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import sys | |
import logging | |
import os | |
import json | |
import torch | |
import argparse | |
import commons | |
import utils | |
import gradio as gr | |
from models import SynthesizerTrn | |
from text.symbols import symbols | |
from text import cleaned_text_to_sequence, get_bert | |
from text.cleaner import clean_text | |
logging.getLogger("numba").setLevel(logging.WARNING) | |
logging.getLogger("markdown_it").setLevel(logging.WARNING) | |
logging.getLogger("urllib3").setLevel(logging.WARNING) | |
logging.getLogger("matplotlib").setLevel(logging.WARNING) | |
logging.basicConfig( | |
level=logging.INFO, format="| %(name)s | %(levelname)s | %(message)s" | |
) | |
logger = logging.getLogger(__name__) | |
limitation = os.getenv("SYSTEM") == "spaces" # limit text and audio length in huggingface spaces | |
def get_text(text, hps): | |
language_str = "JP" | |
norm_text, phone, tone, word2ph = clean_text(text, language_str) | |
phone, tone, language = cleaned_text_to_sequence(phone, tone, language_str) | |
if hps.data.add_blank: | |
phone = commons.intersperse(phone, 0) | |
tone = commons.intersperse(tone, 0) | |
language = commons.intersperse(language, 0) | |
for i in range(len(word2ph)): | |
word2ph[i] = word2ph[i] * 2 | |
word2ph[0] += 1 | |
bert = get_bert(norm_text, word2ph, language_str, device) | |
del word2ph | |
assert bert.shape[-1] == len(phone), phone | |
ja_bert = bert | |
bert = torch.zeros(1024, len(phone)) | |
assert bert.shape[-1] == len( | |
phone | |
), f"Bert seq len {bert.shape[-1]} != {len(phone)}" | |
phone = torch.LongTensor(phone) | |
tone = torch.LongTensor(tone) | |
language = torch.LongTensor(language) | |
return bert, ja_bert, phone, tone, language | |
def infer(text, sdp_ratio, noise_scale, noise_scale_w, length_scale, sid, net_g_ms, hps): | |
bert, ja_bert, phones, tones, lang_ids = get_text(text, hps) | |
with torch.no_grad(): | |
x_tst = phones.to(device).unsqueeze(0) | |
tones = tones.to(device).unsqueeze(0) | |
lang_ids = lang_ids.to(device).unsqueeze(0) | |
bert = bert.to(device).unsqueeze(0) | |
ja_bert = ja_bert.to(device).unsqueeze(0) | |
x_tst_lengths = torch.LongTensor([phones.size(0)]).to(device) | |
del phones | |
sid = torch.LongTensor([sid]).to(device) | |
audio = ( | |
net_g_ms.infer( | |
x_tst, | |
x_tst_lengths, | |
sid, | |
tones, | |
lang_ids, | |
bert, | |
ja_bert, | |
sdp_ratio=sdp_ratio, | |
noise_scale=noise_scale, | |
noise_scale_w=noise_scale_w, | |
length_scale=length_scale, | |
)[0][0, 0] | |
.data.cpu() | |
.float() | |
.numpy() | |
) | |
del x_tst, tones, lang_ids, bert, x_tst_lengths, sid | |
torch.cuda.empty_cache() | |
return audio | |
def create_tts_fn(net_g_ms, hps): | |
def tts_fn(text, speaker, sdp_ratio, noise_scale, noise_scale_w, length_scale): | |
print(f"{text} | {speaker}") | |
sid = hps.data.spk2id[speaker] | |
text = text.replace('\n', ' ').replace('\r', '').replace(" ", "") | |
if limitation: | |
max_len = 100 | |
if len(text) > max_len: | |
return "Error: Text is too long", None | |
audio = infer(text, sdp_ratio=sdp_ratio, noise_scale=noise_scale, noise_scale_w=noise_scale_w, | |
length_scale=length_scale, sid=sid, net_g_ms=net_g_ms, hps=hps) | |
return "Success", (hps.data.sampling_rate, audio) | |
return tts_fn | |
if __name__ == "__main__": | |
device = ( | |
"cuda:0" | |
if torch.cuda.is_available() | |
else ( | |
"mps" | |
if sys.platform == "darwin" and torch.backends.mps.is_available() | |
else "cpu" | |
) | |
) | |
parser = argparse.ArgumentParser() | |
parser.add_argument("--share", default=False, help="make link public", action="store_true") | |
parser.add_argument("-d", "--debug", action="store_true", help="enable DEBUG-LEVEL log") | |
args = parser.parse_args() | |
if args.debug: | |
logger.info("Enable DEBUG-LEVEL log") | |
logging.basicConfig(level=logging.DEBUG) | |
models = [] | |
with open("pretrained_models/info.json", "r", encoding="utf-8") as f: | |
models_info = json.load(f) | |
for i, info in models_info.items(): | |
if not info['enable']: | |
continue | |
name = info['name'] | |
title = info['title'] | |
example = info['example'] | |
hps = utils.get_hparams_from_file(f"./pretrained_models/{name}/config.json") | |
net_g_ms = SynthesizerTrn( | |
len(symbols), | |
hps.data.filter_length // 2 + 1, | |
hps.train.segment_size // hps.data.hop_length, | |
n_speakers=hps.data.n_speakers, | |
**hps.model) | |
utils.load_checkpoint(f'pretrained_models/{i}/{i}.pth', net_g_ms, None, skip_optimizer=True) | |
_ = net_g_ms.eval().to(device) | |
models.append((name, title, example, list(hps.data.spk2id.keys()), net_g_ms, create_tts_fn(net_g_ms, hps))) | |
with gr.Blocks(theme='NoCrypt/miku') as app: | |
with gr.Tabs(): | |
for (name, title, example, speakers, net_g_ms, tts_fn) in models: | |
with gr.TabItem(name): | |
with gr.Row(): | |
gr.Markdown( | |
'<div align="center">' | |
f'<a><strong>{title}</strong></a>' | |
f'</div>' | |
) | |
with gr.Row(): | |
with gr.Column(): | |
input_text = gr.Textbox(label="Text (100 words limitation)" if limitation else "Text", lines=5, value=example) | |
btn = gr.Button(value="Generate", variant="primary") | |
with gr.Row(): | |
sp = gr.Dropdown(choices=speakers, value=speakers[0], label="Speaker") | |
with gr.Row(): | |
sdpr = gr.Slider(label="SDP Ratio", minimum=0, maximum=1, step=0.1, value=0.2) | |
ns = gr.Slider(label="noise_scale", minimum=0.1, maximum=1.0, step=0.1, value=0.6) | |
nsw = gr.Slider(label="noise_scale_w", minimum=0.1, maximum=1.0, step=0.1, value=0.8) | |
ls = gr.Slider(label="length_scale", minimum=0.1, maximum=2.0, step=0.1, value=1) | |
with gr.Column(): | |
o1 = gr.Textbox(label="Output Message") | |
o2 = gr.Audio(label="Output Audio") | |
btn.click(tts_fn, inputs=[input_text, sp, sdpr, ns, nsw, ls], outputs=[o1, o2]) | |
app.queue(concurrency_count=1).launch(share=args.share) | |