# **DIAMOND**
**A sequence-to-sequence model for speech restoration via an autoregressive RQ-Transformer over neural audio codec tokens**
[**Diamond**](https://huggingface.co/nineninesix/diamond-1.0/resolve/main/TECH_REPORT.pdf) turns degraded audio into near-studio 44.1 kHz speech. A bidirectional Transformer encodes the degraded mel-spectrogram; an autoregressive decoder with cross-attention predicts the tokens of a frozen neural audio codec (Descript Audio Codec, 9-book RVQ), which synthesizes the restored waveform.
Restoration is not denoising. A codec cuts the band above 8 kHz, clipping cuts the peaks, a lost packet zeroes the frame — no mask can return what the observation no longer contains. The missing content has to be **generated** from the surviving formants, which is why Diamond is a generative seq2seq model rather than a filter.
Diamond is trained **from scratch** — no pretrained backbone — and reaches the level of the strongest open restorers while remaining the best from-scratch restorer in its class.
## Highlights
- **Two-transformer read-out.** A *time-transformer* models the frame sequence; a compact *depth-transformer* walks the 9 RVQ codebooks inside each frame. This restores the RVQ causal chain and distributes the output projection — where earlier models read all nine books out of a single frame vector with parallel heads.
- **From scratch, 166.6M trainable params.** No pretrained speech backbone, 63 GPU-hours of training for the released checkpoint.
- **44.1 kHz output.** The frozen DAC decoder synthesizes full-band audio; sibilants and "air" above 8 kHz are regenerated, not merely passed through.
- **Content is measured, not assumed.** A generative restorer can sound clean while smearing words, so CER against the ground-truth transcript is a mandatory second axis alongside DNSMOS.
- **Calibrated degradation.** Inputs come from a DSP augmentor whose codec palette is fitted per-sample to a real degraded-speech distribution, not from a bag of random effects.
## Hear the difference
Real, heavily degraded speech restored to 44.1 kHz. DNSMOS-P.835 OVRL rises by more
than a full point on each clip — audible even without headphones. Try it live in the
[**Diamond Space**](https://huggingface.co/spaces/nineninesix/Diamond).
#
Degraded input
Restored — 44.1 kHz
DNSMOS OVRL
1
2.16 → 3.50 (+1.34)
2
2.83 → 3.59 (+0.76)
3
2.56 → 3.65 (+1.10)
4
3.32 → 3.89 (+0.57)
## Model Details
- **Developed by:** [nineninesix.ai](https://www.nineninesix.ai)
- **Model type:** Autoregressive sequence-to-sequence speech restoration (encoder + RQ-Transformer decoder over codec tokens)
- **Encoder:** Bidirectional Transformer, no downsampling — 20 layers, 512d, 8 heads (63.9M)
- **Decoder / time-transformer:** causal self-attention + cross-attention — 20 layers, 512d, 8 heads (88.7M)
- **Codebook read-out / depth-transformer:** local AR over the 9 RVQ codes of a frame — 4 layers, 384d, 6 heads (14.0M)
- **Parameters:** ≈ 166.6M trainable (the DAC codec, ~74M, is frozen and downloaded at runtime)
- **Audio codec:** [Descript Audio Codec](https://github.com/descriptinc/descript-audio-codec) — 44.1 kHz, 9-codebook RVQ, 86 frames/s
- **Primitives:** RMSNorm, learnable per-layer RoPE, QK-Norm, LayerScale, GELU FFN
- **Input / output sample rate:** any input, resampled to 24 kHz internally → 44,100 Hz output
- **Training data:** 681.5 h of studio speech, ~2.5k speakers, 28% natively wide-band.
- **Languages:** English
- **License:** Apache 2.0
## Benchmarks
Measured on 750 real degraded recordings — identical clips for every model. DNSMOS-P.835 (raw `sig_bak_ovr.onnx`) for perceptual quality, CER against the ground-truth transcript for content preservation.
| Model | DNSMOS OVRL ↑ | SIG ↑ | BAK ↑ | CER (median) ↓ |
|---|---|---|---|---|
| Sidon | **3.923** | **4.129** | **4.416** | 0.016 |
| **Diamond (ours)** | 3.829 | 4.056 | 4.348 | 0.028 |
| RE-USE | 3.789 | 4.003 | 4.344 | **0.014** |
| Resemble Enhance | 3.764 | 3.994 | 4.301 | 0.027 |
| UniSE | 3.752 | 4.014 | 4.261 | 0.027 |
| VoiceFixer | 3.566 | 3.790 | 4.226 | 0.042 |
| input (degraded) | 3.575 | 3.890 | 4.082 | 0.014 |
Diamond improves ≈88% of clips (mean ΔOVRL +0.255) and places second of six on perceptual quality — ahead of RE-USE, which trains on roughly four times the data. Both systems that beat it on CER are **non-autoregressive**: the gap is the exposure bias inherent to AR decoding, not a capacity limit. The one other autoregressive system here, UniSE, lands on exactly Diamond's mean CER (0.131) by a different route — which suggests the tail belongs to the paradigm, not to this particular recipe.
