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import os
import spaces
import sys
import json
import argparse
import subprocess
from functools import lru_cache
from distutils.util import strtobool
now_dir = os.getcwd()
sys.path.append(now_dir)
current_script_directory = os.path.dirname(os.path.realpath(__file__))
logs_path = os.path.join(current_script_directory, "logs")
from rvc.lib.tools.prerequisites_download import prequisites_download_pipeline
from rvc.train.extract.preparing_files import generate_config, generate_filelist
from rvc.train.process.model_blender import model_blender
from rvc.train.process.model_information import model_information
from rvc.train.process.extract_small_model import extract_small_model
from rvc.lib.tools.analyzer import analyze_audio
from rvc.lib.tools.launch_tensorboard import launch_tensorboard_pipeline
from rvc.lib.tools.model_download import model_download_pipeline
python = sys.executable
# Get TTS Voices -> https://speech.platform.bing.com/consumer/speech/synthesize/readaloud/voices/list?trustedclienttoken=6A5AA1D4EAFF4E9FB37E23D68491D6F4
@lru_cache(maxsize=1) # Cache only one result since the file is static
def load_voices_data():
with open(os.path.join("rvc", "lib", "tools", "tts_voices.json")) as f:
return json.load(f)
voices_data = load_voices_data()
locales = list({voice["Locale"] for voice in voices_data})
@lru_cache(maxsize=None)
def import_voice_converter():
from rvc.infer.infer import VoiceConverter
return VoiceConverter()
@lru_cache(maxsize=1)
def get_config():
from rvc.configs.config import Config
return Config()
# Infer
@spaces.GPU(duration=120)
def run_infer_script(
pitch: int,
filter_radius: int,
index_rate: float,
volume_envelope: int,
protect: float,
hop_length: int,
f0_method: str,
input_path: str,
output_path: str,
pth_path: str,
index_path: str,
split_audio: bool,
f0_autotune: bool,
clean_audio: bool,
clean_strength: float,
export_format: str,
upscale_audio: bool,
f0_file: str,
embedder_model: str,
embedder_model_custom: str = None,
):
infer_pipeline = import_voice_converter()
infer_pipeline.convert_audio(
pitch=pitch,
filter_radius=filter_radius,
index_rate=index_rate,
volume_envelope=volume_envelope,
protect=protect,
hop_length=hop_length,
f0_method=f0_method,
audio_input_path=input_path,
audio_output_path=output_path,
model_path=pth_path,
index_path=index_path,
split_audio=split_audio,
f0_autotune=f0_autotune,
clean_audio=clean_audio,
clean_strength=clean_strength,
export_format=export_format,
upscale_audio=upscale_audio,
f0_file=f0_file,
embedder_model=embedder_model,
embedder_model_custom=embedder_model_custom,
)
return f"File {input_path} inferred successfully.", output_path.replace(
".wav", f".{export_format.lower()}"
)
# Batch infer
@spaces.GPU(duration=200)
def run_batch_infer_script(
pitch: int,
filter_radius: int,
index_rate: float,
volume_envelope: int,
protect: float,
hop_length: int,
f0_method: str,
input_folder: str,
output_folder: str,
pth_path: str,
index_path: str,
split_audio: bool,
f0_autotune: bool,
clean_audio: bool,
clean_strength: float,
export_format: str,
upscale_audio: bool,
f0_file: str,
embedder_model: str,
embedder_model_custom: str = None,
):
audio_files = [
f for f in os.listdir(input_folder) if f.endswith((".mp3", ".wav", ".flac"))
]
print(f"Detected {len(audio_files)} audio files for inference.")
for audio_file in audio_files:
if "_output" in audio_file:
pass
else:
input_path = os.path.join(input_folder, audio_file)
output_file_name = os.path.splitext(os.path.basename(audio_file))[0]
output_path = os.path.join(
output_folder,
f"{output_file_name}_output{os.path.splitext(audio_file)[1]}",
)
infer_pipeline = import_voice_converter()
print(f"Inferring {input_path}...")
infer_pipeline.convert_audio(
pitch=pitch,
filter_radius=filter_radius,
index_rate=index_rate,
volume_envelope=volume_envelope,
protect=protect,
hop_length=hop_length,
f0_method=f0_method,
audio_input_path=input_path,
audio_output_path=output_path,
model_path=pth_path,
index_path=index_path,
split_audio=split_audio,
f0_autotune=f0_autotune,
clean_audio=clean_audio,
clean_strength=clean_strength,
export_format=export_format,
upscale_audio=upscale_audio,
f0_file=f0_file,
embedder_model=embedder_model,
embedder_model_custom=embedder_model_custom,
)
return f"Files from {input_folder} inferred successfully."
