This repository has been archived by the owner on Jan 13, 2024. It is now read-only.
-
Notifications
You must be signed in to change notification settings - Fork 62
/
app.py
365 lines (338 loc) · 12 KB
/
app.py
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
import asyncio
import datetime
import logging
import os
import time
import traceback
import edge_tts
import gradio as gr
import librosa
import torch
from fairseq import checkpoint_utils
from config import Config
from lib.infer_pack.models import (
SynthesizerTrnMs256NSFsid,
SynthesizerTrnMs256NSFsid_nono,
SynthesizerTrnMs768NSFsid,
SynthesizerTrnMs768NSFsid_nono,
)
from rmvpe import RMVPE
from vc_infer_pipeline import VC
logging.getLogger("fairseq").setLevel(logging.WARNING)
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)
limitation = os.getenv("SYSTEM") == "spaces"
config = Config()
edge_output_filename = "edge_output.mp3"
tts_voice_list = asyncio.get_event_loop().run_until_complete(edge_tts.list_voices())
tts_voices = [f"{v['ShortName']}-{v['Gender']}" for v in tts_voice_list]
model_root = "weights"
models = [
d for d in os.listdir(model_root) if os.path.isdir(os.path.join(model_root, d))
]
if len(models) == 0:
raise ValueError("No model found in `weights` folder")
models.sort()
def model_data(model_name):
# global n_spk, tgt_sr, net_g, vc, cpt, version, index_file
pth_files = [
os.path.join(model_root, model_name, f)
for f in os.listdir(os.path.join(model_root, model_name))
if f.endswith(".pth")
]
if len(pth_files) == 0:
raise ValueError(f"No pth file found in {model_root}/{model_name}")
pth_path = pth_files[0]
print(f"Loading {pth_path}")
cpt = torch.load(pth_path, 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"])
elif version == "v2":
if if_f0 == 1:
net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=config.is_half)
else:
net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
else:
raise ValueError("Unknown version")
del net_g.enc_q
net_g.load_state_dict(cpt["weight"], strict=False)
print("Model loaded")
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)
# n_spk = cpt["config"][-3]
index_files = [
os.path.join(model_root, model_name, f)
for f in os.listdir(os.path.join(model_root, model_name))
if f.endswith(".index")
]
if len(index_files) == 0:
print("No index file found")
index_file = ""
else:
index_file = index_files[0]
print(f"Index file found: {index_file}")
return tgt_sr, net_g, vc, version, index_file, if_f0
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()
return hubert_model.eval()
print("Loading hubert model...")
hubert_model = load_hubert()
print("Hubert model loaded.")
print("Loading rmvpe model...")
rmvpe_model = RMVPE("rmvpe.pt", config.is_half, config.device)
print("rmvpe model loaded.")
def tts(
model_name,
speed,
tts_text,
tts_voice,
f0_up_key,
f0_method,
index_rate,
protect,
filter_radius=3,
resample_sr=0,
rms_mix_rate=0.25,
):
print("------------------")
print(datetime.datetime.now())
print("tts_text:")
print(tts_text)
print(f"tts_voice: {tts_voice}")
print(f"Model name: {model_name}")
print(f"F0: {f0_method}, Key: {f0_up_key}, Index: {index_rate}, Protect: {protect}")
try:
if limitation and len(tts_text) > 280:
print("Error: Text too long")
return (
f"Text characters should be at most 280 in this huggingface space, but got {len(tts_text)} characters.",
None,
None,
)
tgt_sr, net_g, vc, version, index_file, if_f0 = model_data(model_name)
t0 = time.time()
if speed >= 0:
speed_str = f"+{speed}%"
else:
speed_str = f"{speed}%"
asyncio.run(
edge_tts.Communicate(
tts_text, "-".join(tts_voice.split("-")[:-1]), rate=speed_str
).save(edge_output_filename)
)
t1 = time.time()
edge_time = t1 - t0
audio, sr = librosa.load(edge_output_filename, sr=16000, mono=True)
duration = len(audio) / sr
print(f"Audio duration: {duration}s")
if limitation and duration >= 20:
print("Error: Audio too long")
return (
f"Audio should be less than 20 seconds in this huggingface space, but got {duration}s.",
edge_output_filename,
None,
)
f0_up_key = int(f0_up_key)
if not hubert_model:
load_hubert()
if f0_method == "rmvpe":
vc.model_rmvpe = rmvpe_model
times = [0, 0, 0]
audio_opt = vc.pipeline(
hubert_model,
net_g,
0,
audio,
edge_output_filename,
times,
f0_up_key,
f0_method,
index_file,
# file_big_npy,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
protect,
None,
)
if tgt_sr != resample_sr >= 16000:
tgt_sr = resample_sr
info = f"Success. Time: edge-tts: {edge_time}s, npy: {times[0]}s, f0: {times[1]}s, infer: {times[2]}s"
print(info)
return (
info,
edge_output_filename,
(tgt_sr, audio_opt),
)
except EOFError:
info = (
"It seems that the edge-tts output is not valid. "
"This may occur when the input text and the speaker do not match. "
"For example, maybe you entered Japanese (without alphabets) text but chose non-Japanese speaker?"
