REVIEW 3 major objections 6 minor 1 cited by
DualDub: Video-to-Soundtrack Generation via Joint Speech and Background Audio Synthesis
T0 review · 3 major / 6 minor · reviewed 2026-08-06 · deepseek-v4-flash
Pith's one-line read This paper proposes video-to-soundtrack generation, in which a single multimodal language model produces both intelligible speech and background audio from a video, and reports that this joint approach yields markedly more harmonious and…
desk verdict A credible new task and a well-built unified model, but the harmony metric that carries the headline claim needs external validation before the superiority result lands. read the letter →
The pith
A machine-rendered reading of the paper's core claim, the machinery that carries it, and where it could break.
The reading
What carries the argument
The load-bearing components are three. The cross-modal aligner combines two causal cross-attention heads (audio attends to past speech, speech attends to past audio) with two non-causal heads (audio and speech each attend to the full video), so rhythm and timing cohere without leaking future information during generation. The curriculum schedule trains the model first on video-to-audio, then adds text-to-speech, then fine-tunes on a small amount of fully paired video-text-speech-audio data, which is what makes joint training feasible given data scarcity. The generation head is a multimodal language model with two output heads predicting audio and speech tokens simultaneously, decoded through a flow-matching network rather than the codec's native decoder to recover waveform quality. On the evaluation side, the paper introduces CASP, a contrastive audio-speech pretraining model whose cosine similarity between the audio and speech embeddings defines the DualScore harmony metric.
What would settle it
An audit of the test set would settle it: if listeners can hear intelligible speech leaking into the separated 'background' tracks, or if a blind forced-choice test finds the stitched baselines as harmonious as DualDub as often as not, the reported harmony advantage is not established.
Extended reading notes
Core claim
The central claim is that generating background audio and speech in one autoregressive pass, conditioned on the same video, transcript, and speaker reference, yields soundtracks that are more synchronized and more acoustically harmonious than anything obtainable by running a separate video-to-audio model and a separate video-to-speech model and mixing their outputs. DualDub encodes video with a frozen visual encoder, tokenizes both tracks with a unified codec, lets causal cross-attention pass history between the audio and speech streams while non-causal attention aligns both with the video, and decodes tokens to waveforms through a flow-matching network into a pre-trained audio VAE latent space. The paper reports DualScore 0.59 versus 0.19-0.30 for concatenated baselines and subjective audio-speech correspondence and harmony scores of 3.44 versus 1.44-1.69, while remaining competitive or superior on individual audio and speech quality metrics. DualBench supplies 1,319 filtered test clips and a contrastive audio-speech pretraining metric, DualScore, for measuring harmony between the two generated tracks.
Load-bearing premise
The evaluation's load-bearing premise is that the 1,319 test clips produced by automatic separation and a -40 dB energy cutoff contain clean, correctly paired speech and background audio, and that the learned DualScore matches what humans mean by harmony.
Editorial extensions
If this is right
- For automated dubbing, the result implies a single model can replace separate audio-effects and speech models, removing the timing and level conflicts that appear when two independently generated tracks are mixed.
- Because the curriculum learns from large existing video-audio and text-speech corpora plus a small paired set, other multimodal generation tasks facing the same data scarcity could adopt the same staged recipe.
- DualBench gives later work a common yardstick, so quality, video synchronization, and audio-speech harmony can be measured on the same 1,319 clips.
- Joint conditioning on video means speech prosody and background audio are shaped by the same visual events, enabling scene-driven narration to be produced end-to-end.
Reading between the lines
- An extension the paper leaves implicit is that if DualScore tracks perceived harmony, it could be reused as a training reward for joint audio-speech models rather than only a test-time metric, since it is computed from frozen embeddings.
- The paper's separation-and-filter pipeline suggests the quality ceiling of V2ST evaluation depends on source separation; stronger separation than the current model might enlarge the usable training set and change the measured harmony gap.
- The stated bottleneck of discrete 40-token-per-second codec representation points to a concrete next experiment: replacing discrete tokens with continuous representations should improve speech naturalness and audio fidelity under the same curriculum.
- A testable extension is to vary the speaker reference and transcript while fixing the video, checking whether the generated background audio adapts to the changed speech content rather than staying tied only to the visual scene.
Editorial analysis
A structured set of objections, weighed in public.
