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FastSLM addresses the computational bottleneck of scaling Multimodal Large Language Models to long-form speech by introducing a Hierarchical Temporal Abstractor that drastically reduces token count without sacrificing linguistic fidelity. This architecture achieves a 97% reduction in input tokens, enabling more efficient processing of extended audio contexts compared to existing high-frame-rate projection methods. The approach represents a significant step forward in making speech-language understanding viable for resource-constrained environments while maintaining competitive performance across diverse tasks.
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FastSLM: Hierarchical Temporal Abstraction for Efficient Long-Form Speech Adaptation

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arXiv:2601.06199v4 Announce Type: replace-cross Abstract: Scaling Multimodal Large Language Models (MLLMs) to long-form speech is bottlenecked by the explosive growth of input tokens. Existing speech-language models project high-frame-rate acoustic features directly into the LLM input space, making long-context processing computationally prohibitive. Unlike images or videos, speech lacks spatial redundancy, making extreme token compression particularly challenging. To address this limitation, we propose FastSLM, a token-efficient architecture featuring the Hierarchical Temporal Abstractor (HTA), which progressively distills acoustic features across multiple temporal scales. HTA achieves an extreme compression rate of 1.67 tokens per second (97% reduction) while preserving essential linguistic information for downstream speech-language understanding. Experimental results demonstrate that FastSLM achieves competitive performance across diverse speech-language tasks while requiring substantially fewer speech tokens and FLOPs than existing speech-language models. The source code and model checkpoints are available at https://github.com/Lee-junseok1025/FastSLM.

Abstract:Scaling Multimodal Large Language Models (MLLMs) to long-form speech is bottlenecked by the explosive growth of input tokens. Existing speech-language models project high-frame-rate acoustic features directly into the LLM input space, making long-context processing computationally prohibitive. Unlike images or videos, speech lacks spatial redundancy, making extreme token compression particularly challenging. To address this limitation, we propose FastSLM, a token-efficient architecture featuring the Hierarchical Temporal Abstractor (HTA), which progressively distills acoustic features across multiple temporal scales. HTA achieves an extreme compression rate of 1.67 tokens per second (97% reduction) while preserving essential linguistic information for downstream speech-language understanding. Experimental results demonstrate that FastSLM achieves competitive performance across diverse speech-language tasks while requiring substantially fewer speech tokens and FLOPs than existing speech-language models. The source code and model checkpoints are available at this https URL.

Audio and Speech Processing (eess.AS); Artificial Intelligence (cs.AI); Sound (cs.SD)

arXiv:2601.06199 [eess.AS]

(or arXiv:2601.06199v4 [eess.AS] for this version)

https://doi.org/10.48550/arXiv.2601.06199 Focus to learn more arXiv-issued DOI via DataCite

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