REVIEW 4 major objections 4 minor 1 cited by
Voila: Voice-Language Foundation Models for Real-Time Autonomous Interaction and Voice Role-Play
T0 review · 4 major / 4 minor · reviewed 2026-08-16 · deepseek-v4-flash
Pith's one-line read Voila presents a full-duplex voice-language model family that answers in 195 milliseconds, clones voices from ten-second samples, and unifies ASR, TTS, and conversation in one open-source architecture.
desk verdict The open-source model and LibriSpeech numbers are real, but the paper's headline 195 ms latency and full-duplex autonomy claims have no experimental support anywhere in the manuscript. 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 central object is the hierarchical multi-scale Transformer: a voice-language LLM backbone that handles semantic content and an audio Transformer that predicts acoustic tokens from the backbone's output. Its companion pieces are the Voila tokenizer—a four-layer residual-vector-quantization neural codec with semantic information distilled into the first level—and the word-level text-audio interleaved alignment, in which each text token is paired with its own audio tokens and then embedded, repeated, and averaged before entering the backbone. For full-duplex interaction, Voila-autonomous fuses two streams by tokenizing and embedding both the user's audio and its own audio and averaging the embeddings before the backbone. This machinery lets the pretrained LLM reason in a familiar symbolic space while a separate module handles fine acoustic detail, and it is what turns voice role-play, million-voice libraries, and 195-millisecond latency into properties of a single open-source model rather than an assembled pipeline.
What would settle it
Play the 1,580 spoken benchmark questions to human participants and score their answers with the same automated rubric used for the model outputs, then compare with published accuracy on the original text items; if spoken accuracy falls materially below text accuracy, the difficulty-preservation assumption fails and the reported leaderboard gaps would need rechecking.
Extended reading notes
Core claim
Voila's central discovery is that the reasoning power of a pretrained LLM and the acoustic richness of speech can be combined in one end-to-end architecture, without a text bottleneck, by separating semantic and acoustic modeling across a multi-scale Transformer. The model uses a neural tokenizer whose first residual-vector-quantization level carries semantic content and whose remaining levels carry acoustic detail; the backbone LLM consumes semantic tokens interleaved with text tokens at word level, and a second audio Transformer predicts the acoustic tokens that the tokenizer decodes into waveform. Voila-autonomous, the full-duplex variant, fuses two simultaneous input streams—the user's audio and its own audio—so it can hear itself while speaking and decide to interrupt, backchannel, or stay silent. Trained on speech recognition, synthesis, and instruction-following tasks under a chat format, the same model reaches 30.56 on the new Voila Benchmark (versus 13.29 for SpeechGPT and 11.45 for Moshi), competitive ASR word error rates on LibriSpeech (2.7% when trained on its training split), and a TTS word error rate of 3.2%. The design also treats speaker identity as a plug-in embedding, making voice cloning from ten seconds of audio and persona steering by text direct corollaries of the architecture rather than add-on modules.
Load-bearing premise
The load-bearing premise is that turning text questions into speech and scoring machine transcripts of the answers preserves the difficulty of the original tasks and measures voice-language ability rather than the side effects of transcription and judging.
Editorial extensions
If this is right
- A spoken assistant built on Voila could interject, acknowledge, and respond mid-utterance, matching the timing and overlap of human conversation rather than waiting for a full turn.
- Voice cloning and persona creation would no longer require separate TTS modules or fine-tuning per speaker: one ten-second sample plus a text prompt defines a new character.
- Because ASR, TTS, and dialogue share one backbone, downstream tasks such as multilingual speech translation and emotion-aware response can be added with minimal finetuning, and the same model serves all of them.
- Releasing the models, tokenizer, benchmark, and a million-voice library would give the field a common test bed for measuring whether speech-to-speech models preserve LLM reasoning, not just acoustic fidelity.
- The 195-millisecond latency figure, if reproducible in real deployments, would push voice AI below a human-visible threshold, making full-duplex conversation the default interface for wearable and ambient AI.
Reading between the lines
- A direct test the paper does not run: feed the same questions to the model as text and as speech; if accuracy drops sharply on spoken input, the bottleneck is audio understanding rather than reasoning, which would change how the architecture should be judged.