# TTS
@spaces.GPU(duration=120)
def run_tts_script(
tts_text: str,
tts_voice: str,
tts_rate: int,
pitch: int,
filter_radius: int,
index_rate: float,
volume_envelope: int,
protect: float,
hop_length: int,
f0_method: str,
output_tts_path: str,
output_rvc_path: str,
pth_path: str,
index_path: str,
split_audio: bool,
f0_autotune: bool,
clean_audio: bool,
clean_strength: float,
export_format: str,
upscale_audio: bool,
f0_file: str,
embedder_model: str,
embedder_model_custom: str = None,
):
tts_script_path = os.path.join("rvc", "lib", "tools", "tts.py")
if os.path.exists(output_tts_path):
os.remove(output_tts_path)
command_tts = [
*map(
str,
[
python,
tts_script_path,
tts_text,
tts_voice,
tts_rate,
output_tts_path,
],
),
]
subprocess.run(command_tts)
infer_pipeline = import_voice_converter()
infer_pipeline.convert_audio(
pitch=pitch,
filter_radius=filter_radius,
index_rate=index_rate,
volume_envelope=volume_envelope,
protect=protect,
hop_length=hop_length,
f0_method=f0_method,
audio_input_path=output_tts_path,
audio_output_path=output_rvc_path,
model_path=pth_path,
index_path=index_path,
split_audio=split_audio,
f0_autotune=f0_autotune,
clean_audio=clean_audio,
clean_strength=clean_strength,
export_format=export_format,
upscale_audio=upscale_audio,
f0_file=f0_file,
embedder_model=embedder_model,
embedder_model_custom=embedder_model_custom,
)
return f"Text {tts_text} synthesized successfully.", output_rvc_path.replace(
".wav", f".{export_format.lower()}"
)
# Preprocess
@spaces.GPU(duration=360)
def run_preprocess_script(
model_name: str, dataset_path: str, sample_rate: int, cpu_cores: int
):
config = get_config()
per = 3.0 if config.is_half else 3.7
preprocess_script_path = os.path.join("rvc", "train", "preprocess", "preprocess.py")
command = [
python,
preprocess_script_path,
*map(
str,
[
os.path.join(logs_path, model_name),
dataset_path,
sample_rate,
per,
cpu_cores,
],
),
]
os.makedirs(os.path.join(logs_path, model_name), exist_ok=True)
subprocess.run(command)
return f"Model {model_name} preprocessed successfully."
# Extract
@spaces.GPU(duration=360)
def run_extract_script(
model_name: str,
rvc_version: str,
f0_method: str,
pitch_guidance: bool,
hop_length: int,
cpu_cores: int,
gpu: int,
sample_rate: int,
embedder_model: str,
embedder_model_custom: str = None,
):
config = get_config()
model_path = os.path.join(logs_path, model_name)
pitch_extractor = os.path.join("rvc", "train", "extract", "pitch_extractor.py")
embedding_extractor = os.path.join(
"rvc", "train", "extract", "embedding_extractor.py"
)
command_1 = [
python,
pitch_extractor,
*map(
str,
[
model_path,
f0_method,
hop_length,
cpu_cores,
gpu,
],
),
]
command_2 = [
python,
embedding_extractor,
*map(
str,
[
model_path,
rvc_version,
gpu,
embedder_model,
embedder_model_custom,
],
),
]
subprocess.run(command_1)
subprocess.run(command_2)
generate_config(rvc_version, sample_rate, model_path)
generate_filelist(pitch_guidance, model_path, rvc_version, sample_rate)
return f"Model {model_name} extracted successfully."
# Train
@spaces.GPU(duration=360)
def run_train_script(
model_name: str,
rvc_version: str,
save_every_epoch: int,
save_only_latest: bool,
save_every_weights: bool,
total_epoch: int,
sample_rate: int,
batch_size: int,
gpu: int,
pitch_guidance: bool,
overtraining_detector: bool,
overtraining_threshold: int,
pretrained: bool,
sync_graph: bool,
cache_data_in_gpu: bool,
custom_pretrained: bool = False,
g_pretrained_path: str = None,
d_pretrained_path: str = None,
):
if pretrained == True:
from rvc.lib.tools.pretrained_selector import pretrained_selector
if custom_pretrained == False:
pg, pd = pretrained_selector(bool(pitch_guidance))[str(rvc_version)][
int(sample_rate)
]
else:
if g_pretrained_path is None or d_pretrained_path is None:
raise ValueError(
"Please provide the path to the pretrained G and D models."
)
pg, pd = g_pretrained_path, d_pretrained_path
else:
pg, pd = "", ""
train_script_path = os.path.join("rvc", "train", "train.py")
command = [
python,
train_script_path,
*map(
str,
[
model_name,
save_every_epoch,
total_epoch,
pg,
pd,
rvc_version,
gpu,
batch_size,
sample_rate,
pitch_guidance,
save_only_latest,
save_every_weights,
cache_data_in_gpu,
overtraining_detector,
overtraining_threshold,
sync_graph,
],
),
]
subprocess.run(command)
run_index_script(model_name, rvc_version)
return f"Model {model_name} trained successfully."
