Yaqi Chen
Current Organizations / Institutes
The organizations/institutes from the scholar's latest article
Research Subjects
Yaqi Chen has a total of 22 co-authors. Here are the top co-authors by number of shared publications:
- Dan Qu
Information Engineering University, Zhengzhou, China
11 shared publications - Hao Zhang
Information Engineering University, Zhengzhou, China
8 shared publications - Xukui Yang
School of Information Systems Engineering, University of information Engineering, Zhengzhou, China
6 shared publications - Wenlin Zhang
Information Engineering University, Zhengzhou, China
5 shared publications Dan Qu
Information Engineering University
3 shared publicationsWenlin Zhang
3 shared publications
Latest Publications
- Open Access
- 25 February 2026
- Published by Springer Nature in Peer-to-Peer Networking and Applications
- Vol. 19 (2) , 52
SCNNTraffic: A lightweight and energy-efficient traffic classification method based on spiking convolution neural networks
- 25 February 2026
- Published by Springer Nature in Peer-to-Peer Networking and Applications
- Vol. 19 (2) , 52
Open AccessSCNNTraffic: A lightweight and energy-efficient traffic classification method based on spiking convolution neural networks
- Open Access
- 13 October 2025
- Published by Springer Nature in Journal of King Saud University Computer and Information Sciences
- Vol. 37 (8)
Quantum autoencoder implementation of high-dimensional steganographic encoding for arbitrary quantum states
- 13 October 2025
- Published by Springer Nature in Journal of King Saud University Computer and Information Sciences
- Vol. 37 (8)
Open AccessQuantum autoencoder implementation of high-dimensional steganographic encoding for arbitrary quantum states
- Open Access
- 25 April 2025
- Published by Springer Nature in EURASIP Journal on Audio, Speech, and Music Processing
- Vol. 2025 (1) , 1-10
Parameter-efficient adaptation with multi-channel adversarial training for far-field speech recognition
- 25 April 2025
- Published by Springer Nature in EURASIP Journal on Audio, Speech, and Music Processing
- Vol. 2025 (1) , 1-10
Open AccessParameter-efficient adaptation with multi-channel adversarial training for far-field speech recognition
- Open Access
- 7 April 2025
- Published by MDPI AG in Applied Sciences
- Vol. 15 (7) , 4073
Enhancing Far-Field Speech Recognition with Mixer: A Novel Data Augmentation Approach
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.
- 7 April 2025
- Published by MDPI AG in Applied Sciences
- Vol. 15 (7) , 4073
Open AccessEnhancing Far-Field Speech Recognition with Mixer: A Novel Data Augmentation Approach
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.
- 21 October 2024
- Published by Institute of Electrical and Electronics Engineers (IEEE) in IEEE Signal Processing Letters
- Vol. 31 (10709908) , 3039-3043
Meta-Prompt: Boosting Whisper's Performance in Low-Resource Speech Recognition
Recent advancements in large-scale pre-trained automatic speech recognition (ASR) foundation models (e.g., Whisper) have exhibited remarkable performance in speech processing tasks. A recently emerging paradigm, prompt tuning, offers a parameter-efficient approach for fine-tuning, which has proven to be effective in enhancing the adaptation of pre-trained models to downstream tasks. In this paper, we first explore the prompting method for low-resource speech recognition based on Whisper. Although effective, it poses a challenge in the few-shot scenario due to its high sensitivity to initialization. To address this problem, we propose a novel meta-prompt for low-resource speech recognition that leverages the benefits of meta-learning for fast learning. Moreover, we further present a lightweight version of meta-prompt that omits the learning of encoder-prompt, reducing computational and storage costs. Extensive experiments on FLEURS datasets demonstrate consistent improvements across eleven target languages, showing better generalizability. Notably, meta-prompt achieves similar performance with a 20 $\%$ -shot compared to prompt tuning with a 50 $\%$ -shot setting, suggesting excellent few-shot learning ability.
- 21 October 2024
- Published by Institute of Electrical and Electronics Engineers (IEEE) in IEEE Signal Processing Letters
- Vol. 31 (10709908) , 3039-3043
Meta-Prompt: Boosting Whisper's Performance in Low-Resource Speech Recognition
Recent advancements in large-scale pre-trained automatic speech recognition (ASR) foundation models (e.g., Whisper) have exhibited remarkable performance in speech processing tasks. A recently emerging paradigm, prompt tuning, offers a parameter-efficient approach for fine-tuning, which has proven to be effective in enhancing the adaptation of pre-trained models to downstream tasks. In this paper, we first explore the prompting method for low-resource speech recognition based on Whisper. Although effective, it poses a challenge in the few-shot scenario due to its high sensitivity to initialization. To address this problem, we propose a novel meta-prompt for low-resource speech recognition that leverages the benefits of meta-learning for fast learning. Moreover, we further present a lightweight version of meta-prompt that omits the learning of encoder-prompt, reducing computational and storage costs. Extensive experiments on FLEURS datasets demonstrate consistent improvements across eleven target languages, showing better generalizability. Notably, meta-prompt achieves similar performance with a 20 $\%$ -shot compared to prompt tuning with a 50 $\%$ -shot setting, suggesting excellent few-shot learning ability.
- Open Access
- 14 October 2024
- Published by Springer Nature in Research Square
Enhancing Far-Field Speech Recognition with Mixer: A Novel Data Augmentation Approach
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.
- 14 October 2024
- Published by Springer Nature in Research Square
Open AccessEnhancing Far-Field Speech Recognition with Mixer: A Novel Data Augmentation Approach
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.
- Open Access
- 5 July 2024
- Published by Springer Nature in Neural Processing Letters
- Vol. 56 (4) , 1-14
A Lightweight Task-Agreement Meta Learning for Low-Resource Speech Recognition
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.- 5 July 2024
- Published by Springer Nature in Neural Processing Letters
- Vol. 56 (4) , 1-14
Open AccessA Lightweight Task-Agreement Meta Learning for Low-Resource Speech Recognition
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. - 9 April 2024
- Published by Elsevier in Computer Speech & Language
- Vol. 87
Improving cross-lingual low-resource speech recognition by Task-based Meta PolyLoss
- 9 April 2024
- Published by Elsevier in Computer Speech & Language
- Vol. 87
Improving cross-lingual low-resource speech recognition by Task-based Meta PolyLoss
- Open Access
- 20 November 2023
- Published by Association for Computing Machinery (ACM) in ACM Transactions on Asian and Low-Resource Language Information Processing
- Vol. 22 (11) , 1-17
Task-based Meta Focal Loss for Multilingual Low-resource Speech Recognition
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.
- 20 November 2023
- Published by Association for Computing Machinery (ACM) in ACM Transactions on Asian and Low-Resource Language Information Processing
- Vol. 22 (11) , 1-17
Open AccessTask-based Meta Focal Loss for Multilingual Low-resource Speech Recognition
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.
- 19 October 2023
- Published by Elsevier in Computer Speech & Language
- Vol. 84
Meta adversarial learning improves low-resource speech recognition
- 19 October 2023
- Published by Elsevier in Computer Speech & Language
- Vol. 84
Meta adversarial learning improves low-resource speech recognition