 1974. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/1957/45574.

Council of Science Editors:

Herr AJ. Pattern recognition of speech using hybrid computer feature extraction. [Doctoral Dissertation]. Oregon State University; 1974. Available from: http://hdl.handle.net/1957/45574

University of Ottawa

15. Li, Dongfu. Deep Neural Network Approach for Single Channel Speech Enhancement Processing .

Degree: 2016, University of Ottawa

URL: http://hdl.handle.net/10393/34472

► Speech intelligibility represents how comprehensible a speech is. It is more important than speech quality in some applications. Single channel speech intelligibility enhancement is much… (more)

Subjects/Keywords: DNN; GMM; MRCG; Single-channel speech processing

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Li, D. (2016). Deep Neural Network Approach for Single Channel Speech Enhancement Processing . (Thesis). University of Ottawa. Retrieved from http://hdl.handle.net/10393/34472

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

Li, Dongfu. “Deep Neural Network Approach for Single Channel Speech Enhancement Processing .” 2016. Thesis, University of Ottawa. Accessed January 17, 2021. http://hdl.handle.net/10393/34472.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

Li, Dongfu. “Deep Neural Network Approach for Single Channel Speech Enhancement Processing .” 2016. Web. 17 Jan 2021.

Vancouver:

Li D. Deep Neural Network Approach for Single Channel Speech Enhancement Processing . [Internet] [Thesis]. University of Ottawa; 2016. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/10393/34472.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

Li D. Deep Neural Network Approach for Single Channel Speech Enhancement Processing . [Thesis]. University of Ottawa; 2016. Available from: http://hdl.handle.net/10393/34472

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

16. Kocagoncu, Ece. Dynamic speech networks in the brain: Dual contribution of incrementality and constraints in access to semantics.

Degree: PhD, 2017, University of Cambridge

URL: https://www.repository.cam.ac.uk/handle/1810/270309

► This thesis explores the spatiotemporal network dynamics underlying natural speech comprehension, as measured by electro-magnetoencephalography (E/MEG). I focus on the transient effects of incrementality and… (more)

▼ This thesis explores the spatiotemporal network dynamics underlying natural speech comprehension, as measured by electro-magnetoencephalography (E/MEG). I focus on the transient effects of incrementality and constraints in speech on access to lexical semantics. Through three E/MEG experiments I address two core issues in systems neuroscience of language: 1) What are the network dynamics underpinning cognitive computations that take place when we map sounds to rich semantic representations? 2) How do the prior semantic and syntactic contextual constraints facilitate this mapping? Experiment 1 investigated the cognitive processes and relevant networks that come online prior to a word’s recognition point (e.g. “f” for butterfly) as we access meaning through speech in isolation. The results revealed that 300 ms before the word is recognised, the speech incrementally activated matching phonological and semantic representations resulting in transient competition. This competition recruited LIFG, and modality specific regions (LSMG, LSTG for the phonological; LAG and MTG for the semantic domain). Immediately after the word’s recognition point the semantic representation of the target concept was boosted, and rapidly accessed recruiting bilateral MTG and AG. Experiment 2 explored the cortical networks underpinning contextual semantic processing in speech. Participant listened to two-word spoken phrases where the semantic constraint provided by the modifier was manipulated. To separate out cognitive networks that are modulated by semantic constraint from task positive networks I performed a temporal independent component analysis. Among 14 networks extracted, only the activity of bilateral AG was modulated by semantic constraint between -400 to -300 ms before the noun’s recognition point. Experiment 3 addressed the influence of sentential syntactic constraint on anticipation and activation of upcoming syntactic frames in speech. Participants listened to sentences with local syntactic ambiguities. The analysis of the connectivity dynamics in the left frontotemporal syntax network showed that the processing of sentences that contained the less anticipated syntactic structure showed early increased feedforward information flow in 0-100 ms, followed by increased recurrent connectivity between LIFG and LpMTG from the 200-500 ms from the verb onset. Altogether the three experiments reveal novel insights into transient cognitive networks recruited incrementally over time both in the absence of and with context, as the speech unfolds, and how the activation of these networks are modulated by contextual syntactic and semantic constraints. Further I provide neural evidence that contextual constraints serve to facilitate speech comprehension, and how the speech networks recover from failed anticipations.

Subjects/Keywords: Semantics; Constraint; MEG; network; speech; predictive processing

10 more images…

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Kocagoncu, E. (2017). Dynamic speech networks in the brain: Dual contribution of incrementality and constraints in access to semantics. (Doctoral Dissertation). University of Cambridge. Retrieved from https://www.repository.cam.ac.uk/handle/1810/270309

Chicago Manual of Style (16^th Edition):

Kocagoncu, Ece. “Dynamic speech networks in the brain: Dual contribution of incrementality and constraints in access to semantics.” 2017. Doctoral Dissertation, University of Cambridge. Accessed January 17, 2021. https://www.repository.cam.ac.uk/handle/1810/270309.