Referee Report
Summary. The paper proposes a new task, video-to-soundtrack (V2ST) generation, and presents DualDub, a multimodal language model that jointly generates background audio and speech from video, transcript, and a reference speech clip. The model consists of a multimodal encoder, a cross-modal aligner with causal and non-causal attention, dual decoding heads, and a curriculum learning strategy. The authors also introduce DualBench, a benchmark with a curated test set built from V2C-Animation via Mel-RoFormer separation and energy filtering, plus metrics including the newly proposed DualScore for audio-speech harmony. Experiments compare DualDub against concatenated V2A and V2S baselines, reporting higher objective harmony scores and higher subjective audio-speech correspondence and harmony (ASCH), along with V2A/V2S quality comparisons and ablations.
Significance. If the central empirical claim is accepted, the paper makes a useful contribution by defining a practical and underexplored task, proposing a coherent architecture for joint speech and background audio generation, and releasing a benchmark with code. The consistent subjective trend (Table 5, ASCH 3.44 vs 1.44-1.69) is a genuine strength, as is the open benchmark and the detailed curriculum learning design. However, the main quantitative evidence for the 'harmony' advantage is DualScore, a metric introduced and trained in this paper on private data processed by the same separation pipeline used to build the benchmark test set; no human correlation validation is provided. This makes the objective pillar of the central claim currently unverified, although the issue is addressable with additional evaluation.
major comments (3)
- [§4.3, §5.1, Table 5] The load-bearing objective evidence for the harmony advantage is DualScore, but its validity as a measure of perceived audio-speech harmony is not established. The CASP model is trained on 1,500 hours of private audio-speech pairs created with the same Mel-RoFormer separation and -40 dB filtering pipeline used to build the V2ST test set (§4.2, §5.1), and its only reported validation is retrieval accuracy on that same V2ST test set. No per-clip correlation between DualScore and human ASCH ratings is reported, so the large DualScore gap (0.59 vs 0.19-0.30) could reflect separator-specific artifacts, speech bleed, or other correlated features rather than harmony that humans perceive. I ask the authors to validate DualScore against human harmony judgments on a held-out set (including per-item correlation), to release the CASP training data or otherwise remove the training/evaluation overlap, and to re-report the system comparison using an independently separated or manually verified test set.
- [Table 3, §8] The ground-truth word error rate on the DualBench test set is 35.62%, while DualDub's generated speech reaches 12.74%; this indicates that the separated speech references in the test set are heavily corrupted or mislabeled, contradicting the assumption of clean, correctly paired ground-truth speech. Section 8 itself concedes that Mel-RoFormer 'sometimes separates empty audio or speech segments.' This does not necessarily invalidate the comparison across systems, but it undermines the reliability of ground-truth-anchored metrics and makes the test set unsuitable as a benchmark reference without further quality control. The authors should quantify the proportion of corrupted clips, provide a manual verification subset, and report the main results on that subset.
- [Tables 2-5] Objective metrics are reported as point estimates without error bars, confidence intervals, or significance tests. For instance, the DualScore differences in Table 5 (0.59 vs 0.19-0.30) and the FD/FAD/KLD differences in Tables 2 and 4 are presented without any measure of variance, so the reader cannot assess whether the differences are within noise. The subjective ASCH comparison is based on only 16 clips, even though 1,319 test clips are available. I request bootstrap confidence intervals or significance tests on all objective metrics and a subjective evaluation on a larger, diverse sample, or at least a justification for the small subjective sample size.
minor comments (6)
- [Figure 2] The label 'Casual Non-Casual' should be 'Causal Non-Causal'.
- [§4.3] The phrase 'we access audio and speech quality' should be 'we assess audio and speech quality'.
- [References] The reference list appears twice in the manuscript, duplicating references [1] through [58] verbatim; this should be corrected.
- [Table 4] The column header 'SSIM↑' under 'Audio-Video alignment' appears to be a typo for 'SIM↑' (speaker similarity), since the value 0.84 matches the SIM column in Table 3 and SSIM is not defined in the text.
- [Eq. (6)] Equation (6) contains a stray period and line break after the differential equation; the equation and surrounding punctuation should be cleaned up.
- [§5.1] The sentence 'Notably, following the data process of DualBench, we separate all audio-speech streams to avoid speech leaks into the background audio' is grammatically awkward and should be revised for clarity.
