REVIEW 3 major objections 7 minor 44 references
RT-VC: Real-Time Zero-Shot Voice Conversion with Speech Articulatory Coding
T0 review · 3 major / 7 minor · reviewed 2026-08-07 · deepseek-v4-flash
Pith's one-line read RT-VC claims a real-time zero-shot voice conversion system that matches StreamVC's quality while cutting CPU latency to 61.4 ms, a 13.3 percent reduction.
desk verdict A credible streaming articulatory VC system whose headline latency win over StreamVC is not yet established as a like-for-like comparison. 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 object is the articulatory feature space defined by Speech Articulatory Coding (SPARC): a speaker-agnostic, normalized set of vocal tract kinematics (EMA) that is supposed to carry content, plus a separate speaker embedding for timbre. Around it, three causal components make real-time operation possible: a source extractor that predicts pitch, periodicity, and loudness at 200 Hz; an EMA inverter that maps MFCCs to pseudo-EMA labels through 11 dilated causal convolutions; and a DDSP harmonic-plus-noise vocoder that synthesizes speech from these articulatory features, modulated by the speaker embedding via FiLM layers. The latency equation is $L = 32\ \mathrm{ms} + 15\ \mathrm{ms} + 14.4\ \mathrm{ms} = 61.4\ \mathrm{ms}$, where 32 ms is the spectral feature lookahead, 15 ms is the input chunk size, and 14.4 ms is the measured average processing time per chunk on an Apple M3 CPU.
What would settle it
Measure RT-VC and StreamVC on the same CPU, with the same audio chunking and the same intelligibility, similarity, f0 consistency, and latency protocol; if RT-VC's end-to-end latency is not below StreamVC's on that common benchmark, or if its WER and speaker similarity are worse than the reported margins, the central claim fails.
Extended reading notes
Core claim
The paper's central claim is that a causal articulatory coding pipeline can match the quality of the best real-time zero-shot voice conversion system while being faster on CPU. RT-VC builds on the Speech Articulatory Coding (SPARC) idea: linguistic content is encoded as normalized, speaker-agnostic vocal tract kinematics (EMA), and speaker identity is captured separately by a speaker embedding, so swapping the embedding transfers voice while preserving content. To make this streamable, RT-VC trains a causal source extractor and a causal acoustic-to-articulatory inversion model using pseudo-EMA labels from SPARC, and uses a DDSP harmonic-plus-noise vocoder for synthesis. Evaluated on unseen LibriTTS speakers with VCTK targets, it reports WER 6.69 percent versus StreamVC's 6.22 percent, Resemblyzer similarity 76.65 percent versus 77.81 percent, f0 Pearson correlation 0.865 versus 0.842, and 61.4 ms CPU latency versus 70.8 ms, with naturalness scores above 3.8.
Load-bearing premise
The comparison assumes StreamVC's published metrics (70.8 ms latency on its own hardware, plus WER, CER, Resemblyzer, and f0 values) are directly comparable to RT-VC's self-measured numbers, even though the two systems were not evaluated under identical hardware and protocols and the naturalness metrics differ.
Editorial extensions
If this is right
- Zero-shot voice conversion can run at interactive speed on CPU-only servers, making browser-based conversion with dynamic target switching practical.
- The causal source extractor and EMA inverter form a streaming articulatory front-end that could be reused for real-time articulatory feedback or speech-to-EMA applications beyond voice conversion.
- DDSP vocoding directly from articulatory features is enough to match a strong learned-vocoder baseline in intelligibility, similarity, and f0 consistency, so heavy neural vocoding is not necessary for real-time quality.
- The same articulatory representation that carries content may also transfer pitch range naturally through the median-pitch ratio scaling, preserving prosody while changing timbre.
Reading between the lines
- Because the paper compares its own latency measurement on an Apple M3 CPU to StreamVC's published 70.8 ms figure, a same-benchmark replication on identical hardware would be needed to know whether the 13.3 percent gap holds or shifts.
- The pseudo-EMA labels come from an English-only, English-articulation probe, so conversion quality on non-English or heavily accented input likely degrades; a multilingual evaluation would test how far the articulatory representation really generalizes.