- The benchmark relies on a text-to-speech rewrite of math and code and on an automated judge scoring speech-recognition transcripts, so the numbers likely understate acoustic capability and overstate reasoning under noisy conditions; a human-listener evaluation on the same 1,580 items would settle which.
- If the word-level text-audio interleaved alignment holds up, it suggests that explicit sub-word synchronization, not just a streaming encoder, is the active ingredient for stable full-duplex training—an insight transferable to other speech-language models.
- The million-voice library, built from an embedding extractor, implies that voice identity may be separable from linguistic content in the learned representation; that hypothesis could be tested by cross-lingual voice cloning and by removing the speaker embedding at inference.
Signed reviews
Editorial analysis
A structured set of objections, weighed in public.
Referee Report
Summary. The paper introduces Voila, a family of end-to-end voice-language models built on a hierarchical multi-scale Transformer. The architecture combines a voice tokenizer with semantic and acoustic RVQ levels, a text-audio interleaved alignment strategy, and a two-stream full-duplex variant (Voila-autonomous) that averages embeddings of the user and system audio streams. The authors claim 195 ms response latency, support for over a million voices, and unified ASR/TTS/dialogue capabilities in a single model. The experimental section describes a new audio benchmark (Voila Benchmark) constructed by rewriting text questions with GPT-4o and synthesizing them with Google TTS, with scoring by GPT-4o on Whisper transcriptions; it also reports LibriSpeech ASR and TTS WER numbers. Voila outperforms SpeechGPT and Moshi on the new benchmark and is competitive with strong ASR/TTS baselines.
Significance. The paper has tangible strengths: the models, code, tokenizer, benchmark, and a million-voice library are released open-source; the LibriSpeech ASR and TTS results are concrete and competitive; the proposed interleaved text-audio alignment with a four-level RVQ tokenizer is a plausible architectural contribution. If the headline claims of 195 ms latency and full-duplex autonomous interaction were properly measured and validated, the work would be significant for real-time voice AI. However, the current manuscript does not provide such measurements, and the new benchmark's scoring protocol is unvalidated, so the benchmark numbers cannot substitute for evidence of the paper's central claims. The value of the contribution is therefore primarily in the released artifacts and the component-level results, not in the demonstrated real-time autonomous capabilities featured in the title and abstract.
major comments (4)
- [Abstract and Section 1] The abstract's headline quantitative claim, 'response latency of just 195 milliseconds, surpassing the average human response time,' is never defined or measured in the paper. Section 4 contains no latency experiment, no definition of the latency interval (e.g., end of user speech to start of system audio, or time-to-first-token), no hardware or streaming configuration, and no comparison against pipeline or other end-to-end baselines. As the central quantitative promise of the paper, its complete absence from the evaluation is a load-bearing omission.
- [Section 3.2 and Figure 5] The 'full-duplex' and 'autonomous interaction' claims are not supported by the presented mechanism or by any experiment. The description only states that the user audio stream and Voila's own audio stream are tokenized, embedded, averaged, and fed into the backbone LLM; it does not explain or evaluate how the model decides when to speak, how it handles barge-in, whether it produces backchannels, or how it avoids self-interference. No results on overlapping speech, interruption handling, or proactive initiation are provided, despite the title and abstract promising 'real-time autonomous interaction.'
- [Sections 4.1 and 4.2] The Voila Benchmark's validity is not established. The benchmark is constructed by rewriting text questions with GPT-4o and synthesizing speech with Google TTS, and responses are scored by GPT-4o on Whisper transcriptions. No evidence is given that TTS rendering preserves the difficulty of the original tasks, that GPT-4o scoring correlates with human judgment, or that Whisper transcription errors affect all compared models equally. Without such validation, the numerical comparisons in Tables 1 and 2 cannot be interpreted as reliable measurements of voice-language capability.
- [Section 3.3] The paper claims 'over one million pre-built voices' and customization from audio samples 'as short as 10 seconds,' but it provides no experimental evaluation of voice similarity, naturalness, or intelligibility of the cloned voices. This is a stated key feature of the system, and its absence from the experimental section weakens the claim.
minor comments (4)
- [Throughout] The text contains repeated formatting artifacts such as 'V oila' with a space; these should be fixed in the final version.
