On the Computation of the Kullback-Leibler Measure for Spectral Distances

Efficient algorithms for the exact and approximate computation of the symmetrical Kullback-Leibler (1998) measure for spectral distances are presented for linear predictive coding (LPC) spectra. A interpretation of this measure is given in terms of the poles of the spectra. The performances of the algorithms in terms of accuracy and computational complexity are assessed for the application of computing concatenation costs in unit-selection-based speech synthesis. With the same complexity and storage requirements, the exact method is superior in terms of accuracy

Meta Learning Approach to Phone Duration Modeling

One of the essential prerequisites for achieving the naturalness of synthesized speech is the possibility of the automatic prediction of phone duration, due to the high importance of segmental duration in speech perception. In this paper we present a new phone duration prediction model for the Serbian language using meta learning approach. Based on the data obtained from the analysis of a large speech database, we used a feature set of 21 parameters describing phones and their contexts. These include attributes related to the segmental identity, manner of articulation (for consonants), attributes related to phonological context, such as segment types and voicing values of neighboring phones, presence or absence of lexical stress, morphological attributes, such as part-of-speech, and prosodic attributes, such as phonological word length, the position of the segment in the syllable, the position of the syllable in a word, the position of a word in a phrase, phrase break level, etc. Phone duration model obtained using meta learning algorithm outperformed the best individual model by approximately 2,0% and 1,7% in terms of the relative reduction of the root-mean-squared error and the mean absolute error, respectively

Tagging Prosody and Discourse Structure in Elicited Spontaneous Speech

This paper motivates and describes the annotation and analysis of prosody and discourse structure for several large spoken language corpora. The annotation schema are of two types: tags for prosody and intonation, and tags for several aspects of discourse structure. The choice of the particular tagging schema in each domain is based in large part on the insights they provide in corpus-based studies of the relationship between discourse structure and the accenting of referring expressions in American English. We first describe these results and show that the same models account for the accenting of pronouns in an extended passage from one of the Speech Warehouse hotel-booking dialogues. We then turn to corpora described in Venditti [Ven00], which adapts the same models to Tokyo Japanese. Japanese is interesting to compare to English, because accent is lexically specified and so cannot mark discourse focus in the same way. Analyses of these corpora show that local pitch range expansion serves the analogous focusing function in Japanese. The paper concludes with a section describing several outstanding questions in the annotation of Japanese intonation which corpus studies can help to resolve.Work reported in this paper was supported in part by a grant from the Ohio State University Office of Research, to Mary E. Beckman and co-principal investigators on the OSU Speech Warehouse project, and by an Ohio State University Presidential Fellowship to Jennifer J. Venditti

A survey on mouth modeling and analysis for Sign Language recognition

© 2015 IEEE.Around 70 million Deaf worldwide use Sign Languages (SLs) as their native languages. At the same time, they have limited reading/writing skills in the spoken language. This puts them at a severe disadvantage in many contexts, including education, work, usage of computers and the Internet. Automatic Sign Language Recognition (ASLR) can support the Deaf in many ways, e.g. by enabling the development of systems for Human-Computer Interaction in SL and translation between sign and spoken language. Research in ASLR usually revolves around automatic understanding of manual signs. Recently, ASLR research community has started to appreciate the importance of non-manuals, since they are related to the lexical meaning of a sign, the syntax and the prosody. Nonmanuals include body and head pose, movement of the eyebrows and the eyes, as well as blinks and squints. Arguably, the mouth is one of the most involved parts of the face in non-manuals. Mouth actions related to ASLR can be either mouthings, i.e. visual syllables with the mouth while signing, or non-verbal mouth gestures. Both are very important in ASLR. In this paper, we present the first survey on mouth non-manuals in ASLR. We start by showing why mouth motion is important in SL and the relevant techniques that exist within ASLR. Since limited research has been conducted regarding automatic analysis of mouth motion in the context of ALSR, we proceed by surveying relevant techniques from the areas of automatic mouth expression and visual speech recognition which can be applied to the task. Finally, we conclude by presenting the challenges and potentials of automatic analysis of mouth motion in the context of ASLR

