FireRedTeam 开源 FireRedAudio 通用音频语言模型
GitHub - FireRedTeam/FireRedAudio · GitHub
FireRedTeam 开源 FireRedAudio,一个基于共享 9B 参数 LLM 的通用音频语言模型,采用解耦的连续表示,Audio Encoder 负责理解、RedAE 通路负责生成。单模型覆盖 ASR、音频理解、零样本 TTS、Instruct TTS、语义与声学语音编辑,并支持对最长约一小时录音做精确时间对齐。项目以 Apache-2.0 许可发布,需 Python 3.10 与 uv,默认面向 CUDA 12.8;除 ASR 外,语音生成与音频理解仅支持中英文。
推荐理由
精选AI 评分 60
FireRedAudio 用共享 9B 主干加解耦表示把 ASR、理解、TTS 与语音编辑收进一个模型,可据此判断统一音频栈的可行路径。
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当前显示原文,中文全文译文尚未就绪。
FireRedTeam; FireRedAudio
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One model to listen, understand, reason, speak, and edit.
FireRedAudio is a general-purpose audio language model built on a shared 9B-parameter LLM with decoupled continuous representations: an Audio Encoder handles understanding, while a RedAE pathway handles generation. A single model supports ASR, audio understanding, zero-shot TTS, instruct TTS, semantic/acoustic speech editing, and accurate temporal grounding over recordings up to one hour long.
Highlights ✨
🧩 Purpose-built representations, one shared backbone — The Audio Encoder pathway serves understanding, while the RedAE-Patch pathway serves speech generation. Their representations remain decoupled but share the same language and reasoning backbone. To the best of our knowledge, this is the first publicly disclosed design of its kind in a unified audio-language model; 📊 One model, a full audio stack — FireRedAudio spans ASR, broad and fine-grained audio understanding, zero-shot TTS, Instruct TTS, and free-form speech editing, achieving competitive or leading results across MMAU, MMSU, Seed-TTS-Eval, InstructTTSEval, and Ming-Freeform-Audio-Edit; 🎙️ Create and edit speech with natural language — Clone a voice from a reference clip, design a voice from a description, or edit what was said and how it sounds through one continuous-latent generation pathway; ⏱️ Go from minutes to hour-long recordings — Understand recordings up to one hour with precise time-to-content alignment. Organize audio into timestamped structures, produce grounded summaries, retrieve content by time (or time by content), and reason over evidence distributed across the recording
[2026.09.08] We release a Gradio web demo for zero-shot TTS, instruct TTS, speech editing, ASR, and audio understanding; [2026.08.21] We release the FireRedAudio code and model
Requires Python 3.10 and uv. System prerequisites: a CUDA toolkit (GPU inference + compiling the causal-conv1d / flash-attn kernels) and ffmpeg (audio decoding via torchaudio/torchcodec).
Wheels target CUDA 12.8 (cu128) by default. For a different CUDA version, change the pytorch-cu128 index URL in pyproject.toml (e.g. cu126 / cu129) and re-run uv sync.
Model Download
Download the pretrained model from Hugging Face with the hf CLI:
Alternatively, download the pretrained model using modelscope CLI:
A single-file Gradio app at app.py wraps all four tasks (zero-shot TTS · instruct TTS · speech editing · listen & understand) with the same weights and prompt templates as the CLI. Gradio is the only extra dep on top of the base install:
Defaults to http://0.0.0.0:7860. The understanding tab exposes the full Qwen3 sampling controls (temperature, top_p, top_k, min_p, repetition_penalty, do_sample) with a think-preset toggle that swaps in Qwen3's recommended (temperature, top_p) for reasoning / non-reasoning modes. If you're serving behind a reverse proxy, add --root-path .
Performance
Audio Understanding
Results marked with * are obtained from our own evaluation
ASR
Results marked with * are obtained from our own evaluation
Zero-Shot TTS
Instruct TTS
These results are obtained from our own evaluation
Third-party ports and integrations of FireRedAudio, maintained by the community.
Limitations
Everything but ASR is Chinese/English only. Speech generation (tts / edit / voice_design) and audio understanding are limited to Chinese and English — tts selects the language via --language zh / en. ASR is the only task that supports more languages; Zero-shot TTS is not deterministic by default. The flow-matching decoder samples random noise, so output varies run to run; pass a fixed seed (set_seed(...) in the API, --seed on the CLI) for reproducibility, and note that quality can differ across seeds; Long-form input is supported up to about one hour. Beyond that the model is untested and time-to-content alignment may degrade
Usage Disclaimer
The project incorporates zero-shot voice cloning functionality; Please note that this capability is intended solely for academic research purposes; DO NOT use this model for ANY illegal activities❗️❗️; The developers assume no liability for any misuse of this model; If you identify any instances of abuse, misuse, or fraudulent activities related to this project, please report them to our team immediately
Citation
Acknowledgements
Qwen3.5 for the language model foundation; Whisper-large-v3 for the Audio Encoder initialization; x-transformers for RotaryEmbedding; vocos for ISTFT implementation
License
Released under the Apache-2.0 license.