Parametric speech synthesis has received increased attention in recent years following the development of statistical HMM-based speech synthesis. However, the speech produced using this method still does not sound as natural as human speech and there is limited parametric flexibility to replicate voice quality aspects, such as breathiness. The hypothesis of this thesis is that speech naturalness and voice quality can be more accurately replicated by a HMM-based speech synthesiser using an acoustic glottal source model, the Liljencrants-Fant (LF) model, to represent the source component of speech instead of the traditional impulse train. Two different analysis-synthesis methods were developed during this thesis, in order to integrate the LF-model into a baseline HMM-based speech synthesiser, which is based on the popular HTS system and uses the STRAIGHT vocoder. The first method, which is called Glottal Post-Filtering (GPF), consists of passing a chosen LF-model signal through a glottal post-filter to obtain the source signal and then generating speech, by passing this source signal through the spectral envelope filter. The system which uses the GPF method (HTS-GPF system) is similar to the baseline system, but it uses a different source signal instead of the impulse train used by STRAIGHT. The second method, called Glottal Spectral Separation (GSS), generates speech by passing the LF-model signal through the vocal tract filter. The major advantage of the synthesiser which incorporates the GSS method, named HTS-LF, is that the acoustic properties of the LF-model parameters are automatically learnt by the HMMs. In this thesis, an initial perceptual experiment was conducted to compare the LFmodel to the impulse train. The results showed that the LF-model was significantly better, both in terms of speech naturalness and replication of two basic voice qualities (breathy and tense). In a second perceptual evaluation, the HTS-LF system was better than the baseline system, although the difference between the two had been expected to be more significant. A third experiment was conducted to evaluate the HTS-GPF system and an improved HTS-LF system, in terms of speech naturalness, voice similarity and intelligibility. The results showed that the HTS-GPF system performed similarly to the baseline. However, the HTS-LF system was significantly outperformed by the baseline. Finally, acoustic measurements were performed on the synthetic speech to investigate the speech distortion in the HTS-LF system. The results indicated that a problem in replicating the rapid variations of the vocal tract filter parameters at transitions between voiced and unvoiced sounds is the most significant cause of speech distortion. This problem encourages future work to further improve the system.

This thesis explores, through a number of production andperception studies, the nature of the voice source signal andhow it varies in spoken communication. Research is alsopresented that deals with the techniques and methodologies foranalysing and synthesising the voice source. The main analytictechnique involves interactive inverse filtering for obtainingthe source signal, which is then parameterised to permit thequantification of source characteristics. The parameterisationis carried by means of model matching, using the four-parameterLF model of differentiated glottal flow. The first three analytic studies focus on segmental andsuprasegmental determinants of source variation. As part of theprosodic variation of utterances, focal stress shows for theglottal excitation an enhancement between the stressed voweland the surrounding consonants. At a segmental level, the voicesource characteristics of a vowel show potentially majordifferences as a function of the voiced/voiceless nature of anadjacent stop. Cross-language differences in the extent anddirectionality of the observed effects suggest differentunderlying control strategies in terms of the timing of thelaryngeal and supralaryngeal gestures, as well as in thelaryngeal tensions settings. Different classes of voicedconsonants also show differences in source characteristics:here the differences are likely to be passive consequences ofthe aerodynamic conditions that are inherent to the consonants.Two further analytic studies present voice source correlatesfor six different voice qualities as defined by Laver'sclassification system. Data from stressed and unstressedcontexts clearly show that the transformation from one voicequality to another does not simply involve global changes ofthe source parameters. As well as providing insights into theseaspects of speech production, the analytic studies providequantitative measures useful in technology applications,particularly in speech synthesis. The perceptual experiments use the LF source implementationin the KLSYN88 synthesiser to test some of the analytic resultsand to harness them to explore the paralinguistic dimension ofspeech communication. A study of the perceptual salience ofdifferent parameters associated with breathy voice indicatesthat the source spectral slope is critically important andthat, surprisingly, aspiration noise contributes relativelylittle. Further perceptual tests using stimuli with differentvoice qualities explore the mapping between voice quality andits paralinguistic function of expressing emotion, mood andattitude. The results of these studies highlight the crucialrole of voice quality in expressing affect as well as providingpointers to how it combines withf0for this purpose. The last section of the thesis focuses on the techniquesused for the analysis and synthesis of the source. Asemi-automatic method for inverse filtering is presented, whichis novel in that it optimises the inverse filter by exploitingthe knowledge that is typically used by the experimenter whencarrying out manual interactive inverse filtering. A furtherstudy looks at the properties of the modified LF model in theKLSYN88 synthesiser: it highlights how it differs from thestandard LF model and discusses the implications forsynthesising the glottal source signal from LF model data.Effective and robust source parameterisation for the analysisof voice quality is the topic of the final paper: theeffectiveness of global, amplitude-based, source parameters isexamined across speech tokens with large differences inf0. Additional amplitude-based parameters areproposed to enable a more detailed characterisation of theglottal pulse. <b>Keywords:</b>Voice source dynamics, glottal sourceparameters, source-filter interaction, voice quality,phonation, perception, affect, emotion, mood, attitude,paralinguistic, inverse filtering, knowledge-based, formantsynthesis, LF model, fundamental frequency,f0.

