Transforming Perceived Vocal Effort and Breathiness Using Adaptive Pre-Emphasis Linear Prediction

This paper presents a technique to transform high-effort voices into breathy voices using adaptive pre-emphasis linear prediction (APLP). The primary benefit of this technique is that it estimates a spectral emphasis filter that can be used to manipulate the perceived vocal effort. The other benefit of APLP is that it estimates a formant filter that is more consistent across varying voice qualities. This paper describes how constant pre-emphasis linear prediction (LP) estimates a voice source with a constant spectral envelope even though the spectral envelope of the true voice source varies over time. A listening experiment demonstrates how differences in vocal effort and breathiness are audible in the formant filter estimated by constant pre-emphasis LP. APLP is presented as a technique to estimate a spectral emphasis filter that captures the combined influence of the glottal source and the vocal tract upon the spectral envelope of the voice. A final listening experiment demonstrates how APLP can be used to effectively transform high-effort voices into breathy voices. The techniques presented here are relevant to researchers in voice conversion, voice quality, singing, and emotion.     

一种使用声调映射码本的汉语声音转换方法

在使用高斯混合模型实现说话人语音频谱包络变换的同时,提出了一种汉语声调码本映射技术来进一步提高转换语音目标说话人特征倾向性的方法。从源语音和目标语音分别提取汉语单音节的基频曲线作为基频变换单元,作预处理和聚类后分别形成源、目标声调码本,根据时间对准原则建立了一个由源特征空间到目标特征空间的声调模式映射码本。声音转换实验评估了声调码本映射算法的性能。实验结果表明,该算法较好地反映出源说话人与目标说话人基频曲线之间的映射关系,改善了声音转换性能。     

基于遗传径向基神经网络的声音转换

声音转换技术可以将一个人的语音模式转换为与其特性不同的另一个人语音模式,使转换语音保持源说话人原有语音信息内容不变,而具有目标说话人的声音特点。本文研究了由遗传算法训练的RBF神经网络捕获说话人的语音频谱包络映射关系,以实现不同说话人之间声音特性的转换。实验对六个普通话单元音音素的转换语音质量分别作了客观和主观评估,结果表明用神经网络方法可以获得所期望的转换语音性能。实验结果还说明,与K-均值法相比,用遗传算法训练神经网络可以增强网络的全局寻优能力,使转换语音与目标语音的平均频谱失真距离减小约10%。     

基于卷积非负矩阵分解的语音转换方法

为了在语音转换过程中充分考虑语音的帧间相关性,提出了一种基于卷积非负矩阵分解的语音转换方法.卷积非负矩阵分解得到的时频基可较好地保存语音信号中的个人特征信息及帧间相关性.利用这一特性,在训练阶段,通过卷积非负矩阵分解从训练数据中提取源说话人和目标说话人相匹配的时频基.在转换阶段,通过时频基替换实现对源说话人语音的转换.相对于传统方法,本方法能够更好地保存和转换语音帧间相关性.实验仿真及主、客观评价结果表明,与基于高斯混合模型、状态空间模型的语音转换方法相比,该方法具有更好的转换语音质量和转换相似度.     

Nonparallel training for voice conversion based on a parameter adaptation approach

The objective of voice conversion algorithms is to modify the speech by a particular source speaker so that it sounds as if spoken by a different target speaker. Current conversion algorithms employ a training procedure, during which the same utterances spoken by both the source and target speakers are needed for deriving the desired conversion parameters. Such a (parallel) corpus, is often difficult or impossible to collect. Here, we propose an algorithm that relaxes this constraint, i.e., the training corpus does not necessarily contain the same utterances from both speakers. The proposed algorithm is based on speaker adaptation techniques, adapting the conversion parameters derived for a particular pair of speakers to a different pair, for which only a nonparallel corpus is available. We show that adaptation reduces the error obtained when simply applying the conversion parameters of one pair of speakers to another by a factor that can reach 30%. A speaker identification measure is also employed that more insightfully portrays the importance of adaptation, while listening tests confirm the success of our method. Both the objective and subjective tests employed, demonstrate that the proposed algorithm achieves comparable results with the ideal case when a parallel corpus is available.     

