A Log Domain Pulse Model for Parametric Speech Synthesis

Most of the degradation in current Statistical Parametric Speech Synthesis (SPSS) results from the form of the vocoder. One of the main causes of degradation is the reconstruction of the noise. In this article, a new signal model is proposed that leads to a simple synthesizer, without the need for ad-hoc tuning of model parameters. The model is not based on the traditional additive linear source-filter model, it adopts a combination of speech components that are additive in the log domain. Also, the same representation for voiced and unvoiced segments is used, rather than relying on binary voicing decisions. This avoids voicing error discontinuities that can occur in many current vocoders. A simple binary mask is used to denote the presence of noise in the time-frequency domain, which is less sensitive to classification errors. Four experiments have been carried out to evaluate this new model. The first experiment examines the noise reconstruction issue. Three listening tests have also been carried out that demonstrate the advantages of this model: comparison with the STRAIGHT vocoder; the direct prediction of the binary noise mask by using a mixed output configuration; and partial improvements of creakiness using a mask correction mechanism.European Union's Horizon 2020 research and innovation programme under the Marie Sklodowska-Curie; 10.13039/501100000266-EPSR

Voice source characterization for prosodic and spectral manipulation

The objective of this dissertation is to study and develop techniques to decompose the speech signal into its two main components: voice source and vocal tract. Our main efforts are on the glottal pulse analysis and characterization. We want to explore the utility of this model in different areas of speech processing: speech synthesis, voice conversion or emotion detection among others. Thus, we will study different techniques for prosodic and spectral manipulation. One of our requirements is that the methods should be robust enough to work with the large databases typical of speech synthesis. We use a speech production model in which the glottal flow produced by the vibrating vocal folds goes through the vocal (and nasal) tract cavities and its radiated by the lips. Removing the effect of the vocal tract from the speech signal to obtain the glottal pulse is known as inverse filtering. We use a parametric model fo the glottal pulse directly in the source-filter decomposition phase. In order to validate the accuracy of the parametrization algorithm, we designed a synthetic corpus using LF glottal parameters reported in the literature, complemented with our own results from the vowel database. The results show that our method gives satisfactory results in a wide range of glottal configurations and at different levels of SNR. Our method using the whitened residual compared favorably to this reference, achieving high quality ratings (Good-Excellent). Our full parametrized system scored lower than the other two ranking in third place, but still higher than the acceptance threshold (Fair-Good). Next we proposed two methods for prosody modification, one for each of the residual representations explained above. The first method used our full parametrization system and frame interpolation to perform the desired changes in pitch and duration. The second method used resampling on the residual waveform and a frame selection technique to generate a new sequence of frames to be synthesized. The results showed that both methods are rated similarly (Fair-Good) and that more work is needed in order to achieve quality levels similar to the reference methods. As part of this dissertation, we have studied the application of our models in three different areas: voice conversion, voice quality analysis and emotion recognition. We have included our speech production model in a reference voice conversion system, to evaluate the impact of our parametrization in this task. The results showed that the evaluators preferred our method over the original one, rating it with a higher score in the MOS scale. To study the voice quality, we recorded a small database consisting of isolated, sustained Spanish vowels in four different phonations (modal, rough, creaky and falsetto) and were later also used in our study of voice quality. Comparing the results with those reported in the literature, we found them to generally agree with previous findings. Some differences existed, but they could be attributed to the difficulties in comparing voice qualities produced by different speakers. At the same time we conducted experiments in the field of voice quality identification, with very good results. We have also evaluated the performance of an automatic emotion classifier based on GMM using glottal measures. For each emotion, we have trained an specific model using different features, comparing our parametrization to a baseline system using spectral and prosodic characteristics. The results of the test were very satisfactory, showing a relative error reduction of more than 20% with respect to the baseline system. The accuracy of the different emotions detection was also high, improving the results of previously reported works using the same database. Overall, we can conclude that the glottal source parameters extracted using our algorithm have a positive impact in the field of automatic emotion classification