## Links
- **Checkpoint:** [huggingface.co/nineninesix/diamond-1.0](https://huggingface.co/nineninesix/diamond-1.0)
- **Technical Report:** [TECH_REPORT.pdf](https://huggingface.co/nineninesix/diamond-1.0/resolve/main/TECH_REPORT.pdf)
- **Training:** [github.com/nineninesix-ai/diamond-train](https://github.com/nineninesix-ai/diamond-train)
- **Website:** [nineninesix.ai](https://www.nineninesix.ai)
## What it's good for
Offline restoration and **dataset cleansing** — turning large volumes of degraded recordings (podcasts, interviews, archival and user-generated audio that has passed through lossy codecs) into material clean enough to train on. This is the task Diamond was built for and the one it is measured on.
**Keep in mind:**
- **A thin CER tail.** As an autoregressive decoder, Diamond occasionally smears words on hard clips. Inference safeguards (chunked decoding, repetition penalty) hold it down, but the tail is real — check the transcript when content fidelity is critical.
- **Decoding is serial, not real-time.** One frame at a time, 86 frames per second of audio, plus the depth pass. This is offline restoration, not a live filter.
- **English-centric.** Training data is English; other languages are untested.
- **It generates, it does not filter.** Content above the codec's cutoff is invented from context — plausible, not recovered. Do not treat the output as forensic evidence of what was said.
> **Use it responsibly.** Restored speech is synthesized speech. Don't present it as an unaltered recording, and don't use it to fabricate or misattribute what someone said.
## Acknowledgments
Built on the [Descript Audio Codec](https://github.com/descriptinc/descript-audio-codec) for the frozen token space. Trained on [LibriTTS-R](https://www.openslr.org/141/), [Hi-Fi TTS](https://www.openslr.org/109/), [VCTK](https://doi.org/10.7488/ds/2645). Evaluated with [DNSMOS P.835](https://github.com/microsoft/DNS-Challenge) and [faster-whisper](https://github.com/SYSTRAN/faster-whisper).
## Citation
If you use this work in your research, please cite:
```bibtex
@software{diamond_2026,
author = {Almaz Zholdoshbek uulu, Ulanbek Abdurazakov, Denis Pavlov and Nursultan Bakashov},
title = {Diamond: A Sequence-to-Sequence Model for Speech Restoration via an Autoregressive RQ-Transformer over Neural Audio Codec Tokens},
year = {2026},
publisher = {Hugging Face},
howpublished = {\url{https://huggingface.co/nineninesix/diamond-1.0}},
note = {Trained from scratch; no pretrained backbone}
}
```
## References
```bibtex
@inproceedings{kumar2023dac,
title={High-Fidelity Audio Compression with Improved RVQGAN},
author={Kumar, Rithesh and Seetharaman, Prem and Luebs, Alejandro and Kumar, Ishaan and Kumar, Kundan},
booktitle={NeurIPS},
year={2023},
note={arXiv:2306.06546}
}
@inproceedings{lee2022rqtransformer,
title={Autoregressive Image Generation using Residual Quantization},
author={Lee, Doyup and Kim, Chiheon and Kim, Saehoon and Cho, Minsu and Han, Wook-Shin},
booktitle={CVPR},
year={2022},
note={arXiv:2203.01941}
}
@article{defossez2024moshi,
title={Moshi: a speech-text foundation model for real-time dialogue},
author={D{\'e}fossez, Alexandre and Mazar{\'e}, Laurent and Orsini, Manu and Royer, Am{\'e}lie and P{\'e}rez, Patrick and J{\'e}gou, Herv{\'e} and Grave, Edouard and Zeghidour, Neil},
journal={arXiv preprint arXiv:2410.00037},
year={2024}
}
@inproceedings{copet2023musicgen,
title={Simple and Controllable Music Generation},
author={Copet, Jade and Kreuk, Felix and Gat, Itai and Remez, Tal and Kant, David and Synnaeve, Gabriel and Adi, Yossi and D{\'e}fossez, Alexandre},
booktitle={NeurIPS},
year={2023},
note={arXiv:2306.05284}
}
@inproceedings{koizumi2023librittsr,
title={LibriTTS-R: A Restored Multi-Speaker Text-to-Speech Corpus},
author={Koizumi, Yuma and Zen, Heiga and Karita, Shigeki and Ding, Yifan and Yatabe, Kohei and Morioka, Nobuyuki and Bacchiani, Michiel and Zhang, Yu and Han, Wei and Bapna, Ankur},
booktitle={Interspeech},
year={2023},
note={arXiv:2305.18802}
}
@inproceedings{reddy2022dnsmos,
title={DNSMOS P.835: A Non-Intrusive Perceptual Objective Speech Quality Metric to Evaluate Noise Suppressors},
author={Reddy, Chandan K. A. and Gopal, Vishak and Cutler, Ross},
booktitle={ICASSP},
year={2022},
note={arXiv:2110.01763}
}
```
## License
Apache 2.0 — this model and its weights are released under the [Apache License 2.0](https://opensource.org/licenses/Apache-2.0).
The frozen [Descript Audio Codec](https://github.com/descriptinc/descript-audio-codec) is downloaded at runtime and carries its own license (MIT).