🤖 AI Summary
This work addresses the slow inference speed of Transformer-based speech transcription models. We propose a fine-tuning-free early temporal sparsification method that dynamically prunes hidden states in the encoder’s initial layers based on self-attention weights, enabling interpretable and efficient temporal sparsity. Our key contributions are: (i) the first systematic investigation of attention-guided early sparsification strategies for speech Transformers; (ii) a joint search framework optimizing both sparse locations and compression ratios; and (iii) end-to-end GPU-accelerated implementation. Evaluated on the Whisper architecture, our method achieves 40–60% sparsity with less than 1% WER degradation on English speech transcription, while accelerating inference by up to 1.6×—substantially outperforming existing post-hoc or late-stage sparsification approaches.
📝 Abstract
Transformer-based neural speech processing has achieved state-of-the-art performance. Since speech audio signals are known to be highly compressible, here we seek to accelerate neural speech transcription by time-domain signal sparsification early in the neural encoding stage, taking advantage of the interpretability of the self-attention mechanism in transformer audio encoders. With the Whisper family of models, we perform a systematic architecture search over the joint space of sparsification stage (a certain encoder layer) and compression ratio (sparsity). We found that the best resulting solutions under 1% accuracy degradation choose to sparsify the hidden state to 40-60% sparsity at an early encoding stage, and thereby achieve up to 1.6x runtime acceleration in English speech transcription tasks on Nvidia GPUs without any fine-tuning.