REVIEW 4 major objections 6 minor 16 cited by
FireRedChat claims a pluggable turn-taking controller makes full-duplex voice interaction robust—fewer false interruptions, accurate end-of-turn detection, and latency within 0.3 s of a commercial assistant.
Reviewed by Pith at T0; open to challenge. T0 means a machine referee read the full paper against a public rubric. the ladder, T0–T4 →
T0 review · deepseek-v4-flash
2026-08-04 23:28 UTC pith:5NZERKDH
load-bearing objection Solid engineering contribution to full-duplex voice interaction, held back by missing pVAD enrollment details and thin evaluation. the 4 major comments →
FireRedChat: A Pluggable, Full-Duplex Voice Interaction System with Cascaded and Semi-Cascaded Implementations
The pith
A machine-rendered reading of the paper's core claim, the machinery that carries it, and where it could break.
Core claim
At the core is the turn-taking controller: a streaming personalized VAD (pVAD) that conditions on a target-speaker embedding from an ECAPA-TDNN encoder, processes 10ms chunks causally, and outputs a speaking probability for the primary speaker only, suppressing interference from noise and other voices. Rather than feeding denoised audio downstream, the system uses pVAD timestamps to slice the original unprocessed audio, preserving acoustic cues for the response model. A fine-tuned BERT classifier serves as the end-of-turn detector, deciding from all accumulated ASR transcripts whether the user has reached a semantic stopping point. On top of this controller, the authors build cascaded (FireR
What carries the argument
The central object is the turn-taking controller, composed of two learned modules. The streaming personalized VAD (pVAD) is a causal convolutional-recurrent network conditioned on a target-speaker embedding, producing a 10ms-resolution speaking probability for the primary speaker; it is what suppresses false barge-ins and enables precise interruption. The end-of-turn detector is a fine-tuned BERT classifier that reads the accumulated transcript and predicts 'continue' or 'stop'—the semantic decision that prevents premature responses. Around these, the semi-cascaded architecture pairs an AudioLLM (which hears speech directly rather than through ASR text) with a dialogue-conditioned TTS, letti
Load-bearing premise
The system assumes a target-speaker embedding is available for the primary user at inference time; the paper does not say how that embedding is enrolled or refreshed in a live session, and the entire barge-in suppression benefit rests on this unstated step.
What would settle it
Run the pVAD in a live two-speaker conversation with an enrolled primary user and measure the false barge-in rate when a second speaker, of the same gender and similar voice, talks at the same loudness; if false barge-ins increase to the level of an unpersonalized VAD, the claimed suppression depends on a degree of speaker discrimination the model may not achieve. A simpler check: measure barge-in success for the primary speaker with and without the target-speaker embedding—if the gap is small, the pVAD's advantage over a plain VAD is not actually due to personalization.
If this is right
- Any half-duplex pipeline—cascaded, semi-cascaded, or end-to-end speech-to-speech—can be upgraded to full duplex by inserting the turn-taking controller, without retraining the pipeline itself.
- The pVAD's timestamps preserve the original audio for downstream models, so paralinguistic information in the user's voice survives barge-in handling.
- The semi-cascaded configuration replaces ASR+LLM with a single AudioLLM, reducing error propagation and simplifying the chain while keeping TTS separately optimizable.
- The three proposed metrics (barge-in, EoT accuracy, end-to-end latency) give voice-interaction systems a common yardstick for naturalness, control accuracy, and efficiency.
- At the reported T90 of 170 ms with a false barge-in rate of 10.2%, the system shows that suppressing spurious interruptions costs only about 30 ms of extra response delay compared with a more trigger-happy baseline.
Where Pith is reading between the lines
- The paper's implicit promise is that pVAD's speaker conditioning can be deployed in any multi-speaker setting; a natural extension would be to test whether enrollment from a short sample (e.g., a user's first sentence) keeps false barge-in low over a long session as the background and the user's voice drift.
- Because the EoT detector works on transcripts, it should transfer to any ASR output; one testable extension is whether combining acoustic pause features with the semantic classifier further reduces 'continue' errors on incomplete but paused utterances.
- The same turn-taking controller could be applied to an end-to-end speech-to-speech model, which the paper mentions as possible but does not implement; the metrics proposed here would allow a direct comparison of controllability versus end-to-end latency.