Chapter 2: The Original ToBI System and the Evolution of the ToBI Framework

In this chapter, the authors will try to identify the essential properties of a ToBI framework annotation system by describing the development and design of the original ToBI conventions. In this description, the authors will overview the general phonological theory and the specific theory of Mainstream American English intonation and prosody that the authors decided to incorporate in the original ToBI tags. The authors will also state the practical principles that led us to make the decisions that the authors did. The chapter is organised as follows. Section 2.2 briefly chronicles how the MAE_ToBI system came into being. Section 2.3 briefly describes the consensus account of English intonation and prosody on which the MAE_ToBI system is based. Section 2.4 catalogues the different components of a MAE_ToBI transcription and lists the salient rules which constrain the relationships between different components. This section also expands upon the theoretical foundations and practical consequences of adopting the general structure of multiple labelling tiers, and particularly the separation of the labels for tones from the labels for indexing prosodic boundary strength. Section 2.5 then describes some of the extensions of the basic ToBI tiers that have been adopted by some sites. This section also compares our decisions about the number of tiers and about inter-tier constraints with the analogous decisions for some of the other ToBI systems described in this book. Section 2.6 discusses the status of the symbolic labels relative to the continuous phonetic records that are also an obligatory component of the MAE_ToBI transcription. Section 2.7 then closes by listing several open research questions that the authors would like to see addressed by MAE_ToBI users and the larger ToBI community

Adding expressiveness to unit selection speech synthesis and to numerical voice production