Voice conversion using duration-embedded bi-HMMs for expressive speech synthesis

This paper presents an expressive voice conversion model (DeBi-HMM) as the post processing of a text-to-speech (TTS) system for expressive speech synthesis. DeBi-HMM is named for its duration-embedded characteristic of the two HMMs for modeling the source and target speech signals, respectively. Joint estimation of source and target HMMs is exploited for spectrum conversion from neutral to expressive speech. Gamma distribution is embedded as the duration model for each state in source and target HMMs. The expressive style-dependent decision trees achieve prosodic conversion. The STRAIGHT algorithm is adopted for the analysis and synthesis process. A set of small-sized speech databases for each expressive style is designed and collected to train the DeBi-HMM voice conversion models. Several experiments with statistical hypothesis testing are conducted to evaluate the quality of synthetic speech as perceived by human subjects. Compared with previous voice conversion methods, the proposed method exhibits encouraging potential in expressive speech synthesis.     

High quality voice conversion using prosodic and high-resolution spectral features

Voice conversion methods have advanced rapidly over the last decade. Studies have shown that speaker characteristics are captured by spectral feature as well as various prosodic features. Most existing conversion methods focus on the spectral feature as it directly represents the timbre characteristics, while some conversion methods have focused only on the prosodic feature represented by the fundamental frequency. In this paper, a comprehensive framework using deep neural networks to convert both timbre and prosodic features is proposed. The timbre feature is represented by a high-resolution spectral feature. The prosodic features include F0, intensity and duration. It is well known that DNN is useful as a tool to model high-dimensional features. In this work, we show that DNN initialized by our proposed autoencoder pretraining yields good quality DNN conversion models. This pretraining is tailor-made for voice conversion and leverages on autoencoder to capture the generic spectral shape of source speech. Additionally, our framework uses segmental DNN models to capture the evolution of the prosodic features over time. To reconstruct the converted speech, the spectral feature produced by the DNN model is combined with the three prosodic features produced by the DNN segmental models. Our experimental results show that the application of both prosodic and high-resolution spectral features leads to quality converted speech as measured by objective evaluation and subjective listening tests.     

基于话者无关模型的说话人转换方法

提出一种基于话者无关模型的说话人转换方法.考虑到音素信息共同存在于所有说话人的语音中,假设存在一个可以用高斯混合模型来描述的话者无关空间,且可用分段线性变换来描述该空间到各说话人相关空间之间的映射关系.在一个多说话人的数据库上,用话者自适应训练算法来训练模型,并在转换阶段使用源目标说话人空间到话者无关空间的变换关系来构造源与目标之间的特征变换关系,快速、灵活的构造说话人转换系统.通过主观测听实验来验证该算法相对于传统的基于话者相关模型方法的优点.     

语音转换关键技术研究

语音转换技术在语音处理领域是一个比较新的研究方向,也是近年来语音领域的研究热点。语音转换技术是指改变源说话人的语音特征使之具有目标说话人特征的一项技术。本文说明了语音转换的定义,介绍了语音的个性特征,列举了频谱包络的几种主要的转换算法以及韵律转换的主要算法。最后说明了语音转换今后的研究方向。     

A Spectral Conversion Approach to Single-Channel Speech Enhancement

In this paper, a novel method for single-channel speech enhancement is proposed, which is based on a spectral conversion feature denoising approach. Spectral conversion has been applied previously in the context of voice conversion, and has been shown to successfully transform spectral features with particular statistical properties into spectral features that best fit (with the constraint of a piecewise linear transformation) different target statistics. This spectral transformation is applied as an initialization step to two well-known single channel enhancement methods, namely the iterative Wiener filter (IWF) and a particular iterative implementation of the Kalman filter. In both cases, spectral conversion is shown here to provide a significant improvement as opposed to initializations using the spectral features directly from the noisy speech. In essence, the proposed approach allows for applying these two algorithms in a user-centric manner, when "clean" speech training data are available from a particular speaker. The extra step of spectral conversion is shown to offer significant advantages regarding output signal-to-noise ratio (SNR) improvement over the conventional initializations, which can reach 2 dB for the IWF and 6 dB for the Kalman filtering algorithm, for low input SNRs and for white and colored noise, respectively     

