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[电气工程与系统科学] 基于生成的统计参数语音合成 对抗性网络 [推广有奖]

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mingdashike22 在职认证  发表于 2022-3-4 22:35:30 来自手机 |只看作者 |坛友微信交流群|倒序 |AI写论文

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摘要翻译:
提出了一种基于生成对抗网络的统计参数语音合成方法。虽然deep neural networks(DNNs)技术可以用于人工合成语音波形,但与自然语音相比,人工合成的语音质量较低。导致质量下降的问题之一是在生成的语音参数中经常观察到的过平滑效应。本文介绍的一个GAN由两个神经网络组成:鉴别器,用于区分自然样本和生成样本;生成器,用于欺骗鉴别器。在该框架中,识别器被训练以区分自然和生成的语音参数,而声学模型被训练以使传统的最小生成损失和欺骗识别器的对抗性损失的加权和最小化。由于GANs的目标是最小化自然语音参数和生成语音参数之间的散度(即分布差异),因此该方法有效地缓解了生成语音参数的过平滑效应。我们对文本-语音和语音转换的有效性进行了评估,发现该方法与传统的最小生成误差训练算法相比,无论其超参数设置如何,都能生成更多的自然谱参数和f_0$。此外,我们还研究了各种GAN的发散度对合成语音质量的影响,发现最小化推土机距离的Wasserstein GAN在提高合成语音质量方面效果最好。
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英文标题:
《Statistical Parametric Speech Synthesis Incorporating Generative
  Adversarial Networks》
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作者:
Yuki Saito, Shinnosuke Takamichi, Hiroshi Saruwatari
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最新提交年份:
2017
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分类信息:

一级分类:Computer Science        计算机科学
二级分类:Sound        声音
分类描述:Covers all aspects of computing with sound, and sound as an information channel. Includes models of sound, analysis and synthesis, audio user interfaces, sonification of data, computer music, and sound signal processing. Includes ACM Subject Class H.5.5, and intersects with H.1.2, H.5.1, H.5.2, I.2.7, I.5.4, I.6.3, J.5, K.4.2.
涵盖了声音计算的各个方面,以及声音作为一种信息通道。包括声音模型、分析和合成、音频用户界面、数据的可听化、计算机音乐和声音信号处理。包括ACM学科类H.5.5,并与H.1.2、H.5.1、H.5.2、I.2.7、I.5.4、I.6.3、J.5、K.4.2交叉。
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一级分类:Computer Science        计算机科学
二级分类:Machine Learning        机器学习
分类描述:Papers on all aspects of machine learning research (supervised, unsupervised, reinforcement learning, bandit problems, and so on) including also robustness, explanation, fairness, and methodology. cs.LG is also an appropriate primary category for applications of machine learning methods.
关于机器学习研究的所有方面的论文(有监督的,无监督的,强化学习,强盗问题,等等),包括健壮性,解释性,公平性和方法论。对于机器学习方法的应用,CS.LG也是一个合适的主要类别。
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一级分类:Electrical Engineering and Systems Science        电气工程与系统科学
二级分类:Audio and Speech Processing        音频和语音处理
分类描述:Theory and methods for processing signals representing audio, speech, and language, and their applications. This includes analysis, synthesis, enhancement, transformation, classification and interpretation of such signals as well as the design, development, and evaluation of associated signal processing systems. Machine learning and pattern analysis applied to any of the above areas is also welcome.  Specific topics of interest include: auditory modeling and hearing aids; acoustic beamforming and source localization; classification of acoustic scenes; speaker separation; active noise control and echo cancellation; enhancement; de-reverberation; bioacoustics; music signals analysis, synthesis and modification; music information retrieval;  audio for multimedia and joint audio-video processing; spoken and written language modeling, segmentation, tagging, parsing, understanding, and translation; text mining; speech production, perception, and psychoacoustics; speech analysis, synthesis, and perceptual modeling and coding; robust speech recognition; speaker recognition and characterization; deep learning, online learning, and graphical models applied to speech, audio, and language signals; and implementation aspects ranging from system architecture to fast algorithms.
处理代表音频、语音和语言的信号的理论和方法及其应用。这包括分析、合成、增强、转换、分类和解释这些信号,以及相关信号处理系统的设计、开发和评估。机器学习和模式分析应用于上述任何领域也是受欢迎的。感兴趣的具体主题包括:听觉建模和助听器;声波束形成与声源定位;声场景分类;说话人分离;有源噪声控制和回声消除;增强;去混响;生物声学;音乐信号的分析、合成与修饰;音乐信息检索;多媒体音频和联合音视频处理;口语和书面语建模、切分、标注、句法分析、理解和翻译;文本挖掘;言语产生、感知和心理声学;语音分析、合成、感知建模和编码;鲁棒语音识别;说话人识别与特征描述;应用于语音、音频和语言信号的深度学习、在线学习和图形模型;以及从系统架构到快速算法的实现方面。
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英文摘要:
  A method for statistical parametric speech synthesis incorporating generative adversarial networks (GANs) is proposed. Although powerful deep neural networks (DNNs) techniques can be applied to artificially synthesize speech waveform, the synthetic speech quality is low compared with that of natural speech. One of the issues causing the quality degradation is an over-smoothing effect often observed in the generated speech parameters. A GAN introduced in this paper consists of two neural networks: a discriminator to distinguish natural and generated samples, and a generator to deceive the discriminator. In the proposed framework incorporating the GANs, the discriminator is trained to distinguish natural and generated speech parameters, while the acoustic models are trained to minimize the weighted sum of the conventional minimum generation loss and an adversarial loss for deceiving the discriminator. Since the objective of the GANs is to minimize the divergence (i.e., distribution difference) between the natural and generated speech parameters, the proposed method effectively alleviates the over-smoothing effect on the generated speech parameters. We evaluated the effectiveness for text-to-speech and voice conversion, and found that the proposed method can generate more natural spectral parameters and $F_0$ than conventional minimum generation error training algorithm regardless its hyper-parameter settings. Furthermore, we investigated the effect of the divergence of various GANs, and found that a Wasserstein GAN minimizing the Earth-Mover's distance works the best in terms of improving synthetic speech quality.
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PDF链接:
https://arxiv.org/pdf/1709.08041
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关键词:Applications Conventional distribution Segmentation localization 传统 参数 proposed 方法 neural

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