La parla és una de les formes de comunicació més naturals i directes entre éssers humans, ja que codifica un missatge i també claus paralingüístiques sobre l’estat emocional del locutor, el to o la seva intenció, esdevenint així fonamental en la consecució d’una interacció humà-màquina (HCI) més natural. En aquest context, la generació de parla expressiva pel canal de sortida d’HCI és un element clau en el desenvolupament de tecnologies assistencials o assistents personals entre altres aplicacions. La parla sintètica pot ser generada a partir de parla enregistrada utilitzant mètodes basats en corpus com la selecció d’unitats (US), que poden aconseguir resultats d’alta qualitat però d’expressivitat restringida a la pròpia del corpus. A fi de millorar la qualitat de la sortida de la síntesi, la tendència actual és construir bases de dades de veu cada cop més grans, seguint especialment l’aproximació de síntesi anomenada End-to-End basada en tècniques d’aprenentatge profund. Tanmateix, enregistrar corpus ad-hoc per cada estil expressiu desitjat pot ser extremadament costós o fins i tot inviable si el locutor no és capaç de realitzar adequadament els estils requerits per a una aplicació donada (ex: cant en el domini de la narració de contes). Alternativament, nous mètodes basats en la física de la producció de veu s’han desenvolupat a la darrera dècada gràcies a l’increment en la potència computacional. Per exemple, vocals o diftongs poden ser obtinguts utilitzant el mètode d’elements finits (FEM) per simular la propagació d’ones acústiques a través d’una geometria 3D realista del tracte vocal obtinguda a partir de ressonàncies magnètiques (MRI). Tanmateix, atès que els principals esforços en aquests mètodes de producció numèrica de veu s’han focalitzat en la millora del modelat del procés de generació de veu, fins ara s’ha prestat poca atenció a la seva expressivitat. A més, la col·lecció de dades per aquestes simulacions és molt costosa, a més de requerir un llarg postprocessament manual com el necessari per extreure geometries 3D del tracte vocal a partir de MRI. L’objectiu de la tesi és afegir expressivitat en un sistema que genera veu neutra, sense haver d’adquirir dades expressives del locutor original. Per un costat, s’afegeixen capacitats expressives a un sistema de conversió de text a parla basat en selecció d’unitats (US-TTS) dotat d’un corpus de veu neutra, per adreçar necessitats específiques i concretes en l’àmbit de la narració de contes, com són la veu cantada o situacions de suspens. A tal efecte, la veu és parametritzada utilitzant un model harmònic i transformada a l’estil expressiu desitjat d’acord amb un sistema expert. Es presenta una primera aproximació, centrada en la síntesi de suspens creixent per a la narració de contes, i es demostra la seva viabilitat pel que fa a naturalitat i qualitat de narració de contes. També s’afegeixen capacitats de cant al sistema US-TTS mitjançant la integració de mòduls de transformació de parla a veu cantada en el pipeline del TTS, i la incorporació d’un mòdul de generació de prosòdia expressiva que permet al mòdul de US seleccionar unitats més properes a la prosòdia cantada obtinguda a partir de la partitura d’entrada. Això resulta en un framework de síntesi de conversió de text a parla i veu cantada basat en selecció d’unitats (US-TTS&S) que pot generar veu parlada i cantada a partir d'un petit corpus de veu neutra (~2.6h). D’acord amb els resultats objectius, l’estratègia de US guiada per la partitura permet reduir els factors de modificació de pitch requerits per produir veu cantada a partir de les unitats de veu parlada seleccionades, però en canvi té una efectivitat limitada amb els factors de modificació de les durades degut a la curta durada de les vocals parlades neutres. Els resultats dels tests perceptius mostren que tot i òbviament obtenir una naturalitat inferior a la oferta per un sintetitzador professional de veu cantada, el framework pot adreçar necessitats puntuals de veu cantada per a la síntesis de narració de contes amb una qualitat raonable. La incorporació d’expressivitat s’investiga també en la simulació numèrica 3D de vocals basada en FEM mitjançant modificacions de les senyals d’excitació glotal utilitzant una aproximació font-filtre de producció de veu. Aquestes senyals es generen utilitzant un model Liljencrants-Fant (LF) controlat amb el paràmetre de forma del pols Rd, que permet explorar el continu de fonació lax-tens a més del rang de freqüències fonamentals, F0, de la veu parlada. S’analitza la contribució de la font glotal als modes d’alt ordre en la síntesis FEM de les vocals cardinals [a], [i] i [u] mitjançant la comparació dels valors d’energia d’alta freqüència (HFE) obtinguts amb geometries realistes i simplificades del tracte vocal. Les simulacions indiquen que els modes d’alt ordre es preveuen perceptivament rellevants d’acord