- The 0.3 s latency gap to the commercial system is attributed to missing streaming ASR, streaming TTS input, and chunk size; these are engineering knobs, suggesting the gap is closable rather than architectural.
Editorial analysis
A structured set of objections, weighed in public.
Referee Report
Summary. The paper presents FireRedChat, a modular full-duplex voice interaction system centered on a turn-taking controller that combines a streaming personalized voice activity detector (pVAD) with a semantic end-of-turn (EoT) detector. The controller is designed to upgrade half-duplex pipelines (cascaded, semi-cascaded, or speech-to-speech) to full duplex. The authors instantiate cascaded (FireRedASR, Qwen2.5, FireRedTTS-1s) and semi-cascaded (AudioLLM, FireRedTTS-2) variants, and propose three system-level metrics: barge-in, EoT detection accuracy, and end-to-end latency. Experiments against LiveKit, Ten, and DouBao report lower false barge-in rates, competitive EoT accuracy, and latency approaching an industrial system. The central claim is that pVAD and the semantic EoT detector enable more robust, natural, real-time full-duplex interaction than existing open frameworks.
Significance. If the claims hold, the paper makes a useful practical contribution: a pluggable, internally consistent full-duplex stack with open-sourced ASR/TTS components, a concrete set of system-level evaluation metrics, and a design that allows heterogeneous half-duplex pipelines to be upgraded to full duplex. The proposed pVAD idea is appealing because it addresses a real problem—false barge-ins from noise and non-primary speakers—and the EoT detector is a sensible semantic complement to acoustic VAD. The paper also reports reproducible artifacts (demos, FireRedASR, FireRedTTS) and is honest about standardizing the LLM across frameworks. However, several load-bearing experimental and methodological gaps currently prevent the strong conclusions from being fully supported: the enrollment mechanism for the personalized VAD is unspecified, the semi-cascaded advantages are not evaluated at all, and the quantitative results lack variance estimates and are based on small synthetic test sets.
major comments (4)
- [§2.2.1, §3.1] The pVAD is conditioned on a target-speaker ECAPA-TDNN embedding, but the manuscript never states how this embedding is obtained at inference time. A live full-duplex session must know which speaker is primary before suppressing noise/non-primary speakers; no enrollment procedure, adaptation, or fallback is given. Section 3.1 also does not state how the embedding was generated for the 1,000 test utterances per language. If the embedding is extracted from the same utterance that is later tested (an oracle condition), the reported 10.2% false barge-in rate and T90=170 ms are not reproducible in practice. Since pVAD is the main differentiator in the barge-in comparison, this is load-bearing for the headline claim of fewer false interruptions. Please specify the enrollment protocol (e.g., short enrollment utterance, speaker diarization, or voiceprint database) and report barge-in results und
- [§2.3.2, §3] Several central claims about the semi-cascaded implementation are not evaluated. The abstract and §2.3.2 state that the semi-cascaded variant 'captures emotional and paralinguistic cues, yields more coherent responses, lowers latency and error propagation, and improves robustness.' However, Table 1 and all experiments use the cascaded configuration (FireRedASR+Qwen2.5+FireRedTTS-1s); no experiment compares cascaded vs. semi-cascaded on paralinguistic awareness, response coherence, error propagation, or latency. The latency results in Table 4 cannot be attributed to the semi-cascaded design. Either add a head-to-head evaluation of the two implementations (e.g., emotion/paralinguistic metrics, human coherence ratings, ASR-error injection, per-stage latency), or restrict the claims to the cascaded system and present the semi-cascaded advantages as design hypotheses.
- [§3.1, §3.3] The quantitative evidence consists of single point estimates without variance. Barge-in T90 and false barge-in rates in Table 2, EoT accuracies in Table 3, and P50/P95 latencies in Table 4 are each reported once with no confidence intervals, number of repeated trials, or significance tests. Latency is based on 25 synthetic utterances only (Table 4) and measured via screen recording of a phone, which introduces uncontrolled jitter. With n=25 and no repeated runs, a 0.3 s advantage over DouBao is not established. Please report means/standard deviations or confidence intervals over multiple runs and, where possible, larger and more diverse test sets, including real conversational data.
