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import json
import os
import psutil
import time
from threading import Timer
import librosa
import numpy as np
import torch
from torch import no_grad, LongTensor
import commons
import utils
import gradio as gr
from models import SynthesizerTrn
from text import text_to_sequence, _clean_text
from mel_processing import spectrogram_torch
limitation = os.getenv("SYSTEM") == "spaces" # limit text and audio length in huggingface spaces
def get_text(text, hps, is_phoneme):
text_norm = text_to_sequence(text, hps.symbols, [] if is_phoneme else hps.data.text_cleaners)
if hps.data.add_blank:
text_norm = commons.intersperse(text_norm, 0)
text_norm = LongTensor(text_norm)
return text_norm
def create_tts_fn(model, hps, speaker_ids):
def tts_fn(text, speaker, speed, is_phoneme):
if limitation and ((len(text) > 60 and not is_phoneme) or (len(text) > 120 and is_phoneme)):
return "Error: Text is too long", None
speaker_id = speaker_ids[speaker]
stn_tst = get_text(text, hps, is_phoneme)
with no_grad():
x_tst = stn_tst.unsqueeze(0)
x_tst_lengths = LongTensor([stn_tst.size(0)])
sid = LongTensor([speaker_id])
audio = model.infer(x_tst, x_tst_lengths, sid=sid, noise_scale=.667, noise_scale_w=0.8,
length_scale=1.0 / speed)[0][0, 0].data.cpu().float().numpy()
del stn_tst, x_tst, x_tst_lengths, sid
return "Success", (hps.data.sampling_rate, audio)
return tts_fn
def create_vc_fn(model, hps, speaker_ids):
def vc_fn(original_speaker, target_speaker, input_audio):
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 limitation and duration > 15:
return "Error: Audio is too long", None
original_speaker_id = speaker_ids[original_speaker]
target_speaker_id = speaker_ids[target_speaker]
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 != hps.data.sampling_rate:
audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=hps.data.sampling_rate)
with no_grad():
y = torch.FloatTensor(audio)
y = y.unsqueeze(0)
spec = spectrogram_torch(y, hps.data.filter_length,
hps.data.sampling_rate, hps.data.hop_length, hps.data.win_length,
center=False)
spec_lengths = LongTensor([spec.size(-1)])
sid_src = LongTensor([original_speaker_id])
sid_tgt = LongTensor([target_speaker_id])
audio = model.voice_conversion(spec, spec_lengths, sid_src=sid_src, sid_tgt=sid_tgt)[0][
0, 0].data.cpu().float().numpy()
del y, spec, spec_lengths, sid_src, sid_tgt
return "Success", (hps.data.sampling_rate, audio)
return vc_fn
def system_monitor():
def get_size(bs, suffix="B"):
"""
Scale bytes to its proper format
e.g:
1253656 => '1.20MB'
1253656678 => '1.17GB'
"""
factor = 1024
for unit in ["", "K", "M", "G", "T", "P"]:
if bs < factor:
return f"{bs:.2f}{unit}{suffix}"
bs /= factor
def read_int(path):
with open(path, "r") as f:
return int(f.read())
def print_sys_status():
try:
cpu_t1 = read_int("/sys/fs/cgroup/cpu/cpuacct.usage")
t1 = time.time() * 1000000000
time.sleep(1)
cpu_t2 = read_int("/sys/fs/cgroup/cpu/cpuacct.usage")
t2 = time.time() * 1000000000
cpu_percent = (cpu_t2 - cpu_t1) / (t2 - t1) * 100
mem_total = get_size(read_int("/sys/fs/cgroup/memory/memory.limit_in_bytes"))
mem_usage = get_size(read_int("/sys/fs/cgroup/memory/memory.usage_in_bytes"))
print("=" * 10, "CPU & Mem Information", "=" * 10)
print(f"CPU: {cpu_percent}%, "
f"Mem: {mem_usage}/{mem_total}")
except FileNotFoundError:
pass
print("=" * 10, "Disk Information", "=" * 10)
# get all disk partitions
partitions = psutil.disk_partitions()
disk_info = ""
for partition in partitions:
disk_info += f"{partition.mountpoint}: "
try:
partition_usage = psutil.disk_usage(partition.mountpoint)
except PermissionError:
continue
disk_info += f"{partition_usage.percent}%({get_size(partition_usage.total)}), "
print(disk_info)