# Index
@spaces.GPU
def run_index_script(model_name: str, rvc_version: str):
index_script_path = os.path.join("rvc", "train", "process", "extract_index.py")
command = [
python,
index_script_path,
os.path.join(logs_path, model_name),
rvc_version,
]
subprocess.run(command)
return f"Index file for {model_name} generated successfully."
# Model extract
@spaces.GPU
def run_model_extract_script(
pth_path: str,
model_name: str,
sample_rate: int,
pitch_guidance: bool,
rvc_version: str,
epoch: int,
step: int,
):
extract_small_model(
pth_path, model_name, sample_rate, pitch_guidance, rvc_version, epoch, step
)
return f"Model {model_name} extracted successfully."
# Model information
@spaces.GPU
def run_model_information_script(pth_path: str):
print(model_information(pth_path))
# Model blender
def run_model_blender_script(
model_name: str, pth_path_1: str, pth_path_2: str, ratio: float
):
message, model_blended = model_blender(model_name, pth_path_1, pth_path_2, ratio)
return message, model_blended
# Tensorboard
@spaces.GPU
def run_tensorboard_script():
launch_tensorboard_pipeline()
# Download
def run_download_script(model_link: str):
model_download_pipeline(model_link)
return f"Model downloaded successfully."
# Prerequisites
def run_prerequisites_script(
pretraineds_v1: bool, pretraineds_v2: bool, models: bool, exe: bool
):
prequisites_download_pipeline(pretraineds_v1, pretraineds_v2, models, exe)
return "Prerequisites installed successfully."
# Audio analyzer
def run_audio_analyzer_script(
input_path: str, save_plot_path: str = "logs/audio_analysis.png"
):
audio_info, plot_path = analyze_audio(input_path, save_plot_path)
print(
f"Audio info of {input_path}: {audio_info}",
f"Audio file {input_path} analyzed successfully. Plot saved at: {plot_path}",
)
return audio_info, plot_path
# API
def run_api_script(ip: str, port: int):
command = [
"env/Scripts/uvicorn.exe" if os.name == "nt" else "uvicorn",
"api:app",
"--host",
ip,
"--port",
port,
]
subprocess.run(command)
# Parse arguments
def parse_arguments():
parser = argparse.ArgumentParser(
description="Run the main.py script with specific parameters."
)
subparsers = parser.add_subparsers(
title="subcommands", dest="mode", help="Choose a mode"
)
# Parser for 'infer' mode
infer_parser = subparsers.add_parser("infer", help="Run inference")
pitch_description = (
"Set the pitch of the audio. Higher values result in a higher pitch."
)
infer_parser.add_argument(
"--pitch",
type=int,
help=pitch_description,
choices=range(-24, 25),
default=0,
)
filter_radius_description = "Apply median filtering to the extracted pitch values if this value is greater than or equal to three. This can help reduce breathiness in the output audio."
infer_parser.add_argument(
"--filter_radius",
type=int,
help=filter_radius_description,
choices=range(11),
default=3,
)
index_rate_description = "Control the influence of the index file on the output. Higher values mean stronger influence. Lower values can help reduce artifacts but may result in less accurate voice cloning."
infer_parser.add_argument(
"--index_rate",
type=float,
help=index_rate_description,
choices=[(i / 10) for i in range(11)],
default=0.3,
)
volume_envelope_description = "Control the blending of the output's volume envelope. A value of 1 means the output envelope is fully used."
infer_parser.add_argument(
"--volume_envelope",
type=float,
help=volume_envelope_description,
choices=[(i / 10) for i in range(11)],
default=1,
)
protect_description = "Protect consonants and breathing sounds from artifacts. A value of 0.5 offers the strongest protection, while lower values may reduce the protection level but potentially mitigate the indexing effect."
infer_parser.add_argument(
"--protect",
type=float,
help=protect_description,
choices=[(i / 10) for i in range(6)],
default=0.33,
)
hop_length_description = "Only applicable for the Crepe pitch extraction method. Determines the time it takes for the system to react to a significant pitch change. Smaller values require more processing time but can lead to better pitch accuracy."
infer_parser.add_argument(
"--hop_length",
type=int,
help=hop_length_description,
choices=range(1, 513),
default=128,
)
f0_method_description = "Choose the pitch extraction algorithm for the conversion. 'rmvpe' is the default and generally recommended."
