 Research
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Gridbased approximation for voice conversion in low resource environments
EURASIP Journal on Audio, Speech, and Music Processing volume 2016, Article number: 3 (2016)
Abstract
The goal of voice conversion is to modify a source speaker’s speech to sound as if spoken by a target speaker. Common conversion methods are based on Gaussian mixture modeling (GMM). They aim to statistically model the spectral structure of the source and target signals and require relatively large training sets (typically dozens of sentences) to avoid overfitting. Moreover, they often lead to muffled synthesized output signals, due to excessive smoothing of the spectral envelopes.
Mobile applications are characterized with low resources in terms of training data, memory footprint, and computational complexity. As technology advances, computational and memory requirements become less limiting; however, the amount of available training data still presents a great challenge, as a typical mobile user is willing to record himself saying just few sentences. In this paper, we propose the gridbased (GB) conversion method for such low resource environments, which is successfully trained using very few sentences (5–10). The GB approach is based on sequential Bayesian tracking, by which the conversion process is expressed as a sequential estimation problem of tracking the target spectrum based on the observed source spectrum. The converted Mel frequency cepstrum coefficient (MFCC) vectors are sequentially evaluated using a weighted sum of the target training vectors used as grid points. The training process includes simple computations of Euclidian distances between the training vectors and is easily performed even in cases of very small training sets.
We use global variance (GV) enhancement to improve the perceived quality of the synthesized signals obtained by the proposed and the GMMbased methods. Using just 10 training sentences, our enhanced GB method leads to converted sentences having closer GV values to those of the target and to lower spectral distances at the same time, compared to enhanced version of the GMMbased conversion method. Furthermore, subjective evaluations show that signals produced by the enhanced GB method are perceived as more similar to the target speaker than the enhanced GMM signals, at the expense of a small degradation in the perceived quality.
Introduction
Voice conversion systems aim to modify the perceived identity of a source speaker saying a sentence to that of a given target speaker. This kind of transformation is useful for personalization of texttospeech (TTS) systems, voice restoration in case of vocal pathology, obtaining a false identity when answering the phone (for safety reasons, for example), and also for entertainment purposes such as online roleplaying games.
The identity of a speaker is associated with the spectral envelope of the speech signal, and with its prosody attributes: pitch, duration, and energy. Most voice conversion methods aim to transform the spectral envelope of the source speaker to the spectral envelope of the target speaker. The pitch contour is commonly converted by a linear transformation based on the global mean and standard deviation values of the pitch frequency.
The classical conversion method, based on modeling the spectral structure of the speech signals using Gaussian mixture model (GMM), is the most commonly used method to date. The conversion function is linear, trained using either least squares (LS) [1], or a joint sourcetarget GMM training (JGMM) [2]. These linear conversion methods produce oversmoothed spectral envelopes leading to muffled synthesized speech [3, 4]. Several modifications of the GMMbased conversion have been proposed since, among these are as follows: GMM with dynamic frequency warping (DFW) [4], GMM and codebook selection [5], and a combined pitch and spectral envelope GMMbased conversion [6]. Still, these GMMbased conversion methods have been reported to produce muffled output signals, probably due to excessive smoothing of the temporal evolution of the spectral envelope. Recently, a different approach aiming to capture the temporal evolution of the spectral envelope was presented [7]. A GMM is trained using concatenated sequences of the source and target spectral features, and the conversion function is evaluated using maximum likelihood (ML) estimation. To reduce the muffling effect, the global variance (GV) of the spectral features was considered in the trained statistical model. A GV enhancement method called CGMM was also proposed, [8], in the framework of the classical GMMbased conversion, where the GV of the converted features is constrained to match the GV of the features related to the target speaker. These two conversion schemes (with integrated GV enhancement) improve the quality of the converted signals, at the expense of some increase in the spectral distance between the converted and target signals. A realtime implementation for the ML approach have also been proposed [9]. This implementation is based on a lowdelay estimation of the conversion parameters [10] using recursive parameter generation and GV enhancement.
In order to estimate a conversion function from a source speaker to a target speaker, voice conversion methods use training sets of both speakers. Most training algorithms require parallel data sets, that is, prerecorded sentences of the source and target speakers saying the same text. In such a setup, evaluation of a conversion function is based on coupled feature vectors—source and target. Alternatively, some methods have been proposed, suggesting training algorithms which avoid the need for prealignment altogether. A probabilistic approach presented by Nankaku et al. includes statistical modeling for optimizing the conversion function and the correspondence between sourcetarget segments [11]. Another method which does not require time alignment as a preprocessing stage is the iterative combination of a nearest neighbour search step and a conversion step alignment method (INCA) [12]. This method uses iterative estimation of the alignment (using nearest neighbour search) and conversion estimation (classical GMM conversion). Recently, we proposed a modified version of this method called temporalcontext INCA (TCINCA), using context vectors instead of single spectral vectors, which lead to improved estimation of the alignment and to higher quality and similarity to the target speaker [13]. Although these methods were designed for a nonparallel setup, they can be used in a parallel setup, when aligned data is unavailable.