tr = Timer(60, print_sys_status)
tr.start()
print_sys_status()
css = """
#advanced-btn {
color: white;
border-color: black;
background: black;
font-size: .7rem !important;
line-height: 19px;
margin-top: 24px;
margin-bottom: 12px;
padding: 2px 8px;
border-radius: 14px !important;
}
#advanced-options {
display: none;
margin-bottom: 20px;
}
"""
if __name__ == '__main__':
if os.getenv("SYSTEM") == "spaces":
system_monitor() # debug
models = []
with open("saved_model/names.json", "r", encoding="utf-8") as f:
models_names = json.load(f)
for i, models_name in models_names.items():
config_path = f"saved_model/{i}/config.json"
model_path = f"saved_model/{i}/model.pth"
cover_path = f"saved_model/{i}/cover.jpg"
hps = utils.get_hparams_from_file(config_path)
model = SynthesizerTrn(
len(hps.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(model_path, model, None)
model.eval()
speaker_ids = [sid for sid, name in enumerate(hps.speakers) if name != "None"]
speakers = [name for sid, name in enumerate(hps.speakers) if name != "None"]
models.append((models_name, cover_path, speakers, hps.symbols,
create_tts_fn(model, hps, speaker_ids), create_vc_fn(model, hps, speaker_ids)))
app = gr.Blocks(css=css)
with app:
gr.Markdown("# Moe Japanese TTS And Voice Conversion Using VITS Model\n\n"
"\n\n"
"unofficial demo for \n\n"
"- [https://github.com/CjangCjengh/MoeGoe](https://github.com/CjangCjengh/MoeGoe)\n"
"- [https://github.com/Francis-Komizu/VITS](https://github.com/Francis-Komizu/VITS)"
)
with gr.Tabs():
with gr.TabItem("TTS"):
with gr.Tabs():
for i, (model_name, cover_path, speakers, symbols, tts_fn, vc_fn) in enumerate(models):
with gr.TabItem(f"model{i}"):
with gr.Column():
gr.Markdown(f"## {model_name}\n\n"
f"")
tts_input1 = gr.TextArea(label="Text (60 words limitation)", value="γγγ«γ‘γ―γ")
tts_input2 = gr.Dropdown(label="Speaker", choices=speakers,
type="index", value=speakers[0])
tts_input3 = gr.Slider(label="Speed", value=1, minimum=0.5, maximum=2, step=0.1)
advanced_button = gr.Button("Advanced options", elem_id="advanced-btn")
advanced_options = gr.Column()
advanced_options.elem_id = "advanced-options"
with advanced_options:
phoneme_input = gr.Checkbox(value=False, label="Phoneme input")
to_phoneme_btn = gr.Button("Covert text to phoneme")
phoneme_list = gr.Json(label="Phoneme list", value=symbols, elem_id="phoneme_list")
tts_submit = gr.Button("Generate", variant="primary")
tts_output1 = gr.Textbox(label="Output Message")
tts_output2 = gr.Audio(label="Output Audio")
advanced_button.click(None, [], [],
_js="""
() => {
const options = document.querySelector("body > gradio-app").querySelector("#advanced-options");
options.style.display = ["none", ""].includes(options.style.display) ? "flex" : "none";
}""")
tts_submit.click(tts_fn, [tts_input1, tts_input2, tts_input3, phoneme_input],
[tts_output1, tts_output2])
to_phoneme_btn.click(lambda x: _clean_text(x, hps.data.text_cleaners) if x != "" else x,
[tts_input1], [tts_input1])
with gr.TabItem("Voice Conversion"):
with gr.Tabs():
for i, (model_name, cover_path, speakers, symbols, tts_fn, vc_fn) in enumerate(models):
with gr.TabItem(f"model{i}"):
gr.Markdown(f"## {model_name}\n\n"
f"")
vc_input1 = gr.Dropdown(label="Original Speaker", choices=speakers, type="index",
value=speakers[0])
vc_input2 = gr.Dropdown(label="Target Speaker", choices=speakers, type="index",
value=speakers[1])
vc_input3 = gr.Audio(label="Input Audio (15s limitation)")
vc_submit = gr.Button("Convert", variant="primary")
vc_output1 = gr.Textbox(label="Output Message")
vc_output2 = gr.Audio(label="Output Audio")
vc_submit.click(vc_fn, [vc_input1, vc_input2, vc_input3], [vc_output1, vc_output2])
app.launch()
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