767 research outputs found

    Statistical text-to-speech synthesis of Spanish subtitles

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    The final publication is available at Springer via http://dx.doi.org/10.1007/978-3-319-13623-3_5Online multimedia repositories are growing rapidly. However, language barriers are often difficult to overcome for many of the current and potential users. In this paper we describe a TTS Spanish sys- tem and we apply it to the synthesis of transcribed and translated video lectures. A statistical parametric speech synthesis system, in which the acoustic mapping is performed with either HMM-based or DNN-based acoustic models, has been developed. To the best of our knowledge, this is the first time that a DNN-based TTS system has been implemented for the synthesis of Spanish. A comparative objective evaluation between both models has been carried out. Our results show that DNN-based systems can reconstruct speech waveforms more accurately.The research leading to these results has received funding from the European Union Seventh Framework Programme (FP7/2007-2013) under grant agreement no 287755 (transLectures) and ICT Policy Support Programme (ICT PSP/2007-2013) as part of the Competitiveness and Innovation Framework Programme (CIP) under grant agreement no 621030 (EMMA), and the Spanish MINECO Active2Trans (TIN2012-31723) research project.Piqueras Gozalbes, SR.; Del Agua Teba, MA.; GimĂ©nez Pastor, A.; Civera Saiz, J.; Juan CĂ­scar, A. (2014). Statistical text-to-speech synthesis of Spanish subtitles. En Advances in Speech and Language Technologies for Iberian Languages: Second International Conference, IberSPEECH 2014, Las Palmas de Gran Canaria, Spain, November 19-21, 2014. Proceedings. Springer International Publishing. 40-48. https://doi.org/10.1007/978-3-319-13623-3_5S4048Ahocoder, http://aholab.ehu.es/ahocoderCoursera, http://www.coursera.orgHMM-Based Speech Synthesis System (HTS), http://hts.sp.nitech.ac.jpKhan Academy, http://www.khanacademy.orgAxelrod, A., He, X., Gao, J.: Domain adaptation via pseudo in-domain data selection. In: Proc. of EMNLP, pp. 355–362 (2011)Bottou, L.: Stochastic gradient learning in neural networks. In: Proceedings of Neuro-NĂźmes 1991. EC2, Nimes, France (1991)Dahl, G.E., Yu, D., Deng, L., Acero, A.: Context-dependent pre-trained deep neural networks for large-vocabulary speech recognition. IEEE Transactions on Audio, Speech, and Language Processing 20(1), 30–42 (2012)Erro, D., Sainz, I., Navas, E., Hernaez, I.: Harmonics plus noise model based vocoder for statistical parametric speech synthesis. IEEE Journal of Selected Topics in Signal Processing 8(2), 184–194 (2014)Fan, Y., Qian, Y., Xie, F., Soong, F.: TTS synthesis with bidirectional LSTM based recurrent neural networks. In: Proc. of Interspeech (submitted 2014)Hinton, G., Deng, L., Yu, D., Dahl, G.E., Mohamed, A.R., Jaitly, N., Senior, A., Vanhoucke, V., Nguyen, P., Sainath, T.N., et al.: Deep neural networks for acoustic modeling in speech recognition: The shared views of four research groups. IEEE Signal Processing Magazine 29(6), 82–97 (2012)Hunt, A.J., Black, A.W.: Unit selection in a concatenative speech synthesis system using a large speech database. In: Proc. of ICASSP, vol. 1, pp. 373–376 (1996)King, S.: Measuring a decade of progress in text-to-speech. Loquens 1(1), e006 (2014)Koehn, P.: Statistical Machine Translation. Cambridge University Press (2010)Kominek, J., Schultz, T., Black, A.W.: Synthesizer voice quality of new languages calibrated with mean mel cepstral distortion. In: Proc. of SLTU, pp. 63–68 (2008)Lopez, A.: Statistical machine translation. ACM Computing Surveys 40(3), 8:1–8:49 (2008)poliMedia: The polimedia video-lecture repository (2007), http://media.upv.esSainz, I., Erro, D., Navas, E., HernĂĄez, I., SĂĄnchez, J., Saratxaga, I.: Aholab speech synthesizer for albayzin 2012 speech synthesis evaluation. In: Proc. of IberSPEECH, pp. 645–652 (2012)Seide, F., Li, G., Chen, X., Yu, D.: Feature engineering in context-dependent dnn for conversational speech transcription. In: Proc. of ASRU, pp. 24–29 (2011)Shinoda, K., Watanabe, T.: MDL-based context-dependent subword modeling for speech recognition. Journal of the Acoustical Society of Japan 21(2), 79–86 (2000)Silvestre-CerdĂ , J.A., et al.: Translectures. In: Proc. of IberSPEECH, pp. 345–351 (2012)TED Ideas worth spreading, http://www.ted.comThe transLectures-UPV Team.: The transLectures-UPV toolkit (TLK), http://translectures.eu/tlkToda, T., Black, A.W., Tokuda, K.: Mapping from articulatory movements to vocal tract spectrum with Gaussian mixture model for articulatory speech synthesis. In: Proc. of ISCA Speech Synthesis Workshop (2004)Tokuda, K., Kobayashi, T., Imai, S.: Speech parameter generation from hmm using dynamic features. In: Proc. of ICASSP, vol. 1, pp. 660–663 (1995)Tokuda, K., Masuko, T., Miyazaki, N., Kobayashi, T.: Multi-space probability distribution HMM. IEICE Transactions on Information and Systems 85(3), 455–464 (2002)transLectures: D3.1.2: Second report on massive adaptation, http://www.translectures.eu/wp-content/uploads/2014/01/transLectures-D3.1.2-15Nov2013.pdfTurrĂł, C., Ferrando, M., Busquets, J., Cañero, A.: Polimedia: a system for successful video e-learning. 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    Lipreading with Long Short-Term Memory