<p>This thesis explores, through a number of production andperception studies, the nature of the voice source signal andhow it varies in spoken communication. Research is alsopresented that deals with the techniques and methodologies foranalysing and synthesising the voice source. The main analytictechnique involves interactive inverse filtering for obtainingthe source signal, which is then parameterised to permit thequantification of source characteristics. The parameterisationis carried by means of model matching, using the four-parameterLF model of differentiated glottal flow.</p><p>The first three analytic studies focus on segmental andsuprasegmental determinants of source variation. As part of theprosodic variation of utterances, focal stress shows for theglottal excitation an enhancement between the stressed voweland the surrounding consonants. At a segmental level, the voicesource characteristics of a vowel show potentially majordifferences as a function of the voiced/voiceless nature of anadjacent stop. Cross-language differences in the extent anddirectionality of the observed effects suggest differentunderlying control strategies in terms of the timing of thelaryngeal and supralaryngeal gestures, as well as in thelaryngeal tensions settings. Different classes of voicedconsonants also show differences in source characteristics:here the differences are likely to be passive consequences ofthe aerodynamic conditions that are inherent to the consonants.Two further analytic studies present voice source correlatesfor six different voice qualities as defined by Laver'sclassification system. Data from stressed and unstressedcontexts clearly show that the transformation from one voicequality to another does not simply involve global changes ofthe source parameters. As well as providing insights into theseaspects of speech production, the analytic studies providequantitative measures useful in technology applications,particularly in speech synthesis.</p><p>The perceptual experiments use the LF source implementationin the KLSYN88 synthesiser to test some of the analytic resultsand to harness them to explore the paralinguistic dimension ofspeech communication. A study of the perceptual salience ofdifferent parameters associated with breathy voice indicatesthat the source spectral slope is critically important andthat, surprisingly, aspiration noise contributes relativelylittle. Further perceptual tests using stimuli with differentvoice qualities explore the mapping between voice quality andits paralinguistic function of expressing emotion, mood andattitude. The results of these studies highlight the crucialrole of voice quality in expressing affect as well as providingpointers to how it combines with<i>f</i>₀for this purpose.</p><p>The last section of the thesis focuses on the techniquesused for the analysis and synthesis of the source. Asemi-automatic method for inverse filtering is presented, whichis novel in that it optimises the inverse filter by exploitingthe knowledge that is typically used by the experimenter whencarrying out manual interactive inverse filtering. A furtherstudy looks at the properties of the modified LF model in theKLSYN88 synthesiser: it highlights how it differs from thestandard LF model and discusses the implications forsynthesising the glottal source signal from LF model data.Effective and robust source parameterisation for the analysisof voice quality is the topic of the final paper: theeffectiveness of global, amplitude-based, source parameters isexamined across speech tokens with large differences in<i>f</i>₀. Additional amplitude-based parameters areproposed to enable a more detailed characterisation of theglottal pulse.</p><p><b>Keywords:</b>Voice source dynamics, glottal sourceparameters, source-filter interaction, voice quality,phonation, perception, affect, emotion, mood, attitude,paralinguistic, inverse filtering, knowledge-based, formantsynthesis, LF model, fundamental frequency,<i>f</i>₀.</p>