MLA Handbook (7^th Edition):

Kocagoncu, Ece. “Dynamic speech networks in the brain: Dual contribution of incrementality and constraints in access to semantics.” 2017. Web. 17 Jan 2021.

Vancouver:

Kocagoncu E. Dynamic speech networks in the brain: Dual contribution of incrementality and constraints in access to semantics. [Internet] [Doctoral dissertation]. University of Cambridge; 2017. [cited 2021 Jan 17]. Available from: https://www.repository.cam.ac.uk/handle/1810/270309.

Council of Science Editors:

Kocagoncu E. Dynamic speech networks in the brain: Dual contribution of incrementality and constraints in access to semantics. [Doctoral Dissertation]. University of Cambridge; 2017. Available from: https://www.repository.cam.ac.uk/handle/1810/270309

17. Al-Darkazali, Mohammed. Image processing methods to segment speech spectrograms for word level recognition.

Degree: PhD, 2017, University of Sussex

URL: http://sro.sussex.ac.uk/id/eprint/71675/ ; https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.731231

► The ultimate goal of automatic speech recognition (ASR) research is to allow a computer to recognize speech in real-time, with full accuracy, independent of vocabulary… (more)

▼ The ultimate goal of automatic speech recognition (ASR) research is to allow a computer to recognize speech in real-time, with full accuracy, independent of vocabulary size, noise, speaker characteristics or accent. Today, systems are trained to learn an individual speaker's voice and larger vocabularies statistically, but accuracy is not ideal. A small gap between actual speech and acoustic speech representation in the statistical mapping causes a failure to produce a match of the acoustic speech signals by Hidden Markov Model (HMM) methods and consequently leads to classification errors. Certainly, these errors in the low level recognition stage of ASR produce unavoidable errors at the higher levels. Therefore, it seems that ASR additional research ideas to be incorporated within current speech recognition systems. This study seeks new perspective on speech recognition. It incorporates a new approach for speech recognition, supporting it with wider previous research, validating it with a lexicon of 533 words and integrating it with a current speech recognition method to overcome the existing limitations. The study focusses on applying image processing to speech spectrogram images (SSI). We, thus develop a new writing system, which we call the Speech-Image Recogniser Code (SIR-CODE). The SIR-CODE refers to the transposition of the speech signal to an artificial domain (the SSI) that allows the classification of the speech signal into segments. The SIR-CODE allows the matching of all speech features (formants, power spectrum, duration, cues of articulation places, etc.) in one process. This was made possible by adding a Realization Layer (RL) on top of the traditional speech recognition layer (based on HMM) to check all sequential phones of a word in single step matching process. The study shows that the method gives better recognition results than HMMs alone, leading to accurate and reliable ASR in noisy environments. Therefore, the addition of the RL for SSI matching is a highly promising solution to compensate for the failure of HMMs in low level recognition. In addition, the same concept of employing SSIs can be used for whole sentences to reduce classification errors in HMM based high level recognition. The SIR-CODE bridges the gap between theory and practice of phoneme recognition by matching the SSI patterns at the word level. Thus, it can be adapted for dynamic time warping on the SIR-CODE segments, which can help to achieve ASR, based on SSI matching alone.

Subjects/Keywords: 620; TK7882.S65 Speech processing systems

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Al-Darkazali, M. (2017). Image processing methods to segment speech spectrograms for word level recognition. (Doctoral Dissertation). University of Sussex. Retrieved from http://sro.sussex.ac.uk/id/eprint/71675/ ; https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.731231

Chicago Manual of Style (16^th Edition):

Al-Darkazali, Mohammed. “Image processing methods to segment speech spectrograms for word level recognition.” 2017. Doctoral Dissertation, University of Sussex. Accessed January 17, 2021. http://sro.sussex.ac.uk/id/eprint/71675/ ; https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.731231.

MLA Handbook (7^th Edition):

Al-Darkazali, Mohammed. “Image processing methods to segment speech spectrograms for word level recognition.” 2017. Web. 17 Jan 2021.

Vancouver:

Al-Darkazali M. Image processing methods to segment speech spectrograms for word level recognition. [Internet] [Doctoral dissertation]. University of Sussex; 2017. [cited 2021 Jan 17]. Available from: http://sro.sussex.ac.uk/id/eprint/71675/ ; https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.731231.

