Seed-TTS: A Family of High-Quality Versatile Speech Generation Models
Paper · 2024 · Philip Anastassiou, Jiawei Chen, Jitong Chen et al.
Summary
ByteDance's Seed Team presents Seed-TTS, a family of large text-to-speech models. The goal is to clone any speaker's voice from a brief reference recording, producing output that listeners cannot tell apart from real human recordings. The main system has four components. A speech tokenizer feeds an autoregressive transformer language model, a diffusion transformer refines acoustic detail, and a separately trained vocoder produces the waveform. Training follows the pattern used for text LLMs: pre-training on data far larger than any earlier TTS corpus, then speaker and instruction fine-tuning, then reinforcement-learning post-training. The authors also add a self-distillation technique that separates timbre from content, which they use for voice conversion. A second variant, Seed-TTS DiT, is fully diffusion-based and does not need explicit phoneme durations, so it can edit both content and speaking rate. In zero-shot tests on English and Mandarin, human listeners showed no meaningful preference between Seed-TTS and real recordings. This makes the work a strong reference point for expressive voice cloning and raises related safety questions.
Key points
- Benchmarks: objective evaluation uses 1,000 Common Voice (EN) and 2,000 DiDiSpeech (ZH) samples, with reference clips of 3–20 s. WER is measured with Whisper-large-v3 and Paraformer-zh, and SIM with WavLM-large. The authors publish the test-set configuration.
- Zero-shot ICL results: in English, WER is 2.249 for Seed-TTS vs. 2.143 for human speech, and SIM is 0.762 vs. 0.730. In Mandarin, WER is 1.115 vs. 1.254, and SIM is 0.796 vs. 0.750.
- Subjective evaluation: CMOS against human speech is −0.07 (EN) and −0.08 (ZH). Both fall below the 0.1 threshold the authors treat as a significant difference.
- Seed-TTS DiT: this variant scores higher still, with EN WER 1.733 / SIM 0.790 and ZH WER 1.178 / SIM 0.809. It stays robust when large parts of the audio are masked for editing and adapts pauses naturally when the duration is changed.
- Limitations and safeguards: the model struggles with singing and with prompts containing background music or heavy noise. To limit misuse, the authors use multi-step speaker verification and multi-level watermarking.
📚 Shelves 7The field of science this work belongs to (OpenAlex fields).Computer Science · 9,086 worksArtificial Intelligence · 8,889 worksArts and Humanities · 9,296 worksLanguage and Linguistics · 9,206 worksSignal Processing · 213 worksMedicine · 80 worksPhysiology · 11 works
Cite
Cite as Philip Anastassiou, Jiawei Chen, Jitong Chen, Yuanzhe Chen, Zhuo Chen, Ziyi Chen, Jian Cong, Lelai Deng, Chuang Ding, Lu Gao, Mingqing Gong, Peisong Huang, Qingqing Huang, Zhiying Huang, Yuanyuan Huo, Dongya Jia, Chumin Li, Feiya Li, Hui Li, Jiaxin Li, Xiaoyang Li, Xingxing Li, Lin Liu, Shouda Liu, Sichao Liu, Xudong Liu, Yuchen Liu, Zhengxi Liu, Lu Lu, Junjie Pan, Xin Wang, Yuping Wang, Yuxuan Wang, Zhen Wei, Jian Wu, Chao Yao, Yifeng Yang, Yuanhao Yi, Junteng Zhang, Qidi Zhang, Shuo Zhang, Wenjie Zhang, Yang Zhang, Zilin Zhao, Dejian Zhong and Xiaobin Zhuang. 2024. Seed-TTS: A Family of High-Quality Versatile Speech Generation Models. arXiv:2406.02430.