Even when a parallel training set is available, matching an analysis frame of the source speaker to one of the analysis frames of the target speaker is not straightforward, since the two speakers generally do not pronounce the text at the exact same rate. A time alignment is usually carried out using dynamic time warping (DTW), constrained by starting and ending of speech utterances [14]. These time stamps are commonly obtained by phonetic labeling, representing the beginning and ending of each phoneme. Since the source and target training sentences are not spoken in exactly the same rate, DTW often replicates or omits feature vectors, artificially producing a match. The importance of correct time alignment was recently demonstrated as having a large influence on the quality of the synthesized converted speech [15]. A different approach was suggested by [16], where a statistical model for an eigenvoice was trained using several parallel data sets. The conversion function is trained using the eigenvoice model and speech sentences related to a target speaker (not necessarily parallel to the source data sets).
The GMMbased conversion methods mentioned above, using either parallel or nonparallel data, typically require several dozens of sentences for training, and therefore when applied in a mobile environment impose a long recording session on the user. Even the low delay GMMbased approach suggested by Toda et al. was reported to be trained using 60–250 mixtures and 50 training sentences [9]. Therefore, applying them in a mobile environment would compel the user to a long recording session.
Some approaches for training a conversion function that are not based on GMM have been proposed, among them training using a statespace representation [17], and using exemplarbased sparserepresentation [18]. Since these methods are closely related to the proposed GB method, we address them and discuss the differences between them and the GB approach in more details after describing the proposed method in this work (see Section 4). Still, these method are also not suitable for mobile environment since they require several hundreds of parallel training sentences and/or very high computational load during conversion and a substantial memory footprint.
In this paper, we propose a method for spectral conversion based on a gridbased (GB) approximation [19]. We express the spectral conversion process as a sequential Bayesian estimation problem of tracking the target spectrum using observed samples from the source spectrum. We propose models for evaluation of the evidence and likelihood probabilities needed for the GB formulation. Using these approximated probabilities, the algorithm sequentially evaluates the converted spectrum as a weighted sum of the target training vectors. Recently, we presented a similar method using GB approximation which requires phonetic labeling during the test stage [20]. In this paper, we propose a modified version of this method, which does not require any labeling for testing. Additionally, as in TCINCA [13], we use context vectors instead of single vectors in order to improve the estimation of the likelihood probability. Altogether, we present here a more thorough description of the GB approximation method and its modified application for voice conversion under low resources constraints, followed by an extended analysis and detailed results.
Furthermore, as opposed to previously proposed methods that use parallel and timealigned training sets, the GB conversion approach does not require a onetoone correspondence between the source and target training vectors. The training process uses parallel sentences but is based on soft correspondence between the source and target vectors, obtained by phonetic labeling of the training sentences without frame alignment, thus eliminating the need for DTW.
Unlike other GMMbased methods that use statistical modeling of the spatial structure of the source and target spectra, the GB method is datadriven, so it is easily trained using merely 5–10 sentences. Its training stage involves simple computations based on the Euclidean distance between the training vectors.
Objective evaluations show that the GB conversion method proposed here leads to GV values that are closer to the GV values of the target speaker than the classical GMM conversion method and to lowest (or very close to it) spectral distance to the target spectra, at the same time. To further improve the quality, we applied a GV enhancement postprocessing block. We recently proposed this GV enhancement approach and examined its effect on signals converted by a classical GMM conversion method [21]. In this paper, we present an overall scheme, enhanced GB (EnGB), consisting of GB conversion, followed by GV enhancement. We used objective measures and performed extensive subjective evaluations to compare our proposed EnGB scheme to joint GMM (JGMM) [2], also followed by the same GV enhancement block (EnJGMM) and to a GMMbased conversion, trained with a GV constraint (CGMM) [8]. Objectively, EnGB leads to better performance than EnJGMM and CGMM in terms of both spectral distance and GV, using 10 sentences. Listening tests show that in terms of similarity to the target, EnGB outperforms the other examined methods. In terms of quality, CGMM was rated as best, where EnGB was rated as comparable to EnGMM.
