Thèses : « Speech transformation »

1

Kain, Alexander Blouke. "High resolution voice transformation /." Full text open access at:, 2001. http://content.ohsu.edu/u?/etd,189.

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Mandal, Arindam. "Transformation sharing strategies for MLLR speaker adaptation /." Thesis, Connect to this title online; UW restricted, 2007. http://hdl.handle.net/1773/6087.

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Crosmer, Joel R. "Very low bit rate speech coding using the line spectrum pair transformation of the LPC coefficients." Diss., Georgia Institute of Technology, 1985. http://hdl.handle.net/1853/15739.

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4

Salor, Ozgul. "Voice Transformation And Development Of Related Speech Analysis Tools For Turkish." Phd thesis, METU, 2005. http://etd.lib.metu.edu.tr/upload/3/12605787/index.pdf.

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In this dissertation, new approaches in the design of a voice transformation (VT) system for Turkish are proposed. Objectives in this thesis are two-fold. The first objective is to develop standard speech corpora and segmentation tools for Turkish speech research. The second objective is to consider new approaches for VT. A triphone-balanced set of 2462 Turkish sentences is prepared for analysis. An audio corpus of 100 speakers, each uttering 40 sentences out of the 2462-sentence set, is used to train a speech recognition system designed for English. This system is ported to Turkish to obtain a phonetic aligner and a phoneme recognizer. The triphone-balanced sentence set and the phonetic aligner are used to develop a speech corpus for VT. A new voice transformation approach based on Mixed Excitation Linear Prediction (MELP) speech coding framework is proposed. Multi-stage vector quantization of MELP is used to obtain speaker-specific line-spectral frequency (LSF) codebooks for source and target speakers. Histograms mapping the LSF spaces of source and target speakers are used for transformation in the baseline system. The baseline system is improved by a dynamic programming approach to estimate the target LSFs. As a second approach to the VT problem, quantizing the LSFs using k-means clustering algorithm is applied with dimension reduction of LSFs using principle component analysis. This approach provides speaker-specific codebooks out of the speech corpus instead of using MELP&#039 s pre-trained LSF codebook. Evaluations show that both dimension reduction and dynamic programming improve the transformation performance.

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Raghunathan, Anusha. "EVALUATION OF INTELLIGIBILITY AND SPEAKER SIMILARITY OF VOICE TRANSFORMATION." UKnowledge, 2011. http://uknowledge.uky.edu/gradschool_theses/101.

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Voice transformation refers to a class of techniques that modify the voice characteristics either to conceal the identity or to mimic the voice characteristics of another speaker. Its applications include automatic dialogue replacement and voice generation for people with voice disorders. The diversity in applications makes evaluation of voice transformation a challenging task. The objective of this research is to propose a framework to evaluate intentional voice transformation techniques. Our proposed framework is based on two fundamental qualities: intelligibility and speaker similarity. Intelligibility refers to the clarity of the speech content after voice transformation and speaker similarity measures how well the modified output disguises the source speaker. We measure intelligibility with word error rates and speaker similarity with likelihood of identifying the correct speaker. The novelty of our approach is, we consider whether similarly transformed training data are available to the recognizer. We have demonstrated that this factor plays a significant role in intelligibility and speaker similarity for both human testers and automated recognizers. We thoroughly test two classes of voice transformation techniques: pitch distortion and voice conversion, using our proposed framework. We apply our results for patients with voice hypertension using video self-modeling and preliminary results are presented.

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Joly, Yvan. "A digital speech transformation system for developing aids for the hearing impaired /." Thesis, McGill University, 1991. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=60047.

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This thesis presents the design of a digital speech transformation system for developing aids for the hearing impaired. The system is required to perform, in real time, four basic speech transformations: time-scale modification, frequency-scale modification, pitch modification, and spectral envelope modification. It is based on the sinusoidal representation of Quartieri and McAulay, but achieves the required deconvolution of the speech signal by liner prediction, rather than homomorphic filtering. The system is implemented in software, and a detailed study is done to determine the type of hardware implementation required to perform the speech transformations in real time. The proposed implementation is a multiprocessor system based on the TMS320C25 digital signal processor. Results obtained on an emulator for the TMS320C25 demonstrate the ability of the system to analyze and synthesize the speech signal in real time, while results obtained by simulation demonstrate its transformation capabilities.

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Stachurski, Marcin. "The verbal transformation effect : an exploration of the perceptual organization of speech." Thesis, Aston University, 2013. http://publications.aston.ac.uk/18941/.

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Six experiments investigated the influence of several grouping cues within the framework of the Verbal Transformation Effect (VTE, Experiments 1 to 4) and Phonemic Transformation Effect (PTE, Experiments 5 and 6), where listening to a repeated word (VTE) or sequence of vowels (PTE) produces verbal transformations (VTs). In Experiment 1, the influence of F0 frequency and lateralization cues (ITDs) was investigated in terms of the pattern of VTs. As the lateralization difference increased between two repeating sequences, the number of forms was significantly reduced with the fewest forms reported in the dichotic condition. Experiment 2 explored whether or not propensity to report more VTs on high pitch was due to the task demands of monitoring two sequences at once. The number of VTs reported was higher when listeners were asked to attend to one sequence only, suggesting smaller attentional constraints on the task requirements. In Experiment 3, consonant-vowel transitions were edited out from two sets of six stimuli words with ‘strong’ and ‘weak’ formant transitions, respectively. Listeners reported more forms in the spliced-out than in the unedited case for the strong-transition words, but not for those with weak transitions. A similar trend was observed for the F0 contour manipulation used in Experiment 4 where listeners reported more VTs and forms for words following a discontinuous F0 contour. In Experiments 5 and 6, the role of F0 frequency and ITD cues was investigated further using a related phenomenon – the PTE. Although these manipulations had relatively little effect on the number of VTs and forms reported, they did influence the particular forms heard. In summary, the current experiments confirmed that it is possible to successfully investigate auditory grouping cues within the VTE framework and that, in agreement with recent studies, the results can be attributed to the perceptual re-grouping of speech sounds.

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Li, Jinyu. "Soft margin estimation for automatic speech recognition." Diss., Atlanta, Ga. : Georgia Institute of Technology, 2008. http://hdl.handle.net/1853/26613.

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Thesis (Ph.D)--Electrical and Computer Engineering, Georgia Institute of Technology, 2009. Committee Chair: Dr. Chin-Hui Lee; Committee Member: Dr. Anthony Joseph Yezzi; Committee Member: Dr. Biing-Hwang (Fred) Juang; Committee Member: Dr. Mark Clements; Committee Member: Dr. Ming Yuan. Part of the SMARTech Electronic Thesis and Dissertation Collection.

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Alexander, Gutkin. "Towards formal structural representation of spoken language : an evolving transformation system (ETS) approach." Thesis, University of Edinburgh, 2006. http://hdl.handle.net/1842/3527.

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Speech recognition has been a very active area of research over the past twenty years. Despite an evident progress, it is generally agreed by the practitioners of the field that performance of the current speech recognition systems is rather suboptimal and new approaches are needed. The motivation behind the undertaken research is an observation that the notion of representation of objects and concepts that once was considered to be central in the early days of pattern recognition, has been largely marginalised by the advent of statistical approaches. As a consequence of a predominantly statistical approach to speech recognition problem, due to the numeric, feature vector-based, nature of representation, the classes inductively discovered from real data using decision-theoretic techniques have little meaning outside the statistical framework. This is because decision surfaces or probability distributions are difficult to analyse linguistically. Because of the later limitation it is doubtful that the gap between speech recognition and linguistic research can be bridged by the numeric representations. This thesis investigates an alternative, structural, approach to spoken language representation and categorisation. The approach pursued in this thesis is based on a consistent program, known as the Evolving Transformation System (ETS), motivated by the development and clarification of the concept of structural representation in pattern recognition and artificial intelligence from both theoretical and applied points of view. This thesis consists of two parts. In the first part of this thesis, a similarity-based approach to structural representation of speech is presented. First, a linguistically well-motivated structural representation of phones based on distinctive phonological features recovered from speech is proposed. The representation consists of string templates representing phones together with a similarity measure. The set of phonological templates together with a similarity measure defines a symbolic metric space. Representation and ETS-inspired categorisation in the symbolic metric spaces corresponding to the phonological structural representation are then investigated by constructing appropriate symbolic space classifiers and evaluating them on a standard corpus of read speech. In addition, similarity-based isometric transition from phonological symbolic metric spaces to the corresponding non-Euclidean vector spaces is investigated. Second part of this thesis deals with the formal approach to structural representation of spoken language. Unlike the approach adopted in the first part of this thesis, the representation developed in the second part is based on the mathematical language of the ETS formalism. This formalism has been specifically developed for structural modelling of dynamic processes. In particular, it allows the representation of both objects and classes in a uniform event-based hierarchical framework. In this thesis, the latter property of the formalism allows the adoption of a more physiologically-concreteapproach to structural representation. The proposed representation is based on gestural structures and encapsulates speech processes at the articulatory level. Algorithms for deriving the articulatory structures from the data are presented and evaluated.

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TAKEDA, Kazuya, Norihide KITAOKA, and Makoto SAKAI. "Acoustic Feature Transformation Combining Average and Maximum Classification Error Minimization Criteria." Institute of Electronics, Information and Communication Engineers, 2010. http://hdl.handle.net/2237/14970.