采用深度信念网络的语音转换方法

对说话人语音个性特征信息的表征和提取进行了深入研究,提出了一种基于深度信念网络（Deep Belief Nets,DBN）的语音转换方法。分别用提取出的源说话人和目标说话人语音频谱参数来训练DBN,分别得到其在高阶空间的语音个性特征表征;通过人工神经网络（Artificial Neural Networks,ANN）来连接这两个高阶空间并进行特征转换;使用基于目标说话人数据训练出的DBN来对转换后的特征信息进行逆处理得到转换后语音频谱参数,合成转换语音。实验结果表明,与传统的基于GMM方法相比,该方法效果更好,转换语音音质和相似度同目标语音更接近。     

一种基于正弦激励的线性预测模型的语音转换方法

在正弦激励模型的线性预测(LP)残差转换的基础上,提出了一种改进语音特征转换性能的语音转换方法.基于线性预测分析和综合的构架,该方法一方面通过谱包络估计声码器提取源说话人的线性预测编码(LPC)倒谱包络,并使用双线性变换函数实现倒谱包络的转换;另一方面由谐波正弦模型对线性预测残差信号建模和分解,采用基音频率变换将源说话人的残差信号转换为近似目标说话人的残差信号.最后由修正后的残差信号激励时变滤波器得到转换语音,滤波器参数通过转换得到的LPC倒谱包络实时更新.实验结果表明,该方法在主观和客观测试中都具有良好的结果,能有效地转换说话人声音特征,获得高相似度的转换语音.     

基于多谱特征生成对抗网络的语音转换算法

语音转换在教育、娱乐、医疗等各个领域都有广泛的应用,为了得到高质量的转换语音,提出了基于多谱特征生成对抗网络的语音转换算法。利用生成对抗网络对由谱特征参数生成的声纹图进行转换,利用特征级多模态融合技术使网络学习来自不同特征域的多种信息,以提高网络对语音信号的感知能力,从而得到具有良好清晰度和可懂度的高质量转换语音。实验结果表明,在主、客观评价指标上,本文算法较传统算法均有明显提升。     

基于GMM和ANN混合模型的语音转换方法

为了克服利用高斯混合模型(GMM)进行语音转换的过程中出现的过平滑现象,考虑到GMM模型参数的均值能够表征转换特征的频谱包络形状,本文提出一种基于GMM与ANN混合模型的语音转换,利用ANN对GMM模型参数的均值进行转换;为了获取连续的转换频谱,采用静态和动态频谱特征相结合来逼近转换频谱序列;鉴于基频对语音转换的重要性,在频谱转换的基础上,对基频也进行了分析和转换。最后,通过主观和客观实验对提出的混合模型的语音转换方法的性能进行测试,实验结果表明,与传统的基于GMM模型的语音转换方法相比,本文提出的方法能够获得更好的转换语音。     

基于自适应加权谱内插的宽带语音编码算法

提出了一种基于自适应加权谱内插(STRAIGHT)的宽带语音编码算法。输入的语音信号首先经过STRAIGHT分析得到精确的基频参数和谱参数,然后通过时域抽取和频域建模实现有效的编码压缩。在时域抽取时采用的区别于传统编码算法固定帧长的自适应可变帧长方法,使得编码存储量可以根据实际语音变化情况得到更加合理的分配。主观测听结果表明,该算法针对16kHz采样的语音信号,在6kbps码率上可以取得与AMR-WB(G．722．2)在8．85kbps时的相当的音质效果。此外,该算法还具有对恢复语音的时长、基频以及谱参数较强的调整能力。     