- [References] The same work by Maiti et al. appears as both 2024a and 2024b in the references and citations; please consolidate.
- [Abstract vs. Section 3.3] The abstract says voice cloning from samples 'as short as 10 seconds,' while Section 3.3 says the audio clip can be 'any length—ranging from a few seconds to several hours.' Please make the statements consistent.
- [Section 3 and 4] The paper does not report model sizes, number of parameters, training data composition beyond 'extensive audio-text data,' or compute budget. For a foundation-model paper, at least the parameter count per model variant should be stated in the main text.
Circularity Check
No significant circularity: Voila's claims are measured on external benchmarks or are unsupported assertions, and the only self-citation is descriptive and not load-bearing.
full rationale
Voila's central evaluations are conducted against external benchmarks and external baselines. The ASR results use LibriSpeech test-clean with standard WER, and the TTS results follow the VALL-E protocol with HuBERT transcription; these are independent, parameter-free measurements with no fitted quantity renamed as a prediction. The Voila Benchmark is assembled from external text datasets (MMLU, MATH, HumanEval, NQ-Open, GSM8K), converted to speech by GPT-4o rewriting plus Google TTS, and scored by GPT-4o on Whisper transcriptions. This protocol raises validity questions about whether spoken rewrites preserve difficulty and whether GPT-4o scores match human judgment, but it does not reduce Voila's accuracy to its training inputs or to any fitted parameter: the reference answers come from the original datasets, and the model being evaluated is distinct from the grading model. The abstract's headline numbers—195 ms latency and full-duplex autonomous interaction—are never defined or measured in the experimental section, but an unsupported claim is an evidentiary gap, not a circular derivation. The only self-citation is LLaSM (Shu et al., 2023) in the related-work survey of Whisper-encoder connector models; it is descriptive, not load-bearing for Voila's architecture or for any result. No equation, benchmark score, or design choice can be shown to be equivalent to its own inputs by construction, so no significant circularity is present.
Assumptions & free parameters
assumptions (4)
- domain assumption Whisper transcription followed by GPT-4o scoring is a valid measure of spoken-response correctness.
- ad hoc to paper Converting text benchmarks to speech with GPT-4o rewriting and Google TTS preserves the reasoning difficulty of the original tasks.
- ad hoc to paper The hierarchical multi-scale architecture with interleaved text-audio tokens is effective for voice-language modeling.
- ad hoc to paper Averaging the embeddings of the user and Voila audio streams in Voila-autonomous enables full-duplex interaction without self-interference.
Cite this review
Pith. "Pith review of Voila: Voice-Language Foundation Models for Real-Time Autonomous Interaction and Voice Role-Play." pith.science (2026). https://pith.science/paper/5QZ2R3MV
@misc{pith2026250502707,
author = {Pith},
title = {Pith review of: Voila: Voice-Language Foundation Models for Real-Time Autonomous Interaction and Voice Role-Play},
year = {2026},
howpublished = {\url{https://pith.science/paper/5QZ2R3MV}},
note = {Machine review of arXiv:2505.02707}
}
read the original abstract
A voice AI agent that blends seamlessly into daily life would interact with humans in an autonomous, real-time, and emotionally expressive manner. Rather than merely reacting to commands, it would continuously listen, reason, and respond proactively, fostering fluid, dynamic, and emotionally resonant interactions. We introduce Voila, a family of large voice-language foundation models that make a step towards this vision. Voila moves beyond traditional pipeline systems by adopting a new end-to-end architecture that enables full-duplex, low-latency conversations while preserving rich vocal nuances such as tone, rhythm, and emotion. It achieves a response latency of just 195 milliseconds, surpassing the average human response time. Its hierarchical multi-scale Transformer integrates the reasoning capabilities of large language models (LLMs) with powerful acoustic modeling, enabling natural, persona-aware voice generation -- where users can simply write text instructions to define the speaker's identity, tone, and other characteristics. Moreover, Voila supports over one million pre-built voices and efficient customization of new ones from brief audio samples as short as 10 seconds. Beyond spoken dialogue, Voila is designed as a unified model for a wide range of voice-based applications, including automatic speech recognition (ASR), Text-to-Speech (TTS), and, with minimal adaptation, multilingual speech translation. Voila is fully open-sourced to support open research and accelerate progress toward next-generation human-machine interactions.