- [§3.3] The latency comparison is not apples-to-apples. Table 1 states that the LLM was standardized to Qwen2.5 for LiveKit, Ten, and FireRedChat, but DouBao is a closed commercial system accessed via mobile app; its ASR, LLM, TTS, and turn-taking stack are unknown. Differences could come from model size, streaming capability, network, or device rather than the compared architectures. The claim that FireRedChat 'narrows the interaction-latency gap with industrial-grade systems' is therefore only weakly supported. I recommend either limiting the comparison with DouBao to an anecdotal reference, or adding a decomposition of latency (VAD, ASR, LLM, TTS, network) and a comparable open industrial baseline.
minor comments (6)
- [§1] Typo: 'two models in paralle' should be 'parallel'. Also, the text repeatedly renders VAD with a spurious space ('V AD', 'pV AD'); please fix throughout.
- [§3.1] The sentence 'we constructed 1,000 utterances each in Chinese and English' is ambiguous: does this mean 1,000 per language or 1,000 total split across languages? Clarify, and state whether the barge-in metrics are averaged over both languages or reported separately.
- [Table 1] The table is labeled 'Configurations between different systems' but only lists component names. To be reproducible, please include model versions, checkpoints, and any hyperparameters that affect latency or accuracy (e.g., chunk sizes, streaming settings).
- [§2.2.1] The pVAD description gives high-level architecture (mel features, causal conv, ECAPA-TDNN embedding, GRU) but no model size, input frame size, GRU hidden units, or training schedule. Provide enough detail to reimplement or, if the model is proprietary, offer a reference implementation or pseudocode.
- [§3.3] The 25 'representative synthetic utterances' are not described in terms of selection criteria, duration distribution, or emotional/pause coverage. A supplementary table listing the utterances or a description of the sampling procedure would make the latency experiment more interpretable.
- [§3.2, References] Reference [40] (TEN_Turn_Detection) lacks a URL, version, or access details. If the test set is publicly available, provide a link; if not, describe how it can be obtained for reproducibility.
Circularity Check
No significant circularity: system claims rest on direct measurement against external frameworks and benchmarks; self-citations are component references only.
full rationale
The paper's central claims are empirical system-performance claims: fewer false interruptions, more accurate semantic end-of-turn detection, and lower latency. These are supported by direct measurements in Tables 2-4 against external systems (LiveKit, Ten, DouBao) and the external TEN_Turn_Detection benchmark [40]. The pVAD and EoT modules are trained on purpose-built data and evaluated on held-out synthetic mixtures and an external test set; no fitted parameter is renamed as a prediction, and no equation in the paper reduces one claimed quantity to another by construction. Self-citations (FireRedASR [2], FireRedTTS-1s [10], FireRedTTS-2 [14]) are used as component references in the implemented pipelines; they do not carry the argument for the headline conclusions, which are measured de novo. The main weakness is an unspecified enrollment procedure for the target-speaker embedding in pVAD (Section 2.2.1), which could affect real-world reproducibility of the barge-in results; this is a correctness/reproducibility concern, not circularity, because the barge-in metric is a behavioral measurement rather than a quantity derived from the embedding. No load-bearing uniqueness theorem, ansatz-by-citation, or self-definitional reduction is present.
Axiom & Free-Parameter Ledger
free parameters (4)
- pVAD mixture SNR range =
0-30 dB
- Interference type probability =
50%
- T90 threshold =
90%
- EoT training corpus size =
830k
axioms (4)
- domain assumption The target-speaker embedding is available at inference for pVAD without describing an enrollment procedure.
- domain assumption FireRedASR is robust enough to noise and overlapping speech that using original audio (not pVAD-denoised) is safe.
- domain assumption Partial spans sampled from complete utterances are a valid proxy for unfinished turns in EoT training.
- domain assumption Screen-recording with fixed phone position and volume yields a fair latency comparison across systems.