infer_parser.add_argument(
"--f0_method",
type=str,
help=f0_method_description,
choices=[
"crepe",
"crepe-tiny",
"rmvpe",
"fcpe",
"hybrid[crepe+rmvpe]",
"hybrid[crepe+fcpe]",
"hybrid[rmvpe+fcpe]",
"hybrid[crepe+rmvpe+fcpe]",
],
default="rmvpe",
)
infer_parser.add_argument(
"--input_path",
type=str,
help="Full path to the input audio file.",
required=True,
)
infer_parser.add_argument(
"--output_path",
type=str,
help="Full path to the output audio file.",
required=True,
)
pth_path_description = "Full path to the RVC model file (.pth)."
infer_parser.add_argument(
"--pth_path", type=str, help=pth_path_description, required=True
)
index_path_description = "Full path to the index file (.index)."
infer_parser.add_argument(
"--index_path", type=str, help=index_path_description, required=True
)
split_audio_description = "Split the audio into smaller segments before inference. This can improve the quality of the output for longer audio files."
infer_parser.add_argument(
"--split_audio",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=split_audio_description,
default=False,
)
f0_autotune_description = "Apply a light autotune to the inferred audio. Particularly useful for singing voice conversions."
infer_parser.add_argument(
"--f0_autotune",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=f0_autotune_description,
default=False,
)
clean_audio_description = "Clean the output audio using noise reduction algorithms. Recommended for speech conversions."
infer_parser.add_argument(
"--clean_audio",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=clean_audio_description,
default=False,
)
clean_strength_description = "Adjust the intensity of the audio cleaning process. Higher values result in stronger cleaning, but may lead to a more compressed sound."
infer_parser.add_argument(
"--clean_strength",
type=float,
help=clean_strength_description,
choices=[(i / 10) for i in range(11)],
default=0.7,
)
export_format_description = "Select the desired output audio format."
infer_parser.add_argument(
"--export_format",
type=str,
help=export_format_description,
choices=["WAV", "MP3", "FLAC", "OGG", "M4A"],
default="WAV",
)
embedder_model_description = (
"Choose the model used for generating speaker embeddings."
)
infer_parser.add_argument(
"--embedder_model",
type=str,
help=embedder_model_description,
choices=[
"contentvec",
"japanese-hubert-base",
"chinese-hubert-large",
"custom",
],
default="contentvec",
)
embedder_model_custom_description = "Specify the path to a custom model for speaker embedding. Only applicable if 'embedder_model' is set to 'custom'."
infer_parser.add_argument(
"--embedder_model_custom",
type=str,
help=embedder_model_custom_description,
default=None,
)
upscale_audio_description = "Upscale the input audio to a higher quality before processing. This can improve the overall quality of the output, especially for low-quality input audio."
infer_parser.add_argument(
"--upscale_audio",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=upscale_audio_description,
default=False,
)
f0_file_description = "Full path to an external F0 file (.f0). This allows you to use pre-computed pitch values for the input audio."
infer_parser.add_argument(
"--f0_file",
type=str,
help=f0_file_description,
default=None,
)
# Parser for 'batch_infer' mode
batch_infer_parser = subparsers.add_parser(
"batch_infer",
help="Run batch inference",
)
batch_infer_parser.add_argument(
"--pitch",
type=int,
help=pitch_description,
choices=range(-24, 25),
default=0,
)
batch_infer_parser.add_argument(
"--filter_radius",
type=int,
help=filter_radius_description,
choices=range(11),
default=3,
)
batch_infer_parser.add_argument(
"--index_rate",
type=float,
help=index_rate_description,
choices=[(i / 10) for i in range(11)],
default=0.3,
)
batch_infer_parser.add_argument(
"--volume_envelope",
type=float,
help=volume_envelope_description,
choices=[(i / 10) for i in range(11)],
default=1,
)
batch_infer_parser.add_argument(
"--protect",
type=float,
help=protect_description,
choices=[(i / 10) for i in range(6)],
default=0.33,
)
batch_infer_parser.add_argument(
"--hop_length",
type=int,
help=hop_length_description,
choices=range(1, 513),
default=128,
)
batch_infer_parser.add_argument(
"--f0_method",
type=str,
help=f0_method_description,
choices=[
"crepe",
"crepe-tiny",
"rmvpe",
"fcpe",
"hybrid[crepe+rmvpe]",
"hybrid[crepe+fcpe]",
"hybrid[rmvpe+fcpe]",
"hybrid[crepe+rmvpe+fcpe]",
],
default="rmvpe",
)
batch_infer_parser.add_argument(
"--input_folder",
type=str,
help="Path to the folder containing input audio files.",
required=True,
)
batch_infer_parser.add_argument(
"--output_folder",
type=str,
help="Path to the folder for saving output audio files.",
required=True,
)
batch_infer_parser.add_argument(
"--pth_path", type=str, help=pth_path_description, required=True