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Works this paper leans on
-
[1]
A. Babu, C. Wang, A. Tjandra, K. Lakhotia, Q. Xu, N. Goyal, K. Singh, P. Von Platen, Y. Saraf, J. Pino, et al. Xls-r: Self-supervised cross-lingual speech representation learning at scale. arXiv preprint arXiv:2111.09296, 2021
arXiv 2021
-
[2]
C. B \'e rub \'e , M. Ni en, R. Vinay, A. Geiger, T. Budig, A. Bhandari, C. R. P. Benito, N. Ibarcena, O. Pistolese, P. Li, et al. Proactive behavior in voice assistants: A systematic review and conceptual model. Computers in Human Behavior Reports, page 100411, 2024
work page 2024
-
[3]
S. Bora. Breaking the silence: How voice AI is shaping the future of human-machine interactions, 2024. Medium, accessed November 8, 2024
work page 2024
-
[4]
Borsos, R
Z. Borsos, R. Marinier, D. Vincent, E. Kharitonov, O. Pietquin, M. Sharifi, D. Roblek, O. Teboul, D. Grangier, M. Tagliasacchi, et al. Audiolm: a language modeling approach to audio generation. IEEE/ACM transactions on audio, speech, and language processing, 31: 0 2523--2533, 2023
2023
-
[5]
S. Buss. Autonomous action: Self-determination in the passive mode. Ethics, 122 0 (4): 0 647--691, 2012
work page 2012
-
[6]
S. Buyukgoz, J. Grosinger, M. Chetouani, and A. Saffiotti. Two ways to make your robot proactive: Reasoning about human intentions or reasoning about possible futures. Frontiers in Robotics and AI, 9: 0 929267, 2022
work page 2022
-
[7]
E. Casanova, J. Weber, C. D. Shulby, A. C. Junior, E. G \"o lge, and M. A. Ponti. Yourtts: Towards zero-shot multi-speaker tts and zero-shot voice conversion for everyone. In International Conference on Machine Learning, pages 2709--2720. PMLR, 2022
work page 2022
-
[8]
M. Chen, J. Tworek, H. Jun, Q. Yuan, H. P. de Oliveira Pinto, J. Kaplan, H. Edwards, Y. Burda, N. Joseph, G. Brockman, A. Ray, R. Puri, G. Krueger, M. Petrov, H. Khlaaf, G. Sastry, P. Mishkin, B. Chan, S. Gray, N. Ryder, M. Pavlov, A. Power, L. Kaiser, M. Bavarian, C. Winter, P. Tillet, F. P. Such, D. Cummings, M. Plappert, F. Chantzis, E. Barnes, A. Herb...
work page 2021
Show all 69 references
-
[9]
Y. Chu, J. Xu, X. Zhou, Q. Yang, S. Zhang, Z. Yan, C. Zhou, and J. Zhou. Qwen-audio: Advancing universal audio understanding via unified large-scale audio-language models. arXiv preprint arXiv:2311.07919, 2023
2023 arXiv
-
[10]
Y. Chu, J. Xu, Q. Yang, H. Wei, X. Wei, Z. Guo, Y. Leng, Y. Lv, J. He, J. Lin, C. Zhou, and J. Zhou. Qwen2-audio technical report. arXiv preprint arXiv:2407.10759, 2024
2024 arXiv
-
[11]
Cobbe, V
K. Cobbe, V. Kosaraju, M. Bavarian, M. Chen, H. Jun, L. Kaiser, M. Plappert, J. Tworek, J. Hilton, R. Nakano, C. Hesse, and J. Schulman. Training verifiers to solve math word problems. arXiv preprint arXiv:2110.14168, 2021
2021 arXiv
-
[12]
D \'e fossez, L
A. D \'e fossez, L. Mazar \'e , M. Orsini, A. Royer, P. P \'e rez, H. J \'e gou, E. Grave, and N. Zeghidour. Moshi: a speech-text foundation model for real-time dialogue. arXiv preprint arXiv:2410.00037, 2024
2024 arXiv
-
[13]
Défossez, J
A. Défossez, J. Copet, G. Synnaeve, and Y. Adi. High fidelity neural audio compression, 2022. URL https://arxiv.org/abs/2210.13438
2022 arXiv
-
[14]
Faruqui and D
M. Faruqui and D. Hakkani-T \"u r. Revisiting the boundary between asr and nlu in the age of conversational dialog systems. Computational Linguistics, 48 0 (1): 0 221--232, 2022
