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LongCat-Audio-Codec Open Source: An Extremely Low-Bitrate Audio Codec for Large Speech Models

LongCat-Audio-Codec Open Source: An Extremely Low-Bitrate Audio Codec for Large Speech Models

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I. Summary

LongCat-Audio-Codec is an open-source audio codec solution developed by the Meituan LongCat team, optimized for the Speech Large Scale Model (LLM). This project utilizes a dual-token architecture to concurrently model semantic and acoustic information, maintaining speech intelligibility and quality at an ultra-low bitrate of just 0.43 kbps. Its real-time streaming decoder maintains latency in the hundreds of milliseconds, supporting voice interaction and embedded deployment. The decoder's integrated super-resolution module further enhances sound quality without requiring additional models, significantly reducing the resource overhead of end-to-end speech systems.

2. Core Features

1. Dual-Token Parallel Encoding : Simultaneously extracts semantic and acoustic tokens, achieving efficient feature modeling at a low frame rate of 16.7 Hz (60 ms).

2. Extremely low bitrate and high-fidelity reconstruction : Maintains high intelligibility at 0.43 kbps, significantly improving bandwidth utilization.

3. Real-time low-latency decoding : Using a streaming architecture, the overall latency is maintained at hundreds of milliseconds, meeting the needs of real-time speech generation and interaction.

4. Decoding-side super-resolution enhancement : An integrated super-resolution module improves sound quality details without the need for an external model.

5. Lightweight and mobile optimization : Architectural optimization to address the computing power limitations of embedded and mobile devices.

3. Installation

1. Clone repository: git clone https://github.com/meituan-longcat/LongCat-Audio-Codec

2. Installation dependency: pip install -r requirements.txt

3. Load the model: You can download the corresponding weights of meituan-longcat/LongCat-Audio-Codec through Hugging Face.

  1. Run the example: Execute the inference script in the repository to perform encoding and decoding verification.

Typical Use Cases

  1. Front-end compression of large speech models: reducing input bandwidth while maintaining intelligibility.
  2. Real-time voice interaction system: Achieve low-latency transmission in conversational AI or voice assistants.
  3. Speech synthesis on edge and mobile devices: Generate or decode speech locally.
  4. Long-distance voice communication: Maintain clear voice transmission quality in extremely low-bandwidth environments.

5. Ecosystem and Competitive Products

1. Ecosystem Integration : LongCat-Audio-Codec is part of the Meituan LongCat series ecosystem, and works collaboratively with models such as LongCat-Flash to optimize speech generation and understanding.

2. Comparison with competitors : Compared with neural codec solutions such as SemantiCodec, UniCodec, and LMCodec, LongCat-Audio-Codec achieves lower bit rates and stronger real-time performance in the voice field.

3. Industry significance : Lowers the deployment threshold of voice LLM and provides infrastructure support for mobile AI assistants and voice services.

VI. Limitations and Precautions

  1. Even at extremely low bit rates, the sound quality may still suffer from loss of details.
  2. Streaming decoding has high requirements for hardware real-time performance.
  3. Different model versions may have a trade-off between latency and sound quality.
  4. Integrating a super-resolution module will increase the computational burden.

7. Project Address

https://github.com/meituan-longcat/LongCat-Audio-Codec

8. Frequently Asked Questions

Q: Does LongCat-Audio-Codec support offline deployment?

A: It can be run completely offline, but you need to prepare the corresponding model weights and dependent environment.

Q: How to integrate this codec on mobile devices?

A: It can be ported to mobile or embedded platforms through quantized models or lightweight inference frameworks.

Q: Can it be used for non-speech audio?

A: The current version is mainly optimized for voice tasks, and other types of audio require additional training.

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