Muitas pesquisas tem sido feitas em processamento digital de sinais (PDS) na tentativa de se avaliar o sinal de fala para diagnosticar doenças da laringe. Medidas acústicas têm sido propostas de forma a avaliar indiretamente o trato glotal por meio do sinal de voz coletado através de microfone convencional. Para isso, o modelo paramétrico Liljencrants-Fant (LF) foi desenvolvido para representar o sinal glotal em condições normais e patológicas. Tais parâmetros apresentam vantagens sobre medidas acústicas por possuírem características fisiológicas reais das pregas vocais. Assim, podendo ser empregados para identificação de doenças da laringe. Além da estimação dos parâmetros LF, no domínio do tempo (parâmetros T), a forma de onda da derivativa glotal também pôde ser quantificada através dos parâmetros identificados na literatura por parâmetros R (Rd, Ra, Rk e Rg), parâmetros quocientes Q (SQ, OQ, CQ, AQ e NAQ), parâmetros B1 e B2 que são as extensões de bandas do pulso derivativo LF, e o parâmetro ece, que relaciona os parâmetros &#946 e Ta. Os parâmetros B1 e B2 e ece apesar de serem propostos na literatura, não são encontrados resultados diferentes a essas duas medidas. Os resultados mostraram que os parâmetros B não foram confiáveis na discriminação entre as vozes, por outro lado, o parâmetro ece mostrou-se ser opção na discriminação entre as vozes normais, nódulo e Reinke. O objetivo deste trabalho é direcionar a atenção sobre o sinal glotal, estimando-o automaticamente mediante técnicas de PDS aplicadas ao sinal de fala, visando extrair parâmetros que identifiquem as condições normais e patológicas da laringe. Por fim foram propostos os parâmetros TRp e TRs, visando dissociar os efeitos de primeira ordem dos de ordem superior na fase de retorno do pulso glotal com a finalidade de estimar a real não-linearidade do sub-sistema glotal, retratando as condições normais e patológicas da laringe. Por fim foram propostos os parâmetros TRp e TRs, visando dissociar os efeitos de primeira ordem dos de ordem superior na fase de retorno do pulso glotal com a finalidade de estimar a real não-linearidade do sub-sistema glotal, retratando as condições fisiológicas do movimento das pregas vocais. Com um nível de confiança de 95%, o parâmetro de primeira ordem (TRp) é efetivo na discriminação do Edema de Reinke, porém mostrou-se ineficaz na detecção do nódulo. Em relação ao parâmetro de ordem superior, conclui-se que o TRs é um excelente detetor de vozes patológicas (nódulo e Edema de Reinke), porém não é capaz de discriminar as patologias. / Many researches has been conducted in digital signal processing (DSP) atempting to evaluate the physiological conditions of larynx. Acoustical parameters have been proposed to evaluate the glotal tract from voice signal. One technique proposed is the Liljencrants-Fant model (LF) developed to represent normal and pathologic conditions of the larynx. Those parameters compare favourably as far as real physiologic characteristic of vocal folds is concerned. So, a primary use of the model is the larynx pathologic identification. Beyond LF parameters estimation, (T parameters in the time domain), the waveform of glotal pulse derivative also can be quantified through, R parameters (Rd, Ra, Rk and Rg), quocient parameters (SQ, OQ, CQ, AQ and NAQ), B parameters (B1 and B2) that are band extension of the LF glotal pulse derivative and the ece parameter that in fact, is a relationship between &#946 and Ta. Although proposed in the literature, no results are found, related to B and ece parameters. Our founds show that B parameters do not present good results in voice discrimination, however, ece parameter seems to be good option to discriminate normal voice, nodulo and Reinke edema. The main purpose of this work is to estimate the glotal signal from the voice signal using DSP techniques in order to obtain parameters that identifies the physiological larynx condition. In order to estimate the shape of return phase of glotal pulse, twoparameters have been proposed in this work. The first one evaluates the pulse (TRp, in other words, the first order component of the return phase. The second is responsible to evaluate superior orders components of the return phase (TRs), i.e, the non-linear component of the glotal pulse. With 95% of confidence level, TRp is effective in Reinke edema discrimination however it is inefficient for nodule e dection. By the other hand, the TRs parameter works well to detect pathologic voice however is unable to discriminated them.

