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Benchmarking Language Modeling for Lossless Compression of Full-Fidelity Audio

Sound 2026-03-10 v1 Artificial Intelligence Machine Learning Audio and Speech Processing

Abstract

Autoregressive "language" models (LMs) trained on raw waveforms can be repurposed for lossless audio compression, but prior work is limited to 8-bit audio, leaving open whether such approaches work for practical settings (16/24-bit) and can compete with existing codecs. We benchmark LM-based compression on full-fidelity audio across diverse domains (music, speech, bioacoustics), sampling rates (16kHz-48kHz), and bit depths (8, 16, 24-bit). Standard sample-level tokenization becomes intractable at higher bit depths due to vocabulary size (65K for 16-bit; 16.7M for 24-bit). We propose Trilobyte, a byte-level tokenization schema for full resolution audio, improving vocabulary scaling from O(2b)O(2^{b}) to O(1)O(1) and enabling the first tractable 24-bit LM-based lossless compression. While LMs consistently outperform FLAC and yield state-of-the-art compression at 8-bit and 16-bit, we observe that compression gains become more modest as bit depth increases beyond 8-bit.

Keywords

Cite

@article{arxiv.2603.08683,
  title  = {Benchmarking Language Modeling for Lossless Compression of Full-Fidelity Audio},
  author = {Phillip Long and Zachary Novack and Chris Donahue},
  journal= {arXiv preprint arXiv:2603.08683},
  year   = {2026}
}

Comments

Submitted for review at Interspeech 2026, 7 pages, 5 figures