Council of Science Editors:

Al-Darkazali M. Image processing methods to segment speech spectrograms for word level recognition. [Doctoral Dissertation]. University of Sussex; 2017. Available from: http://sro.sussex.ac.uk/id/eprint/71675/ ; https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.731231

Hong Kong University of Science and Technology

18. Ko, Yu Ting. Distinct acoustic modeling for automatic speech recognition.

Degree: 2014, Hong Kong University of Science and Technology

URL: http://repository.ust.hk/ir/Record/1783.1-63070 ; https://doi.org/10.14711/thesis-b1301519 ; http://repository.ust.hk/ir/bitstream/1783.1-63070/1/th_redirect.html

► In triphone-based acoustic modeling, it is difficult to robustly model infrequent triphones due to their lack of training samples. Naive maximum-likelihood (ML) estimation of infrequent… (more)

▼ In triphone-based acoustic modeling, it is difficult to robustly model infrequent triphones due to their lack of training samples. Naive maximum-likelihood (ML) estimation of infrequent triphone models produces poor triphone models and eventually affects the overall performance of an automatic speech recognition (ASR) system. Among different techniques proposed to solve the infrequent triphone problem, the most widely used method in current ASR systems is state tying because of its effectiveness in reducing model size and achieving good recognition results. However, state tying inevitably introduces quantization errors since triphones tied to the same state are not distinguishable in that state. This thesis addresses the problem by the use of distinct acoustic modeling where every modeling unit has a unique model and a distinct acoustic score. The main contribution of this thesis is the formulation of the estimation of triphone models as an adaptation problem through our proposed distinct acoustic modeling framework named eigentriphone modeling. The rational behind eigentriphone modeling is that a basis is derived from the frequent triphones and then each triphone is modeled as a point in the space spanned by the basis. The eigenvectors in the basis represent the most important context-dependent characteristics among the triphones and thus the infrequent triphones can be robustly modeled with few training samples. Furthermore, the proposed framework is very flexible and can be applied to other modeling units. Since grapheme-based modeling is useful in automatic speech recognition of under-resourced languages, we further apply our distinct acoustic modeling framework to estimate context-dependent grapheme models and we call our new method eigentrigrapheme modeling. Experimental evaluation of eigentriphone modeling was carried out on the Wall Street Journal word recognition task and the TIMIT phoneme recognition task. Experimental evaluation of eigentrigrapheme modeling was carried out on four official South African under-resourced languages. It is shown that distinct acoustic modeling using the proposed eigentriphone framework consistently performs better than the conventional tied-state HMMs.

Subjects/Keywords: Automatic speech recognition ; Phonetics, Acoustic ; Data processing

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Ko, Y. T. (2014). Distinct acoustic modeling for automatic speech recognition. (Thesis). Hong Kong University of Science and Technology. Retrieved from http://repository.ust.hk/ir/Record/1783.1-63070 ; https://doi.org/10.14711/thesis-b1301519 ; http://repository.ust.hk/ir/bitstream/1783.1-63070/1/th_redirect.html

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

Ko, Yu Ting. “Distinct acoustic modeling for automatic speech recognition.” 2014. Thesis, Hong Kong University of Science and Technology. Accessed January 17, 2021. http://repository.ust.hk/ir/Record/1783.1-63070 ; https://doi.org/10.14711/thesis-b1301519 ; http://repository.ust.hk/ir/bitstream/1783.1-63070/1/th_redirect.html.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

Ko, Yu Ting. “Distinct acoustic modeling for automatic speech recognition.” 2014. Web. 17 Jan 2021.

Vancouver:

Ko YT. Distinct acoustic modeling for automatic speech recognition. [Internet] [Thesis]. Hong Kong University of Science and Technology; 2014. [cited 2021 Jan 17]. Available from: http://repository.ust.hk/ir/Record/1783.1-63070 ; https://doi.org/10.14711/thesis-b1301519 ; http://repository.ust.hk/ir/bitstream/1783.1-63070/1/th_redirect.html.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

Ko YT. Distinct acoustic modeling for automatic speech recognition. [Thesis]. Hong Kong University of Science and Technology; 2014. Available from: http://repository.ust.hk/ir/Record/1783.1-63070 ; https://doi.org/10.14711/thesis-b1301519 ; http://repository.ust.hk/ir/bitstream/1783.1-63070/1/th_redirect.html

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

University of Notre Dame

19. Yuan Gong. Healthcare Applications and Security Concerns of Speech Processing Systems</h1>.

Degree: Computer Science and Engineering, 2020, University of Notre Dame

URL: https://curate.nd.edu/show/8c97kp81j7k

► Conventionally, most research on speech processing focuses on automatic speech recognition (ASR), i.e., transcribing speech to text. However, natural speech does not only contain… (more)

Subjects/Keywords: machine learning; security; speech processing; healthcare

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Gong, Y. (2020). Healthcare Applications and Security Concerns of Speech Processing Systems</h1>. (Thesis). University of Notre Dame. Retrieved from https://curate.nd.edu/show/8c97kp81j7k

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

Gong, Yuan. “Healthcare Applications and Security Concerns of Speech Processing Systems</h1>.” 2020. Thesis, University of Notre Dame. Accessed January 17, 2021. https://curate.nd.edu/show/8c97kp81j7k.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

Gong, Yuan. “Healthcare Applications and Security Concerns of Speech Processing Systems</h1>.” 2020. Web. 17 Jan 2021.