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TAKEDA, Kazuya, Norihide KITAOKA, and Makoto SAKAI. "Acoustic Feature Transformation Based on Discriminant Analysis Preserving Local Structure for Speech Recognition." Institute of Electronics, Information and Communication Engineers, 2010. http://hdl.handle.net/2237/14969.

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Worvill, Romira. "'Seeing' speech : illusion and the transformation of dramatic writing in Diderot and Lessing." Thesis, University of Oxford, 2002. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.270164.

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Calzada, Defez Àngel. "Conveying expressivity and vocal effort transformation in synthetic speech with Harmonic plus Noise Models." Doctoral thesis, Universitat Ramon Llull, 2016. http://hdl.handle.net/10803/360587.

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Aquesta tesi s'ha dut a terme dins del Grup en de Tecnologies Mèdia (GTM) de l'Escola d'Enginyeria i Arquitectura la Salle. El grup te una llarga trajectòria dins del cap de la síntesi de veu i fins i tot disposa d'un sistema propi de síntesi per concatenació d'unitats (US-TTS) que permet sintetitzar diferents estils expressius usant múltiples corpus. De forma que per a realitzar una síntesi agressiva, el sistema usa el corpus de l'estil agressiu, i per a realitzar una síntesi sensual, usa el corpus de l'estil corresponent. Aquesta tesi pretén proposar modificacions del esquema del US-TTS que permetin millorar la flexibilitat del sistema per sintetitzar múltiples expressivitats usant només un únic corpus d'estil neutre. L'enfoc seguit en aquesta tesi es basa en l'ús de tècniques de processament digital del senyal (DSP) per aplicar modificacions de senyal a la veu sintetitzada per tal que aquesta expressi l'estil de parla desitjat. Per tal de dur a terme aquestes modificacions de senyal s'han usat els models harmònic més soroll per la seva flexibilitat a l'hora de realitzar modificacions de senyal. La qualitat de la veu (VoQ) juga un paper important en els diferents estils expressius. És per això que es va estudiar la síntesi de diferents emocions mitjançant la modificació de paràmetres de VoQ de baix nivell. D'aquest estudi es van identificar un conjunt de limitacions que van donar lloc als objectius d'aquesta tesi, entre ells el trobar un paràmetre amb gran impacte sobre els estils expressius. Per aquest fet l'esforç vocal (VE) es va escollir per el seu paper important en la parla expressiva. Primer es va estudiar la possibilitat de transferir l'VE entre dues realitzacions amb diferent VE de la mateixa paraula basant-se en la tècnica de predicció lineal adaptativa del filtre de pre-èmfasi (APLP). La proposta va permetre transferir l'VE correctament però presentava limitacions per a poder generar nivells intermitjos d'VE. Amb la finalitat de millorar la flexibilitat i control de l'VE expressat a la veu sintetitzada, es va proposar un nou model d'VE basat en polinomis lineals. Aquesta proposta va permetre transferir l'VE entre dues paraules qualsevols i sintetitzar nous nivells d'VE diferents dels disponibles al corpus. Aquesta flexibilitat esta alineada amb l'objectiu general d'aquesta tesi, permetre als sistemes US-TTS sintetitzar diferents estils expressius a partir d'un únic corpus d'estil neutre. La proposta realitzada també inclou un paràmetre que permet controlar fàcilment el nivell d'VE sintetitzat. Això obre moltes possibilitats per controlar fàcilment el procés de síntesi tal i com es va fer al projecte CreaVeu usant interfícies gràfiques simples i intuïtives, també realitzat dins del grup GTM. Aquesta memòria conclou presentant el treball realitzat en aquesta tesi i amb una proposta de modificació de l'esquema d'un sistema US-TTS per incloure els blocs de DSP desenvolupats en aquesta tesi que permetin al sistema sintetitzar múltiple nivells d'VE a partir d'un corpus d'estil neutre. Això obre moltes possibilitats per generar interfícies d'usuari que permetin controlar fàcilment el procés de síntesi, tal i com es va fer al projecte CreaVeu, també realitzat dins del grup GTM. Aquesta memòria conclou presentant el treball realitzat en aquesta tesi i amb una proposta de modificació de l'esquema del sistema US-TTS per incloure els blocs de DSP desenvolupats en aquesta tesi que permetin al sistema sintetitzar múltiple nivells d'VE a partir d'un corpus d'estil neutre. Esta tesis se llevó a cabo en el Grup en Tecnologies Mèdia de la Escuela de Ingeniería y Arquitectura la Salle. El grupo lleva una larga trayectoria dentro del campo de la síntesis de voz y cuenta con su propio sistema de síntesis por concatenación de unidades (US-TTS). El sistema permite sintetizar múltiples estilos expresivos mediante el uso de corpus específicos para cada estilo expresivo. De este modo, para realizar una síntesis agresiva, el sistema usa el corpus de este estilo, y para un estilo sensual, usa otro corpus específico para ese estilo. La presente tesis aborda el problema con un enfoque distinto proponiendo cambios en el esquema del sistema con el fin de mejorar la flexibilidad para sintetizar múltiples estilos expresivos a partir de un único corpus de estilo de habla neutro. El planteamiento seguido en esta tesis esta basado en el uso de técnicas de procesamiento de señales (DSP) para llevar a cabo modificaciones del señal de voz para que este exprese el estilo de habla deseado. Para llevar acabo las modificaciones de la señal de voz se han usado los modelos harmónico más ruido (HNM) por su flexibilidad para efectuar modificaciones de señales. La cualidad de la voz (VoQ) juega un papel importante en diferentes estilos expresivos. Por ello se exploró la síntesis expresiva basada en modificaciones de parámetros de bajo nivel de la VoQ. Durante este estudio se detectaron diferentes problemas que dieron pié a los objetivos planteados en esta tesis, entre ellos el encontrar un único parámetro con fuerte influencia en la expresividad. El parámetro seleccionado fue el esfuerzo vocal (VE) por su importante papel a la hora de expresar diferentes emociones. Las primeras pruebas se realizaron con el fin de transferir el VE entre dos realizaciones con diferente grado de VE de la misma palabra usando una metodología basada en un proceso filtrado de pre-émfasis adaptativo con coeficientes de predicción lineales (APLP). Esta primera aproximación logró transferir el nivel de VE entre dos realizaciones de la misma palabra, sin embargo el proceso presentaba limitaciones para generar niveles de esfuerzo vocal intermedios. A fin de mejorar la flexibilidad y el control del sistema para expresar diferentes niveles de VE, se planteó un nuevo modelo de VE basado en polinomios lineales. Este modelo permitió transferir el VE entre dos palabras diferentes e incluso generar nuevos niveles no presentes en el corpus usado para la síntesis. Esta flexibilidad está alineada con el objetivo general de esta tesis de permitir a un sistema US-TTS expresar múltiples estilos de habla expresivos a partir de un único corpus de estilo neutro. Además, la metodología propuesta incorpora un parámetro que permite de forma sencilla controlar el nivel de VE expresado en la voz sintetizada. Esto abre la posibilidad de controlar fácilmente el proceso de síntesis tal y como se hizo en el proyecto CreaVeu usando interfaces simples e intuitivas, también realizado dentro del grupo GTM. Esta memoria concluye con una revisión del trabajo realizado en esta tesis y con una propuesta de modificación de un esquema de US-TTS para expresar diferentes niveles de VE a partir de un único corpus neutro. This thesis was conducted in the Grup en Tecnologies M`edia (GTM) from Escola d’Enginyeria i Arquitectura la Salle. The group has a long trajectory in the speech synthesis field and has developed their own Unit-Selection Text-To-Speech (US-TTS) which is able to convey multiple expressive styles using multiple expressive corpora, one for each expressive style. Thus, in order to convey aggressive speech, the US-TTS uses an aggressive corpus, whereas for a sensual speech style, the system uses a sensual corpus. Unlike that approach, this dissertation aims to present a new schema for enhancing the flexibility of the US-TTS system for performing multiple expressive styles using a single neutral corpus. The approach followed in this dissertation is based on applying Digital Signal Processing (DSP) techniques for carrying out speech modifications in order to synthesize the desired expressive style. For conducting the speech modifications the Harmonics plus Noise Model (HNM) was chosen for its flexibility in conducting signal modifications. Voice Quality (VoQ) has been proven to play an important role in different expressive styles. Thus, low-level VoQ acoustic parameters were explored for conveying multiple emotions. This raised several problems setting new objectives for the rest of the thesis, among them finding a single parameter with strong impact on the expressive style conveyed. Vocal Effort (VE) was selected for conducting expressive speech style modifications due to its salient role in expressive speech. The first approach working with VE was based on transferring VE between two parallel utterances based on the Adaptive Pre-emphasis Linear Prediction (APLP) technique. This approach allowed transferring VE but the model presented certain restrictions regarding its flexibility for generating new intermediate VE levels. Aiming to improve the flexibility and control of the conveyed VE, a new approach using polynomial model for modelling VE was presented. This model not only allowed transferring VE levels between two different utterances, but also allowed to generate other VE levels than those present in the speech corpus. This is aligned with the general goal of this thesis, allowing US-TTS systems to convey multiple expressive styles with a single neutral corpus. Moreover, the proposed methodology introduces a parameter for controlling the degree of VE in the synthesized speech signal. This opens new possibilities for controlling the synthesis process such as the one in the CreaVeu project using a simple and intuitive graphical interfaces, also conducted in the GTM group. The dissertation concludes with a review of the conducted work and a proposal for schema modifications within a US-TTS system for introducing the VE modification blocks designed in this dissertation.