Circularity Check
No derivation-level circularity: DualDub's generation is not fitted to DualScore and the harmony ordering is independently supported by the human ASCH test; the in-paper DualScore sharing the Mel-RoFormer separation pipeline with training/test data is a metric-validity risk, not a circular reduction.
full rationale
The paper's central claim is an empirical model comparison rather than a derived prediction. DualDub is trained with cross-entropy token prediction (Eq. 5) and flow-matching loss (Eq. 7); it is never optimized against DualScore, and the generated samples are not in CASP's training set. The Table 5 DualScore margin (0.59 vs. 0.19-0.30) is therefore an out-of-sample evaluation, not a fitted value. The objective ordering is also reproduced by the independent 20-listener ASCH test (3.44 vs. 1.44-1.69), so the central claim does not reduce to the authors' own metric. The only self-referential element is the proposed DualScore: CASP is trained on 1,500 h of private audio-speech pairs that, like the V2ST test set, are produced by the same Mel-RoFormer separation and -40 dB energy filtering (Sections 4.2, 5.1, Table 1). This shared pipeline is a genuine metric-validity limitation: if separation leaves correlated artifacts, DualScore could reward those artifacts rather than perceived harmony, and Section 8 concedes that Mel-RoFormer 'sometimes separates empty audio or speech segments.' But this is a benchmark-quality/correctness risk, not a circular derivation: no equation equates DualScore to a fitted parameter of DualDub, and no proposed result is true by definition. Self-citations ([52], [57], [58]) appear only as context or as alternatives in the Limitations section and are not load-bearing for the main result. Accordingly, no circular step is exhibited, and the paper is best assessed as largely self-contained, with the DualScore pipeline flagged as an external-validity concern.
Assumptions & free parameters
free parameters (4)
- Energy filtering threshold =
-40 dB
- Curriculum training schedule =
3 stages, 40 epochs each, lr ranges 2e-6 to 2e-4, 2e-7 to 2e-4, 2e-7 to 2e-5
- Speaker reference duration =
3 seconds
- CASP segment length =
5 seconds
assumptions (5)
- standard math Autoregressive factorization of the joint speech-audio token distribution (Eq. 5)
- domain assumption Mel-RoFormer accurately separates speech from background audio
- domain assumption V2C-Animation transcripts match the spoken content
- domain assumption CLIP ViT-B/32 features are sufficient for video-to-audio synchronization
- ad hoc to paper DualScore (CASP embedding cosine similarity) measures audio-speech harmony
invented entities (1)
-
CASP audio-speech embedding space and DualScore metric
independent evidence
Cite this review
Pith. "Pith review of DualDub: Video-to-Soundtrack Generation via Joint Speech and Background Audio Synthesis." pith.science (2026). https://pith.science/paper/6KEIHP2M
@misc{pith2026250710109,
author = {Pith},
title = {Pith review of: DualDub: Video-to-Soundtrack Generation via Joint Speech and Background Audio Synthesis},
year = {2026},
howpublished = {\url{https://pith.science/paper/6KEIHP2M}},
note = {Machine review of arXiv:2507.10109}
}
read the original abstract
While recent video-to-audio (V2A) models can generate realistic background audio from visual input, they largely overlook speech, an essential part of many video soundtracks. This paper proposes a new task, video-to-soundtrack (V2ST) generation, which aims to jointly produce synchronized background audio and speech within a unified framework. To tackle V2ST, we introduce DualDub, a unified framework built on a multimodal language model that integrates a multimodal encoder, a cross-modal aligner, and dual decoding heads for simultaneous background audio and speech generation. Specifically, our proposed cross-modal aligner employs causal and non-causal attention mechanisms to improve synchronization and acoustic harmony. Besides, to handle data scarcity, we design a curriculum learning strategy that progressively builds the multimodal capability. Finally, we introduce DualBench, the first benchmark for V2ST evaluation with a carefully curated test set and comprehensive metrics. Experimental results demonstrate that DualDub achieves state-of-the-art performance, generating high-quality and well-synchronized soundtracks with both speech and background audio.
Figures
Forward citations
Cited by 1 Pith paper
-
AudioGen-Omni: A Unified Multimodal Diffusion Transformer for Video-Synchronized Audio, Speech, and Song Generation
A single multimodal diffusion transformer generates video-synchronized general audio, speech, and song from flexible combinations of video, text, and lyrics inputs.