Vancouver:

Gong Y. Healthcare Applications and Security Concerns of Speech Processing Systems</h1>. [Internet] [Thesis]. University of Notre Dame; 2020. [cited 2021 Jan 17]. Available from: https://curate.nd.edu/show/8c97kp81j7k.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

Gong Y. Healthcare Applications and Security Concerns of Speech Processing Systems</h1>. [Thesis]. University of Notre Dame; 2020. Available from: https://curate.nd.edu/show/8c97kp81j7k

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Arizona State University

20. Berg, Dena Beth. Manifestation of Higher-order Cognitive Processing Deficits Resulting from Concussion.

Degree: MS, Speech and Hearing Science, 2013, Arizona State University

URL: http://repository.asu.edu/items/17846

► Concussion, a subset of mild traumatic brain injury (mTBI), has recently been brought to the forefront of the media due to a large lawsuit filed… (more)

Subjects/Keywords: Speech therapy; Mental health; cognitive processing; concussion

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Berg, D. B. (2013). Manifestation of Higher-order Cognitive Processing Deficits Resulting from Concussion. (Masters Thesis). Arizona State University. Retrieved from http://repository.asu.edu/items/17846

Chicago Manual of Style (16^th Edition):

Berg, Dena Beth. “Manifestation of Higher-order Cognitive Processing Deficits Resulting from Concussion.” 2013. Masters Thesis, Arizona State University. Accessed January 17, 2021. http://repository.asu.edu/items/17846.

MLA Handbook (7^th Edition):

Berg, Dena Beth. “Manifestation of Higher-order Cognitive Processing Deficits Resulting from Concussion.” 2013. Web. 17 Jan 2021.

Vancouver:

Berg DB. Manifestation of Higher-order Cognitive Processing Deficits Resulting from Concussion. [Internet] [Masters thesis]. Arizona State University; 2013. [cited 2021 Jan 17]. Available from: http://repository.asu.edu/items/17846.

Council of Science Editors:

Berg DB. Manifestation of Higher-order Cognitive Processing Deficits Resulting from Concussion. [Masters Thesis]. Arizona State University; 2013. Available from: http://repository.asu.edu/items/17846

21. ELIAS, CHRISTY. Analysis of Speech Parameters as Indicators of Engagement in Conversation.

Degree: School of Computer Science & Statistics. Discipline of Computer Science, 2020, Trinity College Dublin

URL: http://hdl.handle.net/2262/94132

► In human conversations, the words being said may not always contain the whole of the information intended to be conveyed; instead, such information may be… (more)

Subjects/Keywords: Speech Processing; Acoustic-Prosodic Features; Engagement; Conversation

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

ELIAS, C. (2020). Analysis of Speech Parameters as Indicators of Engagement in Conversation. (Thesis). Trinity College Dublin. Retrieved from http://hdl.handle.net/2262/94132

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

ELIAS, CHRISTY. “Analysis of Speech Parameters as Indicators of Engagement in Conversation.” 2020. Thesis, Trinity College Dublin. Accessed January 17, 2021. http://hdl.handle.net/2262/94132.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

ELIAS, CHRISTY. “Analysis of Speech Parameters as Indicators of Engagement in Conversation.” 2020. Web. 17 Jan 2021.

Vancouver:

ELIAS C. Analysis of Speech Parameters as Indicators of Engagement in Conversation. [Internet] [Thesis]. Trinity College Dublin; 2020. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/2262/94132.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

ELIAS C. Analysis of Speech Parameters as Indicators of Engagement in Conversation. [Thesis]. Trinity College Dublin; 2020. Available from: http://hdl.handle.net/2262/94132

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

University of Illinois – Urbana-Champaign

22. Jacobs, Cassandra L. Knowing a thing is "a thing": The use of acoustic features in multiword expression extraction.

Degree: MS, Computer Science, 2016, University of Illinois – Urbana-Champaign

URL: http://hdl.handle.net/2142/92965

► Speakers of a language need to have complex linguistic representations for speaking, often on the level of non-literal, idiomatic expressions like black sheep. Typically, datasets… (more)

