Feature Extraction
Transformers
Safetensors
moss-audio-tokenizer
audio
audio-tokenizer
neural-codec
moss-tts-family
MOSS Audio Tokenizer
speech-tokenizer
trust-remote-code
custom_code
Instructions to use OpenMOSS-Team/MOSS-Audio-Tokenizer-v2 with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use OpenMOSS-Team/MOSS-Audio-Tokenizer-v2 with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("feature-extraction", model="OpenMOSS-Team/MOSS-Audio-Tokenizer-v2", trust_remote_code=True)# Load model directly from transformers import AutoModel model = AutoModel.from_pretrained("OpenMOSS-Team/MOSS-Audio-Tokenizer-v2", trust_remote_code=True, device_map="auto") - Notebooks
- Google Colab
- Kaggle
Commit ·
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Parent(s): f5a0bd8
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- README.md +7 -50
- images/arch.png +3 -0
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README.md
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@@ -21,7 +21,7 @@ This is the code for the 48khz stereo version of MOSS-Audio-Tokenizer presented
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**Key Features:**
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* **Extreme Compression & Variable Bitrate**: It compresses 48kHz stereo audio into a remarkably low frame rate of 12.5Hz. Utilizing a 32-layer Residual Vector Quantization stack, it supports high-fidelity reconstruction across a wide range of bitrates.
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* **Pure Transformer Architecture**: The model features a "CNN-free" homogeneous architecture built entirely from Causal Transformer blocks. With
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* **Large-Scale General Audio Training**: Trained on 3 million hours of diverse audio data, the model excels at encoding and reconstructing all audio domains, including speech, sound effects, and music.
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* **Unified Semantic-Acoustic Representation**: While achieving state-of-the-art reconstruction quality, Cat produces discrete tokens that are "semantic-rich," making them ideal for downstream tasks like speech understanding (ASR) and generation (TTS).
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* **Fully Trained From Scratch**: Cat does not rely on any pretrained encoders (such as HuBERT or Whisper) or distillation from teacher models. All representations are learned autonomously from raw data.
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`transformers.models.moss_audio_tokenizer` module. It is hosted as a Hugging Face Hub model repository and should be
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loaded with `trust_remote_code=True`.
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- **Quantization:** 32-layer residual vector quantization stack.
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- **Checkpoint size:** the safetensors index reports 2,123,701,248 total parameters.
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- **Weight format:** sharded `safetensors` weights with a `model.safetensors.index.json` index.
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## Intended Use
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MOSS-Audio-Tokenizer-v2 is intended for research and development on audio tokenization, neural codec reconstruction,
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native audio foundation models, speech/audio understanding, speech generation, and related downstream modeling. It can
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encode 48 kHz stereo waveforms into discrete audio codes and decode those codes back to waveforms.
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This model is not intended for use in applications that impersonate a real person, reproduce private or copyrighted
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audio without permission, or make high-stakes decisions from reconstructed audio without additional validation.
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## Training Data And Procedure
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The model was trained from scratch on 3 million hours of diverse audio data, covering speech, sound effects, and music,
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as described in the accompanying paper. The training pipeline jointly optimizes the encoder, quantizer, decoder,
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discriminator, and a decoder-only LLM used for semantic alignment.
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The full training data mixture is not included in this repository. For details on dataset composition, filtering, and
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training/evaluation methodology, refer to the paper.
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## Evaluation
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The model is designed to provide high-fidelity reconstruction and semantically rich discrete representations across
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speech, sound effects, and music. Please refer to the paper for the full benchmark setup and quantitative results.
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## Limitations
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- Audio outside the 48 kHz stereo setting may require resampling and channel conversion before inference.
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- Reconstruction quality depends on audio domain, signal quality, selected number of RVQ layers, and inference settings.
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- The repository uses custom Transformers remote code, so users should review the code and pin a trusted revision in
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production deployments.
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- `flash_attention_2` is optional; if it is unavailable, use the default `sdpa` attention implementation.
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## Requirements
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- Python 3.10 or newer.
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- PyTorch.
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- Transformers. This checkpoint was prepared with `transformers_version` set to `4.56.0.dev0`; use a recent Transformers
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build that supports custom remote-code models.
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- `torchaudio` for the examples below.
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- Optional: `flash-attn` if using `model.set_attention_implementation("flash_attention_2")`.
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## Usage
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**Key Features:**
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* **Extreme Compression & Variable Bitrate**: It compresses 48kHz stereo audio into a remarkably low frame rate of 12.5Hz. Utilizing a 32-layer Residual Vector Quantization stack, it supports high-fidelity reconstruction across a wide range of bitrates.
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* **Pure Transformer Architecture**: The model features a "CNN-free" homogeneous architecture built entirely from Causal Transformer blocks. With 2B combined parameters (Encoder + Decoder), it ensures exceptional scalability and supports low-latency streaming inference.
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* **Large-Scale General Audio Training**: Trained on 3 million hours of diverse audio data, the model excels at encoding and reconstructing all audio domains, including speech, sound effects, and music.
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* **Unified Semantic-Acoustic Representation**: While achieving state-of-the-art reconstruction quality, Cat produces discrete tokens that are "semantic-rich," making them ideal for downstream tasks like speech understanding (ASR) and generation (TTS).
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* **Fully Trained From Scratch**: Cat does not rely on any pretrained encoders (such as HuBERT or Whisper) or distillation from teacher models. All representations are learned autonomously from raw data.
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`transformers.models.moss_audio_tokenizer` module. It is hosted as a Hugging Face Hub model repository and should be
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loaded with `trust_remote_code=True`.
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<br>
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<p align="center">
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<img src="images/arch.png" width="95%"> <br>
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Architecture of MossAudioTokenizer
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</p>
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## Usage
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