amb valors de referència de la literatura, particularment per a fonacions tenses i/o F0s altes. En canvi, per a vocals amb una fonació laxa i/o F0s baixes els nivells d’HFE poden resultar inaudibles, especialment si no hi ha soroll d’aspiració en la font glotal. Després d’aquest estudi preliminar, s’han analitzat les característiques d’excitació de vocals alegres i agressives d’un corpus paral·lel de veu en castellà amb l’objectiu d’incorporar aquests estils expressius de veu tensa en la simulació numèrica de veu. Per a tal efecte, s’ha usat el vocoder GlottDNN per analitzar variacions d’F0 i pendent espectral relacionades amb l’excitació glotal en vocals [a]. Aquestes variacions es mapegen mitjançant la comparació amb vocals sintètiques en valors d’F0 i Rd per simular vocals que s’assemblin als estils alegre i agressiu. Els resultats mostren que és necessari incrementar l’F0 i disminuir l’Rd respecte la veu neutra, amb variacions majors per a alegre que per agressiu, especialment per a vocals accentuades. Els resultats aconseguits en les investigacions realitzades validen la possibilitat d’afegir expressivitat a la síntesi basada en corpus US-TTS i a la simulació numèrica de veu basada en FEM. Tanmateix, encara hi ha marge de millora. Per exemple, l’estratègia aplicada a la producció numèrica de veu es podria millorar estudiant i desenvolupant mètodes de filtratge invers així com incorporant modificacions del tracte vocal, mentre que el framework US-TTS&S es podria beneficiar dels avenços en tècniques de transformació de veu incloent transformacions de la qualitat de veu, aprofitant l’experiència adquirida en la simulació numèrica de vocals expressives.El habla es una de las formas de comunicación más naturales y directas entre seres humanos, ya que codifica un mensaje y también claves paralingüísticas sobre el estado emocional del locutor, el tono o su intención, convirtiéndose así en fundamental en la consecución de una interacción humano-máquina (HCI) más natural. En este contexto, la generación de habla expresiva para el canal de salida de HCI es un elemento clave en el desarrollo de tecnologías asistenciales o asistentes personales entre otras aplicaciones. El habla sintética puede ser generada a partir de habla gravada utilizando métodos basados en corpus como la selección de unidades (US), que pueden conseguir resultados de alta calidad, pero de expresividad restringida a la propia del corpus. A fin de mejorar la calidad de la salida de la síntesis, la tendencia actual es construir bases de datos de voz cada vez más grandes, siguiendo especialmente la aproximación de síntesis llamada End-to-End basada en técnicas de aprendizaje profundo. Sin embargo, gravar corpus ad-hoc para cada estilo expresivo deseado puede ser extremadamente costoso o incluso inviable si el locutor no es capaz de realizar adecuadamente los estilos requeridos para una aplicación dada (ej: canto en el dominio de la narración de cuentos). Alternativamente, nuevos métodos basados en la física de la producción de voz se han desarrollado en la última década gracias al incremento en la potencia computacional. Por ejemplo, vocales o diptongos pueden ser obtenidos utilizando el método de elementos finitos (FEM) para simular la propagación de ondas acústicas a través de una geometría 3D realista del tracto vocal obtenida a partir de resonancias magnéticas (MRI). Sin embargo, dado que los principales esfuerzos en estos métodos de producción numérica de voz se han focalizado en la mejora del modelado del proceso de generación de voz, hasta ahora se ha prestado poca atención a su expresividad. Además, la colección de datos para estas simulaciones es muy costosa, además de requerir un largo postproceso manual como el necesario para extraer geometrías 3D del tracto vocal a partir de MRI. El objetivo de la tesis es añadir expresividad en un sistema que genera voz neutra, sin tener que adquirir datos expresivos del locutor original. Per un lado, se añaden capacidades expresivas a un sistema de conversión de texto a habla basado en selección de unidades (US-TTS) dotado de un corpus de voz neutra, para abordar necesidades específicas y concretas en el ámbito de la narración de cuentos, como son la voz cantada o situaciones de suspense. Para ello, la voz se parametriza utilizando un modelo harmónico y se transforma al estilo expresivo deseado de acuerdo con un sistema experto. Se presenta una primera aproximación, centrada en la síntesis de suspense creciente para la narración de cuentos, y se demuestra su viabilidad en cuanto a naturalidad y calidad de narración de cuentos. También se añaden capacidades de canto al sistema US-TTS mediante la integración de módulos de transformación de habla a voz cantada en el pipeline del TTS, y la incorporación de un módulo de generación de prosodia expresiva que permite al módulo de US seleccionar unidades más cercanas a