Subjects/Keywords: Collocations; Phrases; Speech processing; Language models

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Jacobs, C. L. (2016). Knowing a thing is "a thing": The use of acoustic features in multiword expression extraction. (Thesis). University of Illinois – Urbana-Champaign. Retrieved from http://hdl.handle.net/2142/92965

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

Jacobs, Cassandra L. “Knowing a thing is "a thing": The use of acoustic features in multiword expression extraction.” 2016. Thesis, University of Illinois – Urbana-Champaign. Accessed January 17, 2021. http://hdl.handle.net/2142/92965.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

Jacobs, Cassandra L. “Knowing a thing is "a thing": The use of acoustic features in multiword expression extraction.” 2016. Web. 17 Jan 2021.

Vancouver:

Jacobs CL. Knowing a thing is "a thing": The use of acoustic features in multiword expression extraction. [Internet] [Thesis]. University of Illinois – Urbana-Champaign; 2016. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/2142/92965.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

Jacobs CL. Knowing a thing is "a thing": The use of acoustic features in multiword expression extraction. [Thesis]. University of Illinois – Urbana-Champaign; 2016. Available from: http://hdl.handle.net/2142/92965

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Georgia Tech

23. Marr, James Douglas. Two dimensional prediction for data rate compression of LPC parameters.

Degree: PhD, Electric engineering, 1980, Georgia Tech

URL: http://hdl.handle.net/1853/32798

Subjects/Keywords: Speech processing systems

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Marr, J. D. (1980). Two dimensional prediction for data rate compression of LPC parameters. (Doctoral Dissertation). Georgia Tech. Retrieved from http://hdl.handle.net/1853/32798

Chicago Manual of Style (16^th Edition):

Marr, James Douglas. “Two dimensional prediction for data rate compression of LPC parameters.” 1980. Doctoral Dissertation, Georgia Tech. Accessed January 17, 2021. http://hdl.handle.net/1853/32798.

MLA Handbook (7^th Edition):

Marr, James Douglas. “Two dimensional prediction for data rate compression of LPC parameters.” 1980. Web. 17 Jan 2021.

Vancouver:

Marr JD. Two dimensional prediction for data rate compression of LPC parameters. [Internet] [Doctoral dissertation]. Georgia Tech; 1980. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/1853/32798.

Council of Science Editors:

Marr JD. Two dimensional prediction for data rate compression of LPC parameters. [Doctoral Dissertation]. Georgia Tech; 1980. Available from: http://hdl.handle.net/1853/32798

McGill University

24. Pinnell, Richard James. Adaptive Transform Coding of Speech Signals.

Degree: M. Eng., Department of Electrical Engineering, 1982, McGill University

URL: https://escholarship.mcgill.ca/downloads/qj72p970t.pdf ; https://escholarship.mcgill.ca/concern/theses/mc87ps98r

►

Les techniques de codage dans le domaine fréquentiel ont récemment fait l'objet d'une attention considérable. Le codage de transformées par adaptation y occupe une place… (more)

Subjects/Keywords: Speech processing systems.

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Pinnell, R. J. (1982). Adaptive Transform Coding of Speech Signals. (Masters Thesis). McGill University. Retrieved from https://escholarship.mcgill.ca/downloads/qj72p970t.pdf ; https://escholarship.mcgill.ca/concern/theses/mc87ps98r

Chicago Manual of Style (16^th Edition):

Pinnell, Richard James. “Adaptive Transform Coding of Speech Signals.” 1982. Masters Thesis, McGill University. Accessed January 17, 2021. https://escholarship.mcgill.ca/downloads/qj72p970t.pdf ; https://escholarship.mcgill.ca/concern/theses/mc87ps98r.

MLA Handbook (7^th Edition):

Pinnell, Richard James. “Adaptive Transform Coding of Speech Signals.” 1982. Web. 17 Jan 2021.

Vancouver:

Pinnell RJ. Adaptive Transform Coding of Speech Signals. [Internet] [Masters thesis]. McGill University; 1982. [cited 2021 Jan 17]. Available from: https://escholarship.mcgill.ca/downloads/qj72p970t.pdf ; https://escholarship.mcgill.ca/concern/theses/mc87ps98r.

