AudioLM: a Language Modeling Approach to Audio Generation
Written for the Research library (claude-opus-5-5) from the paper text, in our own words: the license "arXiv — non-exclusive license to distribute" does not allow redistributing the text itself. Original — arxiv.org/pdf/2209.03143
Summary
AudioLM tackles a long-standing tension in audio synthesis: models that produce realistic-sounding waveforms tend to drift into incoherent output (such as babbling speech) without strong conditioning, while earlier "textless" spoken language models stay coherent but offer limited quality and only one clean-speech speaker. The Google Research authors frame audio generation as language modeling over discrete tokens and combine two kinds of tokenizers. Coarse semantic tokens come from k-means clustering of intermediate w2v-BERT activations and carry long-range structure. Fine acoustic tokens come from the residual quantizer of the SoundStream codec and preserve waveform detail. A multi-stage, decoder-only Transformer predicts these tokens hierarchically, and SoundStream decodes them back into audio. Trained only on raw audio, with no transcripts or symbolic scores, the system continues short prompts. Speech continuations are plausible in syntax and meaning and keep the voice, prosody and acoustic conditions of unseen speakers. Piano continuations stay consistent in melody, harmony and rhythm. The work matters because it reaches both high fidelity and long-term coherence without supervision.
Key points
- Hybrid tokenization: w2v-BERT semantic tokens score well on phonetic discriminability but reconstruct audio poorly, while SoundStream acoustic tokens reconstruct well. The authors show the two complement each other.
- Speech prompting: a prompt of only 3 seconds from a speaker not seen in training is enough for AudioLM to continue in the same voice, intonation and recording conditions, including reverberation and background noise.
- Compact sequences: token sequences are 2–3 orders of magnitude shorter than the raw waveform, which makes long contexts feasible despite the quadratic cost of self-attention.
- Comparison with prior systems: AudioLM improves on GSLM, which is limited to a single speaker, and on Jukebox and Perceiver AR, which trade off either audio quality or temporal structure.
- Misuse mitigation: the authors trained a classifier that detects AudioLM-generated speech with very high accuracy. Audio samples are published at google-research.github.io/seanet/audiolm/examples.
Cited by
- Seamless: Multilingual Expressive and Streaming Speech Translation
- SeamlessM4T: Massively Multilingual & Multimodal Machine Translation
- Moshi: a speech-text foundation model for real-time dialogue
- A Language Modeling Approach to Diacritic-Free Hebrew TTS
- AudioPaLM: A Large Language Model That Can Speak and Listen
- Neural Codec Language Models are Zero-Shot Text to Speech Synthesizers
- SpeechGPT: Empowering Large Language Models with Intrinsic Cross-Modal Conversational Abilities
Cite
@misc{borsos2022audiolm,
title = {{AudioLM: a Language Modeling Approach to Audio Generation}},
author = {Borsos, Zalan and Marinier, Raphael and Vincent, Damien and Kharitonov, Eugene and Pietquin, Olivier and Sharifi, Matt and Roblek, Dominik and Teboul, Olivier and Grangier, David and Tagliasacchi, Marco and Zeghidour, Neil},
year = {2022},
doi = {10.48550/arxiv.2209.03143},
eprint = {2209.03143},
archivePrefix = {arXiv},
url = {https://doi.org/10.48550/arxiv.2209.03143}
}References
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