la prosodia cantada obtenida a partir de la partitura de entrada. Esto resulta en un framework de síntesis de conversión de texto a habla y voz cantada basado en selección de unidades (US-TTS&S) que puede generar voz hablada y cantada a partir del mismo pequeño corpus de voz neutra (~2.6h). De acuerdo con los resultados objetivos, la estrategia de US guiada por la partitura permite reducir los factores de modificación de pitch requeridos para producir voz cantada a partir de las unidades de voz hablada seleccionadas, pero en cambio tiene una efectividad limitada con los factores de modificación de duraciones debido a la corta duración de las vocales habladas neutras. Los resultados de las pruebas perceptivas muestran que, a pesar de obtener una naturalidad obviamente inferior a la ofrecida por un sintetizador profesional de voz cantada, el framework puede abordar necesidades puntuales de voz cantada para la síntesis de narración de cuentos con una calidad razonable. La incorporación de expresividad se investiga también en la simulación numérica 3D de vocales basada en FEM mediante modificaciones en las señales de excitación glotal utilizando una aproximación fuente-filtro de producción de voz. Estas señales se generan utilizando un modelo Liljencrants-Fant (LF) controlado con el parámetro de forma del pulso Rd, que permite explorar el continuo de fonación laxo-tenso además del rango de frecuencias fundamentales, F0, de la voz hablada. Se analiza la contribución de la fuente glotal a los modos de alto orden en la síntesis FEM de las vocales cardinales [a], [i] y [u] mediante la comparación de los valores de energía de alta frecuencia (HFE) obtenidos con geometrías realistas y simplificadas del tracto vocal. Las simulaciones indican que los modos de alto orden se prevén perceptivamente relevantes de acuerdo con valores de referencia de la literatura, particularmente para fonaciones tensas y/o F0s altas. En cambio, para vocales con una fonación laxa y/o F0s bajas los niveles de HFE pueden resultar inaudibles, especialmente si no hay ruido de aspiración en la fuente glotal. Después de este estudio preliminar, se han analizado las características de excitación de vocales alegres y agresivas de un corpus paralelo de voz en castellano con el objetivo de incorporar estos estilos expresivos de voz tensa en la simulación numérica de voz. Para ello, se ha usado el vocoder GlottDNN para analizar variaciones de F0 y pendiente espectral relacionadas con la excitación glotal en vocales [a]. Estas variaciones se mapean mediante la comparación con vocales sintéticas en valores de F0 y Rd para simular vocales que se asemejen a los estilos alegre y agresivo. Los resultados muestran que es necesario incrementar la F0 y disminuir la Rd respecto la voz neutra, con variaciones mayores para alegre que para agresivo, especialmente para vocales acentuadas. Los resultados conseguidos en las investigaciones realizadas validan la posibilidad de añadir expresividad a la síntesis basada en corpus US-TTS y a la simulación numérica de voz basada en FEM. Sin embargo, hay margen de mejora. Por ejemplo, la estrategia aplicada a la producción numérica de voz se podría mejorar estudiando y desarrollando métodos de filtrado inverso, así como incorporando modificaciones del tracto vocal, mientras que el framework US-TTS&S desarrollado se podría beneficiar de los avances en técnicas de transformación de voz incluyendo transformaciones de la calidad de la voz, aprovechando la experiencia adquirida en la simulación numérica de vocales expresivas.Speech is one of the most natural and direct forms of communication between human beings, as it codifies both a message and paralinguistic cues about the emotional state of the speaker, its mood, or its intention, thus becoming instrumental in pursuing a more natural Human Computer Interaction (HCI). In this context, the generation of expressive speech for the HCI output channel is a key element in the development of assistive technologies or personal assistants among other applications. Synthetic speech can be generated from recorded speech using corpus-based methods such as Unit-Selection (US), which can achieve high quality results but whose expressiveness is restricted to that available in the speech corpus. In order to improve the quality of the synthesis output, the current trend is to build ever larger speech databases, especially following the so-called End-to-End synthesis approach based on deep learning techniques. However, recording ad-hoc corpora for each and every desired expressive style can be extremely costly, or even unfeasible if the speaker is unable to properly perform the styles required for a given application (e.g., singing in the storytelling domain). Alternatively, new methods based on the physics of voice production have been developed in the last decade thanks to the increase in computing