Cite this review
Pith. "Pith review of FireRedChat: A Pluggable, Full-Duplex Voice Interaction System with Cascaded and Semi-Cascaded Implementations." pith.science (2026). https://pith.science/paper/5NZERKDH
@misc{pith2026250906502,
author = {Pith},
title = {Pith review of: FireRedChat: A Pluggable, Full-Duplex Voice Interaction System with Cascaded and Semi-Cascaded Implementations},
year = {2026},
howpublished = {\url{https://pith.science/paper/5NZERKDH}},
note = {Machine review of arXiv:2509.06502}
}
read the original abstract
Full-duplex voice interaction allows users and agents to speak simultaneously with controllable barge-in, enabling lifelike assistants and customer service. Existing solutions are either end-to-end, difficult to design and hard to control, or modular pipelines governed by turn-taking controllers that ease upgrades and per-module optimization; however, prior modular frameworks depend on non-open components and external providers, limiting holistic optimization. In this work, we present a complete, practical full-duplex voice interaction system comprising a turn-taking controller, an interaction module, and a dialogue manager. The controller integrates streaming personalized VAD (pVAD) to suppress false barge-ins from noise and non-primary speakers, precisely timestamp primary-speaker segments, and explicitly enable primary-speaker barge-ins; a semantic end-of-turn detector improves stop decisions. It upgrades heterogeneous half-duplex pipelines, cascaded, semi-cascaded, and speech-to-speech, to full duplex. Using internal models, we implement cascaded and semi-cascaded variants; the semi-cascaded one captures emotional and paralinguistic cues, yields more coherent responses, lowers latency and error propagation, and improves robustness. A dialogue manager extends capabilities via tool invocation and context management. We also propose three system-level metrics, barge-in, end-of-turn detection accuracy, and end-to-end latency, to assess naturalness, control accuracy, and efficiency. Experiments show fewer false interruptions, more accurate semantic ends, and lower latency approaching industrial systems, enabling robust, natural, real-time full-duplex interaction. Demos: https://fireredteam.github.io/demos/firered_chat.
Figures
Forward citations
Cited by 16 Pith papers
-
A Survey of Full-Duplex Spoken Dialogue Systems: Architectural Hierarchy, Interaction Ontology, and Decision State Machine
A survey proposing an L0-L3 architectural hierarchy, T×I×R interaction ontology, and IDLE/LISTEN/SPEAK/WAIT/DUAL decision state machine for full-duplex spoken dialogue systems, documenting a realization gap between ar...
-
PolySpeech-100: A Large-Scale Benchmark for Speech Understanding Across 100+ Languages and Dialects
PolySpeech-100 is a new benchmark for native-level speech comprehension across 110 linguistic variants that evaluates 22 models and reports E2E advantages on dialects, robustness gaps on low-resource languages, and de...
-
Character Beyond Speech: Leveraging Role-Playing Evaluation in Audio Large Language Models via Reinforcement Learning
RoleJudge is a multidimensional evaluation framework for speech-character alignment in audio LLMs, backed by the RoleChat dataset and multi-stage RL training with standard alignment to reduce reward issues.
-
Hierarchical Acoustic-Semantic Modeling: Modality Separation and Semantic Coherence for Full-Duplex SLMs
Lychee-FD resolves modality interference in full-duplex spoken language models by separating acoustic and semantic parameters in deep layers and adding a dense semantic alignment channel, achieving state-of-the-art pe...
-
Next-Turn: Duration-Aware Streaming Endpoint Detection via Time-to-Next-Speech-Onset Prediction
Next-Turn introduces time-to-next-speech-onset prediction for duration-aware streaming endpoint detection, reporting a 25.9% improvement in accuracy within 320 ms.
-
UAF: A Unified Audio Front-end LLM for Full-Duplex Speech Interaction
UAF is the first unified audio front-end LLM that turns multiple front-end tasks into one sequence prediction model processing streaming audio chunks and reference prompts to output semantic and control tokens for ful...
-
FastTurn: Unifying Acoustic and Streaming Semantic Cues for Low-Latency and Robust Turn Detection
FastTurn unifies acoustic features and streaming CTC decoding for low-latency, robust turn detection in full-duplex dialogue systems and releases a realistic human-dialogue test set.
-
X$^3$-OPD: Distilling Reasoning into Large Audio-Language Models via On-Policy Alignment
X3-OPD improves audio-grounded reasoning by training the audio student on its own rollouts with token-level teacher feedback, using a three-tier paired text-audio corpus.
-
DuplexOmni: Real-Time Listening, Seeing, Thinking, and Speaking for Full-Duplex Interaction
DuplexOmni achieves real-time full-duplex multimodal interaction by separating an interaction layer from a pluggable thinking layer, supported by a Writer-Director pipeline for continuous-interaction training data.