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Keyvani, Alireza. "Robustness in ASR : an experimental study of the interrelationship between discriminant feature-space transformation, speaker normalization and environment compensation." Thesis, McGill University, 2007. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=99772.

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This thesis addresses the general problem of maintaining robust automatic speech recognition (ASR) performance under diverse speaker populations, channel conditions, and acoustic environments. To this end, the thesis analyzes the interactions between environment compensation techniques, frequency warping based speaker normalization, and discriminant feature-space transformation (DFT). These interactions were quantified by performing experiments on the connected digit utterances comprising the Aurora 2 database, using continuous density hidden Markov models (HMM) representing individual digits. Firstly, given that the performance of speaker normalization techniques degrades in the presence of noise, it is shown that reducing the effects of noise through environmental compensation, prior to speaker normalization, leads to substantial improvements in ASR performance. The speaker normalization techniques considered here were vocal tract length normalization (VTLN) and the augmented state-space acoustic decoder (MATE). Secondly, given that discriminant feature-space transformation (DFT) are known to increase class separation, it is shown that performing speaker normalization using VTLN in a discriminant feature-space leads to improvements in the performance of this technique. Classes, in our experiments, corresponded to HMM states. Thirdly, an effort was made to achieve higher class discrimination by normalizing the speech data used to estimate the discriminant feature-space transform. Normalization, in our experiments, corresponded to reducing the variability within each class through the use of environment compensation and speaker normalization. Significant ASR performance improvements were obtained when normalization was performed using environment compensation, while our results were inconclusive for the case where normalization consisted of speaker normalization. Finally, aimed at increasing its noise robustness, a simple modification of MATE is presented. This modification consisted of using, during recognition, knowledge of the distribution of warping factors selected by MATE during training.

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TAKEDA, Kazuya, Seiichi NAKAGAWA, Yuya HATTORI, Norihide KITAOKA, and Makoto SAKAI. "Evaluation of Combinational Use of Discriminant Analysis-Based Acoustic Feature Transformation and Discriminative Training." Institute of Electronics, Information and Communication Engineers, 2010. http://hdl.handle.net/2237/14968.

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Mayer, Peter. "Transformation, restoration and migration : illusions in the perception of speech, music and other complex sounds." Thesis, City University London, 1994. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.339696.

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Lai, Jonathan Ping Wah. "A study of the main character's speech transformation in the Cantonese movie : the Great lover." HKBU Institutional Repository, 1995. http://repository.hkbu.edu.hk/etd_ra/38.

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Fux, Thibaut. "Vers un système indiquant la distance d'un locuteur par transformation de sa voix." Thesis, Grenoble, 2012. http://www.theses.fr/2012GRENT120/document.

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Cette thèse porte sur la transformation de la voix d’un locuteur dans l’objectif d’indiquer la distance de celui-ci : une transformation en voix chuchotée pour indiquer une distance proche et une transformation en voix criée pour une distance plutôt éloignée. Nous effectuons dans un premier temps des analyses approfondies pour déterminer les paramètres les plus pertinentes dans une voix chuchotée et surtout dans une voix criée (beaucoup plus difficile). La contribution principale de cette partie est de montrer la pertinence des paramètres prosodiques dans la perception de l’effort vocal dans une voix criée. Nous proposons ensuite des descripteurs permettant de mieux caractériser les contours prosodiques. Pour la transformation proprement dite, nous proposons plusieurs nouvelles règles de transformation qui contrôlent de manière primordiale la qualité des voix transformées. Les résultats ont montré une très bonne qualité des voix chuchotées transformées ainsi que pour des voix criées pour des structures linguistiques relativement simples (CVC, CVCV, etc.) This thesis focuses on speaker voice transformation in the aim to indicate the distance of it: a spokento-whispered voice transformation to indicate a close distance and a spoken-to-shouted voicetransformation for a rather far distance. We perform at first, in-depth analysis to determine mostrelevant features in whispered voices and especially in shouted voices (much harder). The maincontribution of this part is to show the relevance of prosodic parameters in the perception of vocaleffort in a shouted voice. Then, we propose some descriptors to better characterize the prosodiccontours. For the actual transformation, we propose several new transformation rules whichimportantly control the quality of transformed voice. The results showed a very good quality oftransformed whispered voices and transformed shouted voices for relatively simple linguisticstructures (CVC, CVCV, etc.)

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Panchapagesan, Sankaran. "Frequency warping by linear transformation, and vocal tract inversion for speaker normalization in automatic speech recognition." Diss., Restricted to subscribing institutions, 2008. http://proquest.umi.com/pqdweb?did=1610480121&sid=1&Fmt=2&clientId=1564&RQT=309&VName=PQD.

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Nieuwoudt, Christoph. "Cross-language acoustic adaptation for automatic speech recognition." Thesis, Pretoria : [s.n.], 2000. http://upetd.up.ac.za/thesis/available/etd-01062005-071829.

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Sooful, Jayren Jugpal. "Automated phoneme mapping for cross-language speech recognition." Diss., Pretoria [s.n.], 2004. http://upetd.up.ac.za/thesis/available/etd-01112005-131128.

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Huber, Stefan. "Voice Conversion by modelling and transformation of extended voice characteristics." Electronic Thesis or Diss., Paris 6, 2015. https://accesdistant.sorbonne-universite.fr/login?url=https://theses-intra.sorbonne-universite.fr/2015PA066750.pdf.

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La Conversion de la Voix (VC) vise à transformer les caractéristiques de la voix d’un locuteur source de manière qu’il sera perçu comme étant prononcé par un locuteur cible. Le principe de la VC est de définir des fonctions du transposition pour la conversion de la voix de l’un locuteur source à la voix de l’un locuteur cible. Les fonctions de transformation de VC systèmes "State-Of-The-Art" (START) adapte instantanément aux caractéristiques de la voix source. Cependant, la qualité est pas encore suffisant. Des améliorations considérables sont nécessaires que les techniques VC peuvent être utilisés dans un environnement industriel professionnel. L’objectif de cette thèse est d’augmenter la qualité de la conversion de la voix pour faciliter son applicabilité industrielle dans une mesure raisonnable. Les propriétés de base de différentes START algorithmes de la conversion de la voix sont discutés sur leurs avantages intrinsèques et ses déficits. Basé sur des évaluations expérimentales avec un GMM VC système la conclusion est que la plupart des systèmes VC START qui reposent sur des modèles statistiques sont, en raison de l’effet en moyenne de la régression linéaire, moins appropriées pour atteindre un score du similitude assez élevé avec le haut-parleur cible requise pour l’utilisation industrielle. Les contributions établies pendant de ce travail de thèse se trouvent dans les moyens étendus à a) modéliser l’excitation du source glottique, b) modéliser des descripteurs de la voix en utilisant un nouveau système de parole basée sur un modèle élargie de source-filtre, et c) avancer une nouveau système VC de l’Ircam en le combinant avec les contributions de a) et b) Voice Conversion (VC) aims at transforming the characteristics of a source speaker’s voice in such a way that it will be perceived as being uttered by a target speaker. The principle of VC is to define mapping functions for the conversion from one source speaker’s voice to one target speaker’s voice. The transformation functions of common State-Of-The-Art (START) VC system adapt instantaneously to the characteristics of the source voice. While recent VC systems have made considerable progress over the conversion quality of initial approaches, the quality is nevertheless not yet sufficient. Considerable improvements are required before VC techniques can be used in an professional industrial environment. The objective of this thesis is to augment the quality of Voice Conversion to facilitate its industrial applicability to a reasonable extent. The basic properties of different START algorithms for Voice Conversion are discussed on their intrinsic advantages and shortcomings. Based on experimental evaluations of one GMM-based State-Of-The-Art VC approach the conclusion is that most VC systems which rely on statistical models are, due to averaging effect of the linear regression, less appropriate to achieve a high enough similarity score to the target speaker required for industrial usage. The contributions established throughout this thesis work lie in the extended means to a) model the glottal excitation source, b) model a voice descriptor set using a novel speech system based on an extended source-filter model, and c) to further advance IRCAM’s novel VC system by combining it with the contributions of a) and b)

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Bous, Frederik. "A neural voice transformation framework for modification of pitch and intensity." Electronic Thesis or Diss., Sorbonne université, 2023. http://www.theses.fr/2023SORUS382.

