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Dive into the research topics where Fernando Villavicencio is active.

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Featured researches published by Fernando Villavicencio.


conference of the international speech communication association | 2016

The Voice Conversion Challenge 2016

Tomoki Toda; Ling-Hui Chen; Daisuke Saito; Fernando Villavicencio; Mirjam Wester; Zhizheng Wu; Junichi Yamagishi

This paper describes the Voice Conversion Challenge 2016 devised by the authors to better understand different voice conversion (VC) techniques by comparing their performance on a common dataset. The task of the challenge was speaker conversion, i.e., to transform the voice identity of a source speaker into that of a target speaker while preserving the linguistic content. Using a common dataset consisting of 162 utterances for training and 54 utterances for evaluation from each of 5 source and 5 target speakers, 17 groups working in VC around the world developed their own VC systems for every combination of the source and target speakers, i.e., 25 systems in total, and generated voice samples converted by the developed systems. These samples were evaluated in terms of target speaker similarity and naturalness by 200 listeners in a controlled environment. This paper summarizes the design of the challenge, its result, and a future plan to share views about unsolved problems and challenges faced by the current VC techniques.


Odyssey 2016 | 2016

Voice Liveness Detection for Speaker Verification based on a Tandem Single/Double-channel Pop Noise Detector.

Sayaka Shiota; Fernando Villavicencio; Junichi Yamagishi; Nobutaka Ono; Isao Echizen; Tomoko Matsui

This paper presents an algorithm for detecting spoofing attacks against automatic speaker verification (ASV) systems. While such systems now have performances comparable to those of other biometric modalities, spoofing techniques used against them have progressed drastically. Several techniques can be used to generate spoofing materials (e.g., speech synthesis and voice conversion techniques), and detecting them only on the basis of differences at an acoustic speaker modeling level is a challenging task. Moreover, differences between “live” and artificially generated material are expected to gradually decrease in the near future due to advances in synthesis technologies. A previously proposed “voice liveness” detection framework aimed at validating whether speech signals were generated by a person or artificially created uses elementary algorithms to detect pop noise. Detection is taken as evidence of liveness. A more advanced detection algorithm has now been developed that combines singleand double-channel pop noise detection. Experiments demonstrated that this tandem algorithm detects pop noise more effectively: the detection error rate was up to 80% less that those achieved with the elementary algorithms.


international workshop on machine learning for signal processing | 2015

Observation-model error compensation for enhanced spectral envelope transformation in voice conversion

Fernando Villavicencio; Jordi Bonada; Yuji Hisaminato

A strategy to enhance the signal quality and naturalness was designed for performing probabilistic spectral envelope transformation in voice conversion. The existing modeling error of the probabilistic mixture to represent the observed envelope features is translated generally as an averaging of the information in the spectral domain, resulting in over-smoothed spectra. Moreover, a transformation based on poorly modeled features might not be considered reliable. Our strategy consists of a novel definition of the spectral transformation to compensate the effect of both over-smoothing and poor modeling. The results of an experimental evaluation show that the perceived naturalness of converted speech was enhanced.


9th ISCA Speech Synthesis Workshop | 2016

Development of a statistical parametric synthesis system for operatic singing in German

Michael Pucher; Fernando Villavicencio; Junichi Yamagishi

In this paper we describe the development of a Hidden Markov Model (HMM) based synthesis system for operatic singing in German, which is an extension of the HMM-based synthesis system for popular songs in Japanese and English called “Sinsy”. The implementation of this system consists of German text analysis, lexicon and Letter-To-Sound (LTS) conversion, and syllable duplication, which enables us to convert a German MusicXML input into context-dependent labels for acoustic modelling. Using the front-end, we develop two operatic singing voices, female mezzo-soprano and male bass voices, based on our new database, which consists of singing data of professional opera singers based in Vienna. We describe the details of the database and the recording procedure that is used to acquire singing data of four opera singers in German. For HMM training, we adopt a singer (speaker)-dependent training procedure. For duration modelling we propose a simple method that hierarchically constrains note durations by the overall utterance duration and then constrains phone durations by the synthesised note duration. We evaluate the performance of the voices with two vibrato modelling methods that have been proposed in the literature and show that HMM-based vibrato modelling can improve the overall quality.


conference of the international speech communication association | 2010

Applying voice conversion to concatenative singing-voice synthesis.

Fernando Villavicencio; Jordi Bonada


conference of the international speech communication association | 2015

Voice liveness detection algorithms based on pop noise caused by human breath for automatic speaker verification

Sayaka Shiota; Fernando Villavicencio; Junichi Yamagishi; Nobutaka Ono; Isao Echizen; Tomoko Matsui


Odyssey 2018 The Speaker and Language Recognition Workshop | 2018

The Voice Conversion Challenge 2018: Promoting Development of Parallel and Nonparallel Methods

Jaime Lorenzo-Trueba; Junichi Yamagishi; Tomoki Toda; Daisuke Saito; Fernando Villavicencio; Tomi Kinnunen; Zhen-Hua Ling


Odyssey 2018 The Speaker and Language Recognition Workshop | 2018

A Spoofing Benchmark for the 2018 Voice Conversion Challenge: Leveraging from Spoofing Countermeasures for Speech Artifact Assessment

Tomi Kinnunen; Jaime Lorenzo-Trueba; Junichi Yamagishi; Tomoki Toda; Daisuke Saito; Fernando Villavicencio; Zhen-Hua Ling


Gautham J. Mysore, DAPS (Device and Produced Speech) Dataset - A dataset of professional production quality speech and corresponding aligned speech recorded on common consumer device, https://archive.org/details/daps_dataset | 2018

The Voice Conversion Challenge 2018: database and results

Tomoki Toda; Daisuke Saito; Zhen-Hua Ling; Fernando Villavicencio; Junichi Yamagishi; Jaime Lorenzo-Trueba; Tomi Kinnunen


conference of the international speech communication association | 2016

Applying spectral normalisation and efficient envelope estimation and statistical transformation for the voice conversion challenge 2016

Fernando Villavicencio; Junichi Yamagishi; Jordi Bonada; Felipe Espic

Collaboration


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Junichi Yamagishi

National Institute of Informatics

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Tomoki Toda

National Institute of Information and Communications Technology

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Jordi Bonada

Pompeu Fabra University

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Isao Echizen

National Institute of Informatics

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Nobutaka Ono

National Institute of Informatics

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Sayaka Shiota

Nagoya Institute of Technology

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Tomoko Matsui

International Christian University

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Jaime Lorenzo-Trueba

Technical University of Madrid

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Ling-Hui Chen

University of Science and Technology of China

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