- The reported noise robustness applies to static noise above 20 dB SNR; babble, reverberation, or lower-SNR conditions are plausible failure points and are natural test extensions.
- If the articulatory front-end is speaker-independent enough, manipulating EMA coordinates directly could enable speech edits such as emotion, speaking-rate, or clarity control, which the paper does not attempt.
Editorial analysis
A structured set of objections, weighed in public.
Referee Report
Summary. RT-VC is a streaming, zero-shot voice conversion system that replaces the offline SPARC encoder with a causal source extractor and EMA inverter, and uses a causal DDSP vocoder conditioned on a target-speaker embedding. The paper evaluates RT-VC on 377 LibriTTS test utterances converted to 6 VCTK target speakers, reporting naturalness (UTMOS/MOS), intelligibility (WER/CER), speaker similarity (Resemblyzer/SMOS), f0 correlation, and CPU latency. The central claim is that RT-VC matches StreamVC's conversion quality while reducing CPU latency from 70.8 ms to 61.4 ms (13.3%). A noise-robustness experiment and a web demo are also presented.
Significance. The potential contribution is a practical real-time VC system built from articulatory coding, with a concrete latency figure and a working demo. The causal, streamable redesign of the SPARC encoder and the integration of a streaming DDSP vocoder are sensible engineering steps, and the noise-robustness evaluation is a useful addition. The authors are transparent about limitations, including the language-specific nature of the pseudo-EMA labels and the dependence on recording quality. However, the headline comparison with StreamVC is currently not a controlled experiment: latency is measured on different hardware and with a different latency decomposition, and the quality comparison lacks a direct naturalness measurement and statistical significance metrics. If these comparison issues are addressed, the system would be a solid contribution to real-time VC.
major comments (3)
- [§3.6, Eq. (2); Table 1] The headline claim of a 13.3% CPU latency reduction is not based on a like-for-like comparison. RT-VC's 61.4 ms is computed as 32 ms lookahead + 15 ms chunk size + 14.4 ms processing time measured on an Apple M3 CPU, while StreamVC's 70.8 ms is taken directly from its publication without specifying the hardware, measurement protocol, or the exact decomposition of that figure. Because processing time is hardware-dependent and the two systems use different lookahead and chunking definitions, the reported ordering could change if both systems were measured on the same machine. The paper should either run StreamVC under the same conditions or explicitly present the comparison as indicative and report StreamVC's measurement details.
- [§5.3, Table 1] The claim that RT-VC maintains synthesis quality comparable to StreamVC is not fully supported. Naturalness is not directly compared: StreamVC has no UTMOS or MOS values in the table, and the paper states it could not reproduce StreamVC's DNSMOS result. In addition, WER, CER, Resemblyzer, and f0 PCC are reported as point estimates without error bars or significance tests, so the small observed differences (e.g., WER +0.47%, Resemblyzer –1.16%) cannot be distinguished from evaluation noise. Please provide confidence intervals or statistical tests for these metrics, and either obtain a comparable naturalness score for StreamVC or explicitly restrict the 'comparable quality' claim to intelligibility, similarity, and f0 consistency.
- [§3.3 and §5.1] The causal EMA inverter is trained on pseudo-EMA labels generated by the SPARC model, and the DDSP vocoder is also from prior work by the same group. The evaluation therefore demonstrates that RT-VC can approximate the offline SPARC pipeline in a streaming setting, but it does not independently validate the articulatory feature space as a conversion representation. The paper should state this limitation explicitly in the discussion, since it bears on how the interpretability and grounding claims are interpreted; the comparison with StreamVC remains valid for the final audio output, but the causal chain from speech to articulatory features to speech is only as strong as the quality of the pseudo labels.
minor comments (7)
- [§3.6] The paper should report the real-time factor (RTF) or throughput on the measured CPU, not just the per-chunk processing time, to allow reproducibility and to clarify what 'real-time' means beyond a single latency figure.
- [§5.2] The sentence 'we adopt the same evaluation protocol for all metrics except for naturalness and speaker similarity' is ambiguous because the table lists both automatic and subjective speaker similarity; please clarify which metrics are directly comparable to StreamVC and which are new measurements.
- [Figure 4] The noise robustness curves lack error bars and the number of utterances per SNR condition is not reported, so the trends cannot be assessed for statistical reliability.