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La voix humaine est une grande source de fascination et un objet de recherche depuis plus de 100 ans. Pendant ce temps, de nombreuses technologies ont germées autour de la voix, comme le vocodeur, qui fournit une représentation paramétrique de la voix, couramment utilisée pour la transformation de la voix. Dans cette tradition, les limites des approches basées uniquement sur le traitement du signal sont évidentes : Pour créer des transformations cohérentes, les dépendances entre les différentes propriétés vocales doivent être bien comprises et modélisées avec précision. Modéliser ces corrélations avec des heuristiques obtenues par des études empiriques ne suffit pas à créer des résultats naturels. Il est nécessaire d'extraire systématiquement des informations sur la voix et d'utiliser automatiquement ces informations lors du processus de transformation. Les progrès récents de la puissance de calcul permettent cette analyse systématique des données au moyen de l'apprentissage automatique. Cette thèse utilise donc l'apprentissage automatique pour créer un système neuronal de transformation de la voix. Le système neuronal de transformation de la voix, présenté ici, fonctionne en deux étapes : Tout d'abord, un vocodeur neuronal permet d'établir une correspondance entre la forme d'onde et une représentation mel-spectrogramme des signaux vocaux. Ensuite, un auto-encodeur avec un goulot d'étranglement permet de démêler différentes propriétés de la voix du reste de l'information. L'auto-encodeur permet de modifier une propriété de la voix tout en ajustant automatiquement d'autres caractéristiques de façon à en conserver le réalisme. Dans la première partie de cette thèse, nous comparons différentes approches du vocodage neuronal et nous expliquons pourquoi la représentation mel-spectrogramme est plus adapté pour la transformation neuronale de la voix plutôt que les espaces paramétriques du vocodeur conventionnels. Dans la deuxième partie, nous présentons l'auto-encodeur avec goulot d'étranglement de l'information. L'auto-encodeur crée un code latent indépendant du conditionnement en entrée. En utilisant ce code latent, le synthétiseur peut effectuer la transformation en combinant le code latent original avec une courbe de paramètres modifiée. Nous transformons la voix en utilisant deux paramètres de contrôle : la fréquence fondamentale et le niveau sonore vocal. La transformation de la fréquence fondamentale est un problème qui a longtemps été abordé : Notre approche est comparable aux techniques existantes puisqu'elles utilisent la fréquence fondamentale comme paramètre. Cela nous permet également d'étudier comment l'auto-encodeur modélise les dépendances entre la fréquence fondamentale et d'autres propriétés de la voix dans un environnement connu. Quant au niveau sonore vocal, nous sommes confrontés au problème de la rareté des annotations. Par conséquent, nous proposons d'abord une nouvelle technique d'estimation du niveau sonore vocal dans de grandes bases de données de voix ; puis nous utilisons ces annotations pour entraîner un auto-encodeur avec goulot d'étranglement permettant de modifier le niveau sonore vocal Human voice has been a great source of fascination and an object of research for over 100 years. During that time numerous technologies have sprouted around the voice, such as the vocoder, which provides a parametric representation of the voice, commonly used for voice transformation. From this tradition, the limitations of purely signal processing based approaches are evident: To create meaningful transformations the codependencies between different voice properties have to be understood well and modelled precisely. Modelling these correlations with heuristics obtained by empiric studies is not sufficient to create natural results. It is necessary to extract information about the voice systematically and use this information during the transformation process automatically. Recent advances in computer hardware permit this systematic analysis of data by means of machine learning. This thesis thus uses machine learning to create a neural voice transformation framework. The proposed neural voice transformation framework works in two stages: First a neural vocoder allows mapping between a raw audio and a mel-spectrogram representation of voice signals. Secondly, an auto-encoder with information bottleneck allows disentangling various voice properties from the remaining information. The auto-encoder allows changing one voice property while automatically adjusting the remaining voice properties. In the first part of this thesis, we discuss different approaches to neural vocoding and reason why the mel-spectrogram is better suited for neural voice transformations than conventional parametric vocoder spaces. In the second part we discuss the information bottleneck auto-encoder. The auto-encoder creates a latent code that is independent of its conditional input. Using the latent code the synthesizer can perform the transformation by combining the original latent code with a modified parameter curve. We transform the voice using two control parameters: the fundamental frequency and the voice level. Transformation of the fundamental frequency is an objective with a long history. Using the fundamental frequency allows us to compare our approach to existing techniques and study how the auto-encoder models the dependency on other properties in a well known environment. For the voice level, we face the problem that annotations hardly exist. Therefore, first we provide a new estimation technique for voice level in large voice databases, and subsequently use the voice level annotations to train a bottleneck auto-encoder that allows changing the voice level

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Zailskas, Vytautas. "Lietuvių šnekos vizemų vizualizavimas." Master's thesis, Lithuanian Academic Libraries Network (LABT), 2011. http://vddb.laba.lt/obj/LT-eLABa-0001:E.02~2011~D_20110615_134342-38011.

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Magistro baigiamajame darbe analizuojama lietuvių šnekos vizemų vizualizacijos problema, tiriami lietuvių kalbos vizemų vizualizacijos požymiai, galimybės koduoti SAMPA kodais analizė, analizuojami programinės įrangos kūrimo metodai ir algoritmai. Sukurtas algoritmas sprendžiantis lietuvių šnekos vizemų vizualizacijos problemą. Pasirinktas kompiuterinės grafikos tipas, tinkantis iškeltiems tikslams įvykdyti – vektorinė grafika. Sukurti du vektorių transformacijos metodai, išanalizuoti jų skirtumai ir praktinio panaudojimo galimybės. Sukurta programinė įranga įgalinanti vartotoją kurti vizemas, jas transformuoti ir derinti jų vaizdavimo trukmę įvairiais koeficientais ir vykdyti animaciją pagal pasirinktą transformacijos metodą. Sudarytos penkios vizemos ir dviejų lietuviškų žodžių animacijos, kurių pagalba atliktas tyrimas parodantis darbo ir metodų realizavimo kokybę bei pritaikomumą. In this final Master's thesis work features of Lithuanian speech visemes visualization are analyzed. Possibility of coding with SAMPA codes, software methods and algorithms are inspected. Type of computer graphics is picked, which is suitable for software objectives – vector graphics. Two transformation methods for vector graphics are created and their differences and practical usability are analyzed. Software for visemes creation, transformation and tuning of their duration and duration of transformation between visemes is created and described. The main purpose of this software is to animate Lithuanian speech by the method selected. Five visemes for two Lithuanian words animation is created. Using these visemes research has been done which is showing the quality of realization and adaptability of this software.

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Deol, Raman Kaur. "The creation of the Khalsa : a study into the rhetorical strategies of collective identity transformation." Scholarly Commons, 2009. https://scholarlycommons.pacific.edu/uop_etds/724.

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The Khalsa is a militant sect of the Sikh religion officially created by Guru Gobind on Baisakhi Day in 1699. Sikhism, as a religion and culture, existed within the overarching structure of lndian society during the reign of the Muslim Mughal Empire. Over the course of its history, Sikhism sought to evolve and adapt to internal and external pressures, and the creation of the Khalsa was a momentous and transformational step in that evolutionary process. Using Kenneth Burke's guilt-redemption cycle as a model, this study analyses the events that created the Khalsa. The study found that historical and social pressures provided the rhetorical exigence for the creation of the Khalsa. Guru Gobind isolated and used the guilt of the Sikhs people, the guilt of being passive observers in the face of external pressures, the guilt of living in caste-organized society, the guilt of living in a bureaucratic system wherein the priests had seized power and control, and the guilt of living without external markers of the faith. These sources of guilt were brought to the forefront by Guru Gobind, and resolved through the symbolic sacrifice of five men, after which Guru Gobind created the Khalsa as an answer. Through the Khalsa, its symbols and rituals, the Sikhs were provided with a way to escape the flaws and guilt of the old order. The creation of the Khalsa was an important milestone in the evolution of the Sikh culture and religion. Through this study, the processes and methods of this identity transformation were isolated. Guru Gobind activated social and collective levels of identity through the medium of performance in order to transform his audience of Sikhs into the Khalsa.

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Morais, Lays Bárbara Vieira. "As aporias do lugar de fala: como a política identitária afetou a esquerda." Universidade Federal de Goiás, 2018. http://repositorio.bc.ufg.br/tede/handle/tede/9104.

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Submitted by Luciana Ferreira (lucgeral@gmail.com) on 2018-11-28T14:37:20Z No. of bitstreams: 2 Dissertação - Lays Bárbara Vieira Morais - 2018.pdf: 1704797 bytes, checksum: 1f0bd574efadd1bca7c251d5d6c8c358 (MD5) license_rdf: 0 bytes, checksum: d41d8cd98f00b204e9800998ecf8427e (MD5) Approved for entry into archive by Luciana Ferreira (lucgeral@gmail.com) on 2018-11-28T14:38:55Z (GMT) No. of bitstreams: 2 Dissertação - Lays Bárbara Vieira Morais - 2018.pdf: 1704797 bytes, checksum: 1f0bd574efadd1bca7c251d5d6c8c358 (MD5) license_rdf: 0 bytes, checksum: d41d8cd98f00b204e9800998ecf8427e (MD5) Made available in DSpace on 2018-11-28T14:38:55Z (GMT). No. of bitstreams: 2 Dissertação - Lays Bárbara Vieira Morais - 2018.pdf: 1704797 bytes, checksum: 1f0bd574efadd1bca7c251d5d6c8c358 (MD5) license_rdf: 0 bytes, checksum: d41d8cd98f00b204e9800998ecf8427e (MD5) Previous issue date: 2018-11-08 The research shows, first, the limits of the use of the speech place and, later, how this use and these limits affect the political action of the left, of the social movements of the left. The character of the work is theoretical, making an extensive bibliographical review and based on Critical Theory. In the first chapter we make a historical, political and theoretical point about the subject who speaks, the political action of a subject in the many visions within the left thinking. Chapter two deals with the limit of the current sense used to the speech place, namely: a liberal emphasis of politics on individuals, these are considered as an ideological category, the speech place as an individualism obliquely. Finally, it was concluded that the contemporary identity politics, represented by the use of the speech place, shifted the axis of action from the left of the collective to the individual. A pesquisa procurou demonstrar, inicialmente, os limites do uso do lugar de fala e, posteriormente, como esse uso e esses limites afetam a atuação política da esquerda, dos movimentos sociais de esquerda. O caráter do trabalho é teórico, fazendo uma extensa revisão bibliográfica e tendo como base a Teoria Crítica. No primeiro capitulo faz-se um apontamento histórico, político e teórico referente ao sujeito que fala, a ação política de um sujeito entendida pelas variadas visões dentro do pensamento de esquerda. O capitulo dois trata da aporia do atual sentido empregado ao lugar de fala, qual seja: uma ênfase liberal da política nos indivíduos, estes tidos como uma categoria ideológica, o lugar de fala como um individualismo por via obliqua. Por fim, conclui-se que a política identitária contemporânea, representada pelo uso do lugar de fala, deslocou o eixo de atuação da esquerda da coletividade para o indivíduo.

