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MOSS-TTS

MOSS-TTS, jointly developed by MOSI.AI and the OpenMOSS team and released in May 2026, is an innovative, locally executable, open-source text-to-speech engine that utilizes a Causal Audio Tokenizer with Transformer as its core architecture. Similar to how a Large Language Model tokenizes text words to complete sentences, this tool uniquely combines the relationship between natural language context and audio waveforms in a single step, using a CNN-free audio tokenizer, to achieve real-time performance.

MOSS-TTS, jointly developed by MOSI.AI and the OpenMOSS team and released in May 2026, is an innovative, locally executable, open-source text-to-speech engine that utilizes a Causal Audio Tokenizer with Transformer as its core architecture. Similar to how a Large Language Model tokenizes text words to complete sentences, this tool uniquely combines natural language context and audio waveforms in a single step through a CNN-free audio tokenizer, thereby achieving exceptional real-time text-to-speech performance. While conventional text-to-speech systems have faced limitations in latency and resource optimization due to complex multi-stage pipelines or heavy convolutional layers, MOSS-TTS adopts a highly lightweight transformer design, enabling smooth streaming output with extremely low latency using only local computing resources.

This model family serves as an optimal alternative for overcoming performance constraints in local desktop and edge environments and is designed to enable researcher-centric, autonomous analysis and transformation tasks without relying on expensive, proprietary cloud services. In particular, the stability of zero-shot voice cloning has been significantly improved, and users can explicitly inject pause control options during the synthesis process, ensuring ultra-high-quality voice output that closely resembles human breathing patterns and conversational tempo. These features provide an excellent foundation for creating multi-party voice agent interfaces or interactive research guides in local environments with minimal latency, where real-time responsiveness is paramount.

In practical research and experimental analysis workflows, MOSS-TTS offers unparalleled value, particularly in environments where confidentiality is emphasized and data security regulations are strict, such as academic literature exploration and laboratory automation guidance systems. For example, research abstracts extracted from a large-scale biomedical literature database can be converted into real-time multilingual audio content for listening, or real-time status changes and safety alert sounds from experimental equipment can be output with extremely low latency (in milliseconds) even on equipment with a single CPU, simultaneously increasing research efficiency and laboratory safety.

💻 System Requirements

🧠RAM

0 (MOSS-TTS-Nano CPU 추론 시) / NVIDIA GPU 8GB+ 권장 (RTX 3060/4060급 이상 실시간 서빙)

💾Storage

모델 가중치 용량 약 200MB ~ 3GB, 전체 패키지 환경 설치 공간 포함 5GB 이내

Installation

4-1. Quick Start

git clone https://github.com/OpenMOSS/MOSS-TTS-Nano.git cd MOSS-TTS-Nano && pip install -r requirements.txt

4-2. 상세 설치

1. Conda 가상 환경 생성 및 활성화

conda create -n moss-tts python=3.12 -y conda activate moss-tts

2. PyTorch 및 의존성 라이브러리 설치 (CUDA 환경 권장)

pip install --extra-index-url https://download.pytorch.org/whl/cu128 -e .

3. 모델 실행 및 단일 추론 예시

python infer.py
--prompt-audio-path assets/audio/zh_1.wav
--text "Welcome to the OpenMOSS TTS service."

FAQ

What is MOSS-TTS?

MOSS-TTS, jointly developed by MOSI.AI and the OpenMOSS team and released in May 2026, is an innovative, locally executable, open-source text-to-speech engine that utilizes a Causal Audio Tokenizer with Transformer as its core architecture. Similar to how a Large Language Model tokenizes text words to complete sentences, this tool uniquely combines natural language context and audio waveforms in a single step through a CNN-free audio tokenizer, thereby achieving exceptional real-time text-to-speech performance. While conventional text-to-speech systems have faced limitations in latency and resource optimization due to complex multi-stage pipelines or heavy convolutional layers, MOSS-TTS adopts a highly lightweight transformer design, enabling smooth streaming output with extremely low latency using only local computing resources. This model family serves as an optimal alternative for overcoming performance constraints in local desktop and edge environments and is designed to enable researcher-centric, autonomous analysis and transformation tasks without relying on expensive, proprietary cloud services. In particular, the stability of zero-shot voice cloning has been significantly improved, and users can explicitly inject pause control options during the synthesis process, ensuring ultra-high-quality voice output that closely resembles human breathing patterns and conversational tempo. These features provide an excellent foundation for creating multi-party voice agent interfaces or interactive research guides in local environments with minimal latency, where real-time responsiveness is paramount. In practical research and experimental analysis workflows, MOSS-TTS offers unparalleled value, particularly in environments where confidentiality is emphasized and data security regulations are strict, such as academic literature exploration and laboratory automation guidance systems. For example, research abstracts extracted from a large-scale biomedical literature database can be converted into real-time multilingual audio content for listening, or real-time status changes and safety alert sounds from experimental equipment can be output with extremely low latency (in milliseconds) even on equipment with a single CPU, simultaneously increasing research efficiency and laboratory safety.

When should I use MOSS-TTS?

MOSS-TTS, jointly developed by MOSI.AI and the OpenMOSS team and released in May 2026, is an innovative, locally executable, open-source text-to-speech engine that utilizes a Causal Audio Tokenizer with Transformer as its core architecture. Similar to how a Large Language Model tokenizes text words to complete sentences, this tool uniquely combines the relationship between natural language context and audio waveforms in a single step, using a CNN-free audio tokenizer, to achieve real-time performance.

📄 Official Docs🐙 GitHub

📝 Update Notes

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