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.
๐ป System Requirements
Minimum: 6GB (NVIDIA GPU); Recommended: 12GB+ (RTX 3060 or higher recommended). On CPU, generation speed will be significantly slower (RTF 1.5~3.0).
Model checkpoint ~1.5GB (based on the base model), within the package and preprocessing directory 5GB
โก Installation
4-1. Quick Start
# Setting up PyTorch environment and installing pip packages
pip install f5-tts
# Execute zero-shot speech synthesis in the CLI environment
f5-tts_infer-cli --model F5TTS_Base --ref_audio reference.wav --ref_text "Reference voice text" --gen_text "Target text to synthesize"
4-2. Detailed installation
# Clone GitHub source code and set up virtual environment
git clone https://github.com/SWivid/F5-TTS.git
cd F5-TTS
python3 -m venv f5-env
source f5-env/bin/activate
# Install package in development mode
pip install -e .
# Run Gradio Web UI interface
f5-tts_infer-gradio
๐งฌ 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.
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