@misc{anastassiou2024seed,
title = {{Seed-TTS: A Family of High-Quality Versatile Speech Generation Models}},
author = {Philip Anastassiou, Jiawei Chen, Jitong Chen, Yuanzhe Chen, Zhuo Chen, Ziyi Chen, Jian Cong, Lelai Deng, Chuang Ding, Lu Gao, Mingqing Gong, Peisong Huang, Qingqing Huang, Zhiying Huang, Yuanyuan Huo, Dongya Jia, Chumin Li, Feiya Li, Hui Li, Jiaxin Li, Xiaoyang Li, Xingxing Li, Lin Liu, Shouda Liu, Sichao Liu, Xudong Liu, Yuchen Liu, Zhengxi Liu, Lu Lu, Junjie Pan, Xin Wang, Yuping Wang, Yuxuan Wang, Zhen Wei, Jian Wu, Chao Yao, Yifeng Yang, Yuanhao Yi, Junteng Zhang, Qidi Zhang, Shuo Zhang, Wenjie Zhang, Yang Zhang, Zilin Zhao, Dejian Zhong and Xiaobin Zhuang},
year = {2024},
doi = {10.48550/arxiv.2406.02430},
eprint = {2406.02430},
archivePrefix = {arXiv},
url = {https://doi.org/10.48550/arxiv.2406.02430}
}References
- Yuanzhe Chen, Ming Tu, Tang Li, Xin Li, Qiuqiang Kong, Jiaxin Li, Zhichao Wang, Qiao Tian, Yuping Wang, and Yuxuan Wang. Streaming voice conversion via intermediate bottleneck features and non-streaming teacher guidance. In ICASSP 2023-2023 IEEE International Conference on Acoustics, Speech and Sign… find
- Zhichao Wang, Yuanzhe Chen, Xinsheng Wang, Zhuo Chen, Lei Xie, Yuping Wang, and Yuxuan Wang. StreamVoice: Streamable context-aware language modeling for real-time zero-shot voice conversion. arXiv preprint arXiv:2401.11053, 2024a. find
- Zhichao Wang, Yuanzhe Chen, Lei Xie, Qiao Tian, and Yuping Wang. LM-VC: Zero-shot voice conversion via speech generation based on language models. IEEE Signal Processing Letters, 2023a. find
- Zeqian Ju, Yuancheng Wang, Kai Shen, Xu Tan, Detai Xin, Dongchao Yang, Yanqing Liu, Yichong Leng, Kaitao Song, Siliang Tang, Zhizheng Wu, Tao Qin, Xiang-Yang Li, Wei Ye, Shikun Zhang, Jiang Bian, Lei He, Jinyu Li, and Sheng Zhao. NaturalSpeech 3: Zero-shot speech synthesis with factorized codec and… find
- Mateusz Łajszczak, Guillermo Cámbara, Yang Li, Fatih Beyhan, Arent van Korlaar, Fan Yang, Arnaud Joly, Álvaro Martín-Cortinas, Ammar Abbas, Adam Michalski, et al. BASE TTS: Lessons from building a billion-parameter text-to-speech model on 100k hours of data. arXiv preprint arXiv:2402.08093, 2024. — [arXiv:2402.08093](https://arxiv.org/abs/2402.08093) find
- Kaizhi Qian, Yang Zhang, Shiyu Chang, Xuesong Yang, and Mark Hasegawa-Johnson. AutoVC: Zero-shot voice style transfer with only autoencoder loss. In International Conference on Machine Learning, pages 5210–5219. PMLR, 2019. find
- Ziyue Jiang, Yi Ren, Zhenhui Ye, Jinglin Liu, Chen Zhang, Qian Yang, Shengpeng Ji, Rongjie Huang, Chunfeng Wang, Xiang Yin, et al. Mega-TTS: Zero-shot text-to-speech at scale with intrinsic inductive bias. arXiv preprint arXiv:2306.03509, 2023. — [arXiv:2306.03509](https://arxiv.org/abs/2306.03509) find
- Leslie Pack Kaelbling, Michael L Littman, and Andrew W Moore. Reinforcement learning: A survey. Journal of artificial intelligence research, 4:237–285, 1996. find
- Yuxi Li. Deep reinforcement learning: An overview. arXiv preprint arXiv:1701.07274, 2017. — [arXiv:1701.07274](https://arxiv.org/abs/1701.07274) find
- Chengyi Wang, Sanyuan Chen, Yu Wu, Ziqiang Zhang, Long Zhou, Shujie Liu, Zhuo Chen, Yanqing Liu, Huaming Wang, Jinyu Li, Lei He, Sheng Zhao, and Furu Wei. Neural codec language models are zero-shot text to speech synthesizers, 2023b.
