Yongmao Zhang

dblp:304/5100 · DBLP profile ↗
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10ranked-venue papers
3as first author
10since 2021 · last 2024
0009-0000-0526-5778ORCID · corroborated

Domains — the database's venue-derived domains; a paper can count in several

Graphics, computer vision, multimedia, augmented reality and games · 9 · 3 first-author · 9 since 2021Artificial intelligence and machine learning · 7 · 2 first-author · 7 since 2021
YearPublicationVenuePosition
2024 Text-aware and Context-aware Expressive Audiobook Speech Synthesis
Dake Guo, Xinfa Zhu, Liumeng Xue, Yongmao Zhang, Lei Xie 0001
INTERSPEECH4
2024 METTS: Multilingual Emotional Text-to-Speech by Cross-Speaker and Cross-Lingual Emotion Transfer
abstract
Previous multilingual text-to-speech (TTS) approaches have considered leveraging monolingual speaker data to enable cross-lingual speech synthesis. However, such data-efficient approaches have ignored synthesizing emotional aspects of speech due to the challenges of cross-speaker cross-lingual emotion transfer – the heavy entanglement ofspeaker timbre,emotionandlanguagefactors in the speech signal will make a system to produce cross-lingual synthetic speech with an undesired foreign accent and weak emotion expressiveness. This paper proposes a Multilingual Emotional TTS (METTS) model to mitigate these problems, realizing both cross-speaker and cross-lingual emotion transfer. Specifically, METTS takes DelightfulTTS as the backbone model and proposes the following designs. First, to alleviate the foreign accent problem, METTS introducesmulti-scale emotion modelingto disentangle speech prosody into coarse-grained and fine-grained scales, producing language-agnostic and language-specific emotion representations, respectively. Second, as a pre-processing step, formant shift basedinformation perturbationis applied to the reference signal for better disentanglement of speaker timbre in the speech. Third, a vector quantization basedemotion matcheris designed for reference selection, leading to decent naturalness and emotion diversity in cross-lingual synthetic speech. Experiments demonstrate the good design of METTS.
Xinfa Zhu, Tao Li 0051, Yongmao Zhang, Hongbin Zhou, Heng Lu 0004, Lei Xie 0001
IEEE ACM Trans. Audio Speech Lang. Process.4
2023 Promptspeaker: Speaker Generation Based on Text Descriptions
abstract
Recently, text-guided content generation has received extensive attention. In this work, we explore the possibility of text description-based speaker generation, i.e., using text prompts to control the speaker generation process. Specifically, we propose PromptSpeaker, a text-guided speaker generation system. PromptSpeaker consists of a prompt encoder, a zero-shot VITS, and a Glow model, where the prompt encoder predicts a prior distribution based on the text description and samples from this distribution to obtain a semantic representation. The Glow model subsequently converts the semantic representation into a speaker representation, and the zero-shot VITS finally synthesizes the speaker’s voice based on the speaker representation. We verify that PromptSpeaker can generate speakers new from the training set by objective metrics, and the synthetic speaker voice has reasonable subjective matching quality with the speaker prompt. Our audio samples are available on the demo website1.1Demo: https://promptspeaker.github.io/demo/
Yongmao Zhang, Guanghou Liu, Yunlin Chen, Lei Xie 0001
ASRU1
2023 DSPGAN: A Gan-Based Universal Vocoder for High-Fidelity TTS by Time-Frequency Domain Supervision from DSP
abstract
Recent development of neural vocoders based on the generative adversarial neural network (GAN) has shown obvious advantages of generating raw waveform conditioned on mel-spectrogram with fast inference speed and lightweight networks. Whereas, it is still challenging to train a universal neural vocoder that can synthesize high-fidelity speech from various scenarios with unseen speakers, languages, and speaking styles. In this paper, we propose DSP- GAN, a GAN-based universal vocoder for high-fidelity speech synthesis by applying the time-frequency domain supervision from digital signal processing (DSP). To eliminate the mismatch problem caused by the ground-truth spectrograms in the training phase and the predicted spectrograms in the inference phase, we leverage the mel-spectrogram extracted from the waveform generated by a DSP module, rather than the predicted mel-spectrogram from the Text-to-Speech (TTS) acoustic model, as the time-frequency domain supervision to the GAN-based vocoder. We also utilize sine excitation as the time-domain supervision to improve the harmonic modeling and eliminate various artifacts of the GAN-based vocoder. Experiments show that DSPGAN significantly outperforms the compared approaches and it can generate high-fidelity speech for various TTS models trained using diverse data.1
Yongmao Zhang, Jian Cong, Hanzhao Li, Lei Xie 0001, Jinfeng Bai
ICASSP2
2023 Multi-Speaker Expressive Speech Synthesis via Multiple Factors Decoupling
abstract