This paper is organized as follows. In Section 2, a brief description of GB approximation is presented. The GB conversion method is described in Section 3. The difference between the GB approach and some related methods is discussed in Section 4. Experimental results, demonstrating the performance of the proposed EnGB scheme compared to EnGMMbased methods, are presented in Section 5. Conclusions and further research suggestions are given in Section 6.
Gridbased formulation
A brief formulation of sequential estimation using Bayesian tracking is presented in Section 2.1. In many practical cases, applying this formulation yields a high computational load, which is sometimes unfeasible. The GB method provides a discrete approximation for Bayesian tracking with much less computational complexity, as described in Section 2.2.
Bayesian tracking
Let y _{ t } denote a hidden state vector that follows a firstorder Markov dynamics as
where f _{ t } is a function (not necessarily linear) of y _{ t−1} and of an i.i.d. noise sequence u _{ t }. The observed signal, x _{ t }, depends on the hidden state and on an i.i.d. measurement noise, v _{ t }:
where h _{ t }(·) may also be nonlinear.
The Bayesian optimal estimate for the state vector y _{ t } in terms of minimizing the mean square error, given t vectors sequentially sampled from the observed process—\(\mathbf {x}_{1:t}\triangleq \{\mathbf {x}_{1},\ldots,\mathbf {x}_{t}\}\), is obtained by^{1}
The posterior probability p(y _{ t }x _{1:t }) can be obtained recursively in two stages:

1.
Prediction: obtain the prior probability
$$ p\left(\mathbf{y}_{t}\mathbf{x}_{1:t1}\right) = \int p\left(\mathbf{y}_{t}\mathbf{y}_{t1}\right) p\left(\mathbf{y}_{t1}\mathbf{x}_{1:t1}\right)d \mathbf{y}_{t1}. $$((4)) 
2.
Update: use the current observation x _{ t } to update the posterior probability
$$ p\left(\mathbf{y}_{t}\mathbf{x}_{1:t}\right) = \frac{p\left(\mathbf{x}_{t}\mathbf{y}_{t}\right) p\left(\mathbf{y}_{t}\mathbf{x}_{1:t1}\right)}{p\left(\mathbf{x}_{t}\mathbf{x}_{1:t1}\right)}, $$((5))where
$$ p\left(\mathbf{x}_{t}\mathbf{x}_{1:t1}\right) = \int{p\left(\mathbf{x}_{t}\mathbf{y}_{t}\right)p\left(\mathbf{y}_{t}\mathbf{x}_{1:t1}\right)d\mathbf{y}_{t}}. $$((6))
This recursion is initialized by setting the prior probability to be equal to the initial probability of the state vector: p(y _{0}x _{0})=p(y _{0}), where p(y _{0}) is assumed to be known (in practice, mostly taken as a uniform distribution). The likelihood function p(x _{ t }y _{ t }) that appears in Eq. (5) is determined according to the measurement model (Eq. (2)) and the statistics of the measurement noise v _{ t }.
When the noise signals u _{ t } and v _{ t } are Gaussian, and the functions f _{ t }(·) and h _{ t }(·) are linear and time invariant (meaning that f _{ t }(·)≡f(·) and h _{ t }(·)≡h(·)), this recursion can be computed analytically, leading to Kalman filtering [22]. Yet, in most practical cases where these conditions are not sustained, this derivation is hard and often performed using approximation methods such as GB approximation or particle filtering [19]. These methods sequentially evaluate the posterior probability as a discrete weighted sum using a given set of samples in case of GB or a randomly drawn set in case of particle filtering.
In this paper, we express the spectral conversion process as a sequential estimation problem tracking the target spectrum, using observed samples from the source spectrum. We propose models for the evidence and likelihood probabilities needed for the GB formulation. Using these approximated probabilities the algorithm sequentially evaluates the converted spectrum as a weighted sum of the target training vectors. It is well known that the performance of particle filtering crucially depends on successful statistical modeling of the statespace temporal evolution. The performance of GB, on the other hand, depends on dense modeling of the state space by a set of predetermined grid points. Nevertheless, in the following sections, we show that 5–10 training sentences alone, which still result in several thousands of spectral feature vectors, are sufficient for training a GB conversion. Our subjective evaluations show that the GB conversion is found to be better or comparable, at least, to the classical GMM conversion method, when trained by this small set.
Gridbased approximation
The main principle of GB approximation is to provide a Bayesian sequential estimation framework while avoiding the integral computations in Eqs. (4) and (6) by using a discrete evaluation of the posterior probability.
