Yaqi Chen

Dr. Yaqi Chen


Research Focus & Metrics

Academic Career Length

5Years (2022-Present)

(Calculated once Scilit publication data is synced: current year − year of first publication + 1)

Representative Publications

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13 February 2023 in IEEE/ACM Transactions on Audio, Speech, and Language Processing
Verified
Journal Article
Closed access
HZHao Zhang
Hao Zhang
NSNianwen Si
Nianwen Si
YCYaqi Chen
Yaqi Chen
WZWenlin Zhang
Wenlin Zhang
XYXukui Yang
Xukui Yang
WZWei-Qiang Zhang
Wei-Qiang Zhang
The end-to-end speech translation (E2E-ST) model has gradually become a mainstream paradigm due to its low latency and less error propagation. However, it is non-trivial to train such a model well due to the task complexity and data scarcity. The speech-and-text modality differences result in the E2E-ST model performance usually inferior to the corresponding machine translation (MT) model. Based on the above observation, existing methods often use sharing mechanisms to carry out implicit knowledge transfer by imposing various constraints. However, the final model often performs worse on the MT task than the MT model trained alone, which means that the knowledge transfer ability of this method is also limited. To deal with these problems, we propose the FCCL ( F ine- and C oarse- Granularity C ontrastive L earning) approach for E2E-ST, which makes explicit knowledge transfer through cross-modal multi-grained contrastive learning. A key ingredient of our approach is applying contrastive learning at both sentence- and frame-level to give the comprehensive guide for extracting speech representations containing rich semantic information. In addition, we adopt a simple whitening method to alleviate the representation degeneration in the MT model, which adversely affects contrast learning. Experiments on the MuST-C benchmark show that our proposed approach significantly outperforms the state-of-the-art E2E-ST baselines on all eight language pairs. Further analysis indicates that FCCL can free up its capacity from learning grammatical structure information and force more layers to learn semantic information.

19 October 2023 in Computer Speech & Language
Verified
Journal Article
Closed access
YCYaqi Chen
Yaqi Chen
XYXukui Yang
Xukui Yang
HZHao Zhang
Hao Zhang
WZWenlin Zhang
Wenlin Zhang
DQDan Qu
Dan Qu
CCCong Chen
Cong Chen

9 April 2024 in Computer Speech & Language
Verified
Journal Article
Closed access
YCYaqi Chen
Yaqi Chen
HZHao Zhang
Hao Zhang
XYXukui Yang
Xukui Yang
WZWenlin Zhang
Wenlin Zhang
DQDan Qu
Dan Qu

5 July 2024 in Neural Processing Letters
Verified
Journal Article
Open access
PDF file
YCYaqi Chen
Yaqi Chen
HZHao Zhang
Hao Zhang
WZWen-Lin Zhang
Wen-Lin Zhang
DQDan Qu
Dan Qu
XYXukui Yang
Xukui Yang
Meta-learning has proven to be a powerful paradigm for transferring knowledge from prior tasks to facilitate the quick learning of new tasks in automatic speech recognition. However, the differences between languages (tasks) lead to variations in task learning directions, causing the harmful competition for model’s limited resources. To address this challenge, we introduce the task-agreement multilingual meta-learning (TAMML), which adopts the gradient agreement algorithm to guide the model parameters towards a direction where tasks exhibit greater consistency. However, the computation and storage cost of TAMML grows dramatically with model’s depth increases. To address this, we further propose a simplification called TAMML-Light which only uses the output layer for gradient calculation. Experiments on three datasets demonstrate that TAMML and TAMML-Light achieve outperform meta-learning approaches, yielding superior results.Furthermore, TAMML-Light can reduce at least 80 $$\%$$% of the relative increased computation expenses compared to TAMML.

20 November 2023 in ACM Transactions on Asian and Low-Resource Language Information Processing
Verified
Journal Article
Open access
YCYaqi Chen
Yaqi Chen
WZWen-Lin Zhang
Wen-Lin Zhang
HZHao Zhang
Hao Zhang
DQDan Qu
Dan Qu
XYXukui Yang
Xukui Yang
Low-resource automatic speech recognition is a challenging task due to a lack of labeled training data. To resolve this issue, multilingual meta-learning learns a better model initialization from many source language tasks for fast adaptation to unseen target languages. However, for diverse source languages, the quantity and difficulty vary greatly because of their different data scales and phonological systems. These differences lead to task-quantity and task-difficulty imbalance issues and thus a failure of multilingual meta-learning ASR. In this work, we propose a task-based meta focal loss (TMFL) approach to address this tough challenge. Specifically, we introduce a hard-task moderator and update the meta-parameters using gradients from both the support set and query set. Our proposed approach focuses more on hard tasks and makes full use of the data from hard tasks. Moreover, we analyze the significance of the hard task moderator and interpret its significance at the sample level. Experiment results show that the proposed method, TMFL, significantly outperforms the state-of-the-art multilingual meta-learning on all target languages for the IARPA BABEL and OpenSLR datasets, especially under very-low-resource conditions. In particular, it can reduce character error rate from 72% to 60% by fine-tuning the pre-trained model with about 22 hours of Vietnamese data.

7 April 2025 in Applied Sciences
Verified
Journal Article
Open access
PDF file
TNTong Niu
Tong Niu
YCYaqi Chen
Yaqi Chen
DQDan Qu
Dan Qu
HHHu Hengbo
Hu Hengbo
Recent advancements in end-to-end (E2E) modeling have notably improved automatic speech recognition (ASR) systems; however, far-field speech recognition (FSR) remains challenging due to signal degradation from factors such as low signal-to-noise ratio, reverberation, and interfering sounds. This requires richer training data and multi-channel speech enhancement. To address this gap, we introduce Mixer, a novel data augmentation technique designed to further enhance the performance of large-scale pre-trained models for FSR. Mixer interpolates and mixes feature representations of speech samples and their corresponding losses, extending the MixSpeech framework to intermediate layers of Whisper. Additionally, we propose Mixer-C, which further leverages multi-channel information by combining speech from different microphone channels using a channel selector. Experimental results demonstrate that Mixer significantly outperforms existing methods, including SpecAugment, achieving a relative word error rate (WER) reduction of 3.6% compared to the baseline. Furthermore, Mixer-C offers an additional WER improvement of 2.2%, showcasing its efficacy in improving FSR accuracy.