- [Table 1] Please use footnotes or formatting to mark which values are taken from the StreamVC publication and which are new measurements, and state the hardware used for the latency row.
- [§3.3] The 'information perturbation technique' is described but the citation to Choi et al. (2021) is not given at the point of use; please add the citation in that sentence.
- [Title and Section 1] The title and the first paragraph contain a spurious space in 'V oice'; please correct the typo.
- [§5.1] The claim that all target speakers are 'unseen during training' should clarify whether the pretrained WavLM backbone used in the speaker encoder may have seen VCTK speakers, since that affects the zero-shot interpretation.
Circularity Check
No significant circularity; RT-VC is an empirical system evaluation against an external baseline, not a derivation that reduces to its own inputs.
full rationale
RT-VC's central claims are empirical: conversion quality is measured with standard external metrics (WER, CER, UTMOS, MOS, Resemblyzer, SMOS, f0 PCC) and latency is computed from a stated additive formula. The causal EMA inverter is trained to minimize an L1 loss against pseudo-EMA labels produced by the authors' prior SPARC inversion model, but this is supervised fitting of a learned predictor, not the derivation of a claimed result from that same result. The DDSP vocoder and SPARC components are cited from prior work by the same group, but the paper does not use those citations as the evidence for its central conclusion; the conclusion rests on comparisons to StreamVC, an external system, using independent audio-quality metrics. No equation in the paper equates a predicted quantity to its fitting target by construction, and no fitted parameter is renamed as a prediction. The reliance on StreamVC's published latency (70.8 ms) and the inability to reproduce its DNSMOS value raise measurement-protocol and comparability concerns, but those are correctness risks, not circularity. Self-citation is present in the construction of the system, but it is not load-bearing in any circular sense.
Assumptions & free parameters
free parameters (5)
- window_size =
1024 samples (64ms at 16kHz)
- chunk_size =
15ms
- speaker_embedding_dim =
128
- num_dilated_conv_layers =
11
- processing_time_per_chunk =
14.4ms on Apple M3 CPU
assumptions (7)
- domain assumption SPARC's articulatory feature space provides a naturally disentangled and interpretable representation separating content from speaker identity.
- domain assumption The pseudo-EMA labels produced by the SPARC acoustic-to-articulatory inversion model are accurate enough to serve as ground truth for training a causal EMA inverter.
- standard math The DDSP vocoder from Liu et al. 2024 achieves fast inference and high quality when conditioned on articulatory features.
- domain assumption The linear affine approximation between individual speakers' articulatory systems is sufficient for a universal articulatory space.
- ad hoc to paper Pitch rescaling by the ratio of median pitches of target and source speakers is an adequate conversion of prosody.
- domain assumption StreamVC's published metrics are reliable and comparable to RT-VC's own measurements.
- domain assumption Causal convolutional networks with dilation provide sufficient context to approximate non-causal feature extraction without notable quality loss.
Cite this review
Pith. "Pith review of RT-VC: Real-Time Zero-Shot Voice Conversion with Speech Articulatory Coding." pith.science (2026). https://pith.science/paper/WKHQCVTC
@misc{pith2026250610289,
author = {Pith},
title = {Pith review of: RT-VC: Real-Time Zero-Shot Voice Conversion with Speech Articulatory Coding},
year = {2026},
howpublished = {\url{https://pith.science/paper/WKHQCVTC}},
note = {Machine review of arXiv:2506.10289}
}
read the original abstract
Voice conversion has emerged as a pivotal technology in numerous applications ranging from assistive communication to entertainment. In this paper, we present RT-VC, a zero-shot real-time voice conversion system that delivers ultra-low latency and high-quality performance. Our approach leverages an articulatory feature space to naturally disentangle content and speaker characteristics, facilitating more robust and interpretable voice transformations. Additionally, the integration of differentiable digital signal processing (DDSP) enables efficient vocoding directly from articulatory features, significantly reducing conversion latency. Experimental evaluations demonstrate that, while maintaining synthesis quality comparable to the current state-of-the-art (SOTA) method, RT-VC achieves a CPU latency of 61.4 ms, representing a 13.3\% reduction in latency.