采用STRAIGHT模型和深度信念网络的语音转换方法

提出一种将STRAIGHT模型和深度信念网络DBN相结合实现语音转换的方式。首先,通过STRAIGHT模型提取出源说话人和目标说话人的语音频谱参数,用提取的频谱参数分别训练两个DBN得到语音高阶空间的个性特征信息;然后,用人工神经网络ANN将两个具有高阶特征的空间连接并进行特征转换;最后,用基于目标说话人数据训练出的DBN来对转换后的特征信息进行逆处理得到语音频谱参数,并用STRAIGHT模型合成具有目标说话人个性化特征的语音。实验结果表明,采用此种方式获得的语音转换效果要比传统的采用GMM实现语音转换更好,转换后的语音音质和相似度与目标语音更接近。     

Voice Conversion Based on Maximum-Likelihood Estimation of Spectral Parameter Trajectory

In this paper, we describe a novel spectral conversion method for voice conversion (VC). A Gaussian mixture model (GMM) of the joint probability density of source and target features is employed for performing spectral conversion between speakers. The conventional method converts spectral parameters frame by frame based on the minimum mean square error. Although it is reasonably effective, the deterioration of speech quality is caused by some problems: 1) appropriate spectral movements are not always caused by the frame-based conversion process, and 2) the converted spectra are excessively smoothed by statistical modeling. In order to address those problems, we propose a conversion method based on the maximum-likelihood estimation of a spectral parameter trajectory. Not only static but also dynamic feature statistics are used for realizing the appropriate converted spectrum sequence. Moreover, the oversmoothing effect is alleviated by considering a global variance feature of the converted spectra. Experimental results indicate that the performance of VC can be dramatically improved by the proposed method in view of both speech quality and conversion accuracy for speaker individuality.     

Parametric Formant Modelling and Transformation in Voice Conversion

This paper presents a method for the estimation and mapping of parametric models of speech resonance at formants for voice conversion. The spectral features at formants that contribute to voice characteristics are the trajectories of the frequencies, the bandwidths and intensities of the resonance at formants. The formant features are extracted from the poles of a linear prediction (LP) model of speech. The statistical distributions of formants are modelled by a two-dimensional hidden Markov model (HMM) spanning the time and frequency dimensions. Experimental results are presented which show a close match between HMM-based formant models and the histograms of formants. For voice conversion two alternative methods are explored for mapping the formants of a source speaker to those of a target speaker. The first method is based on an adaptive formant-tracking warping of the frequency response of the LP model and the second method is based on the rotation of the poles of the LP model of speech. Both methods transform all spectral parameters of the resonance at formants of the source speaker towards those of the target speaker. In addition, the issues affecting the selection of the warping ratios for the mapping functions are investigated. Experimental results of formant estimation and perceptual evaluation of voice morphing based on parametric formant models are presented.     

Voice conversion using General Regression Neural Network

The objective of voice conversion system is to formulate the mapping function which can transform the source speaker characteristics to that of the target speaker. In this paper, we propose the General Regression Neural Network (GRNN) based model for voice conversion. It is a single pass learning network that makes the training procedure fast and comparatively less time consuming. The proposed system uses the shape of the vocal tract, the shape of the glottal pulse (excitation signal) and long term prosodic features to carry out the voice conversion task. In this paper, the shape of the vocal tract and the shape of source excitation of a particular speaker are represented using Line Spectral Frequencies (LSFs) and Linear Prediction (LP) residual respectively. GRNN is used to obtain the mapping function between the source and target speakers. The direct transformation of the time domain residual using Artificial Neural Network (ANN) causes phase change and generates artifacts in consecutive frames. In order to alleviate it, wavelet packet decomposed coefficients are used to characterize the excitation of the speech signal. The long term prosodic parameters namely, pitch contour (intonation) and the energy profile of the test signal are also modified in relation to that of the target (desired) speaker using the baseline method. The relative performances of the proposed model are compared to voice conversion system based on the state of the art RBF and GMM models using objective and subjective evaluation measures. The evaluation measures show that the proposed GRNN based voice conversion system performs slightly better than the state of the art models.