)
batch_infer_parser.add_argument(
"--index_path", type=str, help=index_path_description, required=True
)
batch_infer_parser.add_argument(
"--split_audio",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=split_audio_description,
default=False,
)
batch_infer_parser.add_argument(
"--f0_autotune",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=f0_autotune_description,
default=False,
)
batch_infer_parser.add_argument(
"--clean_audio",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=clean_audio_description,
default=False,
)
batch_infer_parser.add_argument(
"--clean_strength",
type=float,
help=clean_strength_description,
choices=[(i / 10) for i in range(11)],
default=0.7,
)
batch_infer_parser.add_argument(
"--export_format",
type=str,
help=export_format_description,
choices=["WAV", "MP3", "FLAC", "OGG", "M4A"],
default="WAV",
)
batch_infer_parser.add_argument(
"--embedder_model",
type=str,
help=embedder_model_description,
choices=[
"contentvec",
"japanese-hubert-base",
"chinese-hubert-large",
"custom",
],
default="contentvec",
)
batch_infer_parser.add_argument(
"--embedder_model_custom",
type=str,
help=embedder_model_custom_description,
default=None,
)
batch_infer_parser.add_argument(
"--upscale_audio",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=upscale_audio_description,
default=False,
)
batch_infer_parser.add_argument(
"--f0_file",
type=str,
help=f0_file_description,
default=None,
)
# Parser for 'tts' mode
tts_parser = subparsers.add_parser("tts", help="Run TTS inference")
tts_parser.add_argument(
"--tts_text", type=str, help="Text to be synthesized", required=True
)
tts_parser.add_argument(
"--tts_voice",
type=str,
help="Voice to be used for TTS synthesis.",
choices=locales,
required=True,
)
tts_parser.add_argument(
"--tts_rate",
type=int,
help="Control the speaking rate of the TTS. Values range from -100 (slower) to 100 (faster).",
choices=range(-100, 101),
default=0,
)
tts_parser.add_argument(
"--pitch",
type=int,
help=pitch_description,
choices=range(-24, 25),
default=0,
)
tts_parser.add_argument(
"--filter_radius",
type=int,
help=filter_radius_description,
choices=range(11),
default=3,
)
tts_parser.add_argument(
"--index_rate",
type=float,
help=index_rate_description,
choices=[(i / 10) for i in range(11)],
default=0.3,
)
tts_parser.add_argument(
"--volume_envelope",
type=float,
help=volume_envelope_description,
choices=[(i / 10) for i in range(11)],
default=1,
)
tts_parser.add_argument(
"--protect",
type=float,
help=protect_description,
choices=[(i / 10) for i in range(6)],
default=0.33,
)
tts_parser.add_argument(
"--hop_length",
type=int,
help=hop_length_description,
choices=range(1, 513),
default=128,
)
tts_parser.add_argument(
"--f0_method",
type=str,
help=f0_method_description,
choices=[
"crepe",
"crepe-tiny",
"rmvpe",
"fcpe",
"hybrid[crepe+rmvpe]",
"hybrid[crepe+fcpe]",
"hybrid[rmvpe+fcpe]",
"hybrid[crepe+rmvpe+fcpe]",
],
default="rmvpe",
)
tts_parser.add_argument(
"--output_tts_path",
type=str,
help="Full path to save the synthesized TTS audio.",
required=True,
)
tts_parser.add_argument(
"--output_rvc_path",
type=str,
help="Full path to save the voice-converted audio using the synthesized TTS.",
required=True,
)
tts_parser.add_argument(
"--pth_path", type=str, help=pth_path_description, required=True
)
tts_parser.add_argument(
"--index_path", type=str, help=index_path_description, required=True
)
tts_parser.add_argument(
"--split_audio",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=split_audio_description,
default=False,
)
tts_parser.add_argument(
"--f0_autotune",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=f0_autotune_description,
default=False,
)
tts_parser.add_argument(
"--clean_audio",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=clean_audio_description,
default=False,
)
tts_parser.add_argument(
"--clean_strength",
type=float,
help=clean_strength_description,
choices=[(i / 10) for i in range(11)],
default=0.7,
)
tts_parser.add_argument(
"--export_format",
type=str,
help=export_format_description,
choices=["WAV", "MP3", "FLAC", "OGG", "M4A"],
default="WAV",
)
tts_parser.add_argument(
"--embedder_model",
type=str,
help=embedder_model_description,
choices=[
"contentvec",
"japanese-hubert-base",
"chinese-hubert-large",
"custom",
],
default="contentvec",
)
tts_parser.add_argument(
"--embedder_model_custom",
type=str,
help=embedder_model_custom_description,
default=None,
)
tts_parser.add_argument(
"--upscale_audio",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help=upscale_audio_description,
default=False,
)
tts_parser.add_argument(
"--f0_file",
type=str,
help=f0_file_description,
default=None,
)
# Parser for 'preprocess' mode
preprocess_parser = subparsers.add_parser(
"preprocess", help="Preprocess a dataset for training."