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Bourdier, Renaud. "Analyse temps/frequence, filtrage et synthese numeriques de signaux de parole : application au filtrage, a la reduction de bruit et a la restauration d'enregistrements anciens." Le Mans, 1988. http://www.theses.fr/1988LEMA1001.

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Etude des phenomenes temporels et frequentiels apparaissant lors d'une synthese a partir de la modification des spectres deduits de l'analyse par transformee de fourier a court terme (tfct). Les performances de l'implementation par tfct d'une analyse synthese, d'une operation de filtrage invariant ou dependant du temps, et d'une reduction de bruit ont ete caracterisees

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Wu, Si Fan. "Speech analysis by AFD." Thesis, University of Macau, 2012. http://umaclib3.umac.mo/record=b2592945.

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Близнюк, А. С. "Лінгвістичний та перекладацький аспекти слоганів до англомовних фільмів". Master's thesis, Сумський державний університет, 2018. http://essuir.sumdu.edu.ua/handle/123456789/72418.

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Робота фокусується на дослідженні рекламного дискурсу та його складової – слоганів до англомовних фільмів. Вивчаються особливості кінослоганів з точки зору лінгвістики, перекладу та викладання. Було досліджено основні лінгвістичні особливості слоганів до англомовних фільмів, а саме лексичні, синтаксичні та стилістичні. Через призму перекладацьких трансформацій, а саме лексичних та граматичних, було досліджено особливості перекладу кінослоганів з англійської на українську мову. Розглянуто стратегії навчання слоганам на заняттях з практики перекладу. З’ясовано, що кінослогани – це помітне явище реклами, яке являє собою невичерпне джерело для дослідження з точки зору лінгвістики. Дослідження слоганів до фільмів з точки зору лінгвістики та перекладу дозволить виявити основні структурні елементи слоганів та засоби, за допомогою яких відбувається вплив на свідомість реципієнта. Теоретична значущість роботи полягає у тому, що матеріал може застосовуватися у дослідженні реклами та рекламних текстів, а також слоганів до англомовних фільмів. The paper focuses on the study of advertising discourse and its component – taglines to English films. Taglines from the perspective of linguistics, translation and teaching is analyzed. The main linguistic features of movie taglines, namely lexical, syntactic and stylistic, were investigated. Through the prism of translation transformations, namely lexical and grammatical, translation of movie taglines from English to Ukrainian was distinguished. Teaching strategies involving movie taglines in higher education institutions were specified. It was found out that movie taglines are noticeable phenomenon of advertising which are an inexhaustible source for the study from the point of view of linguistics. Investigation of movie taglines from the point of view of linguistics and translation will allow to reveal the main structural elements of taglines and means by which an impact on the consciousness of the recipient is made. Theoretical foundations of the study of taglines to English films have been distinguished. Material can be used in the study of advertising and advertising texts, as well in the study of taglines.

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Teisseire, Denis. "Genèse instrumentale des technologies numériques dans les activités des préfets." Thesis, Paris, CNAM, 2019. http://www.theses.fr/2019CNAM1250/document.

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Les préfets auront pour les prochaines années à poursuivre leur action au sein d’une société française inscrite dans la dynamique planétaire d’un « processus d’appropriation d’une innovation disruptive » et « oblige les utilisateurs à rompre avec leurs manières de faire et de penser antérieures ».Comment ce « personnage » assume ce changement est la question inductrice de cette recherche. Il y-a-t-il transformation, transition, évolution ou évitement ?En ayant recours à la Sémiotique des Transactions Coopératives (STC) sont identifiés dans les narrations recueillies les indices d’une appropriation de cette transition numérique, à travers : - ses impacts cognitivo émotionnels ; - la description des conditions et du contexte d’usage des objets devenus instruments ; - la technique gestionnaire déployée pour réguler la mise en tension sur son bassin de vie.Les principaux apports mettent en évidence : - un nouveau rapport du préfet à la technologie ; - un engagement vis-à-vis de l’innovation en tant qu’appui ou facilitation des initiatives émergentes, plus difficilement en tant que porteur de projet ; - un projet de république numérique qui reste flou, mais qui redistribue le lien social et les modalités d’organisation de l’action collective The prefects will have for the next years to continue their action within a French company inscribed in the global dynamics of a "process of appropriation of a disruptive innovation" and "forces the users to break with their ways of doing and to think earlier ".How this "character" assumes this change is the inductive question of this research. Is there transformation, transition, evolution or avoidance?Using the Semiotics of Cooperative Transactions (STC) are identified in the narrations collected the clues of an appropriation of this digital transition, through : - its cognitive and emotional impacts ; - the description of the conditions and the context of use of objects that became instruments; - the managerial technique deployed to regulate the tensioning in his living area.The main contributions highlight : - a new report from the prefect to technology ; - a commitment to innovation as a support or facilitation of emerging initiatives, more difficult as a project leader ; - a draft of a digital republic that remains unclear, but which redistributes the social link and the organizational methods of collective action

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Charpentier, Francis. "Traitement de la parole par analyse-synthese de fourier : application a la synthese par diphones." Paris, ENST, 1988. http://www.theses.fr/1988ENST0009.

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Ces techniaues sont utilisees dans le but d'obtenir une meilleure qualite de son que celle obtenue par les methodes paramagnetiques habituelles. L'accent est mis sur la double approche suivante: 1) interpretation de la transformee de fourier a court terme comme un banc de filtres et synthese par addition des sorties de ce banc filtre; 2) synthese par superposition et addition de signaux a court terme

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Leggetter, Christopher John. "Improved acoustic modelling for HMMs using linear transformations." Thesis, University of Cambridge, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.361709.

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Rekik, Siwar. "Sécurisation de la communication parlée par une techhnique stéganographique." Thesis, Brest, 2012. http://www.theses.fr/2012BRES0061.

