Mingjie Chen
Author directory2026
FlexiSLM: A Spoken Language Model with Dynamic and Controllable Frame Rates
Jiaqi Li | Chaoren Wang | Xiaohai Tian | Mingjie Chen | Xinyu Liang | Xu Li | Yufan Lin | Junwen Qiu | Jun Zhang | Lu Lu | Zhizheng Wu
Proceedings of the 2026 Conference on Empirical Methods in Natural Language Processing
Jiaqi Li | Chaoren Wang | Xiaohai Tian | Mingjie Chen | Xinyu Liang | Xu Li | Yufan Lin | Junwen Qiu | Jun Zhang | Lu Lu | Zhizheng Wu
Proceedings of the 2026 Conference on Empirical Methods in Natural Language Processing
Spoken language models (SLMs) extend LLMs to speech input and output, but existing systems use fixed frame rates (e.g., 25 or 12.5 Hz), overlooking speech’s time-varying information density and limiting inference-time quality–speed tradeoffs. Recent dynamic-frame-rate audio tokenizers enable very low average frame rates and controllability, yet had not been applied to SLMs. We introduce FlexiSLM, the first SLM with dynamic, controllable frame rates, using pretrained FlexiCodec for dynamic speech output tokens. It integrates this representation into a multi-task speech-to-speech SLM, extends it with input-side frame compression, and adds direct frame-rate conditioning for accurate control during inference. FlexiSLM outperforms fixed-frame-rate 7B models, including Qwen2.5-Omni and Kimi-Audio, at 12.5 and 6.25 Hz; it can be steered down to 4.0 Hz, and at 6.25 Hz roughly halves inference time relative to 12.5 Hz while retaining strong speech-to-speech quality. Audio samples: https://flexislm.github.io; code and data: https://github.com/AmphionTeam/FlexiSLM.
SimulS2ST-Omni: Data-Efficient Streaming Speech-to-Speech Translation via Explicit Trajectory Supervision
Rongshen He | Xinyu Liang | Dekun Chen | Jiaqi Li | Mingjie Chen | Zhizheng Wu
Proceedings of the 2026 Conference on Empirical Methods in Natural Language Processing
Rongshen He | Xinyu Liang | Dekun Chen | Jiaqi Li | Mingjie Chen | Zhizheng Wu
Proceedings of the 2026 Conference on Empirical Methods in Natural Language Processing
Long-form streaming speech-to-speech translation (S2ST) requires incremental, unbounded translation under strict latency constraints. Existing methods typically suffer from sentence-bounded supervision or demand massive paired-S2ST supervision. We introduce a training recipe enabling a speech language model for sentence-level and long-form streaming S2ST using only ∼2k hours of paired cross-lingual S2ST data, layered atop auxiliary supervision. Anchored by auxiliary multitask training, our approach remains robust even when the paired-S2ST budget itself is reduced by 90%. Our core contribution, joint text-code trajectory supervision, schedules target text and acoustic semantic codes as a unified commitment path, eliminating the need for separate, unstable speech-side emission controllers. Furthermore, our two-stream Thinker–Talker factorization significantly outperforms unified-decoder baselines by decoupling linguistic reasoning from dense acoustic prediction to mitigate modality interference. Finally, our system achieves highly competitive quality-latency trade-offs on RealSI and ACL60/60-dev, matching state-of-the-art, closed-source S2ST systems such as LiveInterpret 2.0 on ASR-BLEU.
2024
Automatic Speech Recognition System-Independent Word Error Rate Estimation
Chanho Park | Mingjie Chen | Thomas Hain
Proceedings of the 2024 Joint International Conference on Computational Linguistics, Language Resources and Evaluation (LREC-COLING 2024)
Chanho Park | Mingjie Chen | Thomas Hain
Proceedings of the 2024 Joint International Conference on Computational Linguistics, Language Resources and Evaluation (LREC-COLING 2024)
Word error rate (WER) is a metric used to evaluate the quality of transcriptions produced by Automatic Speech Recognition (ASR) systems. In many applications, it is of interest to estimate WER given a pair of a speech utterance and a transcript. Previous work on WER estimation focused on building models that are trained with a specific ASR system in mind (referred to as ASR system-dependent). These are also domain-dependent and inflexible in real-world applications. In this paper, a hypothesis generation method for ASR System-Independent WER estimation (SIWE) is proposed. In contrast to prior work, the WER estimators are trained using data that simulates ASR system output. Hypotheses are generated using phonetically similar or linguistically more likely alternative words. In WER estimation experiments, the proposed method reaches a similar performance to ASR system-dependent WER estimators on in-domain data and achieves state-of-the-art performance on out-of-domain data. On the out-of-domain data, the SIWE model outperformed the baseline estimators in root mean square error and Pearson correlation coefficient by relative 17.58% and 18.21%, respectively, on Switchboard and CALLHOME. The performance was further improved when the WER of the training set was close to the WER of the evaluation dataset.
2017
Fake news stance detection using stacked ensemble of classifiers
James Thorne | Mingjie Chen | Giorgos Myrianthous | Jiashu Pu | Xiaoxuan Wang | Andreas Vlachos
Proceedings of the 2017 EMNLP Workshop: Natural Language Processing meets Journalism
James Thorne | Mingjie Chen | Giorgos Myrianthous | Jiashu Pu | Xiaoxuan Wang | Andreas Vlachos
Proceedings of the 2017 EMNLP Workshop: Natural Language Processing meets Journalism
Fake news has become a hotly debated topic in journalism. In this paper, we present our entry to the 2017 Fake News Challenge which models the detection of fake news as a stance classification task that finished in 11th place on the leader board. Our entry is an ensemble system of classifiers developed by students in the context of their coursework. We show how we used the stacking ensemble method for this purpose and obtained improvements in classification accuracy exceeding each of the individual models’ performance on the development data. Finally, we discuss aspects of the experimental setup of the challenge.