The pretrained model checkpoints can be reached at 🤗 Hugging Face and 🤖 Model Scope, or will be automatically downloaded when running inference scripts.
Currently support 30s for a single generation, which is the total length including both prompt and output audio. However, you can provide infer_cli
and infer_gradio
with longer text, will automatically do chunk generation. Long reference audio will be clip short to ~15s.
To avoid possible inference failures, make sure you have seen through the following instructions.
- Use reference audio <15s and leave some silence (e.g. 1s) at the end. Otherwise there is a risk of truncating in the middle of word, leading to suboptimal generation.
- Uppercased letters will be uttered letter by letter, so use lowercased letters for normal words.
- Add some spaces (blank: " ") or punctuations (e.g. "," ".") to explicitly introduce some pauses.
- Preprocess numbers to Chinese letters if you want to have them read in Chinese, otherwise in English.
Currently supported features:
- Basic TTS with Chunk Inference
- Multi-Style / Multi-Speaker Generation
- Voice Chat powered by Qwen2.5-3B-Instruct
The cli command f5-tts_infer-gradio
equals to python src/f5_tts/infer/infer_gradio.py
, which launches a Gradio APP (web interface) for inference.
The script will load model checkpoints from Huggingface. You can also manually download files and update the path to load_model()
in infer_gradio.py
. Currently only load TTS models first, will load ASR model to do transcription if ref_text
not provided, will load LLM model if use Voice Chat.
Could also be used as a component for larger application.
import gradio as gr
from f5_tts.infer.infer_gradio import app
with gr.Blocks() as main_app:
gr.Markdown("# This is an example of using F5-TTS within a bigger Gradio app")
# ... other Gradio components
app.render()
main_app.launch()
The cli command f5-tts_infer-cli
equals to python src/f5_tts/infer/infer_cli.py
, which is a command line tool for inference.
The script will load model checkpoints from Huggingface. You can also manually download files and use --ckpt_file
to specify the model you want to load, or directly update in infer_cli.py
.
For change vocab.txt use --vocab_file
to provide your vocab.txt
file.
Basically you can inference with flags:
# Leave --ref_text "" will have ASR model transcribe (extra GPU memory usage)
f5-tts_infer-cli \
--model "F5-TTS" \
--ref_audio "ref_audio.wav" \
--ref_text "The content, subtitle or transcription of reference audio." \
--gen_text "Some text you want TTS model generate for you."
# Choose Vocoder
f5-tts_infer-cli --vocoder_name bigvgan --load_vocoder_from_local --ckpt_file <YOUR_CKPT_PATH, eg:ckpts/F5TTS_Base_bigvgan/model_1250000.pt>
f5-tts_infer-cli --vocoder_name vocos --load_vocoder_from_local --ckpt_file <YOUR_CKPT_PATH, eg:ckpts/F5TTS_Base/model_1200000.safetensors>
And a .toml
file would help with more flexible usage.
f5-tts_infer-cli -c custom.toml
For example, you can use .toml
to pass in variables, refer to src/f5_tts/infer/examples/basic/basic.toml
:
# F5-TTS | E2-TTS
model = "F5-TTS"
ref_audio = "infer/examples/basic/basic_ref_en.wav"
# If an empty "", transcribes the reference audio automatically.
ref_text = "Some call me nature, others call me mother nature."
gen_text = "I don't really care what you call me. I've been a silent spectator, watching species evolve, empires rise and fall. But always remember, I am mighty and enduring."
# File with text to generate. Ignores the text above.
gen_file = ""
remove_silence = false
output_dir = "tests"
You can also leverage .toml
file to do multi-style generation, refer to src/f5_tts/infer/examples/multi/story.toml
.
# F5-TTS | E2-TTS
model = "F5-TTS"
ref_audio = "infer/examples/multi/main.flac"
# If an empty "", transcribes the reference audio automatically.
ref_text = ""
gen_text = ""
# File with text to generate. Ignores the text above.
gen_file = "infer/examples/multi/story.txt"
remove_silence = true
output_dir = "tests"
[voices.town]
ref_audio = "infer/examples/multi/town.flac"
ref_text = ""
[voices.country]
ref_audio = "infer/examples/multi/country.flac"
ref_text = ""
You should mark the voice with [main]
[town]
[country]
whenever you want to change voice, refer to src/f5_tts/infer/examples/multi/story.txt
.
To test speech editing capabilities, use the following command:
python src/f5_tts/infer/speech_edit.py
To communicate with socket server you need to run
python src/f5_tts/socket_server.py
Then create client to communicate
import socket
import numpy as np
import asyncio
import pyaudio
async def listen_to_voice(text, server_ip='localhost', server_port=9999):
client_socket = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
client_socket.connect((server_ip, server_port))
async def play_audio_stream():
buffer = b''
p = pyaudio.PyAudio()
stream = p.open(format=pyaudio.paFloat32,
channels=1,
rate=24000, # Ensure this matches the server's sampling rate
output=True,
frames_per_buffer=2048)
try:
while True:
chunk = await asyncio.get_event_loop().run_in_executor(None, client_socket.recv, 1024)
if not chunk: # End of stream
break
if b"END_OF_AUDIO" in chunk:
buffer += chunk.replace(b"END_OF_AUDIO", b"")
if buffer:
audio_array = np.frombuffer(buffer, dtype=np.float32).copy() # Make a writable copy
stream.write(audio_array.tobytes())
break
buffer += chunk
if len(buffer) >= 4096:
audio_array = np.frombuffer(buffer[:4096], dtype=np.float32).copy() # Make a writable copy
stream.write(audio_array.tobytes())
buffer = buffer[4096:]
finally:
stream.stop_stream()
stream.close()
p.terminate()
try:
# Send only the text to the server
await asyncio.get_event_loop().run_in_executor(None, client_socket.sendall, text.encode('utf-8'))
await play_audio_stream()
print("Audio playback finished.")
except Exception as e:
print(f"Error in listen_to_voice: {e}")
finally:
client_socket.close()
# Example usage: Replace this with your actual server IP and port
async def main():
await listen_to_voice("my name is jenny..", server_ip='localhost', server_port=9998)
# Run the main async function
asyncio.run(main())