Reference graph
Works this paper leans on
-
[1]
Kevin Cai, Chonghua Liu, and David M Chan. 2024. Anim-400K: A Large-Scale Dataset for Automated End to End Dubbing of Video. In ICASSP 2024-2024 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP). IEEE, 11796–11800
work page 2024
-
[2]
Honglie Chen, Weidi Xie, Andrea Vedaldi, and Andrew Zisserman. 2020. Vg- gsound: A large-scale audio-visual dataset. InICASSP 2020-2020 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) . IEEE, 721–725
work page 2020
-
[3]
Qi Chen, Mingkui Tan, Yuankai Qi, Jiaqiu Zhou, Yuanqing Li, and Qi Wu. 2022. V2C: Visual voice cloning. InProceedings of the IEEE/CVF Conference on Computer Vision and Pattern Recognition . 21242–21251
work page 2022
-
[4]
Sanyuan Chen, Yu Wu, Chengyi Wang, Shujie Liu, Daniel Tompkins, Zhuo Chen, and Furu Wei. 2022. Beats: Audio pre-training with acoustic tokenizers. arXiv preprint arXiv:2212.09058 (2022)
arXiv 2022
-
[5]
Yushen Chen, Zhikang Niu, Ziyang Ma, Keqi Deng, Chunhui Wang, Jian Zhao, Kai Yu, and Xie Chen. 2024. F5-TTS: A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching. CoRR abs/2410.06885 (2024)
arXiv 2024
-
[6]
Zhengyang Chen, Sanyuan Chen, Yu Wu, Yao Qian, Chengyi Wang, Shujie Liu, Yanmin Qian, and Michael Zeng. 2022. 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) . IEEE, 6147–6151
work page 2022
-
[7]
Ho Kei Cheng, Masato Ishii, Akio Hayakawa, Takashi Shibuya, Alexander G. Schwing, and Yuki Mitsufuji. 2024. Taming Multimodal Joint Training for High- Quality Video-to-Audio Synthesis. CoRR abs/2412.15322 (2024)
arXiv 2024
-
[8]
Jeongsoo Choi, Joanna Hong, and Yong Man Ro. 2023. Diffv2s: Diffusion-based video-to-speech synthesis with vision-guided speaker embedding. In Proceedings of the IEEE/CVF International Conference on Computer Vision . 7812–7821
work page 2023
Show all 62 references
-
[9]
Jeongsoo Choi, Ji-Hoon Kim, Jinyu Li, Joon Son Chung, and Shujie Liu. 2024. V2SFlow: Video-to-Speech Generation with Speech Decomposition and Rectified Flow. CoRR abs/2411.19486 (2024)
2024 arXiv
-
[10]
Gaoxiang Cong, Liang Li, Yuankai Qi, Zheng-Jun Zha, Qi Wu, Wenyu Wang, Bin Jiang, Ming-Hsuan Yang, and Qingming Huang. 2023. Learning to dub movies via hierarchical prosody models. In Proceedings of the IEEE/CVF Conference on Computer Vision and Pattern Recognition . 14687–14697
2023
-
[11]
Gaoxiang Cong, Yuankai Qi, Liang Li, Amin Beheshti, Zhedong Zhang, Anton van den Hengel, Ming-Hsuan Yang, Chenggang Yan, and Qingming Huang. 2024. StyleDubber: Towards Multi-Scale Style Learning for Movie Dubbing. InFindings of the Association for Computational Linguistics, AC...
2024
-
[12]
Russell, and Andrew Owens
Yuexi Du, Ziyang Chen, Justin Salamon, Bryan C. Russell, and Andrew Owens
-
[13]
Zhihao Du, Yuxuan Wang, Qian Chen, Xian Shi, Xiang Lv, Tianyu Zhao, Zhifu Gao, Yexin Yang, Changfeng Gao, Hui Wang, Fan Yu, Huadai Liu, Zhengyan Sheng, Yue Gu, Chong Deng, Wen Wang, Shiliang Zhang, Zhijie Yan, and Jingren Zhou. 2024. CosyVoice 2: Scalable Streaming Speech Synt...