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Une des préoccupations dans le domaine des communications sécurisées est le concept de sécurité de l'information. Aujourd’hui, la réalité a encore prouvé que la communication entre deux parties sur de longues distances a toujours été sujet au risque d'interception. Devant ces contraintes, de nombreux défis et opportunités s’ouvrent pour l'innovation. Afin de pouvoir fournir une communication sécurisée, cela a conduit les chercheurs à développer plusieurs schémas de stéganographie. La stéganographie est l’art de dissimuler un message de manière secrète dans un support anodin. L’objectif de base de la stéganographie est de permettre une communication secrète sans que personne ne puisse soupçonner son existence, le message secret est dissimulé dans un autre appelé medium de couverture qui peut être image, video, texte, audio,…. Cette propriété a motivé les chercheurs à travailler sur ce nouveau champ d’étude dans le but d’élaborer des systèmes de communication secrète résistante à tout type d’attaques. Cependant, de nombreuses techniques ont été développées pour dissimuler un message secret dans le but d’assurer une communication sécurisée. Les contributions majeures de cette thèse sont en premier lieu, de présenter une nouvelle méthode de stéganographie permettant la dissimulation d’un message secret dans un signal de parole. La dissimulation c’est le processus de cacher l’information secrète de façon à la rendre imperceptible pour une partie tierce, sans même pas soupçonner son existence. Cependant, certaines approches ont été étudiées pour aboutir à une méthode de stéganogaraphie robuste. En partant de ce contexte, on s’est intéressé à développer un système de stéganographie capable d’une part de dissimuler la quantité la plus élevée de paramètre tout en gardant la perceptibilité du signal de la parole. D’autre part nous avons opté pour la conception d’un algorithme de stéganographie assez complexe afin d’assurer l’impossibilité d’extraction de l’information secrète dans le cas ou son existence été détecter. En effet, on peut également garantir la robustesse de notre technique de stéganographie à l’aptitude de préservation du message secret face aux tentatives de détection des systèmes de stéganalyse. Notre technique de dissimulation tire son efficacité de l’utilisation de caractéristiques spécifiques aux signaux de parole et àl’imperfection du système auditif humain. Des évaluations comparatives sur des critères objectifs et subjectifs de qualité sont présentées pour plusieurs types de signaux de parole. Les résultats ont révélé l'efficacité du système développé puisque la technique de dissimulation proposée garantit l’imperceptibilité du message secret voire le soupçon de son existence. Dans la suite expérimentale et dans le même cadre de ce travail, la principale application visée par la thèse concerne la transmission de parole sécurisée par un algorithme de stéganographie. Dans ce but il s’est avéré primordial d’utiliser une des techniques de codage afin de tester la robustesse de notre algorithme stéganographique face au processus de codage et de décodage. Les résultats obtenus montrent la possibilité de reconstruction du signal original (contenant des informations secrètes) après codage. Enfin une évaluation de la robustesse de notre technique de stéganographie vis à vis des attaques est faite de façon à optimiser la technique afin d'augmenter le taux de sécurisation. Devant cette nécessité nous avons proposé une nouvelle technique de stéganalyse basée sur les réseaux de neurones AR-TDNN. La technique présentée ici ne permet pas d'extraire l'éventuel message caché, mais simplement de mettre en évidence sa présence One of the concerns in the field of secure communication is the concept of information security. Today’s reality is still showing that communication between two parties over long distances has always been subject to interception. Providing secure communication has driven researchers to develop several cryptography schemes. Cryptography methods achieve security in order to make the information unintelligible to guarantee exclusive access for authenticated recipients. Cryptography consists of making the signal look garbled to unauthorized people. Thus, cryptography indicates the existence of a cryptographic communication in progress, which makes eavesdroppers suspect the existence of valuable data. They are thus incited to intercept the transmitted message and to attempt to decipher the secret information. This may be seen as weakness in cryptography schemes. In contrast to cryptography, steganography allows secret communication by camouflaging the secret signal in another signal (named the cover signal), to avoid suspicion. This quality motivated the researchers to work on this burning field to develop schemes ensuring better resistance to hostile attackers. The word steganography is derived from two Greek words: Stego (means cover) and graphy (means writing). The two combined words constitute steganography, which means covert writing, is the art of hiding written communications. Several steganography techniques were used to send message secretly during wars through the territories of enemies. The major contributions of this thesis are the following ones. We propose a new method to secure speech communication using the Discrete Wavelet Transforms (DWT) and the Fast Fourier Transform (FFT). Our method exploits first the high frequencies using a DWT, then exploits the low-pass spectral properties of the speech magnitude spectrum to hide another speech signal in the low-amplitude high-frequencies region of the cover speech signal. The proposed method allows hiding a large amount of secret information while rendering the steganalysis more complex. Comparative evaluation based on objective and subjective criteria is introduced for original speech signal, stego-signal and reconstructed secret speech signal after the hiding process. Experimental simulations on both female and male speakers revealed that our approach is capable of producing a stego speech that is indistinguishable from the cover speech. The receiver is still able to recover an intelligible copy of the secret speech message. We used an LPC10 coder to test the effect of the coding techniques on the stego-speech signals. Experimental results prove the efficiency of the used coding technique since intelligibility of the stego-speech is maintained after the encoding and decoding processes. We also advocate a new steganalysis technique to ensure the robustness of our steganography method. The proposed classifier is called Autoregressive time delay neural network (ARTDNN). The purpose of this steganalysis system is to identify the presence or not of embedded information, and does not actually attempt to extract or decode the hidden data. The low detecting rate prove the robustness of our hiding technique

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Megyesi, Beata. "Data-driven syntactic analysis." Doctoral thesis, KTH, Speech Transmission and Music Acoustics, 2002. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-3433.

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Спаська, Г. А. "Засоби реалізації стратегії театральності в спічах американських політичних діячів в перекладі українською". Master's thesis, Сумський державний університет, 2019. http://essuir.sumdu.edu.ua/handle/123456789/75285.

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Мета: виявлення мовних особливостей реалізації стратегії театральності в американському політичному дискурсі та вивчення аспектів їх передачі українською мовою. Теоретичне значення: робота дозволяє вивчити особливості збереження засобів реалізації стратегії театральності американського політичного дискурсу у гендерному аспекті під час перекладу українською, що є внеском у перекладознавство, враховуючи велику кількість жінок в американській політиці та зародження цього феномену в Україні. Всебічне дослідження стратегії театральності слугуватиме підґрунтям для наступних розвідок, теоретичних праць і узагальнень новітніх теоретичних положень у галузі перекладознавства. Цель: выявление языковых особенностей реализации стратегии театральности в американском политическом дискурсе и изучение аспектов их передачи на украинском языке. Теоретическое значение: работа позволяет изучить особенности сохранения средств реализации стратегии театральности американского политического дискурса в гендерном аспекте при переводе украинской, что является вкладом в переводоведение, учитывая большое количество женщин в американской политике и зарождения этого феномена в Украине. Всестороннее исследование стратегии театральности служить основой для последующих исследований, теоретических работ и обобщений новейших теоретических положений в области переводоведения. Goal: to identify the linguistic features of the theatrical strategy implementation in the American political discourse and to study the aspects of their realization in the Ukrainian translation. Theoretical meaning: The work allows to study the ways of preserving the means of theatrical strategy implementation in gender aspect via translation into Ukrainian, which is a contribution to translation studies. A comprehensive study of the theatrical strategy will serve as a basis for further theoretical works and the newest theoretical papers in the field of translation studies. The article analyzes means of theatrical strategy implementation the in the speeches of American politicians and ways of their rendering in Ukrainian translation. The main gender differences in the use of the theatrical strategy means are identified. The main focus is on the analysis of translation transformations aimed at preserving communicative and pragmatic effect of the source text. In addition, the paper outlined possible paradigms for the use of US politicians’ speeches to teach consecutive translation.

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Caillat, Isabelle. "Développement d'outils de management et actes de langage dans les entreprises de spectacle vivant." Thesis, Lyon 3, 2011. http://www.theses.fr/2011LYO30052/document.

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Les entreprises de spectacle vivant évoluent dans un contexte de diminution de ressources externes et de modification des modalités d’attribution des subventions par l’application de la LOLF (Loi Organique relative aux Lois de Finances). Leur problématique repose sur les moyens à mettre en œuvre pour faire face à ces contraintes. Nous nous attachons à démontrer que leur développement dépend de l’amélioration de la logique de coopération entre les acteurs à partir du projet artistique et de l’appropriation des contraintes d’évaluation de la LOLF, pour révéler les performances cachées et développer des ressources internes. Cette recherche se construit à partir de l’hypothèse que l’amélioration de la performance globale des organisations dépend d’une action transformative qui opère selon trois axes interdépendants : l’intervention, les outils de management, le langage–acteur. A partir d’une recherche-Intervention dans un théâtre et d’un diagnostic qualitatif dans un autre, nous accompagnons les acteurs dans une conduite de changement et étudions les conditions de développement managérial dans ce type d’organisation. Nous analysons comment l’utilisation du langage dans le cadre de l’Intervention Socio-Economique modifie les représentations et contribue à l’élaboration d’un nouveau dispositif managérial. Nous proposons d’associer les outils de management socio-Économiques, utilisés comme des matrices de lecture de l’organisation, aux critères d’évaluation et de subvention des pouvoirs publics pour en faire des outils de réflexion sur le processus de réalisation et de diffusion de spectacle et construire une représentation partagée entre les différents acteurs Performing arts companies operate in a context of declining resources and changes in the rules governing the allocation of grants, in accordance with the LOLF (Organic Law relative to the Laws of Finance). Their challenge lies in finding ways to address these constraints. We aim to demonstrate that their development depends on improving the manner in which all parties cooperate, based on the artistic project and taking into account the constraints resulting from the evaluation used by the LOLF, to reveal hidden costs and to develop internal resources. This research is based on the hypothesis that improving the overall performance of organizations depends on a transformative action that operates in three interdependent areas: intervention, management tools, and the speech-Actor. Based on intervention-Research in one theatre and on a qualitative study in another, we have assisted actors in the process of change management and examined the conditions in which managerial development takes place in this type of organization. We analyse how language use in the context of Socio-Economic Intervention modifies the representations and contributes to the elaboration of a new managerial instrument. We propose the use of socio-Economic management tools, serving as a framework for analyzing the organisation, combined with the criteria of evaluation and allocation of public funding as a basis of reflexion on the process of creation and dissemination of shows, as well as a means of building a shared representation between the different actors

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Yeates, Stuart Andrew. "Text Augmentation: Inserting markup into natural language text with PPM Models." The University of Waikato, 2006. http://hdl.handle.net/10289/2600.

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This thesis describes a new optimisation and new heuristics for automatically marking up XML documents, and CEM, a Java implementation, using PPM models. CEM is significantly more general than previous systems, marking up large numbers of hierarchical tags, using n-gram models for large n and a variety of escape methods. Four corpora are discussed, including the bibliography corpus of 14682 bibliographies laid out in seven standard styles using the BibTeX system and marked up in XML with every field from the original BibTeX. Other corpora include the ROCLING Chinese text segmentation corpus, the Computists' Communique corpus and the Reuters' corpus. A detailed examination is presented of the methods of evaluating mark up algorithms, including computation complexity measures and correctness measures from the fields of information retrieval, string processing, machine learning and information theory. A new taxonomy of markup complexities is established and the properties of each taxon are examined in relation to the complexity of marked up documents. The performance of the new heuristics and optimisation are examined using the four corpora.

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Olovsson, Hanna. "Communication in Employee Volunteering Programmes : Cross-sector dialogue - A strategic or idealistic approach?" Thesis, Umeå universitet, Sociologiska institutionen, 2015. http://urn.kb.se/resolve?urn=urn:nbn:se:umu:diva-110644.