)
preprocess_parser.add_argument(
"--model_name", type=str, help="Name of the model to be trained.", required=True
)
preprocess_parser.add_argument(
"--dataset_path", type=str, help="Path to the dataset directory.", required=True
)
preprocess_parser.add_argument(
"--sample_rate",
type=int,
help="Target sampling rate for the audio data.",
choices=[32000, 40000, 48000],
required=True,
)
preprocess_parser.add_argument(
"--cpu_cores",
type=int,
help="Number of CPU cores to use for preprocessing.",
choices=range(1, 65),
)
# Parser for 'extract' mode
extract_parser = subparsers.add_parser(
"extract", help="Extract features from a dataset."
)
extract_parser.add_argument(
"--model_name", type=str, help="Name of the model.", required=True
)
extract_parser.add_argument(
"--rvc_version",
type=str,
help="Version of the RVC model ('v1' or 'v2').",
choices=["v1", "v2"],
default="v2",
)
extract_parser.add_argument(
"--f0_method",
type=str,
help="Pitch extraction method to use.",
choices=[
"crepe",
"crepe-tiny",
"rmvpe",
],
default="rmvpe",
)
extract_parser.add_argument(
"--pitch_guidance",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help="Enable or disable pitch guidance during feature extraction.",
default=True,
)
extract_parser.add_argument(
"--hop_length",
type=int,
help="Hop length for feature extraction. Only applicable for Crepe pitch extraction.",
choices=range(1, 513),
default=128,
)
extract_parser.add_argument(
"--cpu_cores",
type=int,
help="Number of CPU cores to use for feature extraction (optional).",
choices=range(1, 65),
default=None,
)
extract_parser.add_argument(
"--gpu",
type=int,
help="GPU device to use for feature extraction (optional).",
default="-",
)
extract_parser.add_argument(
"--sample_rate",
type=int,
help="Target sampling rate for the audio data.",
choices=[32000, 40000, 48000],
required=True,
)
extract_parser.add_argument(
"--embedder_model",
type=str,
help=embedder_model_description,
choices=[
"contentvec",
"japanese-hubert-base",
"chinese-hubert-large",
"custom",
],
default="contentvec",
)
extract_parser.add_argument(
"--embedder_model_custom",
type=str,
help=embedder_model_custom_description,
default=None,
)
# Parser for 'train' mode
train_parser = subparsers.add_parser("train", help="Train an RVC model.")
train_parser.add_argument(
"--model_name", type=str, help="Name of the model to be trained.", required=True
)
train_parser.add_argument(
"--rvc_version",
type=str,
help="Version of the RVC model to train ('v1' or 'v2').",
choices=["v1", "v2"],
default="v2",
)
train_parser.add_argument(
"--save_every_epoch",
type=int,
help="Save the model every specified number of epochs.",
choices=range(1, 101),
required=True,
)
train_parser.add_argument(
"--save_only_latest",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help="Save only the latest model checkpoint.",
default=False,
)
train_parser.add_argument(
"--save_every_weights",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help="Save model weights every epoch.",
default=True,
)
train_parser.add_argument(
"--total_epoch",
type=int,
help="Total number of epochs to train for.",
choices=range(1, 10001),
default=1000,
)
train_parser.add_argument(
"--sample_rate",
type=int,
help="Sampling rate of the training data.",
choices=[32000, 40000, 48000],
required=True,
)
train_parser.add_argument(
"--batch_size",
type=int,
help="Batch size for training.",
choices=range(1, 51),
default=8,
)
train_parser.add_argument(
"--gpu",
type=str,
help="GPU device to use for training (e.g., '0').",
default="0",
)
train_parser.add_argument(
"--pitch_guidance",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help="Enable or disable pitch guidance during training.",
default=True,
)
train_parser.add_argument(
"--pretrained",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help="Use a pretrained model for initialization.",
default=True,
)
train_parser.add_argument(
"--custom_pretrained",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help="Use a custom pretrained model.",
default=False,
)
train_parser.add_argument(
"--g_pretrained_path",
type=str,
nargs="?",
default=None,
help="Path to the pretrained generator model file.",
)
train_parser.add_argument(
"--d_pretrained_path",
type=str,
nargs="?",
default=None,
help="Path to the pretrained discriminator model file.",
)
train_parser.add_argument(
"--overtraining_detector",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help="Enable overtraining detection.",
default=False,
)
train_parser.add_argument(
"--overtraining_threshold",
type=int,
help="Threshold for overtraining detection.",
choices=range(1, 101),
default=50,
)
train_parser.add_argument(
"--sync_graph",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help="Enable graph synchronization for distributed training.",
default=False,
)
train_parser.add_argument(
"--cache_data_in_gpu",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help="Cache training data in GPU memory.",
default=False,
)
# Parser for 'index' mode
index_parser = subparsers.add_parser(
"index", help="Generate an index file for an RVC model."