-
A Survey of Large Audio Language Models: Generalization, Trustworthiness, and Outlook
A survey of Large Audio Language Models that establishes a taxonomy of trustworthiness vulnerabilities and proposes a Defense-in-Depth roadmap for audio intelligence.
-
Full-Duplex Interaction in Spoken Dialogue Systems: A Comprehensive Study from the ICASSP 2026 HumDial Challenge
A new HumDial-FDBench benchmark and real human-recorded dual-channel dataset are released to assess full-duplex dialogue systems on interruptions and conversational flow.
-
FastTurn: Unifying Acoustic and Streaming Semantic Cues for Low-Latency and Robust Turn Detection
FastTurn fuses streaming CTC semantics with acoustic cues for lower-latency, more accurate turn detection in full-duplex dialogue and releases a real-dialogue test set.
-
FastTurn: Unifying Acoustic and Streaming Semantic Cues for Low-Latency and Robust Turn Detection
FastTurn fuses streaming CTC semantics with acoustic cues for lower-latency, more accurate turn detection and ships a real-dialogue test set.
-
Sharp spectral estimates for free boundary problems arising in plasma physics
For a constrained superlinear free-boundary plasma model, the non-local first eigenvalue σ₁ is always positive on balls in every dimension N≥2, despite lacking a general Faber–Krahn property.
-
IRAF: Interference-Resilient Adaptive Fusion for Noise-Robust End-to-End Full-Duplex Spoken Dialogue Systems
IRAF introduces an adaptive fusion module that uses a predicted scalar reliability gate to reduce the impact of interfering speakers on user audio representations in end-to-end full-duplex spoken dialogue systems, wit...
-
Toward Native Multimodal Modeling: A Roadmap
A roadmap that defines architectural nativity for multimodal models and categorizes them into Multi-to-Text, Multi-to-Target, and Multi-to-Multi types while outlining an industrial pipeline toward unified transformer-...
Reference graph
Works this paper leans on
-
[1]
Language model can listen while speaking
Ziyang Ma, Yakun Song, Chenpeng Du, Jian Cong, Zhuo Chen, Yuping Wang, Yuxuan Wang, and Xie Chen. Language model can listen while speaking. InProceedings of the AAAI Conference on Artificial Intelligence, volume 39, pages 24831–24839, 2025
2025
-
[2]
Kai-Tuo Xu, Feng-Long Xie, Xu Tang, and Yao Hu. Fireredasr: Open-source industrial-grade mandarin speech recognition models from encoder-decoder to llm integration.arXiv preprint arXiv:2501.14350, 2025
Pith/arXiv arXiv 2025
-
[3]
Robust speech recognition via large-scale weak supervision
Alec Radford, Jong Wook Kim, Tao Xu, Greg Brockman, Christine McLeavey, and Ilya Sutskever. Robust speech recognition via large-scale weak supervision. InInternational conference on machine learning, pages 28492–28518. PMLR, 2023
2023
-
[4]
Llama: Open and efficient foundation language models.arXiv preprint arXiv:2302.13971, 2023
Hugo Touvron, Thibaut Lavril, Gautier Izacard, Xavier Martinet, Marie-Anne Lachaux, Timo- thée Lacroix, Baptiste Rozière, Naman Goyal, Eric Hambro, Faisal Azhar, et al. Llama: Open and efficient foundation language models.arXiv preprint arXiv:2302.13971, 2023
Pith/arXiv arXiv 2023
-
[5]
Rohan Anil, Andrew M Dai, Orhan Firat, Melvin Johnson, Dmitry Lepikhin, Alexandre Passos, Siamak Shakeri, Emanuel Taropa, Paige Bailey, Zhifeng Chen, et al. Palm 2 technical report. arXiv preprint arXiv:2305.10403, 2023. 9
