High Fidelity Neural Audio Compression
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/2210.13438
Summary
Lossy audio codecs must lower bitrate without audible degradation, but neural codecs risk artifacts on unfamiliar signals and can be computationally heavy. The paper introduces EnCodec, an end-to-end trained neural codec. It pairs a convolutional encoder with a mirrored decoder, both using LSTM layers, and discretizes the latent output with Residual Vector Quantization (RVQ). Training combines time- and frequency-domain reconstruction terms with one multi-scale STFT discriminator (MS-STFTD). This single adversary suppresses artifacts and trains faster than stacking several discriminators. A new gradient balancer treats each loss weight as that loss's share of the total gradient, so weights no longer depend on loss magnitude and training becomes more stable. An optional small Transformer language model then entropy-codes the discrete codes, reducing bandwidth by up to 40%. In MUSHRA listening tests on clean speech, noisy-reverberant speech and music, EnCodec beats baselines such as Opus for both 24 kHz mono and 48 kHz stereo audio. Because it runs in real time on a single CPU core and its code and models are public, it is both a practical codec and a source of discrete audio representations.
Key points
- Supported bitrates are 1.5, 3, 6 and 12 kbps for 24 kHz mono and 6, 12 and 24 kbps for 48 kHz stereo. The encoder produces 75 latent frames per second at 24 kHz and 150 at 48 kHz.
- On 48 kHz stereo music, EnCodec at 6 kbps scores 82.9 MUSHRA. This compares with 17.7 for Opus at 6 kbps and 82.7 for MP3 at 64 kbps. At 12 kbps EnCodec reaches 88.0, roughly matching its 24 kbps result.
- Entropy coding with the Transformer saves about 20–30% on stereo music and up to 40% overall.
- The streaming 24 kHz model has an initial latency of about 13 ms and runs roughly 10× faster than real time on one MacBook Pro 2019 CPU thread, though it is slower than Lyra v2. The non-streaming 48 kHz model has 1 s latency and drops below real time when entropy coding is used.
- Ablations show that MS-STFTD alone is enough for high-quality output, with adding MPD giving only a slight gain. Code and models are released at github.com/facebookresearch/encodec.
Cited by
- 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
- 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{defossez2022high,
title = {{High Fidelity Neural Audio Compression}},
author = {Defossez, Alexandre and Copet, Jade and Synnaeve, Gabriel and Adi, Yossi},
year = {2022},
doi = {10.48550/arxiv.2210.13438},
eprint = {2210.13438},
archivePrefix = {arXiv},
url = {https://doi.org/10.48550/arxiv.2210.13438}
}References
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