power. For instance, vowels or diphthongs can be obtained using the Finite Element Method (FEM) to simulate the propagation of acoustic waves through a 3D realistic vocal tract geometry obtained from Magnetic Resonance Imaging (MRI). However, since the main efforts in these numerical voice production methods have been focused on improving the modelling of the voice generation process, little attention has been paid to its expressiveness up to now. Furthermore, the collection of data for such simulations is very costly, besides requiring manual time-consuming postprocessing like that needed to extract 3D vocal tract geometries from MRI. The aim of the thesis is to add expressiveness into a system that generates neutral voice, without having to acquire expressive data from the original speaker. One the one hand, expressive capabilities are added to a Unit-Selection Text-to-Speech (US-TTS) system fed with a neutral speech corpus, to address specific and timely needs in the storytelling domain, such as for singing or in suspenseful situations. To this end, speech is parameterised using a harmonic-based model and subsequently transformed to the target expressive style according to an expert system. A first approach dealing with the synthesis of storytelling increasing suspense shows the viability of the proposal in terms of naturalness and storytelling quality. Singing capabilities are also added to the US-TTS system through the integration of Speech-to-Singing (STS) transformation modules into the TTS pipeline, and by incorporating an expressive prosody generation module that allows the US to select units closer to the target singing prosody obtained from the input score. This results in a Unit Selection based Text-to-Speech-and-Singing (US-TTS&S) synthesis framework that can generate both speech and singing from the same neutral speech small corpus (~2.6 h). According to the objective results, the score-driven US strategy can reduce the pitch scaling factors required to produce singing from the selected spoken units, but its effectiveness is limited regarding the time-scale requirements due to the short duration of the spoken vowels. Results from the perceptual tests show that although the obtained naturalness is obviously far from that given by a professional singing synthesiser, the framework can address eventual singing needs for synthetic storytelling with a reasonable quality. The incorporation of expressiveness is also investigated in the 3D FEM-based numerical simulation of vowels through modifications of the glottal flow signals following a source-filter approach of voice production. These signals are generated using a Liljencrants-Fant (LF) model controlled with the glottal shape parameter Rd, which allows exploring the tense-lax continuum of phonation besides the spoken vocal range of fundamental frequency values, F0. The contribution of the glottal source to higher order modes in the FEM synthesis of cardinal vowels [a], [i] and [u] is analysed through the comparison of the High Frequency Energy (HFE) values obtained with realistic and simplified 3D geometries of the vocal tract. The simulations indicate that higher order modes are expected to be perceptually relevant according to reference values stated in the literature, particularly for tense phonations and/or high F0s. Conversely, vowels with a lax phonation and/or low F0s can result in inaudible HFE levels, especially if aspiration noise is not present in the glottal source. After this preliminary study, the excitation characteristics of happy and aggressive vowels from a Spanish parallel speech corpus are analysed with the aim of incorporating this tense voice expressive styles into the numerical production of voice. To that effect, the GlottDNN vocoder is used to analyse F0 and spectral tilt variations associated with the glottal excitation on vowels [a]. These variations are mapped through the comparison with synthetic vowels into F0 and Rd values to simulate vowels resembling happy and aggressive styles. Results show that it is necessary to increase F0 and decrease Rd with respect to neutral speech, with larger variations for happy than aggressive style, especially for the stressed [a] vowels. The results achieved in the conducted investigations validate the possibility of adding expressiveness to both corpus-based US-TTS synthesis and FEM-based numerical simulation of voice. Nevertheless, there is still room for improvement. For instance, the strategy applied to the numerical voice production could be improved by studying and developing inverse filtering approaches as well as incorporating modifications of the vocal tract, whereas the developed US-TTS&S framework could benefit from advances in voice transformation techniques including voice quality modifications, taking advantage of the experience gained in the numerical simulation of expressive vowels

