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import argparse
import os
import sys
import tempfile
import logging
from pathlib import Path
import os
import shutil
import glob
import gradio as gr
import librosa.display
import numpy as np
from datetime import datetime
from pydub import AudioSegment
import pysrt
import torch
import torchaudio
import traceback
from utils.formatter import format_audio_list, find_latest_best_model
from utils.gpt_train import train_gpt
from TTS.tts.configs.xtts_config import XttsConfig
from TTS.tts.models.xtts import Xtts
from openvoice_cli.downloader import download_checkpoint
from openvoice_cli.api import ToneColorConverter
import openvoice_cli.se_extractor as se_extractor
logging.basicConfig(level=logging.INFO)
logger = logging.getLogger(__name__)
# Clear logs
def remove_log_file(file_path):
log_file = Path(file_path)
if log_file.exists() and log_file.is_file():
log_file.unlink()
# remove_log_file(str(Path.cwd() / "log.out"))
def clear_gpu_cache():
# clear the GPU cache
if torch.cuda.is_available():
torch.cuda.empty_cache()
XTTS_MODEL = None
def load_model(xtts_checkpoint, xtts_config, xtts_vocab,xtts_speaker):
global XTTS_MODEL
clear_gpu_cache()
if not xtts_checkpoint or not xtts_config or not xtts_vocab:
return "You need to run the previous steps or manually set the `XTTS checkpoint path`, `XTTS config path`, and `XTTS vocab path` fields !!"
config = XttsConfig()
config.load_json(xtts_config)
XTTS_MODEL = Xtts.init_from_config(config)
print("Loading XTTS model! ")
XTTS_MODEL.load_checkpoint(config, checkpoint_path=xtts_checkpoint, vocab_path=xtts_vocab,speaker_file_path=xtts_speaker, use_deepspeed=False)
if torch.cuda.is_available():
XTTS_MODEL.cuda()
print("Model Loaded!")
return "Model Loaded!"
def run_tts(lang, tts_text, speaker_audio_file, output_file_path, temperature, length_penalty, repetition_penalty, top_k, top_p, speed, sentence_split, use_config):
if XTTS_MODEL is None:
raise Exception("XTTS_MODEL is not loaded. Please load the model before running TTS.")
if not tts_text.strip():
raise ValueError("Text for TTS is empty.")
if not os.path.exists(speaker_audio_file):
raise FileNotFoundError(f"Speaker audio file not found: {speaker_audio_file}")
gpt_cond_latent, speaker_embedding = XTTS_MODEL.get_conditioning_latents(audio_path=speaker_audio_file, gpt_cond_len=XTTS_MODEL.config.gpt_cond_len, max_ref_length=XTTS_MODEL.config.max_ref_len, sound_norm_refs=XTTS_MODEL.config.sound_norm_refs)
if use_config:
out = XTTS_MODEL.inference(
text=tts_text,
language=lang,
gpt_cond_latent=gpt_cond_latent,
speaker_embedding=speaker_embedding,
temperature=XTTS_MODEL.config.temperature, # Add custom parameters here
length_penalty=XTTS_MODEL.config.length_penalty,
repetition_penalty=XTTS_MODEL.config.repetition_penalty,
top_k=XTTS_MODEL.config.top_k,
top_p=XTTS_MODEL.config.top_p,
speed=speed,
enable_text_splitting = True
)
else:
out = XTTS_MODEL.inference(
text=tts_text,
language=lang,
gpt_cond_latent=gpt_cond_latent,
speaker_embedding=speaker_embedding,
temperature=temperature, # Add custom parameters here
length_penalty=length_penalty,
repetition_penalty=float(repetition_penalty),
top_k=top_k,
top_p=top_p,
speed=speed,
enable_text_splitting = sentence_split
)
with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as fp:
out["wav"] = torch.tensor(out["wav"]).unsqueeze(0)
out_path = fp.name
torchaudio.save(out_path, out["wav"], 24000)