This paper aims to synthesize the target speaker’s speech with desired speaking style and emotion by transferring the style and emotion from reference speech recorded by other speakers. We address this challenging problem with a two-stage framework composed of a text-to-style-and-emotion (Text2SE) module and a style-and- emotion-to-wave (SE2Wave) module, bridging by neural bottleneck (BN) features. To further solve the multi-factor (speaker timbre, speaking style and emotion) decoupling problem, we adopt the multi-label binary vector (MBV) and mutual information (MI) minimization to respectively discretize the extracted embeddings and disentangle these highly entangled factors in both Text2SE and SE2Wave modules. Moreover, we introduce a semi-supervised training strategy to leverage data from multiple speakers, including emotion-labeled data, style-labeled data, and unlabeled data. To better transfer the fine-grained expression from references to the target speaker in non-parallel transfer, we introduce a reference-candidate pool and propose an attention-based reference selection approach. Extensive experiments demonstrate the good design of our model.
Xinfa Zhu, Yongmao Zhang, Tao Li 0051, Lei Xie 0001
ICASSP4
2023 PromptStyle: Controllable Style Transfer for Text-to-Speech with Natural Language Descriptions
Guanghou Liu, Yongmao Zhang, Yunlin Chen, Lei Xie 0001
INTERSPEECH2
2023 VISinger2: High-Fidelity End-to-End Singing Voice Synthesis Enhanced by Digital Signal Processing Synthesizer
Yongmao Zhang, Heyang Xue, Hanzhao Li, Lei Xie 0001, Tingwei Guo, Ruixiong Zhang, Caixia Gong
INTERSPEECH1
2022 VISinger: Variational Inference with Adversarial Learning for End-to-End Singing Voice Synthesis
abstract
In this paper, we propose VISinger, a complete end-to-end high-quality singing voice synthesis (SVS) system that directly generates singing audio from lyrics and musical score. Our approach is inspired by VITS [1], an end-to-end speech generation model which adopts VAE-based posterior encoder augmented with normalizing flow based prior encoder and adversarial decoder. VISinger follows the main architecture of VITS, but makes substantial improvements to the prior encoder according to the characteristics of singing. First, instead of using phoneme-level mean and variance of acoustic features, we introduce a length regulator and a frame prior network to get the frame-level mean and variance on acoustic features, modeling the rich acoustic variation in singing. Second, we further introduce an F0 predictor to guide the frame prior network, leading to stabler singing performance. Finally, to improve the singing rhythm, we modify the duration predictor to specifically predict the phoneme to note duration ratio, helped with singing note normalization. Experiments on a professional Mandarin singing corpus show that VISinger significantly outperforms FastSpeech+Neural-Vocoder two-stage approach and the oracle VITS; ablation study demonstrates the effectiveness of different contributions.
Yongmao Zhang, Jian Cong, Heyang Xue, Lei Xie 0001, Pengcheng Zhu 0004, Mengxiao Bi
ICASSP1
2022 Opencpop: A High-Quality Open Source Chinese Popular Song Corpus for Singing Voice Synthesis
abstract
This paper introduces Opencpop, a publicly available highquality Mandarin singing corpus designed for singing voice synthesis (SVS).The corpus consists of 100 popular Mandarin songs performed by a female professional singer.Audio files are recorded with studio quality at a sampling rate of 44,100 Hz and the corresponding lyrics and musical scores are provided.All singing recordings have been phonetically annotated with phoneme boundaries and syllable (note) boundaries.To demonstrate the reliability of the released data and to provide a baseline for future research, we built baseline deep neural network-based SVS models and evaluated them with both objective metrics and subjective mean opinion score (MOS) measure.Experimental results show that the best SVS model trained on our database achieves 3.70 MOS, indicating the reliability of the provided corpus.Opencpop is released to the open-source community WeNet 1 , and the corpus, as well as synthesized demos, can be found on the project homepage 2 .
Pengcheng Zhu 0004, Jie Wu 0017, Hanzhao Li, Heyang Xue, Yongmao Zhang, Lei Xie 0001, Mengxiao Bi
INTERSPEECH7
2022 Learn2Sing 2.0: Diffusion and Mutual Information-Based Target Speaker SVS by Learning from Singing Teacher
abstract
Building a high-quality singing corpus for a person who is not good at singing is non-trivial, thus making it challenging to create a singing voice synthesizer for this person.Learn2Sing is dedicated to synthesizing the singing voice of a speaker without his or her singing data by learning from data recorded by others, i.e., the singing teacher.Inspired by the fact that pitch is the key style factor to distinguish singing from speaking voice, the proposed Learn2Sing 2.0 first generates the preliminary acoustic feature with averaged pitch value in the phone level, which allows the training of this process for different styles, i.e., speaking or singing, share same conditions except for the speaker information.Then, conditioned on the specific style, a diffusion decoder, which is accelerated by a fast sampling algorithm during the inference stage, is adopted to gradually restore the final acoustic feature.During the training, to avoid the information confusion of the speaker embedding and the style embedding, mutual information is employed to restrain the learning of speaker embedding and style embedding.Experiments show that the proposed approach is capable of synthesizing highquality singing voice for the target speaker without singing data with 10 decoding steps.
Heyang Xue, Yongmao Zhang, Lei Xie 0001, Pengcheng Zhu 0004, Mengxiao Bi
INTERSPEECH3