Let \(\left \{\mathbf {y}_{t}^{k}\right \}_{k=1}^{N_{y}}\) be a set of predetermined grid points taken from the state space {y _{ t }}. We divide the state space into cells, so that each cell has a grid point \(\mathbf {y}_{t}^{k}\) as its center. Thus, the posterior probability can be approximated by^{2}
where the posterior weights \(\left \{w_{tt}^{k}\right \}_{k=1}^{N_{y}}\) denote the conditional probabilities
Using this discrete approximation, the prior probability is also approximated as a discrete sum
The prior weights can be estimated sequentially [19]
where \(p\left (\mathbf {y}_{t}^{k}{\mathbf {y}_{t1}^{l}}\right)\), called the evidence probability, is derived from the state space dynamics (Eq. (1)). The posterior weights \(\left \{w_{tt}^{k}\right \}_{k=1}^{N_{y}}\) are evaluated by the following:
where, as stated above, the likelihood probability \(p\left (\mathbf {x}_{t}{\mathbf {y}_{t}^{k}}\right)\) is derived from the measurement model (Eq. (2)).
Finally, the hidden state vector y _{ t } is approximated using the posterior weights
Note that Eqs. (10), (11), and (12) are discrete evaluations of Eqs. (4)–(3), correspondingly. It is known [19] that the estimated terms in Eq. (7) and in Eq. (12) are biased for any finite N _{ y }. Still, as more grid points are taken, the bias gets smaller and the approximation improves, since the state space is more densely represented.
The sequential estimation process is initialized using the initial probability of the state vector \(p\left ({\mathbf {y}_{0}^{k}}\right)\), which as stated above, is assumed to be known
Table 1 summarizes the main stages of sequential Bayesian estimation using GB approximation.
Voice conversion using gridbased approximation
We now use the GB approximation method described above as a framework for spectral voice conversion. We express the conversion as a sequential estimation problem, where the observed process is the source spectrum, and the tracked state space is the target spectrum. We propose models for both likelihood and evidence densities, required for the sequential estimation process, as described in Eqs. (10)–(12).
The GB conversion method proposed here uses a parallel training set but does not require time alignment between the source and target training vectors since it is trained using soft correspondence between them, rather than matched pairs. The training and conversion stages of the proposed GB conversion method are presented below in Sections 3.1 and 3.2, respectively.
Training stage
The training process described here includes precomputation of the evidence and discrete likelihood probabilities. These probabilities are evaluated using all available training data. Note the difference from our previously presented GB method, where these probabilities were evaluated separately for each phoneme [20]. The source and target training sentences are assumed to be parallel and phonetically labeled. The spectral features of the two speakers are extracted from the voiced frames, but, as stated above, no time alignment is performed. Instead, a matching process of the source and target utterances is performed as follows. Each sequence of frames related to a certain phoneme at the source is matched to its corresponding sequence at the target, according to the phonetic labeling. When matching frames extracted from recordings of the word ‘father’, for example, the sequence of frames related to the phoneme ‘f’ at the source is matched to the sequence of frames related to the phoneme ‘f’, taken from the target’s recording of this word. The same is done for ‘a’, ‘th’, etc. Note that although matched sequences mostly have different lengths, our training process does not require using an alignment procedure such as DTW, unlike GMMbased methods do. Based on the matched sequences, we model the discrete likelihood probability used in Eq. (11), as follows:
where \(\{\mathbf {x}^{m}\}_{m=1}^{N_{x}}\) and \(\{\mathbf {y}^{k}\}_{k=1}^{N_{y}}\) are source and target training vectors, respectively. We normalize the obtained discrete likelihood probability so that
The discrete likelihood probability defines a relaxed correspondence between the source and target training vectors, as opposed to a onetoone match defined in other parallel methods, for which p(x _{ t }=x ^{m}y _{ t }=y ^{k})=δ _{ m,k }.
The evidence probability, as mentioned before, expresses the transition probability from state y ^{l} to state y ^{k}. In natural speech, spectral feature vectors related to consecutive time frames are typically similar, but not identical. Motivated by this behaviour, we model the transition probability as having the same value for all the states inside a ball, centered at y ^{k} with a radius R _{ y }. The probability of transitions to farther states, however, is taken as a simple Gaussian distribution, centered at y ^{k}. Altogether, we model the discrete evidence probability, used in Eq. (10), as follows:
where the exponential term in Eq. (16) is the maximum between the Mel cepstral distortion (MCD) of the two states y ^{l} and y ^{k} normalized by a parameter R _{ y }, and 1
where y ^{p}(p) and y ^{l}(p) are the pth elements of y ^{k} and y ^{l}, respectively. An alternative approach would be to take the exponential term, defined in Eq. (17), as a normalized distance. For example, M _{ k,l }=MCD(y ^{k},y ^{l})/R _{ y }, where R _{ y } is a parameter selected by the user. However, in case of a sparse training set, the most substantial probability would be for staying in the same state. Since the training set is fixed, the likelihood and evidence densities are in fact time invariant.