return "Speech generated !", out_path, speaker_audio_file
def load_params_tts(out_path,version):
out_path = Path(out_path)
# base_model_path = Path.cwd() / "models" / version
# if not base_model_path.exists():
# return "Base model not found !","","",""
ready_model_path = out_path / "ready"
vocab_path = ready_model_path / "vocab.json"
config_path = ready_model_path / "config.json"
speaker_path = ready_model_path / "speakers_xtts.pth"
reference_path = ready_model_path / "reference.wav"
model_path = ready_model_path / "model.pth"
if not model_path.exists():
model_path = ready_model_path / "unoptimize_model.pth"
if not model_path.exists():
return "Params for TTS not found", "", "", ""
return "Params for TTS loaded", model_path, config_path, vocab_path,speaker_path, reference_path
def process_srt_and_generate_audio(
srt_file,
lang,
speaker_reference_audio,
temperature,
length_penalty,
repetition_penalty,
top_k,
top_p,
speed,
sentence_split,
use_config ):
try:
subtitles = pysrt.open(srt_file)
audio_files = []
output_dir = create_output_dir(parent_dir='/content/drive/MyDrive/Voice Conversion Result')
for index, subtitle in enumerate(subtitles):
audio_filename = f"audio_{index+1:03d}.wav"
audio_file_path = os.path.join(output_dir, audio_filename)
subtitle_text=remove_endperiod(subtitle.text)
run_tts(lang, subtitle_text, speaker_reference_audio, audio_file_path,
temperature, length_penalty, repetition_penalty, top_k, top_p,
speed, sentence_split, use_config)
logger.info(f"Generated audio file: {audio_file_path}")
audio_files.append(audio_file_path)
output_audio_path = merge_audio_with_srt_timing(subtitles, audio_files, output_dir)
return output_audio_path
except Exception as e:
logger.error(f"Error in process_srt_and_generate_audio: {e}")
raise
def create_output_dir(parent_dir):
try:
# 定义一个基于当前日期和时间的文件夹名称
folder_name = datetime.now().strftime("audio_outputs_%Y-%m-%d_%H-%M-%S")
# 定义父目录,这里假设在Colab的根目录
#parent_dir = "/content/drive/MyDrive/Voice Conversion Result"
# 完整的文件夹路径
output_dir = os.path.join(parent_dir, folder_name)
# 创建文件夹
if not os.path.exists(output_dir):
os.makedirs(output_dir)
logger.info(f"Created output directory at: {output_dir}")
return output_dir
except Exception as e:
logger.error(f"Failed to create output directory: {e}")
raise
def srt_time_to_ms(srt_time):
return (srt_time.hours * 3600 + srt_time.minutes * 60 + srt_time.seconds) * 1000 + srt_time.milliseconds
def merge_audio_with_srt_timing(subtitles, audio_files, output_dir):
try:
combined = AudioSegment.silent(duration=0)
last_position_ms = 0
for subtitle, audio_file in zip(subtitles, audio_files):
start_time_ms = srt_time_to_ms(subtitle.start)
if last_position_ms < start_time_ms:
silence_duration = start_time_ms - last_position_ms
combined += AudioSegment.silent(duration=silence_duration)
last_position_ms = start_time_ms
audio = AudioSegment.from_file(audio_file, format="wav")
combined += audio
last_position_ms += len(audio)
output_path = os.path.join(output_dir, "combined_audio_with_timing.wav")
#combined_with_set_frame_rate = combined.set_frame_rate(24000)
#combined_with_set_frame_rate.export(output_path, format="wav")
combined.export(output_path, format="wav")
logger.info(f"Exported combined audio to: {output_path}")
return output_path
except Exception as e:
logger.error(f"Error merging audio files: {e}")
raise
def remove_endperiod(subtitle):
"""Removes the period (.) at the end of a subtitle.
