Ultrasound-Based Silent Speech Interface Built on a Continuous Vocoder

Recently it was shown that within the Silent Speech Interface (SSI) field, the prediction of F0 is possible from Ultrasound Tongue Images (UTI) as the articulatory input, using Deep Neural Networks for articulatory-to-acoustic mapping. Moreover, text-to-speech synthesizers were shown to produce higher quality speech when using a continuous pitch estimate, which takes non-zero pitch values even when voicing is not present. Therefore, in this paper on UTI-based SSI, we use a simple continuous F0 tracker which does not apply a strict voiced / unvoiced decision. Continuous vocoder parameters (ContF0, Maximum Voiced Frequency and Mel-Generalized Cepstrum) are predicted using a convolutional neural network, with UTI as input. The results demonstrate that during the articulatory-to-acoustic mapping experiments, the continuous F0 is predicted with lower error, and the continuous vocoder produces slightly more natural synthesized speech than the baseline vocoder using standard discontinuous F0.Comment: 5 pages, 3 figures, accepted for publication at Interspeech 201

Application of singing synthesis techniquest to bertsolaritza

237 p

Application of singing synthesis techniquest to bertsolaritza

237 p

DESIGN AND EVALUATION OF HARMONIC SPEECH ENHANCEMENT AND BANDWIDTH EXTENSION

Improving the quality and intelligibility of speech signals continues to be an important topic in mobile communications and hearing aid applications. This thesis explored the possibilities of improving the quality of corrupted speech by cascading a log Minimum Mean Square Error (logMMSE) noise reduction system with a Harmonic Speech Enhancement (HSE) system. In HSE, an adaptive comb filter is deployed to harmonically filter the useful speech signal and suppress the noisy components to noise floor. A Bandwidth Extension (BWE) algorithm was applied to the enhanced speech for further improvements in speech quality. Performance of this algorithm combination was evaluated using objective speech quality metrics across a variety of noisy and reverberant environments. Results showed that the logMMSE and HSE combination enhanced the speech quality in any reverberant environment and in the presence of multi-talker babble. The objective improvements associated with the BWE were found to be minima

Analysis/Synthesis Comparison of Vocoders Utilized in Statistical Parametric Speech Synthesis

Tässä työssä esitetään kirjallisuuskatsaus ja kokeellinen osio tilastollisessa parametrisessa puhesynteesissä käytetyistä vokoodereista. Kokeellisessa osassa kolmen valitun vokooderin (GlottHMM, STRAIGHT ja Harmonic/Stochastic Model) analyysi-synteesi -ominaisuuksia tarkastellaan usealla tavalla. Suoritetut kokeet olivat vokooderiparametrien tilastollisten jakaumien analysointi, puheen tunnetilan tilastollinen vaikutus vokooderiparametrien jakaumiin sekä subjektiivinen kuuntelukoe jolla mitattiin vokooderien suhteellista analyysi-synteesi -laatua. Tulokset osoittavat että STRAIGHT-vokooderi omaa eniten Gaussiset parametrijakaumat ja tasaisimman synteesilaadun. GlottHMM-vokooderin parametrit osoittivat suurinta herkkyyttä puheen tunnetilan funktiona ja vokooderi sai parhaan, mutta laadultaan vaihtelevan kuuntelukoetuloksen. HSM-vokooderin LSF-parametrien havaittiin olevan Gaussisempia kuin GlottHMM-vokooderin LSF parametrit, mutta vokooderin havaittiin kärsivän kohinaherkkyydestä, ja se sai huonoimman kuuntelukoetuloksen.This thesis presents a literature study followed by an experimental part on the state-of-the-art vocoders utilized in statistical parametric speech synthesis. In the experimental part, the analysis/synthesis properties of three selected vocoders (GlottHMM, STRAIGHT and Harmonic/Stochastic Model) are examined. The performed tests were the analysis of vocoder parameter distributions, statistical testing on the effect of emotions to the vocoder parameter distributions, and a subjective listening test evaluating the vocoders' relative analysis/synthesis quality. The results indicate that the STRAIGHT vocoder has the most Gaussian parameter distributions and most robust synthesis quality, whereas the GlottHMM vocoder has the most emotion sensitive parameters and best but unreliable synthesis quality. The HSM vocoder's LSF parameters were found to be more Gaussian than the GlottHMM vocoder's LSF parameters. HSM was found to be sensitive to noise, and it scored the lowest score on the subjective listening test

HMM-based speech synthesis using an acoustic glottal source model

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

Técnicas de personalización de voces sintéticas para su uso por personas con discapacidad oral

151 p.Esta tesis presenta avances realizados en la personalización de voces sintéticas que emplean los sistemas de conversión de texto a voz utilizados por personas con alguna discapacidad oral. Se presenta un nuevo algoritmo de adaptación de locutor para voces sintéticas basadas en síntesis estadístico paramétrica. Este algoritmo hace uso únicamente de fragmentos vocálicos para imitar la voz del locutor objetivo y se ha demostrado que es robusto frente a la escasez de datos y que tiene un desempeño similar a otros algoritmos del estado del arte.También se describe el diseño e implementación de un banco de voces en el cual cualquier persona puede realizar grabaciones de su voz real para generar una voz sintética que posteriormente puede ser empleada por otro usuario. De esta manera las personas pueden ¿donar¿ su voz.Por último, se presenta una metodología que hace uso de diversas medidas objetivas de evaluación de señales de voz para puntuar la calidad de las voces disponibles en el banco de voces