2022
-
[15]
J. L. Flanagan. Speech analysis synthesis and perception. Springer Berlin, Heidelberg, 1972
1972
-
[16]
Grosinger
J. Grosinger. On proactive human-ai systems. In AIC, pages 140--146, 2022
2022
-
[17]
S. Hao, Y. Gu, H. Ma, J. Hong, Z. Wang, D. Wang, and Z. Hu. Reasoning with language model is planning with world model. In Proceedings of the 2023 Conference on Empirical Methods in Natural Language Processing, pages 8154--8173, 2023 a
2023
-
[18]
S. Hao, T. Liu, Z. Wang, and Z. Hu. ToolkenGPT : Augmenting frozen language models with massive tools via tool embeddings. arXiv preprint arXiv:2305.11554, 2023 b
2023 arXiv
-
[19]
Hassid, T
M. Hassid, T. Remez, T. A. Nguyen, I. Gat, A. Conneau, F. Kreuk, J. Copet, A. Defossez, G. Synnaeve, E. Dupoux, et al. Textually pretrained speech language models. Advances in Neural Information Processing Systems, 36, 2024
2024
-
[20]
Hendrycks, C
D. Hendrycks, C. Burns, S. Basart, A. Zou, M. Mazeika, D. Song, and J. Steinhardt. Measuring massive multitask language understanding. Proceedings of the International Conference on Learning Representations (ICLR), 2021 a
2021
-
[21]
Hendrycks, C
D. Hendrycks, C. Burns, S. Kadavath, A. Arora, S. Basart, E. Tang, D. Song, and J. Steinhardt. Measuring mathematical problem solving with the math dataset. arXiv preprint arXiv:2103.03874, 2021 b
2021 arXiv
-
[22]
N. Hoke. Digital assistants vs digital companions: What's the difference?, 2021. URL https://blog.intuitionrobotics.com/digital-assistants-vs-digital-companions-whats-the-difference. Blog post
2021
-
[23]
W.-N. Hsu, B. Bolte, Y.-H. H. Tsai, K. Lakhotia, R. Salakhutdinov, and A. Mohamed. Hubert: Self-supervised speech representation learning by masked prediction of hidden units, 2021. URL https://arxiv.org/abs/2106.07447
2021 arXiv
-
[24]
Huang, B
A. Huang, B. Wu, B. Wang, C. Yan, C. Hu, C. Feng, F. Tian, F. Shen, J. Li, M. Chen, P. Liu, R. Miao, W. You, X. Chen, X. Yang, Y. Huang, Y. Zhang, Z. Gong, Z. Zhang, H. Zhou, J. Sun, B. Li, C. Feng, C. Wan, H. Hu, J. Wu, J. Zhen, R. Ming, S. Yuan, X. Zhang, Y. Zhou, B. Li, B. ...
2025 arXiv
-
[25]
Huang, M
R. Huang, M. Li, D. Yang, J. Shi, X. Chang, Z. Ye, Y. Wu, Z. Hong, J. Huang, J. Liu, Y. Ren, Z. Zhao, and S. Watanabe. Audiogpt: Understanding and generating speech, music, sound, and talking head, 2023. URL https://arxiv.org/abs/2304.12995
2023 arXiv
-
[26]
Hurst, A
A. Hurst, A. Lerer, A. P. Goucher, A. Perelman, A. Ramesh, A. Clark, A. Ostrow, A. Welihinda, A. Hayes, A. Radford, et al. Gpt-4o system card. arXiv preprint arXiv:2410.21276, 2024
2024 arXiv
-
[27]
H. Kim, S. Seo, K. Jeong, O. Kwon, S. Kim, J. Kim, J. Lee, E. Song, M. Oh, J.-W. Ha, S. Yoon, and K. M. Yoo. Integrating paralinguistics in speech-empowered large language models for natural conversation, 2024. URL https://arxiv.org/abs/2402.05706
2024 arXiv
-
[28]
Kumar, P
R. Kumar, P. Seetharaman, A. Luebs, I. Kumar, and K. Kumar. High-fidelity audio compression with improved rvqgan. Advances in Neural Information Processing Systems, 36, 2024
2024
-
[29]
Kwiatkowski, J
T. Kwiatkowski, J. Palomaki, O. Redfield, M. Collins, A. Parikh, C. Alberti, D. Epstein, I. Polosukhin, J. Devlin, K. Lee, K. Toutanova, L. Jones, M. Kelcey, M.-W. Chang, A. M. Dai, J. Uszkoreit, Q. Le, and S. Petrov. Natural questions: A benchmark for question answering resea...