"""
if subtitle.endswith('.'):
subtitle = subtitle[:-1]
return subtitle
def convert_voice(reference_audio, audio_to_convert):
device = "cuda:0" if torch.cuda.is_available() else "cpu"
# 定义输入和输出音频路径
#input_audio_path = audio_to_convert
base_name, ext = os.path.splitext(os.path.basename(audio_to_convert))
new_file_name = base_name + 'convertedvoice' + ext
output_path = os.path.join(os.path.dirname(audio_to_convert), new_file_name)
tune_one(input_file=audio_to_convert, ref_file=reference_audio, output_file=output_path, device=device)
return output_path
def tune_one(input_file,ref_file,output_file,device):
current_dir = os.path.dirname(os.path.realpath(__file__))
checkpoints_dir = os.path.join(current_dir, 'checkpoints')
ckpt_converter = os.path.join(checkpoints_dir, 'converter')
if not os.path.exists(ckpt_converter):
os.makedirs(ckpt_converter, exist_ok=True)
download_checkpoint(ckpt_converter)
device = device
tone_color_converter = ToneColorConverter(os.path.join(ckpt_converter, 'config.json'), device=device)
tone_color_converter.load_ckpt(os.path.join(ckpt_converter, 'checkpoint.pth'))
source_se, _ = se_extractor.get_se(input_file, tone_color_converter, vad=True)
target_se, _ = se_extractor.get_se(ref_file, tone_color_converter, vad=True)
# Ensure output directory exists and is writable
output_dir = os.path.dirname(output_file)
if output_dir:
if not os.path.exists(output_dir):
os.makedirs(output_dir, exist_ok=True)
# Run the tone color converter
tone_color_converter.convert(
audio_src_path=input_file,
src_se=source_se,
tgt_se=target_se,
output_path=output_file,
)
'''
def tune_batch(input_dir, ref_file, output_dir=None, device='cpu', output_format='.wav'):
current_dir = os.path.dirname(os.path.realpath(__file__))
checkpoints_dir = os.path.join(current_dir, 'checkpoints')
ckpt_converter = os.path.join(checkpoints_dir, 'converter')
if not os.path.exists(ckpt_converter):
os.makedirs(ckpt_converter, exist_ok=True)
download_checkpoint(ckpt_converter)
tone_color_converter = ToneColorConverter(os.path.join(ckpt_converter, 'config.json'), device=device)
tone_color_converter.load_ckpt(os.path.join(ckpt_converter, 'checkpoint.pth'))
target_se, _ = se_extractor.get_se(ref_file, tone_color_converter, vad=True)
# Use default output directory 'out' if not provided
if output_dir is None:
output_dir = os.path.join(current_dir, 'out')
os.makedirs(output_dir, exist_ok=True)
# Check for any audio files in the input directory (wav, mp3, flac) using glob
audio_extensions = ('*.wav', '*.mp3', '*.flac')
audio_files = []
for extension in audio_extensions:
audio_files.extend(glob.glob(os.path.join(input_dir, extension)))
for audio_file in tqdm(audio_files,"Tune file",len(audio_files)):
# Extract source SE from audio file
source_se, _ = se_extractor.get_se(audio_file, tone_color_converter, vad=True)
# Run the tone color converter
filename_without_extension = os.path.splitext(os.path.basename(audio_file))[0]
output_filename = f"{filename_without_extension}_tuned{output_format}"
output_file = os.path.join(output_dir, output_filename)
tone_color_converter.convert(
audio_src_path=audio_file,
src_se=source_se,
tgt_se=target_se,
output_path=output_file,
)
print(f"Converted {audio_file} to {output_file}")
return output_dir
def main_single(args):
tune_one(input_file=args.input, ref_file=args.ref, output_file=args.output, device=args.device)
def main_batch(args):
output_dir = tune_batch(
input_dir=args.input_dir,
ref_file=args.ref_file,
output_dir=args.output_dir,
device=args.device,
output_format=args.output_format
)
print(f"Batch processing complete. Converted files are saved in {output_dir}")
'''
# define a logger to redirect
class Logger:
def __init__(self, filename="log.out"):
self.log_file = filename
self.terminal = sys.stdout
self.log = open(self.log_file, "w")
def write(self, message):
self.terminal.write(message)
self.log.write(message)
def flush(self):
self.terminal.flush()
self.log.flush()
def isatty(self):