Conversion stage
The likelihood probability modeled above in Eq. (14) is defined only for a discrete set consisting of the source training vector. In this section, we extend Eq. (14) to model any input vector x _{ t }∈R^{P}, as required by the GB formulation.
In our previous work dealing with GB conversion [20], we modeled the continuous likelihood probability p(x _{ t }y _{ t }=y ^{k}) as a sum of the discrete likelihood probabilities p(x ^{m}y _{ t }=y ^{k}), m=1,…,N _{ x }, (defined in Eqs. (14) and (15)), each weighted by a Gaussian kernel, centered at x ^{m}
where R _{ x } is a parameter determined by the user. The Gaussian term \(e^{{\text {MCD}^{2}\left (\mathbf {x}_{t}, \mathbf {x}^{m}\right)}/{2{R_{x}^{2}}}}\) can be viewed as an interpolation factor from the discrete space represented by the source training vectors to the continuous space of the test source vectors.
Denote X _{ t }=(x _{ t−τ/2},…,x _{ t },…,x _{ t+τ/2}) as context test vector—a sequence of test source vectors. Also, denote \(\{\mathbf {X}_{t}^{m}\}_{m=1}^{N_{x}}\) as training context vectors similarly obtained from the source training set. Previously, in [13], we have shown that Euclidian distance between context vectors leads to improved spectral matching compared with Euclidian distance between single vectors. Although that was shown for matching spectral segments of two different speakers, it is certainly beneficial for matching spectral segments taken from the same speaker. Therefore, we substitute the MCD term in the Gaussian kernel in Eq. (19) with the mean MCD between context vectors, i.e.,
Define \(w_{tt}^{k}\) as the posterior weights corresponding to the training vectors \(\left \{\mathbf {y}^{k}\right \}_{k=1}^{N_{y}}\)
During conversion, the posterior weights are sequentially evaluated, using the corresponding evidence and likelihood probabilities defined in Eqs. (16) and (20), according to Eqs. (10) and (11). The posterior weights are used to obtain the converted outcome as a discrete Bayesian approximation (as defined in Eq. (12))
Due to the sequential update of the posterior weights, the converted spectral outputs evolve smoothly in time, within each phonetic segment, also during transitions between phonemes. Figure 1 demonstrates the obtained time evolution of the first and third Mel frequency cepstrum coefficients (MFCCs) using GB conversion, compared to the classical GMMbased conversion—JGMM [2]. The classical GMMbased conversion is applied frame by frame which may lead to discontinuities. The proposed GB, however, is based on a sequential update leading to a smoother time evolution of the cepstral elements, as seen in Fig. 1.
To conclude, the main stages of converting a sequence of source vectors are summarized in Table 2.
Discussion
The GB approach uses a state space representation of the source and target spectra to obtain a converted spectra as a weighted sum of the target training vectors. In this section, we address two related methods: (1) a method based on state space representation [17] and (2) an exemplarbased approach [18], where the converted spectra is evaluated as a weighted sum of the target training vectors. We discuss here the similarities and differences between these methods and our proposed approach.
In [17], a state space approach for representing speech spectra as an observed process generated from an underling sequence of a hidden Markov process has been proposed. The source and target speech are both modeled using this state space representation. The state space parameters are divided into two parts: a common part related to the uttered speech (assuming a parallel training set) and a differentia part related to the difference between the speakers. These parts are evaluated during training time using an iterative algorithm known as expectation maximization (EM) [23]. During the test, the common parameters related to the test utterance are evaluated using EM and then used, along with the trained differentia part to obtain the converted spectra. Both training and conversion stages include iterative training (EM). Conversion results reported by the authors were obtained using several hundreds of parallel training sentences. Although our method and Xu et al.’s method [17] both use state space for representing the temporal evolution of the speech spectra, in our method, the source and the target spectra are linked through a state space dynamics, while in Xu et al.’s approach, the parallel source and target spectra are each modeled as the observed signals of a shared underlined unobserved Markov process.