)
index_parser.add_argument(
"--model_name", type=str, help="Name of the model.", required=True
)
index_parser.add_argument(
"--rvc_version",
type=str,
help="Version of the RVC model ('v1' or 'v2').",
choices=["v1", "v2"],
default="v2",
)
# Parser for 'model_extract' mode
model_extract_parser = subparsers.add_parser(
"model_extract", help="Extract a specific epoch from a trained model."
)
model_extract_parser.add_argument(
"--pth_path", type=str, help="Path to the main .pth model file.", required=True
)
model_extract_parser.add_argument(
"--model_name", type=str, help="Name of the model.", required=True
)
model_extract_parser.add_argument(
"--sample_rate",
type=int,
help="Sampling rate of the extracted model.",
choices=[32000, 40000, 48000],
required=True,
)
model_extract_parser.add_argument(
"--pitch_guidance",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
help="Enable or disable pitch guidance for the extracted model.",
required=True,
)
model_extract_parser.add_argument(
"--rvc_version",
type=str,
help="Version of the extracted RVC model ('v1' or 'v2').",
choices=["v1", "v2"],
default="v2",
)
model_extract_parser.add_argument(
"--epoch",
type=int,
help="Epoch number to extract from the model.",
choices=range(1, 10001),
required=True,
)
model_extract_parser.add_argument(
"--step",
type=str,
help="Step number to extract from the model (optional).",
required=False,
)
# Parser for 'model_information' mode
model_information_parser = subparsers.add_parser(
"model_information", help="Display information about a trained model."
)
model_information_parser.add_argument(
"--pth_path", type=str, help="Path to the .pth model file.", required=True
)
# Parser for 'model_blender' mode
model_blender_parser = subparsers.add_parser(
"model_blender", help="Fuse two RVC models together."
)
model_blender_parser.add_argument(
"--model_name", type=str, help="Name of the new fused model.", required=True
)
model_blender_parser.add_argument(
"--pth_path_1",
type=str,
help="Path to the first .pth model file.",
required=True,
)
model_blender_parser.add_argument(
"--pth_path_2",
type=str,
help="Path to the second .pth model file.",
required=True,
)
model_blender_parser.add_argument(
"--ratio",
type=float,
help="Ratio for blending the two models (0.0 to 1.0).",
choices=[(i / 10) for i in range(11)],
default=0.5,
)
# Parser for 'tensorboard' mode
subparsers.add_parser(
"tensorboard", help="Launch TensorBoard for monitoring training progress."
)
# Parser for 'download' mode
download_parser = subparsers.add_parser(
"download", help="Download a model from a provided link."
)
download_parser.add_argument(
"--model_link", type=str, help="Direct link to the model file.", required=True
)
# Parser for 'prerequisites' mode
prerequisites_parser = subparsers.add_parser(
"prerequisites", help="Install prerequisites for RVC."
)
prerequisites_parser.add_argument(
"--pretraineds_v1",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
default=True,
help="Download pretrained models for RVC v1.",
)
prerequisites_parser.add_argument(
"--pretraineds_v2",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
default=True,
help="Download pretrained models for RVC v2.",
)
prerequisites_parser.add_argument(
"--models",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
default=True,
help="Download additional models.",
)
prerequisites_parser.add_argument(
"--exe",
type=lambda x: bool(strtobool(x)),
choices=[True, False],
default=True,
help="Download required executables.",
)
# Parser for 'audio_analyzer' mode
audio_analyzer = subparsers.add_parser(
"audio_analyzer", help="Analyze an audio file."
)
audio_analyzer.add_argument(
"--input_path", type=str, help="Path to the input audio file.", required=True
)
# Parser for 'api' mode
api_parser = subparsers.add_parser("api", help="Start the RVC API server.")
api_parser.add_argument(
"--host", type=str, help="Host address for the API server.", default="127.0.0.1"
)
api_parser.add_argument(
"--port", type=int, help="Port for the API server.", default=8000
)
return parser.parse_args()
def main():
if len(sys.argv) == 1:
print("Please run the script with '-h' for more information.")