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Recent years have brought the private and non-profit sectors closer together in terms of cross-sectional collaborations. As businesses have become more involved in communities, initiatives such as employee volunteering (EV)—where employees are encouraged to volunteer by their employers—are becoming more popular and are receiving more scholarly attention. However, the question still remains as to whether the main reasons behind cooperation are related to strategy: does interaction and communication in EV mirror a more ideal- or strategic approach? As EV programmes (EVPs) bring together actors with different world-views and perspectives, much can be learned from studying their interaction. The present study examined the nature of communication in EV and whether this communication reflects a strategic (based on self-interest) or ideal (corresponding to Habermas’s ideal speech situation and stakeholder dialogue) approach. In addition, the study looked at factors that, according to participants, facilitate improved communication and understanding in EV. The findings indicate that communication in EVPs largely reflected the strategic approach. However, the ideal approach is still apparent in some situations and can successfully be used given the right conditions--for example, in situations of long-term collaboration with increased experience when participants invest time, resources and motivation in building relationships. Time and honesty was also important factors. However, a range of barriers made collaboration more difficult. Pursuit of strategic short-term solutions and shortage of resources and time may hinder important dialogue and understanding.

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Андрейчук, Богдан Валерійович, та Bogdan Andreichuk. "Метод розпізнавання голосових сигналів для керування комп’ютерними системами вимірювань". Master's thesis, ТНТУ ім. І. Пулюя, Факультет прикладних інформаційних технологій та електроінженерії, Кафедра біотехнічних систем, м. Тернопіль, Україна, 2021. http://elartu.tntu.edu.ua/handle/lib/36502.

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В кваліфікаційній роботі здійснено порівняльний аналіз застосування різних вимірювань близькості та векторних ознак, який показав, що найбільш придатними для завдань розпізнавання векторами ознак можна вважати: мел-кепстральні коефіцієнти та розподіл інформаційних інтервалів мовного сигналу для керування комп’ютерними системами. In the qualification work, a comparative analysis of the use of different measurements of proximity and vector features, which showed that the most suitable for recognition tasks vector features can be considered: mel-keppstral coefficients and distribution of information intervals of speech signal to control computer systems. ВСТУП 8 РОЗДІ 1. ОСНОВНА ЧАСТИНА 11 1.1. Сучасний стан напряму розпізнавання мовних сигналів 11 1.2. Особливості мовлення та сприйняття мови людиною 16 1.2.1. Мовний апарат 17 1.2.2. Сприйняття мовного сигналу людиною 20 1.3 Методи цифрової обробки сигналів у задачах розпізнавання мовних сигналів 26 1.3.1. Спектральний аналіз 26 1.3.2. Віконний аналіз у базисі Фур'є 27 1.3.3. Вейвлет аналіз 27 1.3.4. Кепстральний аналіз 29 1.4 Субсмуговий підхід до обробки мовних сигналів 32 1.5 Висновки до розділу 1 33 РОЗДІЛ 2. ОСНОВНА ЧАСТИНА 34 2.1. Акустико-фонетичний підхід до розпізнавання мовних сигналів 34 2.2. Обчислювальні аспекти субсмугового аналізу мовних сигналів у задачах ідентифікації 37 2.3. Дослідження просторів ознак у задачах розпізнавання мовних сигналів 43 2.3.1. Декомпозиція сигналу банком фільтрів 43 2.3.2. Розподіл миттєвих енергій відрізка МС 45 2.3.3. Розподіл часток енергії відрізка МС 47 2.3.4. Розподіл інформаційних інтервалів відрізка МС 49 2.3.5. Частота переходів через нуль 52 2.3.6. Ширина частотної області, що займає сигнал 55 2.3.7. Мел-кепстральні коефіцієнти мовного сигналу 60 2.4. Заходи близькості у задачах розпізнавання мовних сигналів 63 2.4.1. Євклідова відстань 63 2.4.2. Середньоквадратичне відхилення 63 2.4.3. Відстань Махаланобіса 64 2.4.4. Кореляція послідовностей 64 2.4.5. Динамічна трансформація тимчасової шкали 65 2.5. Висновки до розділу 2 67 РОЗДІЛ 3. НАУКОВО-ДОСЛІДНА ЧАСТИНА 68 3.1. Методика оцінки методів розпізнавання мовних сигналів 68 3.2 Дослідження підходів до розпізнавання мовних сигналів 72 3.3. Висновки до розділу 3 80 РОЗДІЛ 4. ОХОРОНА ПРАЦІ ТА БЕЗПЕКА В НАДЗВИЧАЙНИХ СИТУАЦІЯХ 81 4.1. Охорона праці 81 4.2. Безпека в надзвичайних ситуаціях 84 4.3. Висновки до розділу 4 86 ВИСНОВКИ 87 СПИСОК ВИКОРИСТАНИХ ДЖЕРЕЛ 88 Додаток А. Копія тези конференції 93

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Bjerva, Johannes. "Genetic Algorithms in the Brill Tagger : Moving towards language independence." Thesis, Stockholms universitet, Avdelningen för datorlingvistik, 2013. http://urn.kb.se/resolve?urn=urn:nbn:se:su:diva-90472.

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The viability of using rule-based systems for part-of-speech tagging was revitalised when a simple rule-based tagger was presented by Brill (1992). This tagger is based on an algorithm which automatically derives transformation rules from a corpus, using an error-driven approach. In addition to performing on par with state of the art stochastic systems for part-of-speech tagging, it has the advantage that the automatically derived rules can be presented in a human-readable format. In spite of its strengths, the Brill tagger is quite language dependent, and performs much better on languages similar to English than on languages with richer morphology. This issue is addressed in this paper through defining rule templates automatically with a search that is optimised using Genetic Algorithms. This allows the Brill GA-tagger to search a large search space for templates which in turn generate rules which are appropriate for various target languages, which has the added advantage of removing the need for researchers to define rule templates manually. The Brill GA-tagger performs significantly better (p<0.001) than the standard Brill tagger on all 9 target languages (Chinese, Japanese, Turkish, Slovene, Portuguese, English, Dutch, Swedish and Icelandic), with an error rate reduction of between 2% -- 15% for each language. Da Brill (1992) presenterte sin enkle regelbaserte ordklasse-tagger ble det igjen aktuelt å bruke regelbaserte system for tagging av ordklasser. Taggerens grunnlag er en algoritme som automatisk lærer seg transformasjonsregler fra et korpus. I tillegg til at taggeren yter like bra som moderne stokastiske metoder for ordklasse-tagging har Brill-taggeren den fordelen at reglene den lærer seg kan presenteres i et format som lett kan oppfattes av mennesker. Til tross for sine styrker er Brill-taggeren relativt språkavhengig ettersom den fungerer mye bedre for språk som ligner engelsk enn språk med rikere morfologi. Denne oppgaven forsøker å løse dette problemet gjennom å definere regelmaler automatisk med et søk som er optimert med Genetiske Algoritmer. Dette lar Brill GA-taggeren søke gjennom et mye større område enn den ellers kunne ha gjort etter maler som i sin tur genererer regler som er tilpasset målspråket, hvilket også har fordelen at forskere ikke trenger å definere regelmaler manuelt. Brill GA-taggeren yter signifikant bedre (p<0.001) enn Brill-taggeren på alle 9 målspråk (Kinesisk, Japansk, Tyrkisk, Slovensk, Portugisisk, Engelsk, Nederlandsk, Svensk og Islandsk), med en feilprosent som er mellom 2% og 15% lavere i alle språk. När Brill (1992) presenterade sin enkla regelbaserade ordklasstaggare blev det återigen aktuellt att använda regelbaserade system för taggning av ordklasser. Taggaren är baserad på en algoritm som automatiskt lär sig transformationsregler från en korpus. Bortsett från att taggaren fungerar lika bra som moderna stokastiska metoder för ordklasstaggning har den också fördelen att reglerna som den lär sig kan presenteras i ett format som lätt kan läsas av människor. Trots sina styrkor är Brill-taggeren relativt språkberoende i och med att den fungerar mycket bättre för språk som liknar engelska än för språk med rikare morfologi. Den här uppsatsen försöker att lösa detta problem genom att definiera regelmallar automatiskt med en sökning som är optimerad med Genetiska Algoritmer. Detta gör att Brill GA-taggaren kan söka genom ett mycket större område än den annars skulle ha kunnat göra efter mallar som i sin tur genererar regler som är anpassade för målspråket. Detta har också fördelen att forskare inte behöver definiera regelmallar manuellt. Brill GA-taggeren får signifikant bättre träffsäkerhet (p<0.001) än Brill-taggeren på alla 9 målspråken (Kinesiska, Japanska, Turkiska, Slovenska, Portugisiska, Engelska, Nederländska, Svenska och Isländska), med en felprocent som är mellan 2% och 15% lägre för alla språk.

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Dey, Ken. "The effectiveness of social media in advancing transformational change." Thesis, Gonzaga University, 2013. http://pqdtopen.proquest.com/#viewpdf?dispub=1537849.