2024 arXiv
-
[14]
Benjamin Elizalde, Soham Deshmukh, Mahmoud Al Ismail, and Huaming Wang
-
[15]
Patrick Esser, Sumith Kulal, Andreas Blattmann, Rahim Entezari, Jonas Müller, Harry Saini, Yam Levi, Dominik Lorenz, Axel Sauer, Frederic Boesel, et al. [n. d.]. Scaling rectified flow transformers for high-resolution image synthesis. In Forty- first international conference o...
-
[16]
In IEEE International Conference on Acoustics, Speech and Signal Processing ICASSP 2023, Rhodes Island, Greece, June 4-10, 2023
CLAP Learning Audio Concepts from Natural Language Supervision. In IEEE International Conference on Acoustics, Speech and Signal Processing ICASSP 2023, Rhodes Island, Greece, June 4-10, 2023 . IEEE, 1–5
2023
-
[17]
Ruibo Fu, Shuchen Shi, Hongming Guo, Tao Wang, Chunyu Qiang, Zhengqi Wen, Jianhua Tao, Xin Qi, Yi Lu, Xiaopeng Wang, Zhiyong Wang, Yukun Liu, Xuefei Liu, Shuai Zhang, and Guanjun Li. 2024. MINT: a Multi-modal Image and Narrative Text Dubbing Dataset for Foley Audio Content Pla...
2024 arXiv
-
[18]
Hawley, and Jordi Pons
Zach Evans, CJ Carr, Josiah Taylor, Scott H. Hawley, and Jordi Pons. 2024. Fast Timing-Conditioned Latent Audio Diffusion. In Forty-first International Confer- ence on Machine Learning, ICML 2024, Vienna, Austria, July 21-27, 2024 . OpenRe- view.net
2024
-
[19]
Yuan Gong, Andrew Rouditchenko, Alexander H Liu, David Harwath, Leonid Karlinsky, Hilde Kuehne, and James Glass. 2022. Contrastive audio-visual masked autoencoder. arXiv preprint arXiv:2210.07839 (2022)
2022 arXiv
-
[20]
Rohit Girdhar, Alaaeldin El-Nouby, Zhuang Liu, Mannat Singh, Kalyan Vasudev Alwala, Armand Joulin, and Ishan Misra. 2023. ImageBind One Embedding Space to Bind Them All. In IEEE/CVF Conference on Computer Vision and Pattern Recognition, CVPR 2023, Vancouver, BC, Canada, June 1...
2023
-
[21]
Shawn Hershey, Sourish Chaudhuri, Daniel PW Ellis, Jort F Gemmeke, Aren Jansen, R Channing Moore, Manoj Plakal, Devin Platt, Rif A Saurous, Bryan Seybold, et al. 2017. CNN architectures for large-scale audio classification. In2017 ieee international conference on acoustics, sp...
2017
-
[22]
Michael Hassid, Michelle Tadmor Ramanovich, Brendan Shillingford, Miaosen Wang, Ye Jia, and Tal Remez. 2022. More than Words: In-the-Wild Visually- Driven Prosody for Text-to-Speech. In IEEE/CVF Conference on Computer Vision and Pattern Recognition, CVPR 2022, New Orleans, LA,...
2022
-
[23]
Vladimir Iashin and Esa Rahtu. 2021. Taming Visually Guided Sound Generation. In British Machine Vision Conference
2021
-
[24]
Rongjie Huang, Mingze Li, Dongchao Yang, Jiatong Shi, Xuankai Chang, Zhen- hui Ye, Yuning Wu, Zhiqing Hong, Jiawei Huang, Jinglin Liu, Yi Ren, Yuexian Zou, Zhou Zhao, and Shinji Watanabe. 2024. AudioGPT: Understanding and Generating Speech, Music, Sound, and Talking Head. In T...
2024
-
[25]
Ziyue Jiang, Jinglin Liu, Yi Ren, Jinzheng He, Chen Zhang, Zhenhui Ye, Pengfei Wei, Chunfeng Wang, Xiang Yin, Zejun Ma, and Zhou Zhao. 2023. Mega-TTS 2: Zero-Shot Text-to-Speech with Arbitrary Length Speech Prompts. CoRR abs/2307.07218 (2023)
2023 arXiv
-
[26]
Shengpeng Ji, Ziyue Jiang, Xize Cheng, Yifu Chen, Minghui Fang, Jialong Zuo, Qian Yang, Ruiqi Li, Ziang Zhang, Xiaoda Yang, Rongjie Huang, Yidi Jiang, Qian Chen, Siqi Zheng, Wen Wang, and Zhou Zhao. 2024. WavTokenizer: an Efficient Acoustic Discrete Codec Tokenizer for Audio L...