An exemplarbased sparse representation approach for voice conversion has been proposed in [18]. Each speech signal is modeled as a linear combination of basis vectors (the training vectors), where the weighting matrix is called an activation matrix. The main assumption used in this method is that the speaker’s identity is modeled by the basis vectors, where the information regarding the uttered text lies entirely in the activation matrix. Therefore, given a test source signal, its activation matrix is evaluated and then multiplied by the target training set, used as the target basis vectors, to obtain the converted spectra. Therefore, this method does not require any training, but its testing stage includes high computational load and a substantial memory footprint. As the exemplarbased method, our proposed GB method also uses a linear combination of the target training vectors. Besides the obvious differences in the models used by the two methods, there are two major differences: (1) We use sequential evaluation of the weights to ensure smooth temporal evolution while in the exemplarbased method, the activation matrix is evaluated as a batch. (2) We use scalar weights while the exemplarbased method uses weighting vectors (the activation matrix).
Experimental results
Experiments setup
In our experiments, we used speech sentences of four US English speakers taken from the CMU ARCTIC database [24]: two males (bdl, rms) and two females (clb, slt). Two different sizes of training sets 5 and 10 parallel sentences were used to demonstrate the performance of the examined methods as a function of training set size. The testing set consisted of 50 additional parallel sentences. All sentences were sampled at 16 kHz and were phonetically labeled.
Analysis and synthesis were both carried out using an available vocoder [25]. This vocoder uses a twoband harmonic/noise parametrization, separated by a maximal voicing frequency for representing each spectral envelope [26]. Twentyfive MFCCs were extracted from the harmonic parameters [27]: the zeroth coefficients, related to the energy, were not converted. The other 24 coefficients were used as spectral feature vectors during training and conversion.
The spectral features of unvoiced frames were not converted but simply copied to the converted sentence, since they do not capture much of the speaker’s individuality [28] and their conversion often leads to quality degradation [29]. The maximal voicing frequency was also not converted but reestimated from the converted parameters by the vocoder. The sequences of the training data set used for GB conversion were matched (without alignment), as described in Section 3.1. The training set used for the other examined methods, and the testing set, were each time aligned using a DTW algorithm based on phonetic labeling [14].
Pitch was converted by a simple linear function using the mean and standard deviation values of the source and target speakers,
where \({f}_{0}^{(x),t}\) and \({\hat {f}}_{0}^{(y),t}\) are the pitch values of the source and converted signals at the tth frame, respectively. The parameters μ ^{(x)} and μ ^{(y)} are the mean pitch values, and σ ^{(x)} and σ ^{(y)} are the standard deviations of the source and target pitch values, respectively. In this case, the mean and standard deviation of the converted pitch contour match the mean and standard deviation of the pitch values of the target speaker.
Objective evaluations
We evaluated the performance of the examined conversion methods by two objective measures: normalized distortion (ND) and normalized GV (NGV), as defined below.
To obtain a fair comparison between different sourcetarget pairs, we normalized the mean spectral distortion between the converted and target signals by the mean spectral distortion between the source and target signals [30]
where MCD is the distance between two cepstral vectors (defined in Section 3, Eq. (18)) and \(\tilde {\mathbf {Y}}_{1:T} \triangleq \left (\tilde {\mathbf {y}}_{1}, \tilde {\mathbf {y}}_{2}, \ldots,~\tilde {\mathbf {y}}_{T}\right)^{\top }\), \(\mathbf {Y}_{1:T} \triangleq \left (\mathbf {y}_{1}, \mathbf {y}_{2}, \ldots,~\mathbf {y}_{T}\right)^{\top }\), and \(\mathbf {X}_{1:T} \triangleq \left (\mathbf {x}_{1}, \mathbf {x}_{2}, \ldots,~\mathbf {x}_{T}\right)^{\top }\) are timealigned sequences of cepstral vectors, related to the converted, target, and source utterances, respectively.
The GV of the pth elements of a sequence, \(\tilde {\mathbf {Y}}_{1:T}\), representing a converted speech utterance, is as follows:
In this paper, we use a normalized global variance (NGV) to measure the variability of a sequence of converted vectors
where \(\sigma ^{2}_{\mathbf {Y}}(p)\) is the empirical GV of the pth elements of the target speaker, obtained from the target training vectors
Note that the target GV defined in Eq. (27) is evaluated by averaging over the entire training corpus. This evaluation of GV is different from the one proposed in [7] for spectral conversion and GV enhancement, where the GV of each utterance of the target is modeled as a random variable drawn from a Gaussian distribution.
The desired values for these measures are ND→0 and NGV→1, indicating that the converted outcome is close to the target signal in terms of spectral similarity and global variance.
The examined GMMbased methods (JGMM and CGMM) were trained using diagonal covariance matrices and 1–4 Gaussian mixtures, due to the low amount of training data.