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    Lipreading, i.e. speech recognition from visual-only recordings of a speaker's face, can be achieved with a processing pipeline based solely on neural networks, yielding significantly better accuracy than conventional methods. Feed-forward and recurrent neural network layers (namely Long Short-Term Memory; LSTM) are stacked to form a single structure which is trained by back-propagating error gradients through all the layers. The performance of such a stacked network was experimentally evaluated and compared to a standard Support Vector Machine classifier using conventional computer vision features (Eigenlips and Histograms of Oriented Gradients). The evaluation was performed on data from 19 speakers of the publicly available GRID corpus. With 51 different words to classify, we report a best word accuracy on held-out evaluation speakers of 79.6% using the end-to-end neural network-based solution (11.6% improvement over the best feature-based solution evaluated).Comment: Accepted for publication at ICASSP 201

    Integrating Articulatory Features into HMM-based Parametric Speech Synthesis

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    This paper presents an investigation of ways to integrate articulatory features into Hidden Markov Model (HMM)-based parametric speech synthesis, primarily with the aim of improving the performance of acoustic parameter generation. The joint distribution of acoustic and articulatory features is estimated during training and is then used for parameter generation at synthesis time in conjunction with a maximum-likelihood criterion. Different model structures are explored to allow the articulatory features to influence acoustic modeling: model clustering, state synchrony and cross-stream feature dependency. The results of objective evaluation show that the accuracy of acoustic parameter prediction can be improved when shared clustering and asynchronous-state model structures are adopted for combined acoustic and articulatory features. More significantly, our experiments demonstrate that modeling the dependency between these two feature streams can make speech synthesis more flexible. The characteristics of synthetic speech can be easily controlled by modifying generated articulatory features as part of the process of acoustic parameter generation

    Linguistic unit discovery from multi-modal inputs in unwritten languages: Summary of the "Speaking Rosetta" JSALT 2017 Workshop

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    We summarize the accomplishments of a multi-disciplinary workshop exploring the computational and scientific issues surrounding the discovery of linguistic units (subwords and words) in a language without orthography. We study the replacement of orthographic transcriptions by images and/or translated text in a well-resourced language to help unsupervised discovery from raw speech.Comment: Accepted to ICASSP 201

    Speech synthesis : Developing a web application implementing speech technology

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    Speech is a natural media of communication for humans. Text-to-speech (TTS) technology uses a computer to synthesize speech. There are three main techniques of TTS synthesis. These are formant-based, articulatory and concatenative. The application areas of TTS include accessibility, education, entertainment and communication aid in mass transit. A web application was developed to demonstrate the application of speech synthesis technology. Existing speech synthesis engines for the Finnish language were compared and two open source text to speech engines, Festival and Espeak were selected to be used with the web application. The application uses a Linux-based speech server which communicates with client devices with the HTTP-GET protocol. The application development successfully demonstrated the use of speech synthesis in language learning. One of the emerging sectors of speech technologies is the mobile market due to limited input capabilities in mobile devices. Speech technologies are not equally available in all languages. Text in the Oromo language was tested using Finnish speech synthesizers; due to similar rules in orthography of germination of consonants and length of vowels, legible results were gained
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