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import os | |
import glob | |
import json | |
import argparse | |
import traceback | |
import logging | |
import gradio as gr | |
import numpy as np | |
import librosa | |
import torch | |
import asyncio | |
import edge_tts | |
from datetime import datetime | |
from fairseq import checkpoint_utils | |
from infer_pack.models import SynthesizerTrnMs256NSFsid, SynthesizerTrnMs256NSFsid_nono | |
from vc_infer_pipeline import VC | |
from config import Config | |
config = Config() | |
logging.getLogger("numba").setLevel(logging.WARNING) | |
limitation = os.getenv("SYSTEM") == "spaces" # limit audio length in huggingface spaces | |
def create_vc_fn(tgt_sr, net_g, vc, if_f0, file_index): | |
def vc_fn( | |
input_audio, | |
f0_up_key, | |
f0_method, | |
index_rate, | |
tts_mode, | |
tts_text, | |
tts_voice | |
): | |
try: | |
if tts_mode: | |
if len(tts_text) > 100 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) | |
else: | |
if input_audio is None: | |
return "You need to upload an audio", None | |
sampling_rate, audio = input_audio | |
duration = audio.shape[0] / sampling_rate | |
if duration > 20 and limitation: | |
return "Please upload an audio file that is less than 20 seconds. If you need to generate a longer audio file, please use Colab.", 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) | |
times = [0, 0, 0] | |
f0_up_key = int(f0_up_key) | |
audio_opt = vc.pipeline( | |
hubert_model, | |
net_g, | |
0, | |
audio, | |
times, | |
f0_up_key, | |
f0_method, | |
file_index, | |
index_rate, | |
if_f0, | |
f0_file=None, | |
) | |
print( | |
f"[{datetime.now().strftime('%Y-%m-%d %H:%M')}]: npy: {times[0]}, f0: {times[1]}s, infer: {times[2]}s" | |
) | |
return (tgt_sr, audio_opt) | |
except: | |
info = traceback.format_exc() | |
print(info) | |
return info, (None, None) | |
return vc_fn | |
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() | |
def change_to_tts_mode(tts_mode): | |
if tts_mode: | |
return gr.Audio.update(visible=False), gr.Textbox.update(visible=True), gr.Dropdown.update(visible=True) | |
else: | |
return gr.Audio.update(visible=True), gr.Textbox.update(visible=False), gr.Dropdown.update(visible=False) | |
if __name__ == '__main__': | |
load_hubert() | |
categories = [] | |
tts_voice_list = asyncio.get_event_loop().run_until_complete(edge_tts.list_voices()) | |
voices = [f"{v['ShortName']}-{v['Gender']}" for v in tts_voice_list] | |
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'] | |
description = category_info['description'] | |
models = [] | |
with open(f"weights/{category_folder}/model_info.json", "r", encoding="utf-8") as f: | |
models_info = json.load(f) | |
for model_name, info in models_info.items(): | |
if not info['enable']: | |
continue | |
model_title = info['title'] | |
model_author = info.get("author", None) | |
model_cover = f"weights/{category_folder}/{model_name}/{info['cover']}" | |
model_index = f"weights/{category_folder}/{model_name}/{info['feature_retrieval_library']}" | |
cpt = torch.load(f"weights/{category_folder}/{model_name}/{model_name}.pth", 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) | |
if if_f0 == 1: | |
net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=config.is_half) | |
else: | |
net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"]) | |
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: {model_name}") | |
models.append((model_name, model_title, model_author, model_cover, create_vc_fn(tgt_sr, net_g, vc, if_f0, model_index))) | |
categories.append([category_title, category_folder, description, models]) | |
with gr.Blocks() as app: | |
gr.Markdown( | |
"# <center> RVC Models\n" | |
"## <center> The input audio should be clean and pure voice without background music.\n" | |
"## <center> [Recommended to use google colab for more feature](https://colab.research.google.com/drive/110kiMZTdP6Ri1lY9-NbQf17GVPPhHyeT?usp=sharing)\n" | |
"### <center> This project was inspired by [zomehwh](https://huggingface.co/spaces/zomehwh/rvc-models) and [ardha27](https://huggingface.co/spaces/ardha27/rvc-models)\n" | |
"[![image](https://colab.research.google.com/assets/colab-badge.svg)](https://colab.research.google.com/drive/110kiMZTdP6Ri1lY9-NbQf17GVPPhHyeT?usp=sharing)\n\n" | |
"[![Original Repo](https://badgen.net/badge/icon/github?icon=github&label=Original%20Repo)](https://github.com/RVC-Project/Retrieval-based-Voice-Conversion-WebUI)" | |
) | |
for (folder_title, folder, description, models) in categories: | |
with gr.TabItem(folder_title): | |
if description: | |
gr.Markdown(f"### <center> {description}") | |
with gr.Tabs(): | |
if not models: | |
gr.Markdown("# <center> No Model Loaded.") | |
gr.Markdown("## <center> Please added the model or fix your model path.") | |
continue | |
with gr.Tabs(): | |
for (name, title, author, cover, vc_fn) in models: | |
with gr.TabItem(name): | |
with gr.Row(): | |
gr.Markdown( | |
'<div align="center">' | |
f'<div>{title}</div>\n'+ | |
(f'<div>Model author: {author}</div>' if author else "")+ | |
(f'<img style="width:auto;height:300px;" src="file/{cover}">' if cover else "")+ | |
'</div>' | |
) | |
with gr.Row(): | |
with gr.Column(): | |
vc_input = gr.Audio(label="Input audio"+' (less than 20 seconds)' if limitation else '') | |
vc_transpose = gr.Number(label="Transpose", value=0, info='Type "12" to change from male to female voice. Type "-12" to change female to male voice') | |
vc_f0method = gr.Radio( | |
label="Pitch extraction algorithm", | |
choices=["pm", "harvest"], | |
value="pm", | |
interactive=True, | |
info="PM is fast but Harvest is better for low frequencies. (Default: PM)" | |
) | |
vc_index_ratio = gr.Slider( | |
minimum=0, | |
maximum=1, | |
label="Retrieval feature ratio", | |
value=0.6, | |
interactive=True, | |
info="(Default: 0.6)" | |
) | |
tts_mode = gr.Checkbox(label="tts (use edge-tts as input)", value=False) | |
tts_text = gr.Textbox(visible=False,label="TTS text (100 words limitation)" if limitation else "TTS text") | |
tts_voice = gr.Dropdown(label="Edge-tts speaker", choices=voices, visible=False, allow_custom_value=False, value="en-US-AnaNeural-Female") | |
vc_submit = gr.Button("Generate", variant="primary") | |
with gr.Column(): | |
vc_output = gr.Audio(label="Output Audio") | |
vc_submit.click(vc_fn, [vc_input, vc_transpose, vc_f0method, vc_index_ratio, tts_mode, tts_text, tts_voice], [vc_output]) | |
tts_mode.change(change_to_tts_mode, [tts_mode], [vc_input, tts_text, tts_voice]) | |
app.queue(concurrency_count=1, max_size=20, api_open=config.api).launch(share=config.colab) |