A Study on Low-Latency Recognition-Synthesis-Based Any-to-One Voice Conversion

Yi-Yang Ding, Li-Juan Liu, Yu Hu, Zhenhua Ling
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Abstract

Some application scenarios of voice conversion, such as identity disguise in voice communication, require low-latency generation of converted speech. In traditional conversion methods, both history and future information in input speech are utilized to predict the converted acoustic features at each frame, which leads to long latency of voice conversion. Therefore, this paper proposes a low-latency recognition-synthesis-based any-to-one voice conversion method. Bottleneck (BN) features are extracted by an automatic speech recognition (ASR) acoustic model for frame-by-frame phoneme classification. A minimum mutual information (MMI) loss is introduced to reduce the speaker information in BNs caused by the low-latency configuration. The BN features are sent into a speaker-dependent low-latency LSTM-based acoustic feature predictor and the speech waveforms are reconstructed by an LPCNet vocoder from predicted acoustic features. The total latency of our proposed voice conversion method is 190ms, which is less than the delay requirement for comfortable communication in ITU-T G.114. The naturalness of converted speech is comparable with the upper-bound model trained without low-latency constraints.
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基于低延迟识别合成的任意对一语音转换研究
语音转换的一些应用场景,如语音通信中的身份伪装,需要低延迟生成转换后的语音。在传统的转换方法中,输入语音的历史信息和未来信息都被用来预测每一帧转换的声学特征,这导致了语音转换的长延迟。因此,本文提出了一种基于低延迟识别合成的任意对一语音转换方法。采用自动语音识别(ASR)声学模型提取瓶颈(BN)特征,进行逐帧音素分类。引入最小互信息损耗(minimum mutual information loss, MMI)来减少低时延配置导致的话音人信息丢失。将BN特征发送到基于扬声器的低延迟lstm声学特征预测器中,由LPCNet声码器根据预测的声学特征重建语音波形。我们提出的语音转换方法的总延迟为190ms,小于ITU-T G.114中舒适通信的延迟要求。转换后的语音的自然度与没有低延迟约束的上界模型相当。
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