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import os
import glob
import json
import traceback
import logging
import gradio as gr
import numpy as np
import librosa
import torch
import asyncio
import edge_tts
import yt_dlp
import ffmpeg
import subprocess
import sys
import io
import wave
from datetime import datetime
from fairseq import checkpoint_utils
from lib.infer_pack.models import (
SynthesizerTrnMs256NSFsid,
SynthesizerTrnMs256NSFsid_nono,
SynthesizerTrnMs768NSFsid,
SynthesizerTrnMs768NSFsid_nono,
)
from vc_infer_pipeline import VC
from config import Config
config = Config()
logging.getLogger("numba").setLevel(logging.WARNING)
limitation = os.getenv("SYSTEM") == "spaces"
audio_mode = []
f0method_mode = []
f0method_info = ""
if limitation is True:
audio_mode = ["Upload audio", "TTS Audio"]
f0method_mode = ["pm", "crepe", "harvest"]
f0method_info = "PM is fast, rmvpe is middle, Crepe or harvest is good but it was extremely slow (Default: PM)"
else:
audio_mode = ["Upload audio", "Youtube", "TTS Audio"]
f0method_mode = ["pm", "crepe", "harvest"]
f0method_info = "PM is fast, rmvpe is middle. Crepe or harvest is good but it was extremely slow (Default: PM))"
if os.path.isfile("rmvpe.pt"):
f0method_mode.insert(2, "rmvpe")
def create_vc_fn(model_title, tgt_sr, net_g, vc, if_f0, version, file_index):
def vc_fn(
vc_audio_mode,
vc_input,
vc_upload,
tts_text,
tts_voice,
f0_up_key,
f0_method,
index_rate,
filter_radius,
resample_sr,
rms_mix_rate,
protect,
):
try:
if vc_audio_mode == "Input path" or "Youtube" and vc_input != "":
audio, sr = librosa.load(vc_input, sr=16000, mono=True)
elif vc_audio_mode == "Upload audio":
if vc_upload is None:
return "You need to upload an audio", None
sampling_rate, audio = vc_upload
duration = audio.shape[0] / sampling_rate
if duration > 360 and limitation:
return "Please upload an audio file that is less than 1 minute.", None
audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32)
if len(audio.shape) > 1:
audio = librosa.to_mono(audio.transpose(1, 0))
if sampling_rate != 16000:
audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000)
elif vc_audio_mode == "TTS Audio":
if len(tts_text) > 600 and limitation:
return "Text is too long", None
if tts_text is None or tts_voice is None:
return "You need to enter text and select a voice", None
asyncio.run(edge_tts.Communicate(tts_text, "-".join(tts_voice.split('-')[:-1])).save("tts.mp3"))
audio, sr = librosa.load("tts.mp3", sr=16000, mono=True)
vc_input = "tts.mp3"
times = [0, 0, 0]
f0_up_key = int(f0_up_key)
audio_opt = vc.pipeline(
hubert_model,
net_g,
0,
audio,
vc_input,
times,
f0_up_key,
f0_method,
file_index,
# file_big_npy,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
protect,
f0_file=None,
)
info = f"[{datetime.now().strftime('%Y-%m-%d %H:%M')}]: npy: {times[0]}, f0: {times[1]}s, infer: {times[2]}s"
print(f"{model_title} | {info}")
return info, (tgt_sr, audio_opt)
except:
info = traceback.format_exc()
print(info)
return info, (None, None)
return vc_fn
def load_model():
categories = []
with open("weights/folder_info.json", "r", encoding="utf-8") as f:
folder_info = json.load(f)
for category_name, category_info in folder_info.items():
if not category_info['enable']:
continue
category_title = category_info['title']
category_folder = category_info['folder_path']
models = []
with open(f"weights/{category_folder}/model_info.json", "r", encoding="utf-8") as f:
models_info = json.load(f)
for character_name, info in models_info.items():
if not info['enable']:
continue
model_title = info['title']
model_name = info['model_path']
model_author = info.get("author", None)
model_cover = f"weights/{category_folder}/{character_name}/{info['cover']}"
model_index = f"weights/{category_folder}/{character_name}/{info['feature_retrieval_library']}"