sys.exit(1)
args = parse_arguments()
try:
if args.mode == "infer":
run_infer_script(
pitch=args.pitch,
filter_radius=args.filter_radius,
index_rate=args.index_rate,
volume_envelope=args.volume_envelope,
protect=args.protect,
hop_length=args.hop_length,
f0_method=args.f0_method,
input_path=args.input_path,
output_path=args.output_path,
pth_path=args.pth_path,
index_path=args.index_path,
split_audio=args.split_audio,
f0_autotune=args.f0_autotune,
clean_audio=args.clean_audio,
clean_strength=args.clean_strength,
export_format=args.export_format,
embedder_model=args.embedder_model,
embedder_model_custom=args.embedder_model_custom,
upscale_audio=args.upscale_audio,
f0_file=args.f0_file,
)
elif args.mode == "batch_infer":
run_batch_infer_script(
pitch=args.pitch,
filter_radius=args.filter_radius,
index_rate=args.index_rate,
volume_envelope=args.volume_envelope,
protect=args.protect,
hop_length=args.hop_length,
f0_method=args.f0_method,
input_folder=args.input_folder,
output_folder=args.output_folder,
pth_path=args.pth_path,
index_path=args.index_path,
split_audio=args.split_audio,
f0_autotune=args.f0_autotune,
clean_audio=args.clean_audio,
clean_strength=args.clean_strength,
export_format=args.export_format,
embedder_model=args.embedder_model,
embedder_model_custom=args.embedder_model_custom,
upscale_audio=args.upscale_audio,
f0_file=args.f0_file,
)
elif args.mode == "tts":
run_tts_script(
tts_text=args.tts_text,
tts_voice=args.tts_voice,
tts_rate=args.tts_rate,
pitch=args.pitch,
filter_radius=args.filter_radius,
index_rate=args.index_rate,
volume_envelope=args.volume_envelope,
protect=args.protect,
hop_length=args.hop_length,
f0_method=args.f0_method,
input_path=args.input_path,
output_path=args.output_path,
pth_path=args.pth_path,
index_path=args.index_path,
split_audio=args.split_audio,
f0_autotune=args.f0_autotune,
clean_audio=args.clean_audio,
clean_strength=args.clean_strength,
export_format=args.export_format,
embedder_model=args.embedder_model,
embedder_model_custom=args.embedder_model_custom,
upscale_audio=args.upscale_audio,
f0_file=args.f0_file,
)
elif args.mode == "preprocess":
run_preprocess_script(
model_name=args.model_name,
dataset_path=args.dataset_path,
sample_rate=args.sample_rate,
cpu_cores=args.cpu_cores,
)
elif args.mode == "extract":
run_extract_script(
model_name=args.model_name,
rvc_version=args.rvc_version,
f0_method=args.f0_method,
pitch_guidance=args.pitch_guidance,
hop_length=args.hop_length,
cpu_cores=args.cpu_cores,
gpu=args.gpu,
sample_rate=args.sample_rate,
embedder_model=args.embedder_model,
embedder_model_custom=args.embedder_model_custom,
)
elif args.mode == "train":
run_train_script(
model_name=args.model_name,
rvc_version=args.rvc_version,
save_every_epoch=args.save_every_epoch,
save_only_latest=args.save_only_latest,
save_every_weights=args.save_every_weights,
total_epoch=args.total_epoch,
sample_rate=args.sample_rate,
batch_size=args.batch_size,
gpu=args.gpu,
pitch_guidance=args.pitch_guidance,
overtraining_detector=args.overtraining_detector,
overtraining_threshold=args.overtraining_threshold,
pretrained=args.pretrained,
custom_pretrained=args.custom_pretrained,
sync_graph=args.sync_graph,
cache_data_in_gpu=args.cache_data_in_gpu,
g_pretrained_path=args.g_pretrained_path,
d_pretrained_path=args.d_pretrained_path,
)
elif args.mode == "index":
run_index_script(
model_name=args.model_name,
rvc_version=args.rvc_version,
)
elif args.mode == "model_extract":
run_model_extract_script(
pth_path=args.pth_path,
model_name=args.model_name,
sample_rate=args.sample_rate,
pitch_guidance=args.pitch_guidance,
rvc_version=args.rvc_version,
epoch=args.epoch,
step=args.step,
)
elif args.mode == "model_information":
run_model_information_script(
pth_path=args.pth_path,
)
elif args.mode == "model_blender":
run_model_blender_script(
model_name=args.model_name,
pth_path_1=args.pth_path_1,
pth_path_2=args.pth_path_2,
ratio=args.ratio,
)
elif args.mode == "tensorboard":
run_tensorboard_script()
elif args.mode == "download":
run_download_script(
model_link=args.model_link,
)
elif args.mode == "prerequisites":
run_prerequisites_script(
pretraineds_v1=args.pretraineds_v1,
pretraineds_v2=args.pretraineds_v2,
models=args.models,
exe=args.exe,
)
elif args.mode == "audio_analyzer":
run_audio_analyzer_script(
input_path=args.input_path,
)
elif args.mode == "api":
run_api_script(
ip=args.host,
port=args.port,
)
except Exception as error:
print(f"An error occurred during execution: {error}")
import traceback
traceback.print_exc()
if __name__ == "__main__":
main()