return False
# redirect stdout and stderr to a file
sys.stdout = Logger()
sys.stderr = sys.stdout
# logging.basicConfig(stream=sys.stdout, level=logging.INFO)
import logging
logging.basicConfig(
level=logging.INFO,
format="%(asctime)s [%(levelname)s] %(message)s",
handlers=[
logging.StreamHandler(sys.stdout)
]
)
def read_logs():
sys.stdout.flush()
with open(sys.stdout.log_file, "r") as f:
return f.read()
if __name__ == "__main__":
parser = argparse.ArgumentParser(
description="""XTTS fine-tuning demo\n\n"""
"""
Example runs:
python3 TTS/demos/xtts_ft_demo/xtts_demo.py --port
""",
formatter_class=argparse.RawTextHelpFormatter,
)
parser.add_argument(
"--port",
type=int,
help="Port to run the gradio demo. Default: 5003",
default=5003,
)
parser.add_argument(
"--out_path",
type=str,
help="Output path (where data and checkpoints will be saved) Default: output/",
default=str(Path.cwd() / "finetune_models"),
)
parser.add_argument(
"--num_epochs",
type=int,
help="Number of epochs to train. Default: 6",
default=6,
)
parser.add_argument(
"--batch_size",
type=int,
help="Batch size. Default: 2",
default=2,
)
parser.add_argument(
"--grad_acumm",
type=int,
help="Grad accumulation steps. Default: 1",
default=1,
)
parser.add_argument(
"--max_audio_length",
type=int,
help="Max permitted audio size in seconds. Default: 11",
default=11,
)
args = parser.parse_args()
with gr.Blocks() as demo:
with gr.Tab("0 - Voice conversion"):
with gr.Column() as col0:
gr.Markdown("## OpenVoice Conversion Tool")
voice_convert_seed = gr.File(label="Upload Reference Speaker Audio being generated")
#pitch_shift_slider = gr.Slider(minimum=-12, maximum=12, step=1, value=0, label="Pitch Shift (Semitones)")
audio_to_convert = gr.Textbox(
label="Input the to-be-convert audio location",
value="",
)
convert_button = gr.Button("Convert Voice")
converted_audio = gr.Audio(label="Converted Audio")
convert_button.click(
convert_voice,
inputs=[voice_convert_seed, audio_to_convert], #, pitch_shift_slider],
outputs=[converted_audio]
)
with gr.Tab("1 - Data processing"):
out_path = gr.Textbox(
label="Output path (where data and checkpoints will be saved):",
value=args.out_path,
)
# upload_file = gr.Audio(
# sources="upload",
# label="Select here the audio files that you want to use for XTTS trainining !",
# type="filepath",
# )
upload_file = gr.File(
file_count="multiple",
label="Select here the audio files that you want to use for XTTS trainining (Supported formats: wav, mp3, and flac)",
)
whisper_model = gr.Dropdown(
label="Whisper Model",
value="large-v3",
choices=[
"large-v3",
"large-v2",
"large",
"medium",
"small"
],
)
lang = gr.Dropdown(
label="Dataset Language",
value="en",
choices=[
"en",
"es",
"fr",
"de",
"it",
"pt",
"pl",
"tr",
"ru",
"nl",
"cs",
"ar",
"zh",
"hu",
"ko",
"ja"
],
)
progress_data = gr.Label(
label="Progress:"
)
# demo.load(read_logs, None, logs, every=1)
prompt_compute_btn = gr.Button(value="Step 1 - Create dataset")
def preprocess_dataset(audio_path, language, whisper_model, out_path,train_csv,eval_csv, progress=gr.Progress(track_tqdm=True)):
clear_gpu_cache()
train_csv = ""
eval_csv = ""
out_path = os.path.join(out_path, "dataset")
os.makedirs(out_path, exist_ok=True)
if audio_path is None:
return "You should provide one or multiple audio files! If you provided it, probably the upload of the files is not finished yet!", "", ""
else:
try:
train_meta, eval_meta, audio_total_size = format_audio_list(audio_path, whisper_model = whisper_model, target_language=language, out_path=out_path, gradio_progress=progress)
except:
traceback.print_exc()
error = traceback.format_exc()
return f"The data processing was interrupted due an error !! Please check the console to verify the full error message! \n Error summary: {error}", "", ""
# clear_gpu_cache()
# if audio total len is less than 2 minutes raise an error
if audio_total_size < 120:
message = "The sum of the duration of the audios that you provided should be at least 2 minutes!"