- Ziqiang Zhang, Long Zhou, Chengyi Wang, Sanyuan Chen, Yu Wu, Shujie Liu, Zhuo Chen, Yanqing Liu, Huaming Wang, Jinyu Li, et al. Speak foreign languages with your own voice: Cross-lingual neural codec language modeling. arXiv preprint arXiv:2303.03926, 2023. — [arXiv:2303.03926](https://arxiv.org/abs/2303.03926) find
- Yuan Gao, Nobuyuki Morioka, Yu Zhang, and Nanxin Chen. E3 TTS: Easy end-to-end diffusion-based text to speech. In 2023 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU), pages 1–8. find
- Zehua Chen, Yihan Wu, Yichong Leng, Jiawei Chen, Haohe Liu, Xu Tan, Yang Cui, Ke Wang, Lei He, Sheng Zhao, et al. ResGrad: Residual denoising diffusion probabilistic models for text to speech. arXiv preprint arXiv:2212.14518, 2022a. — [arXiv:2212.14518](https://arxiv.org/abs/2212.14518) find
- Justin Lovelace, Soham Ray, Kwangyoun Kim, Kilian Q Weinberger, and Felix Wu. Simple-TTS: End-to-end text-to-speech synthesis with latent diffusion. 2023. find
- Xu Tan, Jiawei Chen, Haohe Liu, Jian Cong, Chen Zhang, Yanqing Liu, Xi Wang, Yichong Leng, Yuanhao Yi, Lei He, Frank Soong, Tao Qin, Sheng Zhao, and Tie-Yan Liu. NaturalSpeech: End-to-end text to speech synthesis with human-level quality, 2022. find
- Matthew Le, Apoorv Vyas, Bowen Shi, Brian Karrer, Leda Sari, Rashel Moritz, Mary Williamson, Vimal Manohar, Yossi Adi, Jay Mahadeokar, et al. Voicebox: Text-guided multilingual universal speech generation at scale. Advances in neural information processing systems, 36, 2024. find
- Chenpeng Du, Yiwei Guo, Feiyu Shen, Zhijun Liu, Zheng Liang, Xie Chen, Shuai Wang, Hui Zhang, and Kai Yu. UniCATS: A unified context-aware text-to-speech framework with contextual VQ-diffusion and vocoding. In Proceedings of the AAAI Conference on Artificial Intelligence, volume 38, pages 17924–1793… find
- Yi Ren, Yangjun Ruan, Xu Tan, Tao Qin, Sheng Zhao, Zhou Zhao, and Tie-Yan Liu. FastSpeech: Fast, robust and controllable text to speech. Advances in neural information processing systems, 32, 2019. find
- Yuanhao Yi, Lei He, Shifeng Pan, Xi Wang, and Yuchao Zhang. SoftSpeech: Unsupervised duration model in FastSpeech 2. In INTERSPEECH, pages 1606–1610, 2022a. find
- Yuanhao Yi, Lei He, Shifeng Pan, Xi Wang, and Yujia Xiao. ProsodySpeech: Towards advanced prosody model for neural text-to-speech. In ICASSP 2022-2022 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 7582–7586. find
- Hugo Touvron, Thibaut Lavril, Gautier Izacard, Xavier Martinet, Marie-Anne Lachaux, Timothée Lacroix, Baptiste Rozière, Naman Goyal, Eric Hambro, Faisal Azhar, et al. LLaMA: Open and efficient foundation language models. arXiv preprint arXiv:2302.13971, 2023.
- Ashish Vaswani, Noam Shazeer, Niki Parmar, Jakob Uszkoreit, Llion Jones, Aidan N Gomez, Łukasz Kaiser, and Illia Polosukhin. Attention is all you need. Advances in neural information processing systems, 30, 2017.