Council of Science Editors:

Pinnell RJ. Adaptive Transform Coding of Speech Signals. [Masters Thesis]. McGill University; 1982. Available from: https://escholarship.mcgill.ca/downloads/qj72p970t.pdf ; https://escholarship.mcgill.ca/concern/theses/mc87ps98r

Kansas State University

25. Solanki, Niranjan U. Digital speech processing techniques.

Degree: 1974, Kansas State University

URL: http://hdl.handle.net/2097/8352

Subjects/Keywords: Speech processing systems

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Solanki, N. U. (1974). Digital speech processing techniques. (Thesis). Kansas State University. Retrieved from http://hdl.handle.net/2097/8352

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

Solanki, Niranjan U. “Digital speech processing techniques.” 1974. Thesis, Kansas State University. Accessed January 17, 2021. http://hdl.handle.net/2097/8352.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

Solanki, Niranjan U. “Digital speech processing techniques.” 1974. Web. 17 Jan 2021.

Vancouver:

Solanki NU. Digital speech processing techniques. [Internet] [Thesis]. Kansas State University; 1974. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/2097/8352.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

Solanki NU. Digital speech processing techniques. [Thesis]. Kansas State University; 1974. Available from: http://hdl.handle.net/2097/8352

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Rutgers University

26. Hou, Jun, 1974-. On the use of frame and segment-based methods for the detection and classification of speech sounds and features:.

Degree: PhD, Electrical and Computer Engineering, 2009, Rutgers University

URL: http://hdl.rutgers.edu/1782.2/rucore10001600001.ETD.000051833

►

Statistical data-driven methods and knowledge-based methods are two recent trends in Automatic Speech Recognition (ASR). Hidden Markov Model (HMM)-based speech recognition techniques have achieved great… (more)

▼

Statistical data-driven methods and knowledge-based methods are two recent trends in Automatic Speech Recognition (ASR). Hidden Markov Model (HMM)-based speech recognition techniques have achieved great success for controlled tasks and environments. However, when we require improved accuracy and robustness (closer to Human Speech Recognition (HSR)), HMM algorithms for speech recognition gradually fail. Hence a need has emerged to incorporate higher level linguistic information into ASR systems in order to further discriminate between speech classes or phonemes with high confusion rates. The Automatic Speech Attribute Transcription (ASAT) project is one of the recent research efforts that has tried to bridge the gap between ASR and HSR. In this thesis we focus on the design and optimization of the front end processing of the ASAT system, whose goal is to estimate a set of attribute and phoneme probability lattices which can be combined with information from higher level knowledge sources in a set of speech event verification modules in order to make a final recognition decision. We propose a set of both frame-based methods and segment-based methods to improve the recognition performance of distinctive features and phonemes in English. We also study and evaluate both a parallel speech feature organization and a hierarchical phoneme topology. There are 4 main parts in this thesis work. In the first part, we use frame-based methods to estimate the likelihood of static sounds (e.g., steady vowels, fricatives, etc), and implement the parallel feature detection using Multi-Layer Perceptrons (MLPs) in order to detect the 14 Sound Pattern of English (SPE) features. In the second part, we use segment-based methods to classify dynamic sounds (e.g., stop consonants, diphthongs, etc), and use Time-Delay Neural Networks (TDNNs) to recognize phoneme classes in a hierarchical phoneme and feature organization. In the third part and in the forth part, we combine the frame-based parallel speech feature detection system and the segment-based hierarchical phoneme classification system to improve the overall phoneme classification performance and the speech feature detection performance. The main contribution of this thesis is the creation of a phoneme recognizer that overcomes the disadvantages of pure statistical or knowledge-based systems, and provides a way to incorporate acoustic/phonetic/linguistic knowledge into an existing (HMM-based) automatic speech recognition system.

Advisors/Committee Members: Hou, Jun, 1974- (author), Rabiner, Lawrence (chair), Marsic, Ivan (internal member), Wilder, Joseph (internal member), Rosenberg, Aaron (outside member), Lee, Chin-Hui (outside member).

Subjects/Keywords: Speech processing systems

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Hou, Jun, 1. (2009). On the use of frame and segment-based methods for the detection and classification of speech sounds and features:. (Doctoral Dissertation). Rutgers University. Retrieved from http://hdl.rutgers.edu/1782.2/rucore10001600001.ETD.000051833

Chicago Manual of Style (16^th Edition):

Hou, Jun, 1974-. “On the use of frame and segment-based methods for the detection and classification of speech sounds and features:.” 2009. Doctoral Dissertation, Rutgers University. Accessed January 17, 2021. http://hdl.rutgers.edu/1782.2/rucore10001600001.ETD.000051833.

MLA Handbook (7^th Edition):

Hou, Jun, 1974-. “On the use of frame and segment-based methods for the detection and classification of speech sounds and features:.” 2009. Web. 17 Jan 2021.

Vancouver:

Hou, Jun 1. On the use of frame and segment-based methods for the detection and classification of speech sounds and features:. [Internet] [Doctoral dissertation]. Rutgers University; 2009. [cited 2021 Jan 17]. Available from: http://hdl.rutgers.edu/1782.2/rucore10001600001.ETD.000051833.