2019 doi
-
[30]
Lakhotia, E
K. Lakhotia, E. Kharitonov, W.-N. Hsu, Y. Adi, A. Polyak, B. Bolte, T.-A. Nguyen, J. Copet, A. Baevski, A. Mohamed, and E. Dupoux. Generative spoken language modeling from raw audio, 2021. URL https://arxiv.org/abs/2102.01192
2021 arXiv
-
[31]
Lebourdais, M
M. Lebourdais, M. Tahon, A. Laurent, and S. Meignier. Automatic speech interruption detection: Analysis, corpus, and system. In Joint International Conference on Computational Linguistics, Language Resources and Evaluation (LREC-Coling 2024), pages \`a --para \^ tre, 2024
2024
-
[32]
Y. Li, H. Sun, M. Lin, T. Li, G. Dong, T. Zhang, B. Ding, W. Song, Z. Cheng, Y. Huo, S. Chen, X. Li, D. Pan, S. Zhang, X. Wu, Z. Liang, J. Liu, T. Zhang, K. Lu, Y. Zhao, Y. Shen, F. Yang, K. Yu, T. Lin, J. Xu, Z. Zhou, and W. Chen. Baichuan-omni technical report. arXiv preprin...
-
[33]
T.-E. Lin, Y. Wu, F. Huang, L. Si, J. Sun, and Y. Li. Duplex conversation: Towards human-like interaction in spoken dialogue systems. In Proceedings of the 28th ACM SIGKDD Conference on Knowledge Discovery and Data Mining, pages 3299--3308, 2022
2022
-
[34]
A. H. Liu, H.-J. Chang, M. Auli, W.-N. Hsu, and J. Glass. Dinosr: Self-distillation and online clustering for self-supervised speech representation learning. Advances in Neural Information Processing Systems, 36, 2024
2024
-
[35]
Maiti, Y
S. Maiti, Y. Peng, S. Choi, J.-w. Jung, X. Chang, and S. Watanabe. Voxtlm: Unified decoder-only models for consolidating speech recognition, synthesis and speech, text continuation tasks. In ICASSP 2024-2024 IEEE International Conference on Acoustics, Speech and Signal Process...
