Parallel and Limited Data Voice Conversion Using Stochastic Variational Deep Kernel Learning

Typically, voice conversion is regarded as an engineering problem with limited training data. The reliance on massive amounts of data hinders the practical applicability of deep learning approaches, which have been extensively researched in recent years. On the other hand, statistical methods are effective with limited data but have difficulties in modelling complex mapping functions. This paper proposes a voice conversion method that works with limited data and is based on stochastic variational deep kernel learning (SVDKL). At the same time, SVDKL enables the use of deep neural networks' expressive capability as well as the high flexibility of the Gaussian process as a Bayesian and non-parametric method. When the conventional kernel is combined with the deep neural network, it is possible to estimate non-smooth and more complex functions. Furthermore, the model's sparse variational Gaussian process solves the scalability problem and, unlike the exact Gaussian process, allows for the learning of a global mapping function for the entire acoustic space. One of the most important aspects of the proposed scheme is that the model parameters are trained using marginal likelihood optimization, which considers both data fitting and model complexity. Considering the complexity of the model reduces the amount of training data by increasing the resistance to overfitting. To evaluate the proposed scheme, we examined the model's performance with approximately 80 seconds of training data. The results indicated that our method obtained a higher mean opinion score, smaller spectral distortion, and better preference tests than the compared methods

Mapping Techniques for Voice Conversion

Speaker identity plays an important role in human communication. In addition to the linguistic content, speech utterances contain acoustic information of the speaker characteristics. This thesis focuses on voice conversion, a technique that aims at changing the voice of one speaker (a source speaker) into the voice of another specific speaker (a target speaker) without changing the linguistic information. The relationship between the source and target speaker characteristics is learned from the training data. Voice conversion can be used in various applications and fields: text-to-speech systems, dubbing, speech-to-speech translation, games, voice restoration, voice pathology, etc. Voice conversion offers many challenges: which features to extract from speech, how to find linguistic correspondences (alignment) between source and target features, which machine learning techniques to use for creating a mapping function between the features of the speakers, and finally, how to make the desired modifications to the speech waveform. The features can be any parameters that describe the speech and the speaker identity, e.g. spectral envelope, excitation, fundamental frequency, and phone durations. The main focus of the thesis is on the design of suitable mapping techniques between frame-level source and target features, but also aspects related to parallel data alignment and prosody conversion are addressed. The perception of the quality and the success of the identity conversion are largely subjective. Conventional statistical techniques are able to produce good similarity between the original and the converted target voices but the quality is usually degraded. The objective of this thesis is to design conversion techniques that enable successful identity conversion while maintaining the original speech quality. Due to the limited amount of data, statistical techniques are usually utilized in extracting the mapping function. The most popular technique is based on a Gaussian mixture model (GMM). However, conventional GMM-based conversion suffers from many problems that result in degraded speech quality. The problems are analyzed in this thesis, and a technique that combines GMM-based conversion with partial least squares regression is introduced to alleviate these problems. Additionally, approaches to solve the time-independent mapping problem associated with many algorithms are proposed. The most significant contribution of the thesis is the proposed novel dynamic kernel partial least squares regression technique that allows creating a non-linear mapping function and improves temporal correlation. The technique is straightforward, efficient and requires very little tuning. It is shown to outperform the state-of-the-art GMM-based technique using both subjective and objective tests over a variety of speaker pairs. In addition, quality is further improved when aperiodicity and binary voicing values are predicted using the same technique. The vast majority of the existing voice conversion algorithms concern the transformation of the spectral envelopes. However, prosodic features, such as fundamental frequency movements and speaking rhythm, also contain important cues of identity. It is shown in the thesis that pure prosody alone can be used, to some extent, to recognize speakers that are familiar to the listeners. Furthermore, a prosody conversion technique is proposed that transforms fundamental frequency contours and durations at syllable level. The technique is shown to improve similarity to the target speaker’s prosody and reduce roboticness compared to a conventional frame-based conversion technique. Recently, the trend has shifted from text-dependent to text-independent use cases meaning that there is no parallel data available. The techniques proposed in the thesis currently assume parallel data, i.e. that the same texts have been spoken by both speakers. However, excluding the prosody conversion algorithm, the proposed techniques require no phonetic information and are applicable for a small amount of training data. Moreover, many text-independent approaches are based on extracting a sort of alignment as a pre-processing step. Thus the techniques proposed in the thesis can be exploited after the alignment process