Council of Science Editors:

Hou, Jun 1. On the use of frame and segment-based methods for the detection and classification of speech sounds and features:. [Doctoral Dissertation]. Rutgers University; 2009. Available from: http://hdl.rutgers.edu/1782.2/rucore10001600001.ETD.000051833

University of New South Wales

27. Dang, Ting. Speech based Continuous Emotion Prediction: An investigation of Speaker Variability and Emotion Uncertainty.

Degree: Electrical Engineering & Telecommunications, 2018, University of New South Wales

URL: http://handle.unsw.edu.au/1959.4/60161 ; https://unsworks.unsw.edu.au/fapi/datastream/unsworks:51221/SOURCE02?view=true

► Understanding and describing human emotional state is important for many applications such as interactive human-computer interface design and clinical diagnosis tools. Speech based emotion prediction… (more)

▼ Understanding and describing human emotional state is important for many applications such as interactive human-computer interface design and clinical diagnosis tools. Speech based emotion prediction is generally viewed as a regression problem, where speech waveforms are labelled in terms of affective attributes such as arousal and valence, with numerical values indicating the short-term emotion intensity. Current research on continuous emotion prediction has primarily focused on improving the backend, developing novel features or improving feature selection techniques. However, emotion expressions or perceptions are in general heterogeneous across individuals, depending on a wide range of factors, such as cultural background and speaker’s gender. The impact of these sources of variations on the continuous emotion prediction systems has not been fully explored yet and is the focus of this thesis. Speaker variability, i.e., differences in emotion expression among speakers, has been shown to be one of the most confounding factors in categorical emotion recognition system, but there is limited literature that analyses the effect on continuous emotion prediction systems. In this thesis, a probabilistic framework is proposed to quantify speaker variability in continuous emotion systems in both the feature and the model domains. Furthermore, three compensation techniques for speaker variability are developed and in-depth analyses in both the feature and model spaces are carried out. Another confounding factor is the inter-rater variability, i.e., difference in emotion perception among raters, which is ignored in current approaches by taking the average rating across multiple raters as the ‘true’ representation of the emotion states. However, differences in perception among raters suggest that prediction certainty varies with time. A novel approach for the prediction of emotion uncertainty is proposed and implemented by including the inter-rater variability as a representation of the uncertainty information in a probabilistic model. In addition, Kalman filters are incorporated into this framework to take into account the temporal dependencies of the emotion uncertainty, as well as providing the flexibility to relax the Gaussianity assumption on the emotion distribution that reflects the uncertainty. The proposed frameworks and methods have been extensively evaluated on multiple state-of-the-art databases and the results have demonstrated the potential of the proposed solutions. Advisors/Committee Members: Sethu, Vidhyasaharan, Electrical Engineering & Telecommunications, Faculty of Engineering, UNSW, Ambikairaja, Eliathamby, Electrical Engineering & Telecommunications, Faculty of Engineering, UNSW.

Subjects/Keywords: pattern recognition; speech processing; machine learning

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Dang, T. (2018). Speech based Continuous Emotion Prediction: An investigation of Speaker Variability and Emotion Uncertainty. (Doctoral Dissertation). University of New South Wales. Retrieved from http://handle.unsw.edu.au/1959.4/60161 ; https://unsworks.unsw.edu.au/fapi/datastream/unsworks:51221/SOURCE02?view=true

Chicago Manual of Style (16^th Edition):

Dang, Ting. “Speech based Continuous Emotion Prediction: An investigation of Speaker Variability and Emotion Uncertainty.” 2018. Doctoral Dissertation, University of New South Wales. Accessed January 17, 2021. http://handle.unsw.edu.au/1959.4/60161 ; https://unsworks.unsw.edu.au/fapi/datastream/unsworks:51221/SOURCE02?view=true.

MLA Handbook (7^th Edition):

Dang, Ting. “Speech based Continuous Emotion Prediction: An investigation of Speaker Variability and Emotion Uncertainty.” 2018. Web. 17 Jan 2021.

Vancouver:

Dang T. Speech based Continuous Emotion Prediction: An investigation of Speaker Variability and Emotion Uncertainty. [Internet] [Doctoral dissertation]. University of New South Wales; 2018. [cited 2021 Jan 17]. Available from: http://handle.unsw.edu.au/1959.4/60161 ; https://unsworks.unsw.edu.au/fapi/datastream/unsworks:51221/SOURCE02?view=true.