2024
-
[36]
Maiti, Y
S. Maiti, Y. Peng, S. Choi, J. weon Jung, X. Chang, and S. Watanabe. Voxtlm: unified decoder-only models for consolidating speech recognition/synthesis and speech/text continuation tasks, 2024 b . URL https://arxiv.org/abs/2309.07937
2024 arXiv
-
[37]
A. S. Meyer. Timing in conversation. Journal of Cognition, 6 0 (1), 2023
2023
-
[38]
Mitsui, K
K. Mitsui, K. Mitsuda, T. Wakatsuki, Y. Hono, and K. Sawada. Pslm: Parallel generation of text and speech with llms for low-latency spoken dialogue systems. arXiv preprint arXiv:2406.12428, 2024
2024 arXiv
-
[39]
Nachmani, A
E. Nachmani, A. Levkovitch, R. Hirsch, J. Salazar, C. Asawaroengchai, S. Mariooryad, E. Rivlin, R. Skerry-Ryan, and M. T. Ramanovich. Spoken question answering and speech continuation using spectrogram-powered llm. arXiv preprint arXiv:2305.15255, 2023
2023 arXiv
-
[40]
T. A. Nguyen, B. Muller, B. Yu, M. R. Costa-Jussa, M. Elbayad, S. Popuri, P.-A. Duquenne, R. Algayres, R. Mavlyutov, I. Gat, et al. Spirit-lm: Interleaved spoken and written language model. arXiv preprint arXiv:2402.05755, 2024
2024 arXiv
-
[41]
Ouyang, J
L. Ouyang, J. Wu, X. Jiang, D. Almeida, C. Wainwright, P. Mishkin, C. Zhang, S. Agarwal, K. Slama, A. Ray, et al. Training language models to follow instructions with human feedback. Advances in neural information processing systems, 35: 0 27730--27744, 2022
2022
-
[42]
Panayotov, G
V. Panayotov, G. Chen, D. Povey, and S. 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. IEEE, 2015
2015
-
[43]
Polyak, Y
A. Polyak, Y. Adi, J. Copet, E. Kharitonov, K. Lakhotia, W.-N. Hsu, A. Mohamed, and E. Dupoux. Speech resynthesis from discrete disentangled self-supervised representations, 2021. URL https://arxiv.org/abs/2104.00355
2021 arXiv
-
[44]
Radford, J
A. Radford, J. W. Kim, T. Xu, G. Brockman, C. McLeavey, and I. Sutskever. Robust speech recognition via large-scale weak supervision, 2022. URL https://arxiv.org/abs/2212.04356
2022 arXiv
-
[45]
Rekesh, N
D. Rekesh, N. R. Koluguri, S. Kriman, S. Majumdar, V. Noroozi, H. Huang, O. Hrinchuk, K. Puvvada, A. Kumar, J. Balam, et al. Fast conformer with linearly scalable attention for efficient speech recognition. In 2023 IEEE Automatic Speech Recognition and Understanding Workshop (...
2023
-
[46]
P. K. Rubenstein, C. Asawaroengchai, D. D. Nguyen, A. Bapna, Z. Borsos, F. de Chaumont Quitry, P. Chen, D. E. Badawy, W. Han, E. Kharitonov, H. Muckenhirn, D. Padfield, J. Qin, D. Rozenberg, T. Sainath, J. Schalkwyk, M. Sharifi, M. T. Ramanovich, M. Tagliasacchi, A. Tudor, M. ...
2023 arXiv
-
[47]
R. Schafer. Scientific bases of human-machine communication by voice. Proceedings of the National Academy of Sciences of the United States of America, 92 0 (22): 0 9914--9920, 1995
1995
-
[48]
Schneider, A
S. Schneider, A. Baevski, R. Collobert, and M. Auli. wav2vec: Unsupervised pre-training for speech recognition. arXiv preprint arXiv:1904.05862, 2019
1904 arXiv
-
[49]
Schroeder and N
J. Schroeder and N. Epley. Mistaking minds and machines: How speech affects dehumanization and anthropomorphism. Journal of Experimental Psychology: General, 145 0 (11): 0 1427, 2016
2016
-
[50]
Y. Shao, L. Li, J. Dai, and X. Qiu. Character-llm: A trainable agent for role-playing. arXiv preprint arXiv:2310.10158, 2023
2023 arXiv
-
[51]
Y. Shen, K. Song, X. Tan, D. Li, W. Lu, and Y. Zhuang. Hugginggpt: Solving ai tasks with chatgpt and its friends in hugging face, 2023. URL https://arxiv.org/abs/2303.17580
2023 arXiv
-
[52]
Y. Shu, S. Dong, G. Chen, W. Huang, R. Zhang, D. Shi, Q. Xiang, and Y. Shi. Llasm: Large language and speech model, 2023
2023
-
[53]
G. Skantze. Turn-taking in conversational systems and human-robot interaction: a review. Computer Speech & Language, 67: 0 101178, 2021