Pith/arXiv arXiv 2023
-
[6]
Aakanksha Chowdhery, Sharan Narang, Jacob Devlin, Maarten Bosma, Gaurav Mishra, Adam Roberts, Paul Barham, Hyung Won Chung, Charles Sutton, Sebastian Gehrmann, et al. Palm: Scaling language modeling with pathways.Journal of Machine Learning Research, 24(240):1– 113, 2023
work page 2023
-
[7]
Qwen2 technical report.arXiv preprint arXiv:2407.10671, 2024
Qwen Team. Qwen2 technical report.arXiv preprint arXiv:2407.10671, 2024
Pith/arXiv arXiv 2024
-
[8]
Qwen3 technical report.arXiv preprint arXiv:2505.09388, 2025
An Yang, Anfeng Li, Baosong Yang, Beichen Zhang, Binyuan Hui, Bo Zheng, Bowen Yu, Chang Gao, Chengen Huang, Chenxu Lv, et al. Qwen3 technical report.arXiv preprint arXiv:2505.09388, 2025
Pith/arXiv arXiv 2025
-
[9]
Hao-Han Guo, Yao Hu, Kun Liu, Fei-Yu Shen, Xu Tang, Yi-Chen Wu, Feng-Long Xie, Kun Xie, and Kai-Tuo Xu. Fireredtts: A foundation text-to-speech framework for industry-level generative speech applications.arXiv preprint arXiv:2409.03283, 2024
Pith/arXiv arXiv 2024
-
[10]
Hao-Han Guo, Yao Hu, Fei-Yu Shen, Xu Tang, Yi-Chen Wu, Feng-Long Xie, and Kun Xie. Fireredtts-1s: An upgraded streamable foundation text-to-speech system.arXiv preprint arXiv:2503.20499, 2025
Pith/arXiv arXiv 2025
-
[11]
Zhihao Du, Qian Chen, Shiliang Zhang, Kai Hu, Heng Lu, Yexin Yang, Hangrui Hu, Siqi Zheng, Yue Gu, Ziyang Ma, et al. Cosyvoice: A scalable multilingual zero-shot text-to-speech synthesizer based on supervised semantic tokens.arXiv preprint arXiv:2407.05407, 2024
Pith/arXiv arXiv 2024
-
[12]
Zhihao Du, Changfeng Gao, Yuxuan Wang, Fan Yu, Tianyu Zhao, Hao Wang, Xiang Lv, Hui Wang, Chongjia Ni, Xian Shi, et al. Cosyvoice 3: Towards in-the-wild speech generation via scaling-up and post-training.arXiv preprint arXiv:2505.17589, 2025
Pith/arXiv arXiv 2025
-
[13]
Zhihao Du, Yuxuan Wang, Qian Chen, Xian Shi, Xiang Lv, Tianyu Zhao, Zhifu Gao, Yexin Yang, Changfeng Gao, Hui Wang, et al. Cosyvoice 2: Scalable streaming speech synthesis with large language models.arXiv preprint arXiv:2412.10117, 2024
Pith/arXiv arXiv 2024
-
[14]
Kun Xie, Feiyu Shen, Junjie Li, Fenglong Xie, Xu Tang, and Yao Hu. Fireredtts-2: Towards long conversational speech generation for podcast and chatbot.arXiv preprint arXiv:2509.02020, 2025
Pith/arXiv arXiv 2025
-
[15]
Yunfei Chu, Jin Xu, Xiaohuan Zhou, Qian Yang, Shiliang Zhang, Zhijie Yan, Chang Zhou, and Jingren Zhou. Qwen-audio: Advancing universal audio understanding via unified large-scale audio-language models.arXiv preprint arXiv:2311.07919, 2023
Pith/arXiv arXiv 2023
-
[16]
Qwen2-audio technical report.arXiv preprint arXiv:2407.10759, 2024
Yunfei Chu, Jin Xu, Qian Yang, Haojie Wei, Xipin Wei, Zhifang Guo, Yichong Leng, Yuan- jun Lv, Jinzheng He, Junyang Lin, et al. Qwen2-audio technical report.arXiv preprint arXiv:2407.10759, 2024
Pith/arXiv arXiv 2024
-
[17]
Changli Tang, Wenyi Yu, Guangzhi Sun, Xianzhao Chen, Tian Tan, Wei Li, Lu Lu, Zejun Ma, and Chao Zhang. Salmonn: Towards generic hearing abilities for large language models.arXiv preprint arXiv:2310.13289, 2023
Pith/arXiv arXiv 2023
-
[18]
On decoder-only architecture for speech-to-text and large language model integration
