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LiteASR: Efficient Automatic Speech Recognition with Low-Rank Approximation
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Modern automatic speech recognition (ASR) models, such as OpenAI's Whisper, rely on deep encoder-decoder architectures, and their encoders are a critical bottleneck for efficient deployment due to high computational intensity. We introduce LiteASR, a low-rank compression scheme for ASR encoders that significantly reduces inference costs while maintaining transcription accuracy. Our approach leverages the strong low-rank properties observed in intermediate activations: by applying principal component analysis (PCA) with a small calibration dataset, we approximate linear transformations with a chain of low-rank matrix multiplications, and further optimize self-attention to work in reduced dimensionality. Evaluation results show that our method can compress Whisper large-v3's encoder size by over 50%, matching Whisper medium's size with better transcription accuracy, thereby establishing a new Pareto frontier of accuracy and efficiency. The code of LiteASR is available at https://github.com/efeslab/LiteASR.
Forward citations
Cited by 2 Pith papers
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Early Attentive Sparsification Accelerates Neural Speech Transcription
Attention-based early audio-token sparsification at 40-60% sparsity accelerates Whisper ASR up to 1.6x with under 1% relative WER loss, across ten model variants, with no fine-tuning.
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Adapting Whisper for Lightweight and Efficient Automatic Speech Recognition of Children for On-device Edge Applications
A fine-tuned Whisper tiny.en model reaches 15.9% WER on children's speech and runs in real time on a Raspberry Pi, with low-rank compression trading an 11% relative WER rise for lighter computation.
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