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The goal of this study is to determine the effectiveness of social media in advancing transformational change. Successfully implementing transformational change in an organization is heavily dependent on the support of key stakeholders. But engaging those stakeholders requires effective communication. Transformational efforts often fail because of the lack of credible communication or a failure to define a vision that can be easily communicated (Kotter, 2007). Researchers say that the key to successful transformational change is embracing a communication based in the realm of conversation where there is genuine two-way dialogue that is focused on listening and probing for more information (Dobbs, 2010). Creating conversations is a key component of social media: a platform of online tools designed to connect people and easily share information (Jue, Marr & Kassotakis, 2010). Social media has the potential to achieve employee engagement, enhance productivity and increase collaboration (Ou, C. J., Davison, R. M., Zhong, X., & Liang, Y.,2010). To determine the effectiveness of social media at driving transformational change a study of existing literature related to transformational change and social media was coupled with a qualitative and quantitative study of organizational users of social media and stakeholders of those organizations. The study employed both a questionnaire and interviews. Results showed a clear preference for the use of social media as an effective form of relationship development and effective communication, but a challenge remains on how organizations can best use social media to create and sustain the relationships required to accomplish transformational change.

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Cassone, Marco. "Characteristics of transformative listening enacted by organization development practitioners." Thesis, Pepperdine University, 2015. http://pqdtopen.proquest.com/#viewpdf?dispub=1571597.

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This study examined the listening behaviors of organization development (OD) practitioners that result in client transformation. Interviews conducted with eleven OD consultants with extensive experience in executive coaching pointed to engaged, focused attention as a core characteristic of their listening. OD practitioners regularly use three primary listening approaches (active, empathetic, and expansive listening) to drive insight and help clients transform their perspectives. Practitioners subsequently use two secondary listening approaches (critical and reductive listening) to anchor insight into action and help clients transform their behavior. Transformative listening describes the repeating process of inquiry that blends primary and secondary listening approaches and tends to transform client perspectives and behavior. Conversely, transactional listening describes a listening approach appropriate for the negotiation and execution of agreements in the transaction of routine business. Self-awareness and use of self foster sensitivity to client needs and practitioner agility in blending the listening approaches used in transformative listening.

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Douros, Ioannis. "Towards a 3 dimensional dynamic generic speaker model to study geometry simplifications of the vocal tract using magnetic resonance imaging data." Electronic Thesis or Diss., Université de Lorraine, 2020. http://www.theses.fr/2020LORR0115.

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Dans cette thèse, nous avons utilisé les données de l’IRM du conduit vocal pour étudier la production de la parole. La première partie consiste en l’étude de l’impact que le vélum, l’épiglotte et la position de la tête a sur la phonation de cinq voyelles françaises. Des simulations acoustiques ont été utilisées pour comparer les formants des cas étudiés avec la référence afin de mesurer leur impact. Pour cette partie du travail, nous avons utilisé des IRM statiques en 3D. Comme la parole est généralement une phénomène dynamique une question s’est posée, à savoir s’il serait possible de traiter les données 3D afin d’incorporer des informations temporelles de la parole continue. Par conséquent, la deuxième partie présente quelques algorithmes que l’on peut utiliser pour améliorer les données de production de la parole. Plusieurs transformations d’images ont été combinées afin de générer des estimations des formes du conduit vocal qui sont plus informatives que les originales. À ce stade, nous avons envisagé, outre l’amélioration des données de production de la parole, de créer un modèle de référence générique qui pourrait fournir des informations améliorées non pas pour un sujet spécifique, mais globalement pour la parole. C’est pourquoi nous avons consacré la troisième partie l’étude d’un algorithme permettant de créer un atlas spatio-temporel de l’appareil vocal qui peut être utilisé comme référence ou standard pour l’étude de la parole car il est indépendant du locuteur. Enfin, la dernière partie de la thèse, fait référence à une sélection de questions ouvertes du domaine qui restent encore sans réponse, quelques pistes intéressantes que l’on peut développer à partir de cette thèse et quelques approches potentielles qui pourraient être envisager afin de répondre à ces questions In this thesis we used MRI (Magnetic Resonance Imaging) data of the vocal tract to study speech production. The first part consist of the study of the impact that the velum, the epiglottis and the head position has on the phonation of five french vowels. Acoustic simulations were used to compare the formants of the studied cases with the reference in order to measure their impact. For this part of the work, we used 3D static MR (Magnetic Resonance) images. As speech is usually a dynamic phenomenon, a question arose, whether it would be possible to process the 3D data in order to incorporate dynamic information of continuous speech. Therefore the second part presents some algorithms that one can use in order to enhance speech production data. Several image transformations were combined in order to generate estimations of vocal tract shapes which are more informative than the original ones. At this point, we envisaged apart from enhancing speech production data, to create a generic speaker model that could provide enhanced information not for a specific subject, but globally for speech. As a result, we devoted the third part in the investigation of an algorithm that one can use to create a spatiotemporal atlas of the vocal tract which can be used as a reference or standard speaker for speech studies as it is speaker independent. Finally, the last part of the thesis, refers to a selection of open questions of the field that are still left unanswered, some interesting directions that one can expand this thesis and some potential approaches that could help someone move forward towards these directions

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Noland, Aaron K. "THE RELATIONSHIP BETWEEN TEACHER TRANSFORMATIONAL LEADERSHIP AND STUDENT OUTCOMES." Miami University / OhioLINK, 2005. http://rave.ohiolink.edu/etdc/view?acc_num=miami1123168677.

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Jiad, Abdul-Hadi Hussein. "The employment of figurative language in Arabic political speeches and its transformation into English." Thesis, SOAS, University of London, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.434596.

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46

Tahir, Muhammad Ali Verfasser], Hermann [Akademischer Betreuer] [Ney, and Reinhold [Akademischer Betreuer] Häb-Umbach. "Discriminative training of linear transformations and mixture density splitting for speech recognition / Muhammad Ali Tahir ; Hermann Ney, Reinhold Häb-Umbach." Aachen : Universitätsbibliothek der RWTH Aachen, 2015. http://d-nb.info/113065950X/34.

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Tahir, Muhammad Ali [Verfasser], Hermann [Akademischer Betreuer] Ney, and Reinhold [Akademischer Betreuer] Häb-Umbach. "Discriminative training of linear transformations and mixture density splitting for speech recognition / Muhammad Ali Tahir ; Hermann Ney, Reinhold Häb-Umbach." Aachen : Universitätsbibliothek der RWTH Aachen, 2015. http://d-nb.info/113065950X/34.

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Serna, Calvo Eva Teresa. "Diagnostics of rotor asymmetries in inverter fed, variable speed induction machines." Berlin Logos-Verl, 2009. http://d-nb.info/99519422X/04.

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Rachman, Laura. "The "other-voice" effect : how speaker identity and language familiarity influence the way we process emotional speech." Thesis, Sorbonne université, 2018. http://www.theses.fr/2018SORUS280.

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L’objectif théorique de cette thèse est d’étudier le rôle de la familiarité vocale sur le traitement de la voix émotionnelle. Les chapitres 2 et 3 présentent des études comportementales et électrophysiologiques portant sur les contributions spécifiques de la voix du self et la voix de l’autre sur le traitement de la parole émotionnelle. En comparant le self et l’autre, la familiarité est évaluée ici à un niveau personnel. Les résultats du chapitre 2 montrent une dissociation chez les participants des traitements explicites et implicites de leur propre voix. Alors que la discrimination explicite de leur propre voix émotionnelle est réduite, le traitement implicite de soi entraîne un avantage pour la reconnaissance des émotions et la discrimination du locuteur. Le chapitre 3 montre que les voix inconnues sont priorisées par rapport à la voix du self dans le traitement des changements émotionnels et acoustiques de bas niveau, par des réponses électrophysiologiques (EEG) et comportementales plus rapides. Au chapitre 4, l’effet de la familiarité sur la perception des émotions vocales est évalué au niveau socioculturel en comparant la langue maternelle et étrangère. Au travers de ces études, cette thèse met en évidence les différentes manières par lesquelles «l’étrangeté» d’une voix - qu’il s’agisse d’un locuteur autre que le soi ou d’une langue étrangère - est traitée avec une priorité plus élevée, mais une précision acoustique diminuée The human voice is a powerful tool to convey emotions. Humans hear voices on a daily basis and are able to rapidly extract relevant information to successfully interact with others. The theoretical aim of this thesis is to investigate the role of familiarity on emotional voice processing. Chapters 2 and 3 present behavioral and electrophysiological studies investigating how self- versus non self-produced voices influence the processing of emotional speech utterances. By contrasting self and other, familiarity is here assessed at a personal level. The results of Chapter 2 show a dissociation of explicit and implicit processing of the self-voice. While explicit discrimination of an emotional self-voice and other-voice was somewhat impaired, implicit self-processing prompted a self-advantage in emotion recognition and speaker discrimination. Chapter 3 reports a prioritization for the non-self voice in the processing of emotional and low-level acoustic changes, reflected in faster electrophysiological (EEG) and behavioral responses. In Chapter 4, the effect of voice familiarity on is assessed at a larger sociocultural scale by comparing speech utterances in the native and a foreign language. Taken together, this thesis highlights some ways in which the ‘otherness’ of a voice - whether a non-self speaker or a foreign language speaker - is processed with a higher priority on the one hand, but with less acoustic precision on the other hand

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Pieper, Christoph [Verfasser], Michael [Gutachter] Beckmann, and Eckehard [Gutachter] Specht. "Transformation of the German energy system - Towards photovoltaic and wind power : Technology Readiness Levels 2018 / Christoph Pieper ; Gutachter: Michael Beckmann, Eckehard Specht." Dresden : Technische Universität Dresden, 2019. http://d-nb.info/1226942334/34.

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Thèses sur le sujet « Speech transformation »

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