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README.md CHANGED
@@ -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 1.6B 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.
@@ -34,55 +34,12 @@ This repository contains a lightweight remote-code implementation that mirrors t
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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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- ## Model Details
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-
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- - **Architecture:** Cat (Causal Audio Tokenizer with Transformer), a CNN-free neural audio codec/tokenizer.
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- - **Sampling rate:** 48 kHz.
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- - **Channels:** stereo public waveform interface.
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- - **Token frame rate:** 12.5 Hz.
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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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-
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- ## Intended Use
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-
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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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-
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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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-
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- ## Training Data And Procedure
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-
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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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-
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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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-
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- ## Evaluation
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-
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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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-
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- ## Limitations
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-
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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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-
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- ## Requirements
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-
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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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+ <br>
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
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  ## Usage
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images/arch.png ADDED

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