2021
-
[54]
Introducing hertz-dev, the first open-source base model for conversational audio generation, 2024
Standard Intelligence . Introducing hertz-dev, the first open-source base model for conversational audio generation, 2024. URL https://si.inc/hertz-dev/. Blog post
2024
-
[55]
Stivers, N
T. Stivers, N. J. Enfield, P. Brown, C. Englert, M. Hayashi, T. Heinemann, G. Hoymann, F. Rossano, J. P. De Ruiter, K.-E. Yoon, et al. Universals and cultural variation in turn-taking in conversation. Proceedings of the National Academy of Sciences, 106 0 (26): 0 10587--10592, 2009
2009
-
[56]
W. Tan, H. Inaguma, N. Dong, P. Tomasello, and X. Ma. Ssr: Alignment-aware modality connector for speech language models. arXiv preprint arXiv:2410.00168, 2024
2024 arXiv
-
[57]
C. Tang, W. Yu, G. Sun, X. Chen, T. Tan, W. Li, L. Lu, Z. Ma, and C. Zhang. Salmonn: Towards generic hearing abilities for large language models, 2024. URL https://arxiv.org/abs/2310.13289
2024 arXiv
-
[58]
C. Wang, S. Chen, Y. Wu, Z. Zhang, L. Zhou, S. Liu, Z. Chen, Y. Liu, H. Wang, J. Li, et al. Neural codec language models are zero-shot text to speech synthesizers. arXiv preprint arXiv:2301.02111, 2023 a
2023 arXiv
-
[59]
H. Wang, C. Liang, S. Wang, Z. Chen, B. Zhang, X. Xiang, Y. Deng, and Y. Qian. Wespeaker: A research and production oriented speaker embedding learning toolkit. In ICASSP 2023, IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 1--5. IEEE, 2023 b
2023
-
[60]
Z. M. Wang, Z. Peng, H. Que, J. Liu, W. Zhou, Y. Wu, H. Guo, R. Gan, Z. Ni, J. Yang, et al. Rolellm: Benchmarking, eliciting, and enhancing role-playing abilities of large language models. arXiv preprint arXiv:2310.00746, 2023 c
-
[61]
J. Wei, X. Wang, D. Schuurmans, M. Bosma, F. Xia, E. Chi, Q. V. Le, D. Zhou, et al. Chain-of-thought prompting elicits reasoning in large language models. Advances in neural information processing systems, 35: 0 24824--24837, 2022
2022
-
[62]
W. Wu, H. Wu, L. Jiang, X. Liu, J. Hong, H. Zhao, and M. Zhang. From role-play to drama-interaction: An llm solution. arXiv preprint arXiv:2405.14231, 2024
2024 arXiv
-
[63]
D. Yang, J. Tian, X. Tan, R. Huang, S. Liu, X. Chang, J. Shi, S. Zhao, J. Bian, X. Wu, Z. Zhao, S. Watanabe, and H. Meng. Uniaudio: An audio foundation model toward universal audio generation, 2023. URL https://arxiv.org/abs/2310.00704
2023 arXiv
-
[64]
L. Yang, C. Achard, and C. Pelachaud. Multimodal analysis of interruptions. In International Conference on Human-Computer Interaction, pages 306--325. Springer, 2022
2022
-
[65]
S. Yao, J. Zhao, D. Yu, N. Du, I. Shafran, K. Narasimhan, and Y. Cao. React: Synergizing reasoning and acting in language models. arXiv preprint arXiv:2210.03629, 2022
2022 arXiv
-
[66]
Zeghidour, A
N. Zeghidour, A. Luebs, A. Omran, J. Skoglund, and M. Tagliasacchi. Soundstream: An end-to-end neural audio codec. IEEE/ACM Transactions on Audio, Speech, and Language Processing, 30: 0 495--507, 2021
2021
-
[67]
Zhang, S
D. Zhang, S. Li, X. Zhang, J. Zhan, P. Wang, Y. Zhou, and X. Qiu. Speechgpt: Empowering large language models with intrinsic cross-modal conversational abilities. arXiv preprint arXiv:2305.11000, 2023 a
2023 arXiv
-
[68]
Zhang, D
X. Zhang, D. Zhang, S. Li, Y. Zhou, and X. Qiu. Speechtokenizer: Unified speech tokenizer for speech large language models. arXiv preprint arXiv:2308.16692, 2023 b
2023 arXiv
-
[69]
Y. Zhu, D. Su, L. He, L. Xu, and D. Yu. Generative pre-trained speech language model with efficient hierarchical transformer, 2024. URL https://arxiv.org/abs/2406.00976
2024 arXiv
Reviewed August 16, 2026 · model on record in the stance chip above.
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