- James Betker. Better speech synthesis through scaling. arXiv preprint arXiv:2305.07243, 2023. — [arXiv:2305.07243](https://arxiv.org/abs/2305.07243) find
- Rithesh Kumar, Prem Seetharaman, Alejandro Luebs, Ishaan Kumar, and Kundan Kumar. High-fidelity audio compression with improved RVQGAN. Advances in Neural Information Processing Systems, 36, 2024. find
- Sang-gil Lee, Wei Ping, Boris Ginsburg, Bryan Catanzaro, and Sungroh Yoon. BigVGAN: A universal neural vocoder with large-scale training. arXiv preprint arXiv:2206.04658, 2022. — [arXiv:2206.04658](https://arxiv.org/abs/2206.04658) find
- Jian Cong, Shan Yang, Lei Xie, and Dan Su. Glow-WaveGAN: Learning speech representations from gan-based variational auto-encoder for high fidelity flow-based speech synthesis. arXiv preprint arXiv:2106.10831, 2021. — [arXiv:2106.10831](https://arxiv.org/abs/2106.10831) find
- Zhengxi Liu and Yanmin Qian. Basis-MelGAN: Efficient neural vocoder based on audio decomposition. arXiv preprint arXiv:2106.13419, 2021. — [arXiv:2106.13419](https://arxiv.org/abs/2106.13419) find
- Yuxuan Wang, Daisy Stanton, Yu Zhang, RJ-Skerry Ryan, Eric Battenberg, Joel Shor, Ying Xiao, Ye Jia, Fei Ren, and Rif A Saurous. Style tokens: Unsupervised style modeling, control and transfer in end-to-end speech synthesis. In International conference on machine learning, pages 5180–5189. PMLR, 201… find
- Borsos Zalán, Marinier Raphaël, Vincent Damien, Kharitonov Eugene, Pietquin Olivier, Sharifi Matt, Teboul Olivier, Tagliasacchi Marco, and Zeghidour Neil. AudioLM: A language modeling approach to audio generation, 2022.
- Rosana Ardila, Megan Branson, Kelly Davis, Michael Henretty, Michael Kohler, Josh Meyer, Reuben Morais, Lindsay Saunders, Francis M Tyers, and Gregor Weber. Common Voice: A massively-multilingual speech corpus. arXiv preprint arXiv:1912.06670, 2019.
- Tingwei Guo, Cheng Wen, Dongwei Jiang, Ne Luo, Ruixiong Zhang, Shuaijiang Zhao, Wubo Li, Cheng Gong, Wei Zou, Kun Han, et al. DiDiSpeech: A large scale mandarin speech corpus. In ICASSP 2021-2021 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 6968–6972. find
- Alec Radford, Jong Wook Kim, Tao Xu, Greg Brockman, Christine McLeavey, and Ilya Sutskever. Robust speech recognition via large-scale weak supervision. In International Conference on Machine Learning, pages 28492–28518. PMLR, 2023.
- Zhifu Gao, Zerui Li, Jiaming Wang, Haoneng Luo, Xian Shi, Mengzhe Chen, Yabin Li, Lingyun Zuo, Zhihao Du, Zhangyu Xiao, et al. FunASR: A fundamental end-to-end speech recognition toolkit. arXiv preprint arXiv:2305.11013, 2023b. — [arXiv:2305.11013](https://arxiv.org/abs/2305.11013) find
- Zhengyang Chen, Sanyuan Chen, Yu Wu, Yao Qian, Chengyi Wang, Shujie Liu, Yanmin Qian, and Michael Zeng. Large-scale self-supervised speech representation learning for automatic speaker verification. In ICASSP 2022-2022 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)… find
- Sanyuan Chen, Chengyi Wang, Zhengyang Chen, Yu Wu, Shujie Liu, Zhuo Chen, Jinyu Li, Naoyuki Kanda, Takuya Yoshioka, Xiong Xiao, et al. WavLM: Large-scale self-supervised pre-training for full stack speech processing. IEEE Journal of Selected Topics in Signal Processing, 16(6): 1505–1518, 2022c. find