Muitas pesquisas tem sido feitas em processamento digital de sinais (PDS) na tentativa de se avaliar o sinal de fala para diagnosticar doenças da laringe. Medidas acústicas têm sido propostas de forma a avaliar indiretamente o trato glotal por meio do sinal de voz coletado através de microfone convencional. Para isso, o modelo paramétrico Liljencrants-Fant (LF) foi desenvolvido para representar o sinal glotal em condições normais e patológicas. Tais parâmetros apresentam vantagens sobre medidas acústicas por possuírem características fisiológicas reais das pregas vocais. Assim, podendo ser empregados para identificação de doenças da laringe. Além da estimação dos parâmetros LF, no domínio do tempo (parâmetros T), a forma de onda da derivativa glotal também pôde ser quantificada através dos parâmetros identificados na literatura por parâmetros R (Rd, Ra, Rk e Rg), parâmetros quocientes Q (SQ, OQ, CQ, AQ e NAQ), parâmetros B1 e B2 que são as extensões de bandas do pulso derivativo LF, e o parâmetro ece, que relaciona os parâmetros &#946 e Ta. Os parâmetros B1 e B2 e ece apesar de serem propostos na literatura, não são encontrados resultados diferentes a essas duas medidas. Os resultados mostraram que os parâmetros B não foram confiáveis na discriminação entre as vozes, por outro lado, o parâmetro ece mostrou-se ser opção na discriminação entre as vozes normais, nódulo e Reinke. O objetivo deste trabalho é direcionar a atenção sobre o sinal glotal, estimando-o automaticamente mediante técnicas de PDS aplicadas ao sinal de fala, visando extrair parâmetros que identifiquem as condições normais e patológicas da laringe. Por fim foram propostos os parâmetros TRp e TRs, visando dissociar os efeitos de primeira ordem dos de ordem superior na fase de retorno do pulso glotal com a finalidade de estimar a real não-linearidade do sub-sistema glotal, retratando as condições normais e patológicas da laringe. Por fim foram propostos os parâmetros TRp e TRs, visando dissociar os efeitos de primeira ordem dos de ordem superior na fase de retorno do pulso glotal com a finalidade de estimar a real não-linearidade do sub-sistema glotal, retratando as condições fisiológicas do movimento das pregas vocais. Com um nível de confiança de 95%, o parâmetro de primeira ordem (TRp) é efetivo na discriminação do Edema de Reinke, porém mostrou-se ineficaz na detecção do nódulo. Em relação ao parâmetro de ordem superior, conclui-se que o TRs é um excelente detetor de vozes patológicas (nódulo e Edema de Reinke), porém não é capaz de discriminar as patologias. / Many researches has been conducted in digital signal processing (DSP) atempting to evaluate the physiological conditions of larynx. Acoustical parameters have been proposed to evaluate the glotal tract from voice signal. One technique proposed is the Liljencrants-Fant model (LF) developed to represent normal and pathologic conditions of the larynx. Those parameters compare favourably as far as real physiologic characteristic of vocal folds is concerned. So, a primary use of the model is the larynx pathologic identification. Beyond LF parameters estimation, (T parameters in the time domain), the waveform of glotal pulse derivative also can be quantified through, R parameters (Rd, Ra, Rk and Rg), quocient parameters (SQ, OQ, CQ, AQ and NAQ), B parameters (B1 and B2) that are band extension of the LF glotal pulse derivative and the ece parameter that in fact, is a relationship between &#946 and Ta. Although proposed in the literature, no results are found, related to B and ece parameters. Our founds show that B parameters do not present good results in voice discrimination, however, ece parameter seems to be good option to discriminate normal voice, nodulo and Reinke edema. The main purpose of this work is to estimate the glotal signal from the voice signal using DSP techniques in order to obtain parameters that identifies the physiological larynx condition. In order to estimate the shape of return phase of glotal pulse, twoparameters have been proposed in this work. The first one evaluates the pulse (TRp, in other words, the first order component of the return phase. The second is responsible to evaluate superior orders components of the return phase (TRs), i.e, the non-linear component of the glotal pulse. With 95% of confidence level, TRp is effective in Reinke edema discrimination however it is inefficient for nodule e dection. By the other hand, the TRs parameter works well to detect pathologic voice however is unable to discriminated them.

The work is about the estimation of vocal pulses from the speech record. Contains a description of the process of speech production, description of the instruments for the measurement of vocal pulses, an overview of software tools for estimating vocal pulses from the speech signal. Description of IAIF and Sahoo method for estimating vocal pulses. The Graphic User Interface in MATLAB is created for easier control of mentioned methods.