A Pulse Model in Log-domain for a Uniform Synthesizer

The quality of the vocoder plays a crucial role in the performance of parametric speech synthesis systems. In order to improve the vocoder quality, it is necessary to reconstruct as much of the perceived components of the speech signal as possible. In this paper, we first show that the noise component is currently not accurately modelled in the widely used STRAIGHT vocoder, thus, limiting the voice range that can be covered and also limiting the overall quality. In order to motivate a new, alternative, approach to this issue, we present a new synthesizer, which uses a uniform representation for voiced and unvoiced segments. This synthesizer has also the advantage of using a simple signal model compared to other approaches, thus offering a convenient and controlled alternative for future developments. Experiments analysing the synthesis quality of the noise component shows improved speech reconstruction using the suggested synthesizer compared to STRAIGHT. Additionally an experiment about analysis/resynthesis shows that the suggested synthesizer solves some of the issues of another uniform vocoder, Harmonic Model plus Phase Distortion (HMPD). In text-to-speech synthesis, it outperforms HMPD and exhibits a similar, or only slightly worse, quality to STRAIGHT’s quality, which is encouraging for a new vocoding approach.This project has received funding from the European Union’s Horizon 2020 research and innovation programme under the Marie Sklodowska-Curie grant agreement No 655764. The research for this paper was also partly supported by EPSRC grant EP/I031022/1 (Natural Speech Technology)

New Method for Delexicalization and its Application to Prosodic Tagging for Text-to-Speech Synthesis

This paper describes a new flexible delexicalization method based on glottal excited parametric speech synthesis scheme. The system utilizes inverse filtered glottal flow and all-pole modelling of the vocal tract. The method provides a possibil- ity to retain and manipulate all relevant prosodic features of any kind of speech. Most importantly, the features include voice quality, which has not been properly modeled in earlier delex- icalization methods. The functionality of the new method was tested in a prosodic tagging experiment aimed at providing word prominence data for a text-to-speech synthesis system. The ex- periment confirmed the usefulness of the method and further corroborated earlier evidence that linguistic factors influence the perception of prosodic prominence.Peer reviewe

Synthesis of listener vocalizations : towards interactive speech synthesis

Spoken and multi-modal dialogue systems start to use listener vocalizations, such as uh-huh and mm-hm, for natural interaction. Generation of listener vocalizations is one of the major objectives of emotionally colored conversational speech synthesis. Success in this endeavor depends on the answers to three questions: Where to synthesize a listener vocalization? What meaning should be conveyed through the synthesized vocalization? And, how to realize an appropriate listener vocalization with the intended meaning? This thesis addresses the latter question. The investigation starts with proposing a three-stage approach: (i) data collection, (ii) annotation, and (iii) realization. The first stage presents a method to collect natural listener vocalizations from German and British English professional actors in a recording studio. In the second stage, we explore a methodology for annotating listener vocalizations -- meaning and behavior (form) annotation. The third stage proposes a realization strategy that uses unit selection and signal modification techniques to generate appropriate listener vocalizations upon user requests. Finally, we evaluate naturalness and appropriateness of synthesized vocalizations using perception studies. The work is implemented in the open source MARY text-to-speech framework, and it is integrated into the SEMAINE project\u27s Sensitive Artificial Listener (SAL) demonstrator.Dialogsysteme nutzen zunehmend Hörer-Vokalisierungen, wie z.B. a-ha oder mm-hm, für natürliche Interaktion. Die Generierung von Hörer-Vokalisierungen ist eines der zentralen Ziele emotional gefärbter, konversationeller Sprachsynthese. Ein Erfolg in diesem Unterfangen hängt von den Antworten auf drei Fragen ab: Wo bzw. wann sollten Vokalisierungen synthetisiert werden? Welche Bedeutung sollte in den synthetisierten Vokalisierungen vermittelt werden? Und wie können angemessene Hörer-Vokalisierungen mit der intendierten Bedeutung realisiert werden? Diese Arbeit widmet sich der letztgenannten Frage. Die Untersuchung erfolgt in drei Schritten: (i) Korpuserstellung; (ii) Annotation; und (iii) Realisierung. Der erste Schritt präsentiert eine Methode zur Sammlung natürlicher Hörer-Vokalisierungen von deutschen und britischen Profi-Schauspielern in einem Tonstudio. Im zweiten Schritt wird eine Methodologie zur Annotation von Hörer-Vokalisierungen erarbeitet, die sowohl Bedeutung als auch Verhalten (Form) umfasst. Der dritte Schritt schlägt ein Realisierungsverfahren vor, die Unit-Selection-Synthese mit Signalmodifikationstechniken kombiniert, um aus Nutzeranfragen angemessene Hörer-Vokalisierungen zu generieren. Schließlich werden Natürlichkeit und Angemessenheit synthetisierter Vokalisierungen mit Hilfe von Hörtests evaluiert. Die Methode wurde im Open-Source-Sprachsynthesesystem MARY implementiert und in den Sensitive Artificial Listener-Demonstrator im Projekt SEMAINE integriert