Figures
Figures from the paper (1 more)
Reference graph
Works this paper leans on
-
[1]
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-
[3]
Ahmed Adel Attia, Yashish M Siriwardena, and Carol Espy-Wilson. 2024. Improving speech inversion through self-supervised embeddings and enhanced tract variables. In 2024 32nd European Signal Processing Conference (EUSIPCO), pages 306--310. IEEE
work page 2024
-
[4]
Alexei Baevski, Yuhao Zhou, Abdelrahman Mohamed, and Michael Auli. 2020. wav2vec 2.0: A framework for self-supervised learning of speech representations. Advances in neural information processing systems, 33:12449--12460
2020
-
[5]
Sanyuan Chen, Chengyi Wang, Zhengyang Chen, Yu Wu, Shujie Liu, Zhuo Chen, Jinyu Li, Naoyuki Kanda, Takuya Yoshioka, Xiong Xiao, et al. 2022. Wavlm: Large-scale self-supervised pre-training for full stack speech processing. IEEE Journal of Selected Topics in Signal Processing, 16(6):1505--1518
2022
-
[6]
Yu-Wen Chen, Kuo-Hsuan Hung, Shang-Yi Chuang, Jonathan Sherman, Wen-Chin Huang, Xugang Lu, and Yu Tsao. 2021. Ema2s: An end-to-end multimodal articulatory-to-speech system. In IEEE International Symposium on Circuits and Systems (ISCAS), pages 1--5
work page 2021
-
[7]
Cheol Jun Cho, Abdelrahman Mohamed, Alan W Black, and Gopala K Anumanchipalli. 2024 a . Self-supervised models of speech infer universal articulatory kinematics. In ICASSP 2024-2024 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 12061--12065. IEEE
work page 2024
-
[8]
Cheol Jun Cho, Peter Wu, Abdelrahman Mohamed, and Gopala K Anumanchipalli. 2023. Evidence of vocal tract articulation in self-supervised learning of speech. In ICASSP 2023-2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 1--5. IEEE
2023
Show all 44 references
-
[9]
Prabhune, Dhruv Agarwal, and Gopala K
Cheol Jun Cho, Peter Wu, Tejas S. Prabhune, Dhruv Agarwal, and Gopala K. Anumanchipalli. 2024 b . Coding speech through vocal tract kinematics. IEEE Journal of Selected Topics in Signal Processing, 18(8):1427--1440
2024
-
[10]
Hyeong-Seok Choi, Juheon Lee, Wansoo Kim, Jie Lee, Hoon Heo, and Kyogu Lee. 2021. Neural analysis and synthesis: Reconstructing speech from self-supervised representations. Advances in Neural Information Processing Systems, 34:16251--16265
2021
-
[11]
Ju-chieh Chou, Cheng-chieh Yeh, and Hung-yi Lee. 2019. One-shot voice conversion by separating speaker and content representations with instance normalization. arXiv preprint arXiv:1904.05742
2019 arXiv
-
[12]
Zhihao Du, Qian Chen, Shiliang Zhang, Kai Hu, Heng Lu, Yexin Yang, Hangrui Hu, Siqi Zheng, Yue Gu, Ziyang Ma, et al. 2024 a . Cosyvoice: A scalable multilingual zero-shot text-to-speech synthesizer based on supervised semantic tokens. arXiv preprint arXiv:2407.05407
2024 arXiv
-
[13]
Zhihao Du, Yuxuan Wang, Qian Chen, Xian Shi, Xiang Lv, Tianyu Zhao, Zhifu Gao, Yexin Yang, Changfeng Gao, Hui Wang, et al. 2024 b . Cosyvoice 2: Scalable streaming speech synthesis with large language models. arXiv preprint arXiv:2412.10117
2024 arXiv
-
[14]
Yingming Gao, Peter Birkholz, and Ya Li. 2024. Articulatory copy synthesis based on the speech synthesizer vocaltractlab and convolutional recurrent neural networks. IEEE/ACM Transactions on Audio, Speech, and Language Processing
2024
-
[15]
Wei-Ning Hsu, Benjamin Bolte, Yao-Hung Hubert Tsai, Kushal Lakhotia, Ruslan Salakhutdinov, and Abdelrahman Mohamed. 2021. Hubert: Self-supervised speech representation learning by masked prediction of hidden units. IEEE/ACM transactions on audio, speech, and language processin...