Council of Science Editors:

Dang T. Speech based Continuous Emotion Prediction: An investigation of Speaker Variability and Emotion Uncertainty. [Doctoral Dissertation]. University of New South Wales; 2018. Available from: http://handle.unsw.edu.au/1959.4/60161 ; https://unsworks.unsw.edu.au/fapi/datastream/unsworks:51221/SOURCE02?view=true

University of British Columbia

28. Wickwire, Kenneth Freeman. Design and testing of time-varying inductors and capcitors for an electrical speech synthesizer.

Degree: Master of Applied Science - MASc, Electrical and Computer Engineering, 1967, University of British Columbia

URL: http://hdl.handle.net/2429/36155

► This thesis describes the design and testing of time-varying inductors and capacitors for use in an electrical analogue of the human vocal tract. The inductors… (more)

Subjects/Keywords: Speech processing systems

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Wickwire, K. F. (1967). Design and testing of time-varying inductors and capcitors for an electrical speech synthesizer. (Masters Thesis). University of British Columbia. Retrieved from http://hdl.handle.net/2429/36155

Chicago Manual of Style (16^th Edition):

Wickwire, Kenneth Freeman. “Design and testing of time-varying inductors and capcitors for an electrical speech synthesizer.” 1967. Masters Thesis, University of British Columbia. Accessed January 17, 2021. http://hdl.handle.net/2429/36155.

MLA Handbook (7^th Edition):

Wickwire, Kenneth Freeman. “Design and testing of time-varying inductors and capcitors for an electrical speech synthesizer.” 1967. Web. 17 Jan 2021.

Vancouver:

Wickwire KF. Design and testing of time-varying inductors and capcitors for an electrical speech synthesizer. [Internet] [Masters thesis]. University of British Columbia; 1967. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/2429/36155.

Council of Science Editors:

Wickwire KF. Design and testing of time-varying inductors and capcitors for an electrical speech synthesizer. [Masters Thesis]. University of British Columbia; 1967. Available from: http://hdl.handle.net/2429/36155

Simon Fraser University

29. Lupini, Peter. Harmonic coding of speech at low bit rates.

Degree: 1995, Simon Fraser University

URL: http://summit.sfu.ca/item/6814

Subjects/Keywords: Speech processing systems.

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Lupini, P. (1995). Harmonic coding of speech at low bit rates. (Thesis). Simon Fraser University. Retrieved from http://summit.sfu.ca/item/6814

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

Lupini, Peter. “Harmonic coding of speech at low bit rates.” 1995. Thesis, Simon Fraser University. Accessed January 17, 2021. http://summit.sfu.ca/item/6814.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

Lupini, Peter. “Harmonic coding of speech at low bit rates.” 1995. Web. 17 Jan 2021.

Vancouver:

Lupini P. Harmonic coding of speech at low bit rates. [Internet] [Thesis]. Simon Fraser University; 1995. [cited 2021 Jan 17]. Available from: http://summit.sfu.ca/item/6814.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

Lupini P. Harmonic coding of speech at low bit rates. [Thesis]. Simon Fraser University; 1995. Available from: http://summit.sfu.ca/item/6814

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

30. Thibeault, Corey M. Applications in Neurorobotics.

Degree: 2012, University of Nevada – Reno

URL: http://hdl.handle.net/11714/3765

► The field of neurorobotics is still in its infancy; however, its intersecting motivations are not. On the one hand, theories of neuroscience that require immersion… (more)

Subjects/Keywords: Embodied Modeling; Emotional Speech Processing; Neurorobotics

Record Details Similar Records Cite « Share

❌

APA · Chicago · MLA · Vancouver · CSE | Export to Zotero / EndNote / Reference Manager

APA (6^th Edition):

Thibeault, C. M. (2012). Applications in Neurorobotics. (Thesis). University of Nevada – Reno. Retrieved from http://hdl.handle.net/11714/3765

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Chicago Manual of Style (16^th Edition):

Thibeault, Corey M. “Applications in Neurorobotics.” 2012. Thesis, University of Nevada – Reno. Accessed January 17, 2021. http://hdl.handle.net/11714/3765.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

MLA Handbook (7^th Edition):

Thibeault, Corey M. “Applications in Neurorobotics.” 2012. Web. 17 Jan 2021.

Vancouver:

Thibeault CM. Applications in Neurorobotics. [Internet] [Thesis]. University of Nevada – Reno; 2012. [cited 2021 Jan 17]. Available from: http://hdl.handle.net/11714/3765.

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

Council of Science Editors:

Thibeault CM. Applications in Neurorobotics. [Thesis]. University of Nevada – Reno; 2012. Available from: http://hdl.handle.net/11714/3765

Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation

		in
And / Or
		in
And / Or
		in
And / Or
		in

Open Access Theses and Dissertations