Towards Sub-millisecond Latency Real-Time Speech Enhancement Models on Hearables

Low latency models are critical for real-time speech enhancement applications, such as hearing aids and hearables. However, the sub-millisecond latency space for resource-constrained hearables remains underexplored. We demonstrate speech enhancement using a computationally efficient minimum-phase FIR filter, enabling sample-by-sample processing to achieve mean algorithmic latency of 0.32 ms to 1.25 ms. With a single microphone, we observe a mean SI-SDRi of 4.1 dB. The approach shows generalization with a DNSMOS increase of 0.2 on unseen audio recordings. We use a lightweight LSTM-based model of 626k parameters to generate FIR taps. Using a real hardware implementation on a low-power DSP, our system can run with 376 MIPS and a mean end-to-end latency of 3.35 ms. In addition, we provide a comparison with existing low-latency spectral masking techniques. We hope this work will enable a better understanding of latency and can be used to improve the comfort and usability of hearables.

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Publication Details

Published
2025-03-12
DOI
https://doi.org/10.1109/icassp49660.2025.10889875
Citations
2
Primary Topic
Speech and Audio Processing
Type
article
Field-Weighted Citation Impact
2.97
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article

Towards Sub-millisecond Latency Real-Time Speech Enhancement Models on Hearables

Chandan K. Reddy, Artem Dementyev, Navin Chatlani, Richard F. Lyon et al.
2 citations
Speech and Audio Processing
2.97
article

Towards Sub-millisecond Latency Real-Time Speech Enhancement Models on Hearables

Chandan K. Reddy, Artem Dementyev, Navin Chatlani, Richard F. Lyon, Scott Wisdom, John R. Hershey
article en
2 citations

Abstract

Low latency models are critical for real-time speech enhancement applications, such as hearing aids and hearables. However, the sub-millisecond latency space for resource-constrained hearables remains underexplored. We demonstrate speech enhancement using a computationally efficient minimum-phase FIR filter, enabling sample-by-sample processing to achieve mean algorithmic latency of 0.32 ms to 1.25 ms. With a single microphone, we observe a mean SI-SDRi of 4.1 dB. The approach shows generalization with a DNSMOS increase of 0.2 on unseen audio recordings. We use a lightweight LSTM-based model of 626k parameters to generate FIR taps. Using a real hardware implementation on a low-power DSP, our system can run with 376 MIPS and a mean end-to-end latency of 3.35 ms. In addition, we provide a comparison with existing low-latency spectral masking techniques. We hope this work will enable a better understanding of latency and can be used to improve the comfort and usability of hearables.

Google (United States) (US), Google DeepMind (United Kingdom) (GB)
Openalex Percentile: Top 11%
Speech and Audio Processing
2.97
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