2024 arXiv
-
[27]
Kevin Kilgour, Mauricio Zuluaga, Dominik Roblek, and Matthew Sharifi. 2018. Fr\’echet audio distance: A metric for evaluating music enhancement algorithms. arXiv preprint arXiv:1812.08466 (2018)
2018 arXiv
-
[28]
Zeqian Ju, Yuancheng Wang, Kai Shen, Xu Tan, Detai Xin, Dongchao Yang, Eric Liu, Yichong Leng, Kaitao Song, Siliang Tang, Zhizheng Wu, Tao Qin, Xiangyang Li, Wei Ye, Shikun Zhang, Jiang Bian, Lei He, Jinyu Li, and Sheng Zhao. 2024. NaturalSpeech 3: Zero-Shot Speech Synthesis w...
2024
-
[29]
Salik, Rajiv Ratn Shah, Yifang Yin, and Roger Zimmermann
Yaman Kumar, Rohit Jain, Khwaja Mohd. Salik, Rajiv Ratn Shah, Yifang Yin, and Roger Zimmermann. 2019. Lipper: Synthesizing Thy Speech Using Multi-View Lipreading. In The Thirty-Third AAAI Conference on Artificial Intelligence, AAAI 2019, The Thirty-First Innovative Application...
2019
-
[30]
Qiuqiang Kong, Yin Cao, Turab Iqbal, Yuxuan Wang, Wenwu Wang, and Mark D Plumbley. 2020. Panns: Large-scale pretrained audio neural networks for audio pattern recognition. IEEE/ACM Transactions on Audio, Speech, and Language Processing 28 (2020), 2880–2894
2020
-
[31]
Shun Lei, Yixuan Zhou, Boshi Tang, Max WY Lam, Hangyu Liu, Jingcheng Wu, Shiyin Kang, Zhiyong Wu, Helen Meng, et al. 2024. Songcreator: Lyrics-based universal song generation. Advances in Neural Information Processing Systems 37 (2024), 80107–80140
2024
-
[32]
Jiyoung Lee, Joon Son Chung, and Soo-Whan Chung. 2023. Imaginary Voice: Face-Styled Diffusion Model for Text-to-Speech. In IEEE International Conference on Acoustics, Speech and Signal Processing ICASSP 2023, Rhodes Island, Greece, June 4-10, 2023. IEEE, 1–5
2023
-
[33]
Mandic, Wenwu Wang, and Mark D
Haohe Liu, Zehua Chen, Yi Yuan, Xinhao Mei, Xubo Liu, Danilo P. Mandic, Wenwu Wang, and Mark D. Plumbley. 2023. AudioLDM: Text-to-Audio Gen- eration with Latent Diffusion Models. In International Conference on Machine Learning, ICML 2023, 23-29 July 2023, Honolulu, Hawaii, USA...
2023
-
[34]
Raghavan, Gavin Mischler, and Nima Mesgarani
Yinghao Aaron Li, Cong Han, Vinay S. Raghavan, Gavin Mischler, and Nima Mesgarani. 2023. StyleTTS 2: Towards Human-Level Text-to-Speech through Style Diffusion and Adversarial Training with Large Speech Language Models. In Advances in Neural Information Processing Systems 36: ...
2023
-
[35]
Lingwei Meng, Long Zhou, Shujie Liu, Sanyuan Chen, Bing Han, Shujie Hu, Yanqing Liu, Jinyu Li, Sheng Zhao, Xixin Wu, et al. 2024. Autoregressive speech synthesis without vector quantization. arXiv preprint arXiv:2407.08551 (2024)
2024 arXiv
-
[36]
Simian Luo, Chuanhao Yan, Chenxu Hu, and Hang Zhao. 2023. Diff-foley: Synchronized video-to-audio synthesis with latent diffusion models. Advances in Neural Information Processing Systems 36 (2023), 48855–48876
2023
-
[37]
Alec Radford, Jong Wook Kim, Tao Xu, Greg Brockman, Christine McLeavey, and Ilya Sutskever. 2023. Robust speech recognition via large-scale weak supervision. In International conference on machine learning . PMLR, 28492–28518
2023
-
[38]
Alec Radford, Jong Wook Kim, Chris Hallacy, Aditya Ramesh, Gabriel Goh, Sandhini Agarwal, Girish Sastry, Amanda Askell, Pamela Mishkin, Jack Clark, Gretchen Krueger, and Ilya Sutskever. 2021. Learning Transferable Visual Mod- els From Natural Language Supervision. In Proceedin...
