An Embarrassingly Simple Approach for LLM with Strong ASR Capacity

Fuente: arXiv
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Autori principali: Ma, Ziyang, Yang, Guanrou, Yang, Yifan, Gao, Zhifu, Wang, Jiaming, Du, Zhihao, Yu, Fan, Chen, Qian, Zheng, Siqi, Zhang, Shiliang, Chen, Xie
Natura: Preprint
Pubblicazione: 2024
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author Ma, Ziyang
Yang, Guanrou
Yang, Yifan
Gao, Zhifu
Wang, Jiaming
Du, Zhihao
Yu, Fan
Chen, Qian
Zheng, Siqi
Zhang, Shiliang
Chen, Xie
author_facet Ma, Ziyang
Yang, Guanrou
Yang, Yifan
Gao, Zhifu
Wang, Jiaming
Du, Zhihao
Yu, Fan
Chen, Qian
Zheng, Siqi
Zhang, Shiliang
Chen, Xie
contents In this paper, we focus on solving one of the most important tasks in the field of speech processing, i.e., automatic speech recognition (ASR), with speech foundation encoders and large language models (LLM). Recent works have complex designs such as compressing the output temporally for the speech encoder, tackling modal alignment for the projector, and utilizing parameter-efficient fine-tuning for the LLM. We found that delicate designs are not necessary, while an embarrassingly simple composition of off-the-shelf speech encoder, LLM, and the only trainable linear projector is competent for the ASR task. To be more specific, we benchmark and explore various combinations of LLMs and speech encoders, leading to the optimal LLM-based ASR system, which we call SLAM-ASR. The proposed SLAM-ASR provides a clean setup and little task-specific design, where only the linear projector is trained. To the best of our knowledge, SLAM-ASR achieves the best performance on the Librispeech benchmark among LLM-based ASR models and even outperforms the latest LLM-based audio-universal model trained on massive pair data. Finally, we explore the capability emergence of LLM-based ASR in the process of modal alignment. We hope that our study can facilitate the research on extending LLM with cross-modality capacity and shed light on the LLM-based ASR community.
format Preprint
id arxiv_https___arxiv_org_abs_2402_08846
institution arXiv
publishDate 2024
record_format arxiv
spellingShingle An Embarrassingly Simple Approach for LLM with Strong ASR Capacity
Ma, Ziyang
Yang, Guanrou
Yang, Yifan
Gao, Zhifu
Wang, Jiaming
Du, Zhihao
Yu, Fan
Chen, Qian
Zheng, Siqi
Zhang, Shiliang
Chen, Xie
Computation and Language
Artificial Intelligence
Multimedia
Sound
Audio and Speech Processing
In this paper, we focus on solving one of the most important tasks in the field of speech processing, i.e., automatic speech recognition (ASR), with speech foundation encoders and large language models (LLM). Recent works have complex designs such as compressing the output temporally for the speech encoder, tackling modal alignment for the projector, and utilizing parameter-efficient fine-tuning for the LLM. We found that delicate designs are not necessary, while an embarrassingly simple composition of off-the-shelf speech encoder, LLM, and the only trainable linear projector is competent for the ASR task. To be more specific, we benchmark and explore various combinations of LLMs and speech encoders, leading to the optimal LLM-based ASR system, which we call SLAM-ASR. The proposed SLAM-ASR provides a clean setup and little task-specific design, where only the linear projector is trained. To the best of our knowledge, SLAM-ASR achieves the best performance on the Librispeech benchmark among LLM-based ASR models and even outperforms the latest LLM-based audio-universal model trained on massive pair data. Finally, we explore the capability emergence of LLM-based ASR in the process of modal alignment. We hope that our study can facilitate the research on extending LLM with cross-modality capacity and shed light on the LLM-based ASR community.
title An Embarrassingly Simple Approach for LLM with Strong ASR Capacity
topic Computation and Language
Artificial Intelligence
Multimedia
Sound
Audio and Speech Processing
url https://arxiv.org/abs/2402.08846