Jian Wu, Yashesh Gaur, Zhuo Chen, Long Zhou, Yimeng Zhu, Tianrui Wang, Jinyu Li, Shujie Liu, Bo Ren, Linquan Liu, et al. On decoder-only architecture for speech-to-text and large language model integration. In2023 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU), pages 1–8. IEEE, 2023
work page 2023
-
[19]
Midashenglm: Efficient audio understanding with general audio captions
Horizon Team, MiLM Plus. Midashenglm: Efficient audio understanding with general audio captions. Technical report, Xiaomi Inc., 2025. Contributors: Heinrich Dinkel et al. (listed alphabetically in Appendix B)
work page 2025
-
[20]
Dong Zhang, Shimin Li, Xin Zhang, Jun Zhan, Pengyu Wang, Yaqian Zhou, and Xipeng Qiu. Speechgpt: Empowering large language models with intrinsic cross-modal conversational abilities.arXiv preprint arXiv:2305.11000, 2023
Pith/arXiv arXiv 2023
-
[21]
Zhihao Du, Jiaming Wang, Qian Chen, Yunfei Chu, Zhifu Gao, Zerui Li, Kai Hu, Xiaohuan Zhou, Jin Xu, Ziyang Ma, et al. Lauragpt: Listen, attend, understand, and regenerate audio with gpt.arXiv preprint arXiv:2310.04673, 2023. 10
Pith/arXiv arXiv 2023
-
[22]
Zhifei Xie and Changqiao Wu. Mini-omni: Language models can hear, talk while thinking in streaming.arXiv preprint arXiv:2408.16725, 2024
Pith/arXiv arXiv 2024
-
[23]
Qingkai Fang, Shoutao Guo, Yan Zhou, Zhengrui Ma, Shaolei Zhang, and Yang Feng. Llama-omni: Seamless speech interaction with large language models.arXiv preprint arXiv:2409.06666, 2024
Pith/arXiv arXiv 2024
-
[24]
Kimi-audio technical report.arXiv preprint arXiv:2504.18425, 2025
Ding Ding, Zeqian Ju, Yichong Leng, Songxiang Liu, Tong Liu, Zeyu Shang, Kai Shen, Wei Song, Xu Tan, Heyi Tang, et al. Kimi-audio technical report.arXiv preprint arXiv:2504.18425, 2025
Pith/arXiv arXiv 2025
-
[25]
Ailin Huang, Boyong Wu, Bruce Wang, Chao Yan, Chen Hu, Chengli Feng, Fei Tian, Feiyu Shen, Jingbei Li, Mingrui Chen, et al. Step-audio: Unified understanding and generation in intelligent speech interaction.arXiv preprint arXiv:2502.11946, 2025
Pith/arXiv arXiv 2025
-
[26]
Step-audio 2 technical report.arXiv preprint arXiv:2507.16632, 2025
Boyong Wu, Chao Yan, Chen Hu, Cheng Yi, Chengli Feng, Fei Tian, Feiyu Shen, Gang Yu, Haoyang Zhang, Jingbei Li, et al. Step-audio 2 technical report.arXiv preprint arXiv:2507.16632, 2025
Pith/arXiv arXiv 2025
-
[27]
Jin Xu, Zhifang Guo, Jinzheng He, Hangrui Hu, Ting He, Shuai Bai, Keqin Chen, Jialin Wang, Yang Fan, Kai Dang, et al. Qwen2. 5-omni technical report.arXiv preprint arXiv:2503.20215, 2025
Pith/arXiv arXiv 2025
-
[28]
Generative spoken dialogue language modeling.Transactions of the Association for Computational Linguistics, 11:250–266, 2023
Tu Anh Nguyen, Eugene Kharitonov, Jade Copet, Yossi Adi, Wei-Ning Hsu, Ali Elkahky, Paden Tomasello, Robin Algayres, Benoit Sagot, Abdelrahman Mohamed, et al. Generative spoken dialogue language modeling.Transactions of the Association for Computational Linguistics, 11:250–266, 2023
2023
-
[29]
Moshi: a speech-text foundation model for real-time dialogue.arXiv preprint arXiv:2410.00037, 2024
Alexandre Défossez, Laurent Mazaré, Manu Orsini, Amélie Royer, Patrick Pérez, Hervé Jégou, Edouard Grave, and Neil Zeghidour. Moshi: a speech-text foundation model for real-time dialogue.arXiv preprint arXiv:2410.00037, 2024
Pith/arXiv arXiv 2024
-
[30]