2021
-
[16]
Zeqian Ju, Yuancheng Wang, Kai Shen, Xu Tan, Detai Xin, Dongchao Yang, Yanqing Liu, Yichong Leng, Kaitao Song, Siliang Tang, et al. 2024. Naturalspeech 3: Zero-shot speech synthesis with factorized codec and diffusion models. arXiv preprint arXiv:2403.03100
2024 arXiv
-
[17]
Hirokazu Kameoka, Takuhiro Kaneko, Kou Tanaka, and Nobukatsu Hojo. 2018. Stargan-vc: Non-parallel many-to-many voice conversion using star generative adversarial networks. In 2018 IEEE Spoken Language Technology Workshop (SLT), pages 266--273. IEEE
2018
-
[18]
Takuhiro Kaneko and Hirokazu Kameoka. 2018. Cyclegan-vc: Non-parallel voice conversion using cycle-consistent adversarial networks. In 2018 26th European Signal Processing Conference (EUSIPCO), pages 2100--2104. IEEE
2018
-
[19]
Takuhiro Kaneko, Hirokazu Kameoka, Kou Tanaka, and Nobukatsu Hojo. 2019 a . Cyclegan-vc2: Improved cyclegan-based non-parallel voice conversion. In ICASSP 2019-2019 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 6820--6824. IEEE
2019
-
[20]
Takuhiro Kaneko, Hirokazu Kameoka, Kou Tanaka, and Nobukatsu Hojo. 2019 b . Stargan-vc2: Rethinking conditional methods for stargan-based voice conversion. arXiv preprint arXiv:1907.12279
2019 arXiv
-
[21]
Anton Kashkin, Ivan Karpukhin, and Svyatoslav Shishkin. 2022. Hifi-vc: High quality asr-based voice conversion. arXiv preprint arXiv:2203.16937
2022 arXiv
-
[22]
Jong Wook Kim, Justin Salamon, Peter Li, and Juan Pablo Bello. 2018. Crepe: A convolutional representation for pitch estimation. In 2018 IEEE international conference on acoustics, speech and signal processing (ICASSP), pages 161--165. IEEE
2018
-
[23]
Miseul Kim, Zhenyu Piao, Jihyun Lee, and Hong-Goo Kang. 2023. Style modeling for multi-speaker articulation-to-speech. In ICASSP, pages 1--5
2023
-
[24]
Yuma Koizumi, Heiga Zen, Shigeki Karita, Yifan Ding, Kohei Yatabe, Nobuyuki Morioka, Michiel Bacchiani, Yu Zhang, Wei Han, and Ankur Bapna. 2023. Libritts-r: A restored multi-speaker text-to-speech corpus. arXiv preprint arXiv:2305.18802
2023 arXiv
-
[25]
Jungil Kong, Jaehyeon Kim, and Jaekyoung Bae. 2020. Hifi-gan: Generative adversarial networks for efficient and high fidelity speech synthesis. Advances in neural information processing systems, 33:17022--17033
2020
-
[26]
Jingyi Li, Weiping Tu, and Li Xiao. 2023. Freevc: Towards high-quality text-free one-shot voice conversion. In ICASSP 2023-2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 1--5. IEEE
2023
-
[27]
Jiachen Lian, Chunlei Zhang, and Dong Yu. 2022. Robust disentangled variational speech representation learning for zero-shot voice conversion. In ICASSP 2022-2022 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 6572--6576. IEEE
2022
-
[28]
Yisi Liu, Bohan Yu, Drake Lin, Peter Wu, Cheol Jun Cho, and Gopala Krishna Anumanchipalli. 2024. Fast, high-quality and parameter-efficient articulatory synthesis using differentiable dsp. In 2024 IEEE Spoken Language Technology Workshop (SLT), pages 711--718. IEEE
2024
-
[29]
Seung-won Park, Doo-young Kim, and Myun-chul Joe. 2020. Cotatron: Transcription-guided speech encoder for any-to-many voice conversion without parallel data. arXiv preprint arXiv:2005.03295
2020 arXiv
-
[30]
Ethan Perez, Florian Strub, Harm De Vries, Vincent Dumoulin, and Aaron Courville. 2018. Film: Visual reasoning with a general conditioning layer. In Proceedings of the AAAI conference on artificial intelligence, volume 32