print(message)
return message, "", ""
print("Dataset Processed!")
return "Dataset Processed!", train_meta, eval_meta
with gr.Tab("2 - Fine-tuning XTTS Encoder"):
load_params_btn = gr.Button(value="Load Params from output folder")
version = gr.Dropdown(
label="XTTS base version",
value="v2.0.2",
choices=[
"v2.0.3",
"v2.0.2",
"v2.0.1",
"v2.0.0",
"main"
],
)
train_csv = gr.Textbox(
label="Train CSV:",
)
eval_csv = gr.Textbox(
label="Eval CSV:",
)
custom_model = gr.Textbox(
label="(Optional) Custom model.pth file , leave blank if you want to use the base file.",
value="",
)
num_epochs = gr.Slider(
label="Number of epochs:",
minimum=1,
maximum=100,
step=1,
value=args.num_epochs,
)
batch_size = gr.Slider(
label="Batch size:",
minimum=2,
maximum=512,
step=1,
value=args.batch_size,
)
grad_acumm = gr.Slider(
label="Grad accumulation steps:",
minimum=2,
maximum=128,
step=1,
value=args.grad_acumm,
)
max_audio_length = gr.Slider(
label="Max permitted audio size in seconds:",
minimum=2,
maximum=20,
step=1,
value=args.max_audio_length,
)
clear_train_data = gr.Dropdown(
label="Clear train data, you will delete selected folder, after optimizing",
value="run",
choices=[
"none",
"run",
"dataset",
"all"
])
progress_train = gr.Label(
label="Progress:"
)
# demo.load(read_logs, None, logs_tts_train, every=1)
train_btn = gr.Button(value="Step 2 - Run the training")
optimize_model_btn = gr.Button(value="Step 2.5 - Optimize the model")
def train_model(custom_model,version,language, train_csv, eval_csv, num_epochs, batch_size, grad_acumm, output_path, max_audio_length):
clear_gpu_cache()
run_dir = Path(output_path) / "run"
# # Remove train dir
if run_dir.exists():
os.remove(run_dir)
# Check if the dataset language matches the language you specified
lang_file_path = Path(output_path) / "dataset" / "lang.txt"
# Check if lang.txt already exists and contains a different language
current_language = None
if lang_file_path.exists():
with open(lang_file_path, 'r', encoding='utf-8') as existing_lang_file:
current_language = existing_lang_file.read().strip()
if current_language != language:
print("The language that was prepared for the dataset does not match the specified language. Change the language to the one specified in the dataset")
language = current_language
if not train_csv or not eval_csv:
return "You need to run the data processing step or manually set `Train CSV` and `Eval CSV` fields !", "", "", "", ""
try:
# convert seconds to waveform frames
max_audio_length = int(max_audio_length * 22050)
speaker_xtts_path,config_path, original_xtts_checkpoint, vocab_file, exp_path, speaker_wav = train_gpt(custom_model,version,language, num_epochs, batch_size, grad_acumm, train_csv, eval_csv, output_path=output_path, max_audio_length=max_audio_length)
except:
traceback.print_exc()
error = traceback.format_exc()
return f"The training was interrupted due an error !! Please check the console to check the full error message! \n Error summary: {error}", "", "", "", ""
# copy original files to avoid parameters changes issues
# os.system(f"cp {config_path} {exp_path}")
# os.system(f"cp {vocab_file} {exp_path}")
ready_dir = Path(output_path) / "ready"
ft_xtts_checkpoint = os.path.join(exp_path, "best_model.pth")
shutil.copy(ft_xtts_checkpoint, ready_dir / "unoptimize_model.pth")
# os.remove(ft_xtts_checkpoint)
ft_xtts_checkpoint = os.path.join(ready_dir, "unoptimize_model.pth")
# Reference
# Move reference audio to output folder and rename it
speaker_reference_path = Path(speaker_wav)
speaker_reference_new_path = ready_dir / "reference.wav"
shutil.copy(speaker_reference_path, speaker_reference_new_path)
print("Model training done!")