Efficient Approaches for Voice Change and Voice Conversion Systems

In this thesis, the study and design of Voice Change and Voice Conversion systems are presented. Particularly, a voice change system manipulates a speaker’s voice to be perceived as it is not spoken by this speaker; and voice conversion system modifies a speaker’s voice, such that it is perceived as being spoken by a target speaker. This thesis mainly includes two sub-parts. The first part is to develop a low latency and low complexity voice change system (i.e. includes frequency/pitch scale modification and formant scale modification algorithms), which can be executed on the smartphones in 2012 with very limited computational capability. Although some low-complexity voice change algorithms have been proposed and studied, the real-time implementations are very rare. According to the experimental results, the proposed voice change system achieves the same quality as the baseline approach but requires much less computational complexity and satisfies the requirement of real-time. Moreover, the proposed system has been implemented in C language and was released as a commercial software application. The second part of this thesis is to investigate a novel low-complexity voice conversion system (i.e. from a source speaker A to a target speaker B) that improves the perceptual quality and identity without introducing large processing latencies. The proposed scheme directly manipulates the spectrum using an effective and physically motivated method – Continuous Frequency Warping and Magnitude Scaling (CFWMS) to guarantee high perceptual naturalness and quality. In addition, a trajectory limitation strategy is proposed to prevent the frame-by-frame discontinuity to further enhance the speech quality. The experimental results show that the proposed method outperforms the conventional baseline solutions in terms of either objective tests or subjective tests

Non-Parallel Articulatory-to-Acoustic Conversion Using Multiview-based Time Warping

This work was supported in part by the Spanish State Research Agency (SRA) grant number PID2019-108040RB-C22/SRA/10.13039/501100011033, and the FEDER/Junta de AndalucíaConsejería de Transformación Económica, Industria, Conocimiento y Universidades project no. B-SEJ-570-UGR20.In this paper, we propose a novel algorithm called multiview temporal alignment by dependence maximisation in the latent space (TRANSIENCE) for the alignment of time series consisting of sequences of feature vectors with different length and dimensionality of the feature vectors. The proposed algorithm, which is based on the theory of multiview learning, can be seen as an extension of the well-known dynamic time warping (DTW) algorithm but, as mentioned, it allows the sequences to have different dimensionalities. Our algorithm attempts to find an optimal temporal alignment between pairs of nonaligned sequences by first projecting their feature vectors into a common latent space where both views are maximally similar. To do this, powerful, nonlinear deep neural network (DNN) models are employed. Then, the resulting sequences of embedding vectors are aligned using DTW. Finally, the alignment paths obtained in the previous step are applied to the original sequences to align them. In the paper, we explore several variants of the algorithm that mainly differ in the way the DNNs are trained. We evaluated the proposed algorithm on a articulatory-to-acoustic (A2A) synthesis task involving the generation of audible speech from motion data captured from the lips and tongue of healthy speakers using a technique known as permanent magnet articulography (PMA). In this task, our algorithm is applied during the training stage to align pairs of nonaligned speech and PMA recordings that are later used to train DNNs able to synthesis speech from PMA data. Our results show the quality of speech generated in the nonaligned scenario is comparable to that obtained in the parallel scenario.Spanish State Research Agency (SRA) PID2019-108040RB-C22/SRA/10.13039/501100011033FEDER/Junta de AndalucíaConsejería de Transformación Económica, Industria, Conocimiento y Universidades project no. B-SEJ-570-UGR20

Developing Sparse Representations for Anchor-Based Voice Conversion

Voice conversion is the task of transforming speech from one speaker to sound as if it was produced by another speaker, changing the identity while retaining the linguistic content. There are many methods for performing voice conversion, but oftentimes these methods have onerous training requirements or fail in instances where one speaker has a nonnative accent. To address these issues, this dissertation presents and evaluates a novel “anchor-based” representation of speech that separates speaker content from speaker identity by modeling how speakers form English phonemes. We call the proposed method Sparse, Anchor-Based Representation of Speech (SABR), and explore methods for optimizing the parameters of this model in native-to-native and native-to-nonnative voice conversion contexts. We begin the dissertation by demonstrating how sparse coding in combination with a compact, phoneme-based dictionary can be used to separate speaker identity from content in objective and subjective tests. The formulation of the representation then presents several research questions. First, we propose a method for improving the synthesis quality by using the sparse coding residual in combination with a frequency warping algorithm to convert the residual from the source to target speaker’s space, and add it to the target speaker’s estimated spectrum. Experimentally, we find that synthesis quality is significantly improved via this transform. Second, we propose and evaluate two methods for selecting and optimizing SABR anchors in native-to-native and native-to-nonnative voice conversion. We find that synthesis quality is significantly improved by the proposed methods, especially in native-to- nonnative voice conversion over baseline algorithms. In a detailed analysis of the algorithms, we find they focus on phonemes that are difficult for nonnative speakers of English or naturally have multiple acoustic states. Following this, we examine methods for adding in temporal constraints to SABR via the Fused Lasso. The proposed method significantly reduces the inter-frame variance in the sparse codes over other methods that incorporate temporal features into sparse coding representations. Finally, in a case study, we examine the use of the SABR methods and optimizations in the context of a computer aided pronunciation training system for building “Golden Speakers”, or ideal models for nonnative speakers of a second language to learn correct pronunciation. Under the hypothesis that the optimal “Golden Speaker” was the learner’s voice, synthesized with a native accent, we used SABR to build voice models for nonnative speakers and evaluated the resulting synthesis in terms of quality, identity, and accentedness. We found that even when deployed in the field, the SABR method generated synthesis with low accentedness and similar acoustic identity to the target speaker, validating the use of the method for building “golden speakers”