Xiong Wang, Yangze Li, Chaoyou Fu, Yunhang Shen, Lei Xie, Ke Li, Xing Sun, and Long Ma. Freeze-omni: A smart and low latency speech-to-speech dialogue model with frozen llm.arXiv preprint arXiv:2411.00774, 2024
Pith/arXiv arXiv 2024
-
[31]
Qian Chen, Yafeng Chen, Yanni Chen, Mengzhe Chen, Yingda Chen, Chong Deng, Zhihao Du, Ruize Gao, Changfeng Gao, Zhifu Gao, et al. Minmo: A multimodal large language model for seamless voice interaction.arXiv preprint arXiv:2501.06282, 2025
Pith/arXiv arXiv 2025
-
[32]
TurnGPT: a Transformer-based Language Model for Predicting Turn-taking in Spoken Dialog
Erik Ekstedt and Gabriel Skantze. Turngpt: a transformer-based language model for predicting turn-taking in spoken dialog.arXiv preprint arXiv:2010.10874, 2020
work page internal anchor Pith review Pith/arXiv arXiv 2010
-
[33]
Vita: Towards open-source interactive omni multimodal llm.arXiv preprint arXiv:2408.05211, 2024
Chaoyou Fu, Haojia Lin, Zuwei Long, Yunhang Shen, Yuhang Dai, Meng Zhao, Yi-Fan Zhang, Shaoqi Dong, Yangze Li, Xiong Wang, et al. Vita: Towards open-source interactive omni multimodal llm.arXiv preprint arXiv:2408.05211, 2024
Pith/arXiv arXiv 2024
-
[34]
Flexduo: A pluggable system for enabling full-duplex capabilities in speech dialogue systems
Borui Liao, Yulong Xu, Jiao Ou, Kaiyuan Yang, Weihua Jian, Pengfei Wan, and Di Zhang. Flexduo: A pluggable system for enabling full-duplex capabilities in speech dialogue systems. arXiv preprint arXiv:2502.13472, 2025
Pith/arXiv arXiv 2025
-
[35]
Two-dimensional convo- lutional recurrent neural networks for speech activity detection
Anastasios Vafeiadis, Eleftherios Fanioudakis, Ilyas Potamitis, Konstantinos V otis, Dimitrios Giakoumis, Dimitrios Tzovaras, Liming Chen, and Raouf Hamzaoui. Two-dimensional convo- lutional recurrent neural networks for speech activity detection. International Speech Communi- cation Association, 2019
work page 2019
-
[36]
Brecht Desplanques, Jenthe Thienpondt, and Kris Demuynck. Ecapa-tdnn: Emphasized channel attention, propagation and aggregation in tdnn based speaker verification.arXiv preprint arXiv:2005.07143, 2020
Pith/arXiv arXiv 2005
-
[37]
Gate-variants of gated recurrent unit (gru) neural networks
Rahul Dey and Fathi M Salem. Gate-variants of gated recurrent unit (gru) neural networks. In 2017 IEEE 60th international midwest symposium on circuits and systems (MWSCAS), pages 1597–1600. IEEE, 2017. 11
work page 2017
-
[38]
Musan: A music, speech, and noise corpus
David Snyder, Guoguo Chen, and Daniel Povey. Musan: A music, speech, and noise corpus. arXiv preprint arXiv:1510.08484, 2015
Pith/arXiv arXiv 2015
-
[39]
Icassp 2023 deep noise suppression challenge.IEEE Open Journal of Signal Processing, 5:725–737, 2024
Harishchandra Dubey, Ashkan Aazami, Vishak Gopal, Babak Naderi, Sebastian Braun, Ross Cutler, Alex Ju, Mehdi Zohourian, Min Tang, Mehrsa Golestaneh, et al. Icassp 2023 deep noise suppression challenge.IEEE Open Journal of Signal Processing, 5:725–737, 2024
work page 2023
-
[40]
Ten turn detection: Turn detection for full-duplex dialogue communication, 2025
TEN Team. Ten turn detection: Turn detection for full-duplex dialogue communication, 2025. 12
work page 2025
discussion (0)
Sign in with ORCID, Apple, or X to comment. Anyone can read and Pith papers without signing in.