# clear_gpu_cache()
return "Model training done!", config_path, vocab_file, ft_xtts_checkpoint,speaker_xtts_path, speaker_reference_new_path
def optimize_model(out_path, clear_train_data):
# print(out_path)
out_path = Path(out_path) # Ensure that out_path is a Path object.
ready_dir = out_path / "ready"
run_dir = out_path / "run"
dataset_dir = out_path / "dataset"
# Clear specified training data directories.
if clear_train_data in {"run", "all"} and run_dir.exists():
try:
shutil.rmtree(run_dir)
except PermissionError as e:
print(f"An error occurred while deleting {run_dir}: {e}")
if clear_train_data in {"dataset", "all"} and dataset_dir.exists():
try:
shutil.rmtree(dataset_dir)
except PermissionError as e:
print(f"An error occurred while deleting {dataset_dir}: {e}")
# Get full path to model
model_path = ready_dir / "unoptimize_model.pth"
if not model_path.is_file():
return "Unoptimized model not found in ready folder", ""
# Load the checkpoint and remove unnecessary parts.
checkpoint = torch.load(model_path, map_location=torch.device("cpu"))
del checkpoint["optimizer"]
for key in list(checkpoint["model"].keys()):
if "dvae" in key:
del checkpoint["model"][key]
# Make sure out_path is a Path object or convert it to Path
os.remove(model_path)
# Save the optimized model.
optimized_model_file_name="model.pth"
optimized_model=ready_dir/optimized_model_file_name
torch.save(checkpoint, optimized_model)
ft_xtts_checkpoint=str(optimized_model)
clear_gpu_cache()
return f"Model optimized and saved at {ft_xtts_checkpoint}!", ft_xtts_checkpoint
def load_params(out_path):
path_output = Path(out_path)
dataset_path = path_output / "dataset"
if not dataset_path.exists():
return "The output folder does not exist!", "", ""
eval_train = dataset_path / "metadata_train.csv"
eval_csv = dataset_path / "metadata_eval.csv"
# Write the target language to lang.txt in the output directory
lang_file_path = dataset_path / "lang.txt"
# Check if lang.txt already exists and contains a different language
current_language = None
if os.path.exists(lang_file_path):
with open(lang_file_path, 'r', encoding='utf-8') as existing_lang_file:
current_language = existing_lang_file.read().strip()
clear_gpu_cache()
print(current_language)
return "The data has been updated", eval_train, eval_csv, current_language
with gr.Tab("3 - Inference"):
with gr.Row():
with gr.Column() as col1:
load_params_tts_btn = gr.Button(value="Load params for TTS from output folder")
xtts_checkpoint = gr.Textbox(
label="XTTS checkpoint path:",
value="",
)
xtts_config = gr.Textbox(
label="XTTS config path:",
value="",
)
xtts_vocab = gr.Textbox(
label="XTTS vocab path:",
value="",
)
xtts_speaker = gr.Textbox(
label="XTTS speaker path:",
value="",
)
progress_load = gr.Label(
label="Progress:"
)
load_btn = gr.Button(value="Step 3 - Load Fine-tuned XTTS model")
with gr.Column() as col2:
speaker_reference_audio = gr.Textbox(
label="Speaker reference audio:",
value="",
)
tts_language = gr.Dropdown(
label="Language",
value="en",
choices=[
"en",
"es",
"fr",
"de",
"it",
"pt",
"pl",
"tr",
"ru",
"nl",
"cs",
"ar",
"zh",
"hu",
"ko",