)
print(info)
return info, None, None
except:
info = traceback.format_exc()
print(info)
return info, None, None
initial_md = """
# RVC text-to-speech webui
This is a text-to-speech webui of RVC models.
Input text ➡[(edge-tts)](https://github.com/rany2/edge-tts)➡ Speech mp3 file ➡[(RVC)](https://github.com/RVC-Project/Retrieval-based-Voice-Conversion-WebUI)➡ Final output
"""
app = gr.Blocks()
with app:
gr.Markdown(initial_md)
with gr.Row():
with gr.Column():
model_name = gr.Dropdown(label="Model", choices=models, value=models[0])
f0_key_up = gr.Number(
label="Transpose (the best value depends on the models and speakers)",
value=0,
)
with gr.Column():
f0_method = gr.Radio(
label="Pitch extraction method (pm: very fast, low quality, rmvpe: a little slow, high quality)",
choices=["pm", "rmvpe"], # harvest and crepe is too slow
value="rmvpe",
interactive=True,
)
index_rate = gr.Slider(
minimum=0,
maximum=1,
label="Index rate",
value=1,
interactive=True,
)
protect0 = gr.Slider(
minimum=0,
maximum=0.5,
label="Protect",
value=0.33,
step=0.01,
interactive=True,
)
with gr.Row():
with gr.Column():
tts_voice = gr.Dropdown(
label="Edge-tts speaker (format: language-Country-Name-Gender)",
choices=tts_voices,
allow_custom_value=False,
value="ja-JP-NanamiNeural-Female",
)
speed = gr.Slider(
minimum=-100,
maximum=100,
label="Speech speed (%)",
value=0,
step=10,
interactive=True,
)
tts_text = gr.Textbox(label="Input Text", value="これは日本語テキストから音声への変換デモです。")
with gr.Column():
but0 = gr.Button("Convert", variant="primary")
info_text = gr.Textbox(label="Output info")
with gr.Column():
edge_tts_output = gr.Audio(label="Edge Voice", type="filepath")
tts_output = gr.Audio(label="Result")
but0.click(
tts,
[
model_name,
speed,
tts_text,
tts_voice,
f0_key_up,
f0_method,
index_rate,
protect0,
],
[info_text, edge_tts_output, tts_output],
)
with gr.Row():
examples = gr.Examples(
examples_per_page=100,
examples=[
["これは日本語テキストから音声への変換デモです。", "ja-JP-NanamiNeural-Female"],
[
"This is an English text to speech conversation demo.",
"en-US-AriaNeural-Female",
],
["这是一个中文文本到语音的转换演示。", "zh-CN-XiaoxiaoNeural-Female"],
["한국어 텍스트에서 음성으로 변환하는 데모입니다.", "ko-KR-SunHiNeural-Female"],
[
"Il s'agit d'une démo de conversion du texte français à la parole.",
"fr-FR-DeniseNeural-Female",
],
[
"Dies ist eine Demo zur Umwandlung von Deutsch in Sprache.",
"de-DE-AmalaNeural-Female",
],
[
"Tämä on suomenkielinen tekstistä puheeksi -esittely.",
"fi-FI-NooraNeural-Female",
],
[
"Это демонстрационный пример преобразования русского текста в речь.",
"ru-RU-SvetlanaNeural-Female",
],
[
"Αυτή είναι μια επίδειξη μετατροπής ελληνικού κειμένου σε ομιλία.",
"el-GR-AthinaNeural-Female",
],
[
"Esta es una demostración de conversión de texto a voz en español.",
"es-ES-ElviraNeural-Female",
],
[
"Questa è una dimostrazione di sintesi vocale in italiano.",
"it-IT-ElsaNeural-Female",
],
[
"Esta é uma demonstração de conversão de texto em fala em português.",
"pt-PT-RaquelNeural-Female",
],
[
"Це демонстрація тексту до мовлення українською мовою.",
"uk-UA-PolinaNeural-Female",
],
[
"هذا عرض توضيحي عربي لتحويل النص إلى كلام.",
"ar-EG-SalmaNeural-Female",
],
[
"இது தமிழ் உரையிலிருந்து பேச்சு மாற்ற டெமோ.",
"ta-IN-PallaviNeural-Female",
],
],
inputs=[tts_text, tts_voice],
)
app.launch(inbrowser=True)