2021
-
[39]
Roy Sheffer and Yossi Adi. 2023. I hear your true colors: Image guided audio gen- eration. In ICASSP 2023-2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) . IEEE, 1–5
2023
-
[40]
Takaaki Saeki, Detai Xin, Wataru Nakata, Tomoki Koriyama, Shinnosuke Takamichi, and Hiroshi Saruwatari. 2022. Utmos: Utokyo-sarulab system for voicemos challenge 2022. arXiv preprint arXiv:2204.02152 (2022)
2022 arXiv
-
[41]
Zineng Tang, Ziyi Yang, Mahmoud Khademi, Yang Liu, Chenguang Zhu, and Mohit Bansal. 2024. CoDi-2: In-Context, Interleaved, and Interactive Any-to-Any Generation. In IEEE/CVF Conference on Computer Vision and Pattern Recognition, CVPR 2024, Seattle, W A, USA, June 16-22, 2024. ...
2024
-
[42]
Kai Shen, Zeqian Ju, Xu Tan, Eric Liu, Yichong Leng, Lei He, Tao Qin, Sheng Zhao, and Jiang Bian. 2024. NaturalSpeech 2: Latent Diffusion Models are Natural and Zero-Shot Speech and Singing Synthesizers. In The Twelfth International Conference on Learning Representations, ICLR...
2024
-
[43]
Hui Wang, Shujie Liu, Lingwei Meng, Jinyu Li, Yifan Yang, Shiwan Zhao, Haiyang Sun, Yanqing Liu, Haoqin Sun, Jiaming Zhou, Yan Lu, and Yong Qin. 2025. FELLE: Autoregressive Speech Synthesis with Token-Wise Coarse-to-Fine Flow Match- ing. CoRR abs/2502.11128 (2025)
2025 arXiv
-
[44]
Hugo Touvron, Thibaut Lavril, Gautier Izacard, Xavier Martinet, Marie-Anne Lachaux, Timothée Lacroix, Baptiste Rozière, Naman Goyal, Eric Hambro, Faisal Azhar, Aurélien Rodriguez, Armand Joulin, Edouard Grave, and Guillaume Lam- ple. 2023. LLaMA: Open and Efficient Foundation ...
2023 arXiv
-
[45]
Ju-Chiang Wang, Wei-Tsung Lu, and Minz Won. 2023. Mel-Band RoFormer for Music Source Separation. arXiv preprint arXiv:2310.01809 (2023)
2023 arXiv
-
[46]
Heng Wang, Jianbo Ma, Santiago Pascual, Richard Cartwright, and Weidong Cai
-
[47]
Zhifeng Xie, Shengye Yu, Qile He, and Mengtian Li. 2024. Sonicvisionlm: Playing sound with vision language models. In Proceedings of the IEEE/CVF Conference Wenjie Tian, Xinfa Zhu, Haohe Liu, Zhixian Zhao, Zihao Chen, Chaofan Ding, Xinhan Di, Junjie Zheng, and Lei Xie on Compu...
2024
-
[48]
Yazhou Xing, Yingqing He, Zeyue Tian, Xintao Wang, and Qifeng Chen. 2024. Seeing and Hearing: Open-domain Visual-Audio Generation with Diffusion La- tent Aligners. InIEEE/CVF Conference on Computer Vision and Pattern Recognition, CVPR 2024, Seattle, W A, USA, June 16-22, 2024....
2024
-
[49]
Yongqi Wang, Wenxiang Guo, Rongjie Huang, Jiawei Huang, Zehan Wang, Fuming You, Ruiqi Li, and Zhou Zhao. 2024. Frieren: Efficient Video-to-Audio Generation Network with Rectified Flow Matching. In Advances in Neural Infor- mation Processing Systems 38: Annual Conference on Neu...