Hidden Markov model based Finnish text-to-speech system utilizing glottal inverse filtering

Tässä työssä esitetään uusi Markovin piilomalleihin (hidden Markov model, HMM) perustuva äänilähteen käänteissuodatusta hyödyntävä suomenkielinen puhesynteesijärjestelmä. Uuden puhesynteesimenetelmän päätavoite on tuottaa luonnolliselta kuulostavaa synteettistä puhetta, jonka ominaisuuksia voidaan muuttaa eri puhujien, puhetyylien tai jopa äänen emootiosisällön mukaan. Näiden tavoitteiden mahdollistamiseksi uudessa puhesynteesimenetelmässä mallinnetaan ihmisen äänentuottojärjestelmää äänilähteen käänteissuodatuksen ja HMM-mallinnuksen avulla. Uusi puhesynteesijärjestelmä hyödyntää äänilähteen käänteissuodatusmenetelmää, joka mahdollistaa äänilähteen ominaisuuksien parametrisoinnin erillään muista puheen parametreista, ja siten näiden parametrien mallintamisen erikseen HMM-järjestelmässä. Synteesivaiheessa luonnollisesta puheesta laskettuja glottispulsseja käytetään äänilähteen luomiseen, ja äänilähteen ominaisuuksia muokataan edelleen tilastollisen HMM-järjestelmän tuottaman parametrisen kuvauksen avulla, mikä imitoi oikeassa puheessa esiintyvää luonnollista äänilähteen ominaisuuksien vaihtelua. Subjektiivisten kuuntelukokeiden tulokset osoittavat, että uuden puhesynteesimenetelmän laatu on huomattavasti parempi verrattuna perinteiseen HMM-pohjaiseen puhesynteesijärjestelmään. Lisäksi tulokset osoittavat, että uusi puhesynteesimenetelmä pystyy tuottamaan luonnolliselta kuulostavaa puhetta eri puhujien ominaisuuksilla.In this work, a new hidden Markov model (HMM) based text-to-speech (TTS) system utilizing glottal inverse filtering is described. The primary goal of the new TTS system is to enable producing natural sounding synthetic speech in different speaking styles with different speaker characteristics and emotions. In order to achieve these goals, the function of the real human voice production mechanism is modeled with the help of glottal inverse filtering embedded in a statistical framework of HMM. The new TTS system uses a glottal inverse filtering based parametrization method that enables the extraction of voice source characteristics separate from other speech parameters, and thus the individual modeling of these characteristics in the HMM system. In the synthesis stage, natural glottal flow pulses are used for creating the voice source, and the voice source characteristics are further modified according to the adaptive all-pole model generated by the HMM system in order to imitate the natural variation in the real voice source. Subjective listening tests show that the quality of the new TTS system is considerably better compared to a traditional HMM-based speech synthesizer. Moreover, the new system is clearly able to produce natural sounding synthetic speech with specific speaker characteristics

Vivos Voco: A survey of recent research on voice transformation at IRCAM

cote interne IRCAM: Lanchantin11cInternational audienceIRCAM has a long experience in analysis, synthesis and transformation of voice. Natural voice transformations are of great interest for many applications and can be combine with text-to-speech system, leading to a powerful creation tool. We present research conducted at IRCAM on voice transformations for the last few years. Transformations can be achieved in a global way by modifying pitch, spectral envelope, durations etc. While it sacrifices the possibility to attain a specific target voice, the approach allows the production of new voices of a high degree of naturalness with different gender and age, modified vocal quality, or another speech style. These transformations can be applied in realtime using ircamTools TRAX. Transformation can also be done in a more specific way in order to transform a voice towards the voice of a target speaker. Finally, we present some recent research on the transformation of expressivity