SoundStream: An End-to-End Neural Audio Codec
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/2107.03312
Summary
Classic audio codecs involve a trade-off. Waveform codecs handle any content but produce artifacts at low bitrates. Parametric codecs stay efficient by assuming the input is speech. Earlier neural codecs such as LPCNet and Lyra also focused only on low-bitrate speech. Zeghidour et al. (Google) present SoundStream, a learned codec for speech, music and general audio at 24 kHz. A fully convolutional encoder turns the waveform into lower-rate embeddings. A residual vector quantizer (RVQ), extending the VQ-VAE quantizer, discretizes them. A convolutional decoder then rebuilds the signal. All three parts are trained jointly using reconstruction losses plus adversarial losses from discriminators, following MelGAN/HiFiGAN-style generators. Randomly dropping quantizer stages during training lets one model serve many bitrates. Causal convolutions keep latency low enough for streaming. In listening tests, SoundStream at 3 kbps beat Opus at 12 kbps and came close to EVS at 9.6 kbps. This shows that end-to-end learned compression can beat strong standard codecs across content types while running on a phone.
Key points
- One model covers bitrates from 3 to 18 kbps thanks to "quantizer dropout" on RVQ layers. Its quality is nearly the same as models trained for a single fixed bitrate.
- At 24 kHz, SoundStream at 3 kbps scores above Opus at 12 kbps in subjective tests and nears EVS at 9.6 kbps.
- The architecture is streamable. Its latency depends only on the encoder's downsampling ratio, and it runs in real time on a single smartphone CPU thread.
- Learning the encoder, instead of using fixed mel-spectrogram features as Lyra does, gives a large gain in coding efficiency.
- An optional conditioning signal lets the codec also suppress background noise in speech, at either the encoder or the decoder, with no extra latency.
Cited by
- AudioLM: a Language Modeling Approach to Audio Generation
- Qwen-Audio: Advancing Universal Audio Understanding via Unified Large-Scale Audio-Language Models
- Seamless: Multilingual Expressive and Streaming Speech Translation
- SeamlessM4T: Massively Multilingual & Multimodal Machine Translation
- High Fidelity Neural Audio Compression
- Moshi: a speech-text foundation model for real-time dialogue
- A Language Modeling Approach to Diacritic-Free Hebrew TTS
- Enhancing TTS Stability in Hebrew using Discrete Semantic Units
- AudioPaLM: A Large Language Model That Can Speak and Listen
- Neural Codec Language Models are Zero-Shot Text to Speech Synthesizers
Cite
@misc{zeghidour2021soundstream,
title = {{SoundStream: An End-to-End Neural Audio Codec}},
author = {Zeghidour, Neil and Luebs, Alejandro and Omran, Ahmed and Skoglund, Jan and Tagliasacchi, Marco},
year = {2021},
doi = {10.48550/arxiv.2107.03312},
eprint = {2107.03312},
archivePrefix = {arXiv},
url = {https://doi.org/10.48550/arxiv.2107.03312}
}References
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