4,891 research outputs found
Efficient Gaussian Mixture Model Evaluation in Voice Conversion
Abstract Voice conversion refers to the adaptation of the characteristics of a source speaker's voice to those of a target speaker. Gaussian mixture models (GMM) have been found to be efficient in the voice conversion task. The GMM parameters are estimated from a training set with the goal to minimize the mean squared error (MSE) between the transformed and target vectors. Obviously, the quality of the GMM model plays an important role in achieving better voice conversion quality. This paper presents a very efficient approach for the evaluation of GMM models directly from the model parameters without using any test data, facilitating the improvement of the transformation performance especially in the case of embedded implementations. Though the proposed approach can be used in any application that utilizes GMM based transformation, we take voice conversion as an example application throughout the paper. The proposed approach is experimented with in this context and evaluated against an MSE based evaluation method. The results show that the proposed method is in line with all subjective observations and MSE results
Analysis of a Modern Voice Morphing Approach using Gaussian Mixture Models for Laryngectomees
This paper proposes a voice morphing system for people suffering from
Laryngectomy, which is the surgical removal of all or part of the larynx or the
voice box, particularly performed in cases of laryngeal cancer. A primitive
method of achieving voice morphing is by extracting the source's vocal
coefficients and then converting them into the target speaker's vocal
parameters. In this paper, we deploy Gaussian Mixture Models (GMM) for mapping
the coefficients from source to destination. However, the use of the
traditional/conventional GMM-based mapping approach results in the problem of
over-smoothening of the converted voice. Thus, we hereby propose a unique
method to perform efficient voice morphing and conversion based on GMM,which
overcomes the traditional-method effects of over-smoothening. It uses a
technique of glottal waveform separation and prediction of excitations and
hence the result shows that not only over-smoothening is eliminated but also
the transformed vocal tract parameters match with the target. Moreover, the
synthesized speech thus obtained is found to be of a sufficiently high quality.
Thus, voice morphing based on a unique GMM approach has been proposed and also
critically evaluated based on various subjective and objective evaluation
parameters. Further, an application of voice morphing for Laryngectomees which
deploys this unique approach has been recommended by this paper.Comment: 6 pages, 4 figures, 4 tables; International Journal of Computer
Applications Volume 49, Number 21, July 201
Sampling-based speech parameter generation using moment-matching networks
This paper presents sampling-based speech parameter generation using
moment-matching networks for Deep Neural Network (DNN)-based speech synthesis.
Although people never produce exactly the same speech even if we try to express
the same linguistic and para-linguistic information, typical statistical speech
synthesis produces completely the same speech, i.e., there is no
inter-utterance variation in synthetic speech. To give synthetic speech natural
inter-utterance variation, this paper builds DNN acoustic models that make it
possible to randomly sample speech parameters. The DNNs are trained so that
they make the moments of generated speech parameters close to those of natural
speech parameters. Since the variation of speech parameters is compressed into
a low-dimensional simple prior noise vector, our algorithm has lower
computation cost than direct sampling of speech parameters. As the first step
towards generating synthetic speech that has natural inter-utterance variation,
this paper investigates whether or not the proposed sampling-based generation
deteriorates synthetic speech quality. In evaluation, we compare speech quality
of conventional maximum likelihood-based generation and proposed sampling-based
generation. The result demonstrates the proposed generation causes no
degradation in speech quality.Comment: Submitted to INTERSPEECH 201
A silent speech system based on permanent magnet articulography and direct synthesis
In this paper we present a silent speech interface (SSI) system aimed at restoring speech communication for individuals who have lost their voice due to laryngectomy or diseases affecting the vocal folds. In the proposed system, articulatory data captured from the lips and tongue using permanent magnet articulography (PMA) are converted into audible speech using a speaker-dependent transformation learned from simultaneous recordings of PMA and audio signals acquired before laryngectomy. The transformation is represented using a mixture of factor analysers, which is a generative model that allows us to efficiently model non-linear behaviour and perform dimensionality reduction at the same time. The learned transformation is then deployed during normal usage of the SSI to restore the acoustic speech signal associated with the captured PMA data. The proposed system is evaluated using objective quality measures and listening tests on two databases containing PMA and audio recordings for normal speakers. Results show that it is possible to reconstruct speech from articulator movements captured by an unobtrusive technique without an intermediate recognition step. The SSI is capable of producing speech of sufficient intelligibility and naturalness that the speaker is clearly identifiable, but problems remain in scaling up the process to function consistently for phonetically rich vocabularies
- …