L’analyse des signaux de parole permet de comprendre le fonctionnement de l’appareil vocal, mais aussi de décrire de nouveaux paramètres permettant de qualifier et quantifier la perception de la voix. Dans le cas de la parole expressive, l'intérêt se porte sur des variations importantes de qualité vocales et sur leurs liens avec l’expressivité et l’intention du sujet. Afin de décrire ces liens, il convient de pouvoir estimer les paramètres du modèle de production mais aussi de décomposer le signal vocal en chacune des parties qui contribuent à ce modèle. Le travail réalisé au cours de cette thèse s’axe donc autour de la segmentation et la décomposition des signaux vocaux et de l’estimation des paramètres du modèle de production vocale : Tout d’abord, la décomposition multi-échelles des signaux vocaux est abordée. En reprenant la méthode LoMA qui trace des lignes suivant les amplitudes maximum sur les réponses temporelles au banc de filtre en ondelettes, il est possible d’y détecter un certain nombre de caractéristiques du signal vocal : les instants de fermeture glottique, l’énergie associée à chaque cycle ainsi que sa distribution spectrale, le quotient ouvert du cycle glottique (par l’observation du retard de phase du premier harmonique). Cette méthode est ensuite testée sur des signaux synthétiques et réels. Puis, la décomposition harmonique + bruit des signaux vocaux est abordée. Une méthode existante (PAPD - Périodic/APériodic Décomposition) est adaptée aux variations de fréquence fondamentale par le biais de la variation dynamique de la taille de la fenêtre d’analyse et est appelée PAP-A. Cette nouvelle méthode est ensuite testée sur une base de signaux synthétiques. La sensibilité à la précision d’estimation de la fréquence fondamentale est notamment abordée. Les résultats montrent des décompositions de meilleures qualité pour PAP-A par rapport à PAPD. Ensuite, le problème de la déconvolution source/filtre est abordé. La séparation source/filtre par ZZT (zéros de la transformée en Z) est comparée aux méthodes usuelles à base de prédiction linéaire. La ZZT est utilisée pour estimer les paramètres du modèle de la source glottique via une méthode simple mais robuste qui permet une estimation conjointe de deux paramètres du débit glottique : le quotient ouvert et l'asymétrie. La méthode ainsi développée est testée et combinée à l’estimation du quotient ouvert par ondelettes. Finalement, ces trois méthodes d’estimations sont appliquées à un grand nombre de fichiers d’une base de données comportant différents styles d’élocution. Les résultats de cette analyse sont discutés afin de caractériser le lien entre style, valeur des paramètres de la production vocale et qualité vocale. On constate notamment l’émergence très nette de groupes de styles. / Analysis of speech signals is a good way of understanding how the voice is produced, but it is also important as a way of describing new parameters in order to define the perception of voice quality. This study focuses on expressive speech, where voice quality varies a lot and is explicitly linked to the expressivity or intention of the speaker. In order to define those links, one has to be able to estimate a high number of parameters of the speech production model, but also be able to decompose the speech signal into each parts that contributes to this model. The work presented in this thesis addresses the segmentation of speech signals, their decomposition and the estimation of the voice production model parameters. At first, multi-scale analysis of speech signals is studied. Using the LoMA method that traces lines across scales from one maximum to the other on the time domain response of a wavelet filter bank, it is possible to detect a number of features on voiced speech, namely : the glottal closing instants, the energy associated to each glottal cycle, the open quotient (by estimating the time delay of the first harmonic). This method is then tested on both synthetic and real speech. Secondly, harmonic plus noise decomposition of speech signals is studied. An existing method (PAPD standing for Periodic/Aperiodic Decomposition) is modified to dynamically adapt the analysis window length to the fundamental frequency (F0) of the signal. The new method is then tested on synthetic speech where the sensibility to the estimation error on F0 is also discussed. Decomposition on real speech, along with their audio files, are also discussed. Results shows that this new method provides better quality of decomposition. Thirdly, the problem of source/filter deconvolution is addressed. The ZZT (Zeros of the Z Transform) method is compared to classical methods based on linear prediction. ZZT is then used for the estimation of the glottal flow parameters with a simple but robust method based on the joint estimation of both the open quotient and the asymmetry. The later method is then combined to the estimation of the open quotient using wavelet analysis. Finally, the three estimation methods developed in this thesis are used to analyze a large number of files from a database presenting different speaking styles. Results are discussed in order to characterize the link between style, model parameters and voice quality. We especially notice the neat appearance of speaking style groups