2024
-
[50]
Dongchao Yang, Jinchuan Tian, Xu Tan, Rongjie Huang, Songxiang Liu, Xuankai Chang, Jiatong Shi, Sheng Zhao, Jiang Bian, Xixin Wu, Zhou Zhao, Shinji Watan- abe, and Helen Meng. 2023. UniAudio: An Audio Foundation Model Toward Universal Audio Generation. CoRR abs/2310.00704 (2023)
2023 arXiv
-
[51]
Guy Yariv, Itai Gat, Sagie Benaim, Lior Wolf, Idan Schwartz, and Yossi Adi
-
[52]
An Yang, Baosong Yang, Binyuan Hui, Bo Zheng, Bowen Yu, Chang Zhou, Cheng- peng Li, Chengyuan Li, Dayiheng Liu, Fei Huang, Guanting Dong, Haoran Wei, Huan Lin, Jialong Tang, Jialin Wang, Jian Yang, Jianhong Tu, Jianwei Zhang, Jianxin Ma, Jianxin Yang, Jin Xu, Jingren Zhou, Jin...
2024 arXiv
-
[53]
Heiga Zen, Viet Dang, Rob Clark, Yu Zhang, Ron J Weiss, Ye Jia, Zhifeng Chen, and Yonghui Wu. 2019. Libritts: A corpus derived from librispeech for text-to- speech. arXiv preprint arXiv:1904.02882 (2019)
2019 arXiv
-
[54]
Yiming Zhang, Yicheng Gu, Yanhong Zeng, Zhening Xing, Yuancheng Wang, Zhizheng Wu, and Kai Chen. 2024. FoleyCrafter: Bring Silent Videos to Life with Lifelike and Synchronized Sounds. CoRR abs/2407.01494 (2024)
2024 arXiv
-
[55]
arXiv:2309.16429 [cs.LG]
Diverse and Aligned Audio-to-Video Generation via Text-to-Video Model Adaptation. arXiv:2309.16429 [cs.LG]
-
[56]
Zhen Ye, Xinfa Zhu, Chi-Min Chan, Xinsheng Wang, Xu Tan, Jiahe Lei, Yi Peng, Haohe Liu, Yizhu Jin, Zheqi DAI, et al . 2025. Llasa: Scaling Train-Time and Inference-Time Compute for Llama-based Speech Synthesis. arXiv preprint arXiv:2502.04128 (2025)
2025 arXiv
-
[57]
Xinfa Zhu, Wenjie Tian, Xinsheng Wang, Lei He, Xi Wang, Sheng Zhao, and Lei Xie. 2025. CosyAudio: Improving Audio Generation with Confidence Scores and Synthetic Captions. CoRR abs/2501.16761 (2025)
2025 arXiv
-
[58]
Xinfa Zhu, Wenjie Tian, and Lei Xie. 2024. Autoregressive Speech Synthesis with Next-Distribution Prediction. arXiv preprint arXiv:2412.16846 (2024)
2024
-
[59]
Zhedong Zhang, Liang Li, Gaoxiang Cong, Haibing Yin, Yuhan Gao, Chenggang Yan, Anton van den Hengel, and Yuankai Qi. 2024. From Speaker to Dubber: Movie Dubbing with Prosody and Duration Consistency Learning. InProceedings of the 32nd ACM International Conference on Multimedia...
2024
-
[60]
Yuan Zhao, Zhenqi Jia, Rui Liu, De Hu, Feilong Bao, and Guanglai Gao. 2024. Mcdubber: Multimodal context-aware expressive video dubbing. In National Conference on Man-Machine Speech Communication . Springer, 168–182
2024
-
[2023]
In IEEE/CVF Conference on Computer Vision and Pattern Recognition, CVPR 2023, Vancouver, BC, Canada, June 17-24, 2023
Conditional Generation of Audio from Video via Foley Analogies. In IEEE/CVF Conference on Computer Vision and Pattern Recognition, CVPR 2023, Vancouver, BC, Canada, June 17-24, 2023. IEEE, 2426–2436
2023
-
[2024]
V2A-Mapper: A Lightweight Solution for Vision-to-Audio Generation by Connecting Foundation Models. In Thirty-Eighth AAAI Conference on Artificial Intelligence, AAAI 2024, Thirty-Sixth Conference on Innovative Applications of Artificial Intelligence, IAAI 2024, Fourteenth Sympo...
2024
Reviewed August 6, 2026 · model on record in the stance chip above.
Discussion (0). Sign in to comment.