We begin with a short examination of the influence of each of the three parameters of the proposed GB method (R _{ x },R _{ y }, and τ) on its performance. Figure 2 presents the ND vs. NGV values obtained for the proposed GB method using R _{ x }∈[0.3,2],R _{ y }∈[1,4], and τ=1, trained by 10 sentences, for a maletomale conversion. As the parameter R _{ x } gets higher, more grid points are considered in the weighted sum, so that ND decreases, but the NGV also decreases. Since the evidence probability is solely determined by the training set (see Eq. (16)), we also examined the performance of the GB method using a datadriven value for R _{ y }, specifically, the median of the MCD between all training vector pairs related to the target speaker. These values vary between 2 and 3 dB when using different sourcetarget pairs and data set sizes. As depicted in Fig. 2, the median leads to the best NDNGV values so all results presented from now on were obtained using this value for R _{ y }.
Figure 3 presents the ND vs. NGV values obtained for the proposed GB method using R _{ x }∈[0.3,2],τ=(0,1,2), trained by 10 sentences, for a maletomale conversion. Using τ=1 leads to higher NGV values than using τ=0, with a slight increase in the ND. However, increasing τ further leads to the same NGV values with a minor decrease in the ND.
Table 3 summarizes the ND and NGV values achieved by JGMM [2] and the proposed GB conversion method, for all four gender conversions: maletomale (M2M), maletofemale (M2F), femaletomale (F2M), and femaletofemale (F2F), using 5 and 10 training sentences. The number of mixtures for JGMM and parameters for the GB (R _{ x } and τ) were selected for each method and training set so that a minimal ND was attained, while keeping the NGV as high as possible. As mentioned above, R _{ y } was taken as the median. The proposed GB leads to higher NGV values in all the cases. For five training sentences, JGMM leads to lower ND values (except for F2M), however, using 10 training sentences, the proposed GB achieves lower or very similar ND values. Still, both methods lead to very low NGV values and consequently, muffled sounding synthesized signals.
To further improve the quality of the synthesized speech, we applied the postprocessing method for GV enhancement [21]. This method maximizes the GV of an input sequence, under a spectral distortion constraint. The GV of each enhanced sequence is increased up to the level where the MCD between the converted sequence and its enhanced version reaches a preset threshold value, denoted as θ _{MCD}. We recently showed [21] that this method leads to significant improvement in the perceived quality of signals converted by the classical GMM method [1]. In this work, we applied this GV enhancement method to both JGMM and to our proposed GB conversion outcomes. We also examined the performance of CGMM, which considers GV enhancement at training.
Table 4 summarizes the ND and NGV values achieved by the examined conversion methods, for all four gender conversions using 5 and 10 training sentences.
Again, the GB conversion, followed by GV enhancement with θ _{MCD}=2 dB (EnGB) leads to the highest NGV values. Using 5 training sentences, JGMM leads to the lowest ND values, while EnGB comes in second (except for F2F). Using 10 training sentences, EnGB produces the lowest ND and at the same time, the highest NGV, for M2M and M2F conversion. For F2M and F2F conversion, EnGB leads to the highest NGV with very similar ND values to JGMM, which are the lowest.
To conclude the objective examination, in terms of NGV, the proposed ENGB conversion scheme outperforms all the examined methods. In terms of ND, JGMM leads to lower ND values using 5 training sentences. Using 10 training sentences, EnGB leads to the lowest (or very similar to the lowest) ND values.
In the next section, we present subjective evaluation results comparing the proposed EnGB conversion scheme to the classical GMMbased conversion method (with enhancement) and to CGMM, in terms of perceived quality and similarity to the target speaker.
Subjective evaluations
Listening tests were carried out to subjectively assess the performance of the examined methods (all trained by 10 sentences). In every test, 10 different sentences were examined by 11 listeners (voice samples are available online [31]). The group of listeners included 20–30yearold, nonexperts men and women. The same four speakers (two males and two females) that were used for the objective evaluations were used for the subjective evaluations. The number of mixtures for the GMMbased methods and parameters for the GB conversion (R _{ x } and τ) were set so minimal spectral distortion would be attained while keeping the NGV as high as possible. We used informal listening tests to select the threshold value for GV enhancement from θ _{MCD}=0.5,1,2,4 dB. The best perceived quality was obtained with θ _{MCD}=2 dB, for both JGMM and GB. All four gender conversions were performed using the same parameters values as described above.
