Wonjun Lee


2025

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DyPCL: Dynamic Phoneme-level Contrastive Learning for Dysarthric Speech Recognition
Wonjun Lee | Solee Im | Heejin Do | Yunsu Kim | Jungseul Ok | Gary Lee
Proceedings of the 2025 Conference of the Nations of the Americas Chapter of the Association for Computational Linguistics: Human Language Technologies (Volume 1: Long Papers)

Dysarthric speech recognition often suffers from performance degradation due to the intrinsic diversity of dysarthric severity and extrinsic disparity from normal speech. To bridge these gaps, we propose a Dynamic Phoneme-level Contrastive Learning (DyPCL) method, which leads to obtaining invariant representations across diverse speakers. We decompose the speech utterance into phoneme segments for phoneme-level contrastive learning, leveraging dynamic connectionist temporal classification alignment. Unlike prior studies focusing on utterance-level embeddings, our granular learning allows discrimination of subtle parts of speech. In addition, we introduce dynamic curriculum learning, which progressively transitions from easy negative samples to difficult-to-distinguishable negative samples based on phonetic similarity of phoneme. Our approach to training by difficulty levels alleviates the inherent variability of speakers, better identifying challenging speeches. Evaluated on the UASpeech dataset, DyPCL outperforms baseline models, achieving an average 22.10% relative reduction in word error rate (WER) across the overall dysarthria group.

2024

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Enhancing Dialogue Speech Recognition with Robust Contextual Awareness via Noise Representation Learning
Wonjun Lee | San Kim | Gary Geunbae Lee
Proceedings of the 25th Annual Meeting of the Special Interest Group on Discourse and Dialogue

Recent dialogue systems typically operate through turn-based spoken interactions between users and agents. These systems heavily depend on accurate Automatic Speech Recognition (ASR), as transcription errors can significantly degrade performance in downstream dialogue tasks. To alleviate this challenge, robust ASR is required, and one effective method is to utilize the dialogue context from user and agent interactions for transcribing the subsequent user utterance. This method incorporates the transcription of the user’s speech and the agent’s response as model input, using the accumulated context generated by each turn. However, this context is susceptible to ASR errors because the ASR model generates it auto-regressively. Such noisy context can further degrade the benefits of context input, resulting in suboptimal ASR performance. In this paper, we introduce context noise representation learning to enhance robustness against noisy context, ultimately improving dialogue speech recognition accuracy. To maximize the advantage of context awareness, our approach involves decoder pre-training with text-based dialogue data and noise representation learning for a context encoder. Evaluated on DSTC11 (MultiWoZ 2.1 audio dialogues), it achieves a 24% relative reduction in Word Error Rate (WER) compared to wav2vec2.0 baselines and a 13% reduction compared to Whisper-large-v2. Notably, in noisy environments where user speech is barely audible, our method proves its effectiveness by utilizing contextual information for accurate transcription. Tested on audio data with strong noise level (Signal Noise Ratio of 0dB), our approach shows up to a 31% relative WER reduction compared to the wav2vec2.0 baseline, providing a reassuring solution for real-world noisy scenarios.

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An Investigation into Explainable Audio Hate Speech Detection
Jinmyeong An | Wonjun Lee | Yejin Jeon | Jungseul Ok | Yunsu Kim | Gary Geunbae Lee
Proceedings of the 25th Annual Meeting of the Special Interest Group on Discourse and Dialogue

Research on hate speech has predominantly revolved around the detection and interpretation from textual inputs, leaving verbal content largely unexplored. Moreover, while there has been some limited exploration into hate speech detection within verbal acoustic speech inputs, the aspect of interpretability has been overlooked. As such, we introduce a new task within the audio hate speech detection task domain - we specifically aim to identify specific time frames of hate speech within audio utterances. Towards this, we propose two different approaches, cascading and End-to-End (E2E). The first cascading approach initially converts audio to transcripts, identifies hate speech within these transcripts, and subsequently locates the corresponding audio time frames. Conversely, the second E2E approach processes audio utterances directly, which allows it to pinpoint hate speech within specific time frames. Moreover, due to the lack of explainable audio hate speech datasets that include frame-level rationales, we curated a synthetic audio dataset to train our models. We further validate these models on actual human speech utterances and we find that the E2E approach outperforms the cascading method in terms of audio frame Intersection over Union (IoU) metric. Furthermore, we observe that the inclusion of frame-level rationales significantly enhances hate speech detection accuracy for both E2E and cascading approaches.