AudioPaLM: A Large Language Model That Can Speak and Listen
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/2306.12925
Summary
AudioPaLM, from Google, tackles a split in current systems. Text LLMs hold broad linguistic knowledge but cannot hear or produce speech. Audio language models capture voice and prosody but lack that knowledge. The authors start from a text-only PaLM-2 (or PaLM) checkpoint and enlarge its embedding table with discrete audio tokens. Everything else in the architecture stays the same, giving a single decoder-only Transformer that reads and writes mixed text and audio sequences. Short textual task tags (e.g. ASR, S2ST with language names) let one model train jointly on ASR, AST, TTS and speech-to-speech translation. Generated audio tokens are turned back into waveforms by SoundStorm or the later AudioLM stages. Starting from text-pretrained weights clearly helps speech tasks. The model reaches state-of-the-art AST and S2ST results and stays competitive on ASR. It also translates speech to text zero-shot for unseen language combinations and keeps a speaker's voice across languages from a brief audio prompt. The work shows that text LLM knowledge transfers to spoken-language processing.
Key points
- The architecture merges PaLM-2 and AudioLM ideas. Audio is tokenized (e.g. with USM-v2 tokens), and the whole model must be finetuned rather than kept frozen as in Flamingo.
- Model scale matters. With the Public + YT + WMT/TED mixture, the 128M, 1B and 8B PaLM-2 models score 16.6/31.6/37.8 BLEU on CVSS AST. VoxPopuli ASR WER drops from 16.4 to 9.8.
- The 1B AudioPaLM beats Whisper Large (1.5B) by over 5 BLEU and cuts VoxPopuli ASR WER by about 28%.
- Starting from PaLM-2 8B instead of PaLM 8B mainly boosts AST, which the authors credit to PaLM-2's stronger text translation. ASR effects are mixed.
- Stated limitations are a strong reliance on audio tokenizer quality and immature benchmarks for generative audio tasks.
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
- HebDB: a Weakly Supervised Dataset for Hebrew Speech Processing
- SALMONN: Towards Generic Hearing Abilities for Large Language Models
- A Survey of Large Language Models
Cite
@misc{rubenstein2023audiopalm,
title = {{AudioPaLM: A Large Language Model That Can Speak and Listen}},
author = {Rubenstein, Paul K. and Asawaroengchai, Chulayuth and Nguyen, Duc Dung and Bapna, Ankur and Borsos, Zalan and Quitry, Felix de Chaumont and Chen, Peter and Badawy, Dalia El and Han, Wei and Kharitonov, Eugene and Muckenhirn, Hannah and Padfield, Dirk and Qin, James and Rozenberg, Danny and Sainath, Tara and Schalkwyk, Johan and Sharifi, Matt and Ramanovich, Michelle Tadmor and Tagliasacchi, Marco and Tudor, Alexandru and Velimirovic, Mihajlo and Vincent, Damien and Yu, Jiahui and Wang, Yongqiang and Zayats, Vicky and Zeghidour, Neil and Zhang, Yu and Zhang, Zhishuai and Zilka, Lukas and Frank, Christian},
year = {2023},
doi = {10.48550/arxiv.2306.12925},
eprint = {2306.12925},
archivePrefix = {arXiv},
url = {https://doi.org/10.48550/arxiv.2306.12925}
}References
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