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Audio AIIntermediate

F5-TTS

The F5-TTS model, developed by researchers at Shanghai Jiao Tong University and the Hong Kong University of Science and Technology and released in October 2024, is a next-generation non-autoregressive text-to-speech model that combines Flow Matching technology with the Diffusion Transformer (DiT) architecture. This model offers zero-shot voice cloning capabilities, allowing it to precisely replicate the target's timbre, intonation, emotional tone, and ambient acoustic environment using only a short reference audio sample of 3 to 5 seconds.

F5-TTS, announced in October 2024 by researchers at Shanghai Jiao Tong University and the Hong Kong University of Science and Technology, is a next-generation non-autoregressive text-to-speech (TTS) model that combines Flow Matching technology with the Diffusion Transformer (DiT) architecture. This model offers zero-shot voice cloning capabilities, precisely replicating the target's timbre, intonation, emotional tone, and ambient acoustic environment using only a short reference speech sample of 3 to 5 seconds. Unlike conventional autoregressive models that sequentially generate words and tokens, limiting processing speed, F5-TTS simplifies the entire text input and speech waveform generation process into a single architecture, significantly improving both generation efficiency and naturalness.

Traditional speech synthesis systems required complex phoneme aligners, complex duration models, and separate text encoder modules connected in a multi-stage pipeline, leading to high system complexity and computational costs, as well as frequent pronunciation errors and information loss at intermediate stages. Similar to how an artist smoothly applies paint along guidelines on a canvas, F5-TTS fills in filler tokens to match the text length and then applies Continuous Normalizing Flow technology to smoothly transition from a noise state to the target speech features. By incorporating a ConvNeXt text refinement block and a Sway Sampling technique that optimizes computational load during inference, it achieves natural multilingual code-switching and precise control over speech rate without requiring a phoneme dictionary or complex pre-processing pipeline.

In biomedical research, clinical settings, and multilingual academic presentation environments, F5-TTS can be effectively used to translate academic lectures into multiple languages while preserving the original speaker's voice or to develop assistive technology tools for individuals with speech impairments. By inputting 5 seconds of reference speech data (.wav) collected by the researcher and the academic text to be translated, the model quickly synthesizes speech with a real-time factor (RTF) of 0.15, enabling near real-time inference even in consumer-grade GPU environments. For example, when processing a descriptive text of 1024 tokens, setting 30 steps of refined Sway Sampling parameters allows for the immediate generation of high-quality audio files in a clean environment with background noise removed and without intonation distortion.

Furthermore, F5-TTS is built on a large-scale training dataset of hundreds of hours, enabling it to generate stable frequency spectrograms even in scenarios involving mixed speech from various languages, such as Korean, English, and Chinese, or rapid changes in emotion. This lightweight, single-architecture-centric design can be easily ported to offline server environments or edge computing research environments, providing a powerful infrastructural advantage by allowing sensitive personal voice data or healthcare text to be processed securely in a local, standalone environment without transmitting it to external cloud APIs.

💻 System Requirements

🧠RAM

최소 6GB (NVIDIA GPU), 권장 12GB+ (RTX 3060 이상 추천). CPU 환경 시 생성 속도 저하 (RTF 1.5~3.0)

💾Storage

모델 체크포인트 ~1.5GB (Base 모델 기준), 패키지 및 전처리 디렉터리 5GB 이내

Installation

4-1. Quick Start

# PyTorch 환경 설정 후 pip 패키지 설치
pip install f5-tts

# CLI 환경에서 zero-shot 음성 합성 실행
f5-tts_infer-cli --model F5TTS_Base --ref_audio reference.wav --ref_text "참조 음성 텍스트" --gen_text "합성할 대상 텍스트"

4-2. 상세 설치

# GitHub 소스 코드 클론 및 가상환경 구축
git clone https://github.com/SWivid/F5-TTS.git
cd F5-TTS
python3 -m venv f5-env
source f5-env/bin/activate

# 개발 모드 패키지 설치
pip install -e .

# Gradio Web UI 인터페이스 실행
f5-tts_infer-gradio

🧬 Bio Use Cases

🔬

Multilingual Research and Academic Presentation Voice Synthesis

Input a 5-second sample of the speaker's original voice (ref.wav, 24kHz) and multilingual academic text to automatically generate English and Korean cross-lingual speech audio, maintaining the original speaker's intonation at an RTF of 0.15 speed with Sway Sampling and nfe=30.

🧬

Hearing/Speech Impairment Research and Digital Tutoring Scenario

Using noise-reduced 3-second patient voice text pairs as guidance, perform F5TTS_Base model inference to build a clear, conversational voice guide assistance pipeline without distortion of area and pitch variability metrics.

