Model reference · open weights
dualcodec is an open-weight audio or speech model from amphion. AxForge deploys and operates it for you on dedicated EU-owned hardware — with the licence handled where one is required.
Available as managed deployment — configured and operated for you on dedicated EU hardware, quoted per deployment.
What it is
| Maker | amphion |
|---|---|
| Type | Audio & music |
| Task | Audio→audio |
| Runs with | dualcodec |
| Released | 2025-01-12 |
| Popularity | 0 downloads / month |
| Licence | Unknown |
About
DualCodec is a low-frame-rate (12.5Hz or 25Hz), semantically-enhanced (with SSL feature) Neural Audio Codec designed to extract discrete tokens for efficient speech generation.
You can check out its demo page. The overview of DualCodec system is shown in the following figure:
pip install dualcodec
| Model_ID | Frame Rate | RVQ Quantizers | Semantic Codebook Size (RVQ-1 Size) | Acoustic Codebook Size (RVQ-rest Size) | Training Data |
|---|---|---|---|---|---|
| 12hz_v1 | 12.5Hz | Any from 1-8 (maximum 8) | 16384 | 4096 | 100K hours Emilia |
| 25hz_v1 | 25Hz | Any from 1-12 (maximum 12) | 16384 | 1024 | 100K hours Emilia |
import dualcodec
model_id = "12hz_v1" # select from available Model_IDs, "12hz_v1" or "25hz_v1"
dualcodec_model = dualcodec.get_model(model_id)
dualcodec_inference = dualcodec.Inference(dualcodec_model=dualcodec_model, device="cuda")
# do inference for your wav
import torchaudio
audio, sr = torchaudio.load("YOUR_WAV.wav")
# resample to 24kHz
audio = torchaudio.functional.resample(audio, sr, 24000)
audio = audio.reshape(1,1,-1)
audio = audio.to("cuda")
# extract codes, for example, using 8 quantizers here:
semantic_codes, acoustic_codes = dualcodec_inference.encode(audio, n_quantizers=8)
# semantic_codes shape: torch.Size([B, 1, T])
# acoustic_codes shape: torch.Size([B, n_quantizers-1, T])
# produce output audio
out_audio = dualcodec_inference.decode(semantic_codes, acoustic_codes)
# save output audio
torchaudio.save("out.wav", out_audio.cpu().squeeze(0), 24000)
First, download checkpoints to local:
# export HF_ENDPOINT=https://hf-mirror.com # uncomment this to use huggingface mirror if you're in China
huggingface-cli download facebook/w2v-bert-2.0 --local-dir w2v-bert-2.0
huggingface-cli download amphion/dualcodec dualcodec_12hz_16384_4096.safetensors dualcodec_25hz_16384_1024.safetensors w2vbert2_mean_var_stats_emilia.pt --local-dir dualcodec_ckpts
The second command downloads the two DualCodec model (12hz_v1 and 25hz_v1) checkpoints and a w2v-bert-2 mean and variance statistics to the local directory dualcodec_ckpts.
Then you can use the following code to inference DualCodec with local checkpoints.
import dualcodec
w2v_path = "./w2v-bert-2.0" # your downloaded path
dualcodec_model_path = "./dualcodec_ckpts" # your downloaded path
model_id = "12hz_v1" # select from available Model_IDs, "12hz_v1" or "25hz_v1"
dualcodec_model = dualcodec.get_model(model_id, dualcodec_model_path)
dualcodec_inference = dualcodec.Inference(dualcodec_model=dualcodec_model, dualcodec_path=dualcodec_model_path, w2v_path=w2v_path, device="cuda")
# do inference for your wav
import torchaudio
audio, sr = torchaudio.load("YOUR_WAV.wav")
# resample to 24kHz
audio = torchaudio.functional.resample(audio, sr, 24000)
audio = audio.reshape(1,1,-1)
audio = audio.to("cuda")
# extract codes, for example, using 8 quantizers here:
semantic_codes, acoustic_codes = dualcodec_inference.encode(audio, n_quantizers=8)
# semantic_codes shape: torch.Size([1, 1, T])
# acoustic_codes shape: torch.Size([1, n_quantizers-1, T])
# produce output audio. If `acoustic_codes=None` is passed, will decode only semantic codes (RVQ-1)
out_audio = dualcodec_inference.decode(semantic_codes, acoustic_codes)
# save output audio
torchaudio.save("out.wav", out_audio.cpu().squeeze(0), 24000)
See "example.ipynb" for a running example.
The notebook provides a demo of reconstructing audios using different number of RVQ layers:
If you want to use the Gradio interface, you can run the following command:
python -m dualcodec.app
This will launch an app that allows you to upload a wav file and get the output wav file.
Models available:
To continue, first install other necessary components for training:
pip install "dualcodec[tts]"
Alternatively, if you want to install from source,
pip install -e .[tts]
DualCodec-VALLE is a TTS model based on DualCodec. It is trained with 12Hz sampling rate and 8 quantizers. The model is trained on 100K hours of Emilia data.
python -m dualcodec.infer.valle.cli_valle_infer --ref_audio --ref_text "TEXT OF YOUR REF AUDIO" --gen_text "This is the generated text" --output_dir test -
From the published model card. Full card on the HuggingFace links in the sidebar.
How it works
Using it via the API
Once AxForge deploys dualcodec for you, it answers on the OpenAI-compatible API — the same base URL and keys as every other model. (dualcodec below is illustrative; you get the exact model name on deployment.)
$ curl -sS https://api.axforge.ai/v1/audio/transcriptions \ -H "Authorization: Bearer $AXFORGE_API_KEY" \ -F model="dualcodec" -F file=@audio.mp3
Create an account — your API key is available in the console. 5M tokens/month currently included with every new account at launch.