- Yakun Song, Zhuo Chen, Xiaofei Wang, Ziyang Ma, and Xie Chen. ELLA-V: Stable neural codec language modeling with alignment-guided sequence reordering. arXiv preprint arXiv:2401.07333, 2024. — [arXiv:2401.07333](https://arxiv.org/abs/2401.07333) find
- Yi Ren, Chenxu Hu, Xu Tan, Tao Qin, Sheng Zhao, Zhou Zhao, and Tie-Yan Liu. FastSpeech 2: Fast and high-quality end-to-end text to speech. arXiv preprint arXiv:2006.04558, 2020. find
- Kai Shen, Zeqian Ju, Xu Tan, Yanqing Liu, Yichong Leng, Lei He, Tao Qin, Sheng Zhao, and Jiang Bian. NaturalSpeech 2: Latent diffusion models are natural and zero-shot speech and singing synthesizers, 2023. find
- Zhengxi Liu, Qiao Tian, Chenxu Hu, Xudong Liu, Menglin Wu, Yuping Wang, Hang Zhao, and Yuxuan Wang. Controllable and lossless non-autoregressive end-to-end text-to-speech. arXiv preprint arXiv:2207.06088, 2022. — [arXiv:2207.06088](https://arxiv.org/abs/2207.06088) find
- Vassil Panayotov, Guoguo Chen, Daniel Povey, and Sanjeev Khudanpur. LibriSpeech: an ASR corpus based on public domain audio books. In 2015 IEEE international conference on acoustics, speech and signal processing (ICASSP), pages 5206–5210. find
- Binbin Zhang, Di Wu, Zhendong Peng, Xingchen Song, Zhuoyuan Yao, Hang Lv, Lei Xie, Chao Yang, Fuping Pan, and Jianwei Niu. WeNet 2.0: More productive end-to-end speech recognition toolkit. arXiv preprint arXiv:2203.15455, 2022. — [arXiv:2203.15455](https://arxiv.org/abs/2203.15455) find
- Sehoon Kim, Amir Gholami, Albert Shaw, Nicholas Lee, Karttikeya Mangalam, Jitendra Malik, Michael W Mahoney, and Kurt Keutzer. Squeezeformer: An efficient transformer for automatic speech recognition. Advances in Neural Information Processing Systems, 35:9361–9373, 2022. find
- Jinyu Li, Rui Zhao, Zhuo Chen, Changliang Liu, Xiong Xiao, Guoli Ye, and Yifan Gong. Developing far-field speaker system via teacher-student learning. In 2018 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 5699–5703. find
- Arsha Nagrani, Joon Son Chung, and Andrew Zisserman. VoxCeleb: A large-scale speaker identification dataset. arXiv preprint arXiv:1706.08612, 2017. — [arXiv:1706.08612](https://arxiv.org/abs/1706.08612) find
- Laurens Van der Maaten and Geoffrey Hinton. Visualizing data using t-SNE. Journal of machine learning research, 9(11), 2008. find
- Yuan-Hao Yi, Yang Ai, Zhen-Hua Ling, and Li-Rong Dai. Singing voice synthesis using deep autoregressive neural networks for acoustic modeling. arXiv preprint arXiv:1906.08977, 2019. — [arXiv:1906.08977](https://arxiv.org/abs/1906.08977) find
- Xiaobin Zhuang, Tao Jiang, Szu-Yu Chou, Bin Wu, Peng Hu, and Simon Lui. LiteSing: Towards fast, lightweight and expressive singing voice synthesis. In ICASSP 2021-2021 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 7078–7082. find
- Yan Deng, Long Zhou, Yuanhao Yi, Shujie Liu, and Lei He. Prosody-aware SpeechT5 for expressive neural TTS. In ICASSP 2023-2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 1–5. find
- Tri Dao, Dan Fu, Stefano Ermon, Atri Rudra, and Christopher Ré. FlashAttention: Fast and memory-efficient exact attention with IO-awareness. Advances in Neural Information Processing Systems, 35:16344–16359, 2022. find
- Joshua Ainslie, James Lee-Thorp, Michiel de Jong, Yury Zemlyanskiy, Federico Lebrón, and Sumit Sanghai. GQA: Training generalized multi-query transformer models from multi-head checkpoints. arXiv preprint arXiv:2305.13245, 2023.
