Soundstream Impl
A PyTorch implementation of the SoundStream neural audio codec. Accepts only 16 kHz audio.
Source du modèle
- Type
- Other
- Taille du fichier
- 71,1 MB
- Version
- vRevision 939b5d8
- Mis à jour
- 10 oct. 2026
- Téléchargements
- 6
- Source
- Hugging Face
Description de la source
SoundStream
A PyTorch implementation of the SoundStream neural audio codec. Accepts only 16 kHz audio.
Encodes speech into discrete tokens (8 codebooks × 80 tokens/sec) and decodes them back to audio.
Metrics
Evaluated on LibriSpeech test-clean:
- STOI: 0.804
- NISQA: 2.276
Architecture
- Encoder: Causal convolutions with residual units and strided downsampling (2, 4, 5, 5 = 200x compression)
- Quantizer: Residual Vector Quantizer with 8 codebooks of 1024 entries each
- Decoder: Mirrored encoder with transposed convolutions
- Discriminator (training only): 3 multi-scale waveform discriminators + 1 STFT-based discriminator
Model parameters: 16 kHz, 32 channels, latent dim 512, codebook size 1024, 8 quantizers, 200x downsampling
Usage
import torchaudio
from transformers import AutoModel
# Load model
model = AutoModel.from_pretrained("timofeiiz/soundstream-impl", trust_remote_code=True)
model.eval()
waveform, sr = torchaudio.load("audio.wav")
assert sr == 16000 # Only 16 kHz sample rate is supported
# Encode to discrete tokens
indices = model.encode(waveform.unsqueeze(0)) # (1, 8, T)
# Decode back to audio
reconstructed = model.decode(indices, original_length=waveform.size(-1))
torchaudio.save("reconstructed.wav", reconstructed.squeeze(0).cpu(), 16000)
License
MIT
Sources
1 source
Sources
1 sourceVérifié 15 août
Artefacts du modèle
1 artefact
Artefacts du modèle
1 artefactExtraits de sources
2 extraits
Extraits de sources
2 extraits--- library_name: transformers tags: - audio - soundstream license: mit language: - en pipeline_tag: audio-to-audio --- # SoundStream A PyTorch implementation of the [SoundStream](https://arxiv.org/abs/2107.03312) neural audio codec. Accepts only 16 kHz audio. Encodes speech into discrete tokens (8 codebooks × 80 tokens/sec) and decodes them back to audio. ## Metrics Evaluated on LibriSpeech test-clean: - **STOI**: 0.804 - **NISQA**: 2.276 ## Architecture - **Encoder**: Causal convolutions with residual units and strided downsampling (2, 4, 5, 5 = 200x compression) - **Quantizer**: Residual Vector Quantizer with 8 codebooks of 1024 entries each - **Decoder**: Mirrored encoder with transposed convolutions - **Discriminator** (training only): 3 multi-scale waveform discriminators + 1 STFT-based discriminator **Model parameters**: 16 kHz, 32 channels, latent dim 512, codebook size 1024, 8 quantizers, 200x downsampling ## Usage ```python import torchaudio from transformers import AutoModel # Load model model = AutoModel.from_pretrained("timofeiiz/soundstream-impl", trust_remote_code=True) model.eval() waveform, sr = torchaudio.load("audio.wav") assert sr == 16000 # Only 16 kHz sample rate is supported # Encode to discrete tokens indices = model.encode(waveform.unsqueeze(0)) # (1, 8, T) # Decode back to audio reconstructed = model.decode(indices, original_length=waveform.size(-1)) torchaudio.save("reconstructed.wav", reconstructed.squeeze(0).cpu(), 16000) ``` ## License MIT
Source context: 6 downloads · 0 likes · Pipeline audio-to-audio · Library transformers · Repo timofeiiz/soundstream-impl