"ja",
]
)
tts_text = gr.Textbox(
label="Input Text.",
value="This model sounds really good and above all, it's reasonably fast.",
)
with gr.Accordion("Advanced settings", open=False) as acr:
temperature = gr.Slider(
label="temperature",
minimum=0,
maximum=1,
step=0.05,
value=0.75,
)
length_penalty = gr.Slider(
label="length_penalty",
minimum=-10.0,
maximum=10.0,
step=0.5,
value=1,
)
repetition_penalty = gr.Slider(
label="repetition penalty",
minimum=1,
maximum=10,
step=0.5,
value=5,
)
top_k = gr.Slider(
label="top_k",
minimum=1,
maximum=100,
step=1,
value=50,
)
top_p = gr.Slider(
label="top_p",
minimum=0,
maximum=1,
step=0.05,
value=0.85,
)
speed = gr.Slider(
label="speed",
minimum=0.2,
maximum=4.0,
step=0.05,
value=1.0,
)
sentence_split = gr.Checkbox(
label="Enable text splitting",
value=True,
)
use_config = gr.Checkbox(
label="Use Inference settings from config, if disabled use the settings above",
value=False,
)
tts_btn = gr.Button(value="Step 4 - Inference")
with gr.Column() as col3:
progress_gen = gr.Label(
label="Progress:"
)
tts_output_audio = gr.Audio(label="Generated Audio.")
reference_audio = gr.Audio(label="Reference audio used.")
with gr.Column() as col4:
srt_upload = gr.File(label="Upload SRT File")
generate_srt_audio_btn = gr.Button(value="Generate Audio from SRT")
srt_output_audio = gr.Audio(label="Combined Audio from SRT")
error_message = gr.Textbox(label="Error Message", visible=False) # 错误消息组件,默认不显示
generate_srt_audio_btn.click(
fn=process_srt_and_generate_audio,
inputs=[
srt_upload,
tts_language,
speaker_reference_audio,
temperature,
length_penalty,
repetition_penalty,
top_k,
top_p,
speed,
sentence_split,
use_config
],
outputs=[srt_output_audio]
)
prompt_compute_btn.click(
fn=preprocess_dataset,
inputs=[
upload_file,
lang,
whisper_model,
out_path,
train_csv,
eval_csv
],
outputs=[
progress_data,
train_csv,
eval_csv,
],
)
load_params_btn.click(
fn=load_params,
inputs=[out_path],
outputs=[
progress_train,
train_csv,
eval_csv,
lang
]
)
train_btn.click(
fn=train_model,
inputs=[
custom_model,
version,
lang,
train_csv,
eval_csv,
num_epochs,
batch_size,
grad_acumm,
out_path,
max_audio_length,
],
outputs=[progress_train, xtts_config, xtts_vocab, xtts_checkpoint,xtts_speaker, speaker_reference_audio],
)
optimize_model_btn.click(
fn=optimize_model,
inputs=[
out_path,
clear_train_data
],
outputs=[progress_train,xtts_checkpoint],
)
load_btn.click(
fn=load_model,
inputs=[
xtts_checkpoint,
xtts_config,
xtts_vocab,
xtts_speaker
],
outputs=[progress_load],
)
tts_btn.click(
fn=run_tts,
inputs=[
tts_language,
tts_text,
speaker_reference_audio,
temperature,
length_penalty,
repetition_penalty,
top_k,
top_p,
speed,
sentence_split,
use_config
],
outputs=[progress_gen, tts_output_audio, reference_audio],
)
load_params_tts_btn.click(
fn=load_params_tts,
inputs=[
out_path,
version
],
outputs=[progress_load,xtts_checkpoint,xtts_config,xtts_vocab,xtts_speaker,speaker_reference_audio],
)
demo.launch(
#share=False,
share=True,
debug=False,
server_port=args.port,
#server_name="localhost"
server_name="0.0.0.0"
)
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