💊

Local Edge Environment On-Premise Integration

Without connecting to external Cloud APIs, perform Gradio UI-based offline standalone operation on a 12GB VRAM GPU device to safely synthesize speech waveforms from personal information-protected text data, keeping 100% of the data on-premise.

FAQ

What is F5-TTS?

F5-TTS, announced in October 2024 by researchers at Shanghai Jiao Tong University and the Hong Kong University of Science and Technology, is a next-generation non-autoregressive text-to-speech (TTS) model that combines Flow Matching technology with the Diffusion Transformer (DiT) architecture. This model offers zero-shot voice cloning capabilities, precisely replicating the target's timbre, intonation, emotional tone, and ambient acoustic environment using only a short reference speech sample of 3 to 5 seconds. Unlike conventional autoregressive models that sequentially generate words and tokens, limiting processing speed, F5-TTS simplifies the entire text input and speech waveform generation process into a single architecture, significantly improving both generation efficiency and naturalness. Traditional speech synthesis systems required complex phoneme aligners, complex duration models, and separate text encoder modules connected in a multi-stage pipeline, leading to high system complexity and computational costs, as well as frequent pronunciation errors and information loss at intermediate stages. Similar to how an artist smoothly applies paint along guidelines on a canvas, F5-TTS fills in filler tokens to match the text length and then applies Continuous Normalizing Flow technology to smoothly transition from a noise state to the target speech features. By incorporating a ConvNeXt text refinement block and a Sway Sampling technique that optimizes computational load during inference, it achieves natural multilingual code-switching and precise control over speech rate without requiring a phoneme dictionary or complex pre-processing pipeline. In biomedical research, clinical settings, and multilingual academic presentation environments, F5-TTS can be effectively used to translate academic lectures into multiple languages while preserving the original speaker's voice or to develop assistive technology tools for individuals with speech impairments. By inputting 5 seconds of reference speech data (.wav) collected by the researcher and the academic text to be translated, the model quickly synthesizes speech with a real-time factor (RTF) of 0.15, enabling near real-time inference even in consumer-grade GPU environments. For example, when processing a descriptive text of 1024 tokens, setting 30 steps of refined Sway Sampling parameters allows for the immediate generation of high-quality audio files in a clean environment with background noise removed and without intonation distortion. Furthermore, F5-TTS is built on a large-scale training dataset of hundreds of hours, enabling it to generate stable frequency spectrograms even in scenarios involving mixed speech from various languages, such as Korean, English, and Chinese, or rapid changes in emotion. This lightweight, single-architecture-centric design can be easily ported to offline server environments or edge computing research environments, providing a powerful infrastructural advantage by allowing sensitive personal voice data or healthcare text to be processed securely in a local, standalone environment without transmitting it to external cloud APIs.

When should I use F5-TTS?

The F5-TTS model, developed by researchers at Shanghai Jiao Tong University and the Hong Kong University of Science and Technology and released in October 2024, is a next-generation non-autoregressive text-to-speech model that combines Flow Matching technology with the Diffusion Transformer (DiT) architecture. This model offers zero-shot voice cloning capabilities, allowing it to precisely replicate the target's timbre, intonation, emotional tone, and ambient acoustic environment using only a short reference audio sample of 3 to 5 seconds.

What is a biomedical use case for F5-TTS?

Multilingual Research and Academic Presentation Voice Synthesis: Input a 5-second sample of the speaker's original voice (ref.wav, 24kHz) and multilingual academic text to automatically generate English and Korean cross-lingual speech audio, maintaining the original speaker's intonation at an RTF of 0.15 speed with Sway Sampling and nfe=30.

📄 Official Docs🐙 GitHub

📝 Update Notes

  1. v1.1.227/23/2026

    F5-TTS 1.1.22 업데이트에서는 음성 생성 시 청크 단위의 지속 시간이 비정상적으로 늘어나는 버그가 해결되었습니다. 덕분에 연구 결과 발표용 영상이나 실험 애니메이션을 제작할 때, 음성과 화면의 싱크를 더욱 정밀하게 맞출 수 있게 되었습니다. 데이터 시각화 자료에 정확한 타이밍의 음성 설명을 입히고자 하는 생명공학 연구원분들께 이번 업데이트를 추천합니다.

  2. v1.1.217/22/2026

    이번 F5-TTS 1.1.21 업데이트는 보안 취약점 해결과 하드웨어 호환성 개선에 집중했습니다. 미세 조정(finetune) 과정의 보안 패치를 통해 민감한 연구 데이터를 다루는 연구원분들이 더욱 안전하게 모델을 학습시킬 수 있습니다. 또한 최신 AMD GPU(RDNA 3.5/4) 지원이 확대되어, 특정 하드웨어 환경을 사용하는 연구실의 연산 효율성을 높이는 데 도움이 됩니다.

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