We conducted subjective quality evaluations in a format similar to multistimulus test with hidden reference and anchor (MUSHRA) [32]. The listeners were presented with four test signals: (a) a hidden reference—the target speaker, (b) enhanced JGMM, (c) CGMM, and (d) EnGB. The test signals were randomly ordered, and the listeners were not informed about the hidden reference signals being included in the test set. During evaluation, the listeners were asked to compare the test signals to the reference signal (the target speaker) and rate their quality between 0 and 100, where at least one of the test signals (the hidden reference) must be rated 100. As expected, all the listeners rated the hidden reference as 100. The mean scores of the examined methods for M2M, M2F, F2M, and F2F conversions and also their scores averaged over all four conversions are presented in Figs. 4 and 5, respectively. All subjective results are presented with their 95 % confidence intervals. We evaluated the individuality performance using, again, a similar format to MUSHRA, as conducted by Godony et al. [33]. The listeners were presented with the same test signals (including the hidden reference) and were asked to rate their similarity to the reference signal, in terms of the speaker’s identity, while ignoring their perceived quality. The mean individuality scores of the examined methods for M2M, M2F, F2M, and F2F conversions and also their scores, averaged over all four conversions, are presented in Figs. 6 and 7, respectively.
Except for F2F, the proposed ENGB was rated as most similar to the target speaker (Fig. 6). In terms of perceived quality, CGMM was rated as having the best quality, while ENJGMM and ENGB were rated as comparable (Fig. 4). All in all, considering all four gender conversion, the proposed ENGB was marked as most similar to the target speaker, while CGMM was marked as having the best quality.
Conclusions
Applying voice conversion in low resource environments, such as mobile applications, presents an engineering challenge. While digital processors and memory units become more advanced and less restricting, the amount of available training data remains limited, since most mobile users are not willing to invest much time and effort in recording their own voices. We propose here a GB voice conversion method suitable for such low resource environments. It is based on our recent paper, which presents a GB framework for voice conversion. The modified GB method presented in this paper is successfully trained using very few sentences (5–10) and does not require phonetic labeling of the test signals.
The GB conversion method is based on sequential Bayesian tracking, using a GB formulation. The target spectral evolution is modeled as a hidden Markov process, tracked by using the source spectrum, modeled as the observed process. The training stage is very simple and based on Euclidean distances between the training vectors, and it is successfully performed using very small training sets. Additionally, although GB is trained using a parallel set, time alignment is not needed. During training, the evidence and likelihood probabilities needed for the GB formulation are approximated as discrete densities. During conversion, the converted spectrum is obtained as a weighted sum of the training target vectors, used as grid points. The weights are sequentially evaluated so that a smooth temporal evolution of the converted spectra is produced.
We used a small set of just 10 sentences for training both the classical GMMbased conversion function and our GB method. According to our experiments, the GB conversion method achieves lower spectral distances between the converted and target spectra and GV values which are closer to the target speaker’s values than the classical GMMbased conversion. To further improve the quality of the synthesized speech, we increased the variability of the converted vectors by applying GV enhancement as a postprocessing block. We compared the proposed EnGB scheme to CGMM and to classical GMMbased conversions, with GV enhancement, using listening tests. This comparison showed that EnGB is the best in terms of similarity to the target speaker and comparable to the enhanced GMM conversion, in terms of quality.
The proposed GB conversion, as most other methods, simply replaces the spectral envelopes extracted from the source signal with the converted outcome. As a result, the synthesized output has the same speaking rate as the source speaker. Further improvement can be obtained by modifying the duration of each converted utterance to match, on average, its corresponding value for the target speaker.
Spectral distortion and GV are commonly used as objective measures since they provide a simple and fully automated way for evaluating conversion systems. These objective measures may express significant trends and phenomena, but as shown here, they do not always agree with subjective evaluation results.
Further research is needed to design alternative measures for objective evaluation of conversion systems, with better correspondence to subjective results. In the mean time, subjective listening tests are imperative to properly evaluate and compare conversion methods.
The proposed GB conversion method, as presented here, is based on soft correspondence between the source and target vectors, obtained by using a parallel training set. Further research is needed to evaluate this correspondence for a nonparallel setup.
Endnotes
^{1} In general, any arbitrary integrable function of the state vector y _{ t } can be evaluated [19].
^{2} If the state space is indeed discrete and finite, and the grid points consist of all its states, this evaluation becomes exact.
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Acknowledgements
The authors would like to thank Slava Shechtman, and the speech research group headed by Ron Hoory, at the IBM Research Labs, Haifa, Israel, for fruitful discussions.
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This paper is part of the doctoral research work of HB under the supervision of the other two authors.
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Benisty, H., Malah, D. & Crammer, K. Gridbased approximation for voice conversion in low resource environments. J AUDIO SPEECH MUSIC PROC. 2016, 3 (2016). https://doi.org/10.1186/s1363601600811
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Keywords
 Bayesian tracking
 Global variance (GV)
 Mel cepstral distortion (MCD)
 Gridbased approximation
 Spectral conversion