cpt = torch.load(f"weights/{category_folder}/{character_name}/{model_name}", map_location="cpu")
tgt_sr = cpt["config"][-1]
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk
if_f0 = cpt.get("f0", 1)
version = cpt.get("version", "v1")
if version == "v1":
if if_f0 == 1:
net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=config.is_half)
else:
net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
model_version = "V1"
elif version == "v2":
if if_f0 == 1:
net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=config.is_half)
else:
net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
model_version = "V2"
del net_g.enc_q
print(net_g.load_state_dict(cpt["weight"], strict=False))
net_g.eval().to(config.device)
if config.is_half:
net_g = net_g.half()
else:
net_g = net_g.float()
vc = VC(tgt_sr, config)
print(f"Model loaded: {character_name} / {info['feature_retrieval_library']} | ({model_version})")
models.append((character_name, model_title, model_author, model_cover, model_version, create_vc_fn(model_title, tgt_sr, net_g, vc, if_f0, version, model_index)))
categories.append([category_title, category_folder, models])
return categories
def cut_vocal_and_inst(url, audio_provider, split_model):
if url != "":
if not os.path.exists("dl_audio"):
os.mkdir("dl_audio")
if audio_provider == "Youtube":
ydl_opts = {
'format': 'bestaudio/best',
'postprocessors': [{
'key': 'FFmpegExtractAudio',
'preferredcodec': 'wav',
}],
"outtmpl": 'dl_audio/youtube_audio',
}
with yt_dlp.YoutubeDL(ydl_opts) as ydl:
ydl.download([url])
audio_path = "dl_audio/youtube_audio.wav"
else:
# Spotify doesnt work.
# Need to find other solution soon.
'''
command = f"spotdl download {url} --output dl_audio/.wav"
result = subprocess.run(command.split(), stdout=subprocess.PIPE)
print(result.stdout.decode())
audio_path = "dl_audio/spotify_audio.wav"
'''
if split_model == "htdemucs":
command = f"demucs --two-stems=vocals {audio_path} -o output"
result = subprocess.run(command.split(), stdout=subprocess.PIPE)
print(result.stdout.decode())
return "output/htdemucs/youtube_audio/vocals.wav", "output/htdemucs/youtube_audio/no_vocals.wav", audio_path, "output/htdemucs/youtube_audio/vocals.wav"
else:
command = f"demucs --two-stems=vocals -n mdx_extra_q {audio_path} -o output"
result = subprocess.run(command.split(), stdout=subprocess.PIPE)
print(result.stdout.decode())
return "output/mdx_extra_q/youtube_audio/vocals.wav", "output/mdx_extra_q/youtube_audio/no_vocals.wav", audio_path, "output/mdx_extra_q/youtube_audio/vocals.wav"
else:
raise gr.Error("URL Required!")
return None, None, None, None
def combine_vocal_and_inst(audio_data, audio_volume, split_model):
if not os.path.exists("output/result"):
os.mkdir("output/result")
vocal_path = "output/result/output.wav"
output_path = "output/result/combine.mp3"
if split_model == "htdemucs":
inst_path = "output/htdemucs/youtube_audio/no_vocals.wav"
else:
inst_path = "output/mdx_extra_q/youtube_audio/no_vocals.wav"
with wave.open(vocal_path, "w") as wave_file:
wave_file.setnchannels(1)
wave_file.setsampwidth(2)
wave_file.setframerate(audio_data[0])
wave_file.writeframes(audio_data[1].tobytes())
command = f'ffmpeg -y -i {inst_path} -i {vocal_path} -filter_complex [1:a]volume={audio_volume}dB[v];[0:a][v]amix=inputs=2:duration=longest -b:a 320k -c:a libmp3lame {output_path}'
result = subprocess.run(command.split(), stdout=subprocess.PIPE)
print(result.stdout.decode())
return output_path
def load_hubert():
global hubert_model
models, _, _ = checkpoint_utils.load_model_ensemble_and_task(
["hubert_base.pt"],
suffix="",
)
hubert_model = models[0]
hubert_model = hubert_model.to(config.device)
if config.is_half:
hubert_model = hubert_model.half()
else:
hubert_model = hubert_model.float()
hubert_model.eval() |