- Simian Luo, Yiqin Tan, Longbo Huang, Jian Li, and Hang Zhao. Latent consistency models: Synthesizing high-resolution images with few-step inference. arXiv preprint arXiv:2310.04378, 2023. find
- Ji Lin, Jiaming Tang, Haotian Tang, Shang Yang, Xingyu Dang, and Song Han. AWQ: Activation-aware weight quantization for llm compression and acceleration. arXiv preprint arXiv:2306.00978, 2023. find
- Yang Song, Prafulla Dhariwal, Mark Chen, and Ilya Sutskever. Consistency models. arXiv preprint arXiv:2303.01469, 2023. — [arXiv:2303.01469](https://arxiv.org/abs/2303.01469) find
- Patrick Esser, Sumith Kulal, Andreas Blattmann, Rahim Entezari, Jonas Müller, Harry Saini, Yam Levi, Dominik Lorenz, Axel Sauer, Frederic Boesel, et al. Scaling rectified flow transformers for high-resolution image synthesis. arXiv preprint arXiv:2403.03206, 2024. — [arXiv:2403.03206](https://arxiv.org/abs/2403.03206) find
- Woosuk Kwon, Zhuohan Li, Siyuan Zhuang, Ying Sheng, Lianmin Zheng, Cody Hao Yu, Joseph Gonzalez, Hao Zhang, and Ion Stoica. Efficient memory management for large language model serving with pagedattention. In Proceedings of the 29th Symposium on Operating Systems Principles, pages 611–626, 2023.
- Tri Dao. FlashAttention-2: Faster attention with better parallelism and work partitioning. arXiv preprint arXiv:2307.08691, 2023. find
- Markus Nagel, Marios Fournarakis, Rana Ali Amjad, Yelysei Bondarenko, Mart Van Baalen, and Tijmen Blankevoort. A white paper on neural network quantization. arXiv preprint arXiv:2106.08295, 2021. — [arXiv:2106.08295](https://arxiv.org/abs/2106.08295) find
- Yi Guo, Fanliu Kong, Xiaoyang Li, Hui Li, Wei Chen, Xiaogang Tian, Jinping Cai, Yang Zhang, and Shouda Liu. decoupleq: Towards 2-bit post-training uniform quantization via decoupling parameters into integer and floating points. arXiv preprint arXiv:2404.12759, 2024. — [arXiv:2404.12759](https://arxiv.org/abs/2404.12759) find
- Songxiang Liu, Yuewen Cao, Disong Wang, Xixin Wu, Xunying Liu, and Helen Meng. Any-to-many voice conversion with location-relative sequence-to-sequence modeling. IEEE/ACM Transactions on Audio, Speech, and Language Processing, 29:1717–1728, 2021. find
- Philip Anastassiou, Zhenyu Tang, Kainan Peng, Dongya Jia, Jiaxin Li, Ming Tu, Yuping Wang, Yuxuan Wang, and Mingbo Ma. VoiceShop: A unified speech-to-speech framework for identity-preserving zero-shot voice editing. arXiv preprint arXiv:2404.06674, 2024. — [arXiv:2404.06674](https://arxiv.org/abs/2404.06674) find
- Sang-Hoon Lee, Ha-Yeong Choi, Seung-Bin Kim, and Seong-Whan Lee. HierSpeech++: Bridging the gap between semantic and acoustic representation of speech by hierarchical variational inference for zero-shot speech synthesis. arXiv preprint arXiv:2311.12454, 2023. — [arXiv:2311.12454](https://arxiv.org/abs/2311.12454) find
- Ha-Yeong Choi, Sang-Hoon Lee, and Seong-Whan Lee. DDDM-VC: Decoupled denoising diffusion models with disentangled representation and prior mixup for verified robust voice conversion. In Proceedings of the AAAI Conference on Artificial Intelligence, volume 38, pages 17862–17870, 2024. find
- Dongya Jia, Qiao Tian, Kainan Peng, Jiaxin Li, Yuanzhe Chen, Mingbo Ma, Yuping Wang, and Yuxuan Wang. Zero-shot accent conversion using pseudo siamese disentanglement network. arXiv preprint arXiv:2212.05751, 2022. — [arXiv:2212.05751](https://arxiv.org/abs/2212.05751) find
- Vadim Popov, Ivan Vovk, Vladimir Gogoryan, Tasnima Sadekova, Mikhail Kudinov, and Jiansheng Wei. Diffusion-based voice conversion with fast maximum likelihood sampling scheme. arXiv preprint arXiv:2109.13821, 2021. — [arXiv:2109.13821](https://arxiv.org/abs/2109.13821) find
- John Schulman, Filip Wolski, Prafulla Dhariwal, Alec Radford, and Oleg Klimov. Proximal policy optimization algorithms. arXiv preprint arXiv:1707.06347, 2017. find
- Rafael Rafailov, Archit Sharma, Eric Mitchell, Christopher D Manning, Stefano Ermon, and Chelsea Finn. Direct preference optimization: Your language model is secretly a reward model. Advances in Neural Information Processing Systems, 36, 2024.