2018
-
[31]
Kaizhi Qian, Yang Zhang, Shiyu Chang, Mark Hasegawa-Johnson, and David Cox. 2020. Unsupervised speech decomposition via triple information bottleneck. In International Conference on Machine Learning, pages 7836--7846. PMLR
2020
-
[32]
Kaizhi Qian, Yang Zhang, Shiyu Chang, Xuesong Yang, and Mark Hasegawa-Johnson. 2019. Autovc: Zero-shot voice style transfer with only autoencoder loss. In International Conference on Machine Learning, pages 5210--5219. PMLR
2019
-
[33]
Kaizhi Qian, Yang Zhang, Heting Gao, Junrui Ni, Cheng-I Lai, David Cox, Mark Hasegawa-Johnson, and Shiyu Chang. 2022. Contentvec: An improved self-supervised speech representation by disentangling speakers. In International conference on machine learning, pages 18003--18017. PMLR
2022
-
[34]
Yashish M Siriwardena and Carol Espy-Wilson. 2023. The secret source: Incorporating source features to improve acoustic-to-articulatory speech inversion. In ICASSP 2023-2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 1--5. IEEE
2023
-
[35]
Lifa Sun, Kun Li, Hao Wang, Shiyin Kang, and Helen Meng. 2016. Phonetic posteriorgrams for many-to-one voice conversion without parallel data training. In 2016 IEEE International Conference on Multimedia and Expo (ICME), pages 1--6. IEEE
2016
-
[36]
Benjamin Van Niekerk, Marc-Andr \'e Carbonneau, Julian Za \" di, Matthew Baas, Hugo Seut \'e , and Herman Kamper. 2022. A comparison of discrete and soft speech units for improved voice conversion. In ICASSP 2022-2022 IEEE International Conference on Acoustics, Speech and Sign...
2022
-
[37]
Haojie Wei, Xueke Cao, Tangpeng Dan, and Yueguo Chen. 2023. Rmvpe: A robust model for vocal pitch estimation in polyphonic music. arXiv preprint arXiv:2306.15412
2023 arXiv
-
[38]
Anumanchipalli
Peter Wu, Li-Wei Chen, Cheol Jun Cho, Shinji Watanabe, Louis Goldstein, Alan W Black, and Gopala K. Anumanchipalli. 2023. Speaker-independent acoustic-to-articulatory speech inversion. In ICASSP 2023 - 2023 IEEE International Conference on Acoustics, Speech and Signal Processi...
2023
-
[39]
Peter Wu, Paul Pu Liang, Jiatong Shi, Ruslan Salakhutdinov, Shinji Watanabe, and Louis-Philippe Morency. 2021. Understanding the tradeoffs in client-side privacy for downstream speech tasks. In 2021 Asia-Pacific Signal and Information Processing Association Annual Summit and C...
2021
-
[40]
Peter Wu, Shinji Watanabe, Louis Goldstein, Alan W Black, and Gopala Krishna Anumanchipalli. 2022. Deep speech synthesis from articulatory representations. In Interspeech
2022
-
[41]
Junichi Yamagishi, Christophe Veaux, and Kirsten MacDonald. 2019. https://doi.org/10.7488/ds/2645 Cstr vctk corpus: English multi-speaker corpus for cstr voice cloning toolkit (version 0.92) . [sound]
2019 doi
-
[42]
Yang Yang, Yury Kartynnik, Yunpeng Li, Jiuqiang Tang, Xing Li, George Sung, and Matthias Grundmann. 2024. https://arxiv.org/abs/2401.03078 Streamvc: Real-time low-latency voice conversion . Preprint, arXiv:2401.03078
2024 arXiv
-
[43]
Neil Zeghidour, Alejandro Luebs, Ahmed Omran, Jan Skoglund, and Marco Tagliasacchi. 2021. Soundstream: An end-to-end neural audio codec. IEEE/ACM Transactions on Audio, Speech, and Language Processing, 30:495--507
2021
-
[44]
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 arXiv
Reviewed August 7, 2026 · model on record in the stance chip above.
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