SEA_AP: una herramienta de segmentación y etiquetado para el análisis prosódico

This paper introduces a tool that performs segmentation and labelling of sound chains in phono units, syllables and/or words departing from a sound signal and its corresponding orthographic transcription. In addition, it also integrates acoustic analysis scripts applied to the Praat programme with the aim of reducing the time spent on tasks related to analysis, correction, smoothing and generation of graphics of the melodic curve. The tool is implemented for Galician, Spanish and Brazilian Portuguese. Our goal is to contribute, by means of this application, to automatize some of the tasks of segmentation, labelling and prosodic analysis, since these tasks require a large investment of time and human resources.En este artículo se presenta una herramienta que realiza la segmentación y el etiquetado de cadenas sonoras en unidades de fono, sílaba y/o palabra partiendo de una señal sonora y de su correspondiente transcripción ortográfica. Además, integra scripts de análisis acústico que se ejecutan sobre el programa Praat con el fin de reducir el tiempo invertido en las tareas de análisis, corrección, suavizado y generación de gráficos de la curva melódica. La herramienta está implementada para gallego, español y portugués de Brasil. Nuestro objetivo es contribuir con esta aplicación a automatizar algunas de las labores de segmentación, etiquetado y análisis prosódico, pues constituyen tareas que requieren una gran inversión de tiempo y de recursos humanos.This work would have not been possible without the help of the Spanish Government (Project ‘SpeechTech4All’ TEC2012-38939-C03-01), the European Regional Development Fund (ERDF), the Government of the Autonomous Community of Galicia (GRC2014/024, “Consolidación de Unidades de Investigación: Proyecto AtlantTIC” CN2012/160) and the “Red de Investigación TecAnDAli” from the Council of Culture, Education and University Planning, Xunta de GaliciaS

Relative Salience of Speech Rhythm and Speech Rate on Perceived Foreign Accent in a Second Language

We investigated the independent contribution of speech rate and speech rhythm to perceived foreign accent. To address this issue we used a resynthesis technique that allows neutralizing segmental and tonal idiosyncrasies between identical sentences produced by French learners of English at different proficiency levels and maintaining the idiosyncrasies pertaining to prosodic timing patterns. We created stimuli that (1) preserved the idiosyncrasies in speech rhythm while controlling for the differences in speech rate between the utterances; (2) preserved the idiosyncrasies in speech rate while controlling for the differences in speech rhythm between the utterances; and (3) preserved the idiosyncrasies both in speech rate and speech rhythm. All the stimuli were created in intoned (with imposed intonational contour) and flat (with monotonized, constant F0) conditions. The original and the resynthesized sentences were rated by native speakers of English for degree of foreign accent. We found that both speech rate and speech rhythm influence the degree of perceived foreign accent, but the effect of speech rhythm is larger than that of speech rate. We also found that intonation enhances the perception of fine differences in rhythmic patterns but reduces the perceptual salience of fine differences in speech rate

Automatic Pronunciation Assessment -- A Review

Pronunciation assessment and its application in computer-aided pronunciation training (CAPT) have seen impressive progress in recent years. With the rapid growth in language processing and deep learning over the past few years, there is a need for an updated review. In this paper, we review methods employed in pronunciation assessment for both phonemic and prosodic. We categorize the main challenges observed in prominent research trends, and highlight existing limitations, and available resources. This is followed by a discussion of the remaining challenges and possible directions for future work.Comment: 9 pages, accepted to EMNLP Finding

Development of the Slovak HMM-Based TTS System and Evaluation of Voices in Respect to the Used Vocoding Techniques

This paper describes the development of a Slovak text-to-speech system which applies a technique wherein speech is directly synthesized from hidden Markov models. Statistical models for Slovak speech units are trained by using the newly created female and male phonetically balanced speech corpora. In addition, contextual informations about phonemes, syllables, words, phrases, and utterances were determined, as well as questions for decision tree-based context clustering algorithms. In this paper, recent statistical parametric speech synthesis methods including the conventional, STRAIGHT and AHOcoder speech synthesis systems are implemented and evaluated. Objective evaluation methods (mel-cepstral distortion and fundamental frequency comparison) and subjective ones (mean opinion score and semantically unpredictable sentences test) are carried out to compare these systems with each other and evaluation of their overall quality. The result of this work is a set of text to speech systems for Slovak language which are characterized by very good intelligibility and quite good naturalness of utterances at the output of these systems. In the subjective tests of intelligibility the STRAIGHT based female voice and AHOcoder based male voice reached the highest scores