- Richard S Sutton, David McAllester, Satinder Singh, and Yishay Mansour. Policy gradient methods for reinforcement learning with function approximation. Advances in neural information processing systems, 12, 1999. find
- Bram Wallace, Meihua Dang, Rafael Rafailov, Linqi Zhou, Aaron Lou, Senthil Purushwalkam, Stefano Ermon, Caiming Xiong, Shafiq Joty, and Nikhil Naik. Diffusion model alignment using direct preference optimization. arXiv preprint arXiv:2311.12908, 2023. — [arXiv:2311.12908](https://arxiv.org/abs/2311.12908) find
- Geoffrey Cideron, Sertan Girgin, Mauro Verzetti, Damien Vincent, Matej Kastelic, Zalán Borsos, Brian McWilliams, Victor Ungureanu, Olivier Bachem, Olivier Pietquin, et al. MusicRL: Aligning music generation to human preferences. arXiv preprint arXiv:2402.04229, 2024. — [arXiv:2402.04229](https://arxiv.org/abs/2402.04229) find
- Dong Zhang, Zhaowei Li, Shimin Li, Xin Zhang, Pengyu Wang, Yaqian Zhou, and Xipeng Qiu. SpeechAlign: Aligning speech generation to human preferences. arXiv preprint arXiv:2404.05600, 2024. — [arXiv:2404.05600](https://arxiv.org/abs/2404.05600) find
- Arash Ahmadian, Chris Cremer, Matthias Gallé, Marzieh Fadaee, Julia Kreutzer, Ahmet Üstün, and Sara Hooker. Back to basics: Revisiting REINFORCE style optimization for learning from human feedback in LLMs. arXiv preprint arXiv:2402.14740, 2024. find
- Rohit Prabhavalkar, Tara N Sainath, Yonghui Wu, Patrick Nguyen, Zhifeng Chen, Chung-Cheng Chiu, and Anjuli Kannan. Minimum word error rate training for attention-based sequence-to-sequence models. In 2018 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 4839–4… find
- Zihao Wang, Chirag Nagpal, Jonathan Berant, Jacob Eisenstein, Alex D’Amour, Sanmi Koyejo, and Victor Veitch. Transforming and combining rewards for aligning large language models. arXiv preprint arXiv:2402.00742, 2024b. — [arXiv:2402.00742](https://arxiv.org/abs/2402.00742) find
- Dario Amodei, Chris Olah, Jacob Steinhardt, Paul Christiano, John Schulman, and Dan Mané. Concrete problems in AI safety. arXiv preprint arXiv:1606.06565, 2016.
- Lijun Yu, José Lezama, Nitesh B Gundavarapu, Luca Versari, Kihyuk Sohn, David Minnen, Yong Cheng, Agrim Gupta, Xiuye Gu, Alexander G Hauptmann, et al. Language model beats diffusion - tokenizer is key to visual generation. arXiv preprint arXiv:2310.05737, 2023. — [arXiv:2310.05737](https://arxiv.org/abs/2310.05737) find
- Xiaofei Wang, Manthan Thakker, Zhuo Chen, Naoyuki Kanda, Sefik Emre Eskimez, Sanyuan Chen, Min Tang, Shujie Liu, Jinyu Li, and Takuya Yoshioka. SpeechX: Neural codec language model as a versatile speech transformer. arXiv preprint arXiv:2308.06873, 2023c. — [arXiv:2308.06873](https://arxiv.org/abs/2308.06873) find
- OpenAI. Navigating the challenges and opportunities of synthetic voices. https://openai.com/index/navigating-the-challenges-and-opportunities-of-synthetic-voices, 2024. ↗
- Authors (alphabetical order) Philip Anastassiou Jiawei Chen Jitong Chen Yuanzhe Chen Zhuo Chen Ziyi Chen Jian Cong Lelai Deng Chuang Ding Lu Gao Mingqing Gong Peisong Huang Qingqing Huang Zhiying Huang Yuanyuan Huo Dongya Jia Chumin Li Feiya Li Hui Li Jiaxin Li Xiaoyang Li Xingxing Li Lin Liu Shouda… find
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