Spaces:
Runtime error
Runtime error
File size: 12,778 Bytes
c1bab10 |
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 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409 410 |
# This demo is adopted from https://github.com/coqui-ai/TTS/blob/dev/TTS/demos/xtts_ft_demo/xtts_demo.py
# With some modifications to fit the viXTTS model
import argparse
import hashlib
import logging
import os
import string
import subprocess
import sys
import tempfile
from datetime import datetime
import gradio as gr
import soundfile as sf
import torch
import torchaudio
from huggingface_hub import hf_hub_download, snapshot_download
from underthesea import sent_tokenize
from unidecode import unidecode
from vinorm import TTSnorm
from TTS.tts.configs.xtts_config import XttsConfig
from TTS.tts.models.xtts import Xtts
XTTS_MODEL = None
SCRIPT_DIR = os.path.dirname(os.path.abspath(__file__))
MODEL_DIR = os.path.join(SCRIPT_DIR, "model")
OUTPUT_DIR = os.path.join(SCRIPT_DIR, "output")
FILTER_SUFFIX = "_DeepFilterNet3.wav"
os.makedirs(OUTPUT_DIR, exist_ok=True)
def clear_gpu_cache():
if torch.cuda.is_available():
torch.cuda.empty_cache()
def load_model(checkpoint_dir="model/", repo_id="tuandaodev/xtts-vi-vinai-100h-custom-dvae", use_deepspeed=False):
global XTTS_MODEL
clear_gpu_cache()
os.makedirs(checkpoint_dir, exist_ok=True)
required_files = ["model.pth", "config.json", "vocab.json", "speakers_xtts.pth"]
files_in_dir = os.listdir(checkpoint_dir)
if not all(file in files_in_dir for file in required_files):
yield f"Missing model files! Downloading from {repo_id}..."
snapshot_download(
repo_id=repo_id,
repo_type="model",
local_dir=checkpoint_dir,
)
hf_hub_download(
repo_id="coqui/XTTS-v2",
filename="speakers_xtts.pth",
local_dir=checkpoint_dir,
)
yield f"Model download finished..."
xtts_config = os.path.join(checkpoint_dir, "config.json")
config = XttsConfig()
config.load_json(xtts_config)
XTTS_MODEL = Xtts.init_from_config(config)
yield "Loading model..."
XTTS_MODEL.load_checkpoint(
config, checkpoint_dir=checkpoint_dir, use_deepspeed=use_deepspeed
)
if torch.cuda.is_available():
XTTS_MODEL.cuda()
print("Model Loaded!")
yield "Model Loaded!"
# Define dictionaries to store cached results
cache_queue = []
speaker_audio_cache = {}
filter_cache = {}
conditioning_latents_cache = {}
def invalidate_cache(cache_limit=50):
"""Invalidate the cache for the oldest key"""
if len(cache_queue) > cache_limit:
key_to_remove = cache_queue.pop(0)
print("Invalidating cache", key_to_remove)
if os.path.exists(key_to_remove):
os.remove(key_to_remove)
if os.path.exists(key_to_remove.replace(".wav", "_DeepFilterNet3.wav")):
os.remove(key_to_remove.replace(".wav", "_DeepFilterNet3.wav"))
if key_to_remove in filter_cache:
del filter_cache[key_to_remove]
if key_to_remove in conditioning_latents_cache:
del conditioning_latents_cache[key_to_remove]
def generate_hash(data):
hash_object = hashlib.md5()
hash_object.update(data)
return hash_object.hexdigest()
def get_file_name(text, max_char=50):
filename = text[:max_char]
filename = filename.lower()
filename = filename.replace(" ", "_")
filename = filename.translate(
str.maketrans("", "", string.punctuation.replace("_", ""))
)
filename = unidecode(filename)
current_datetime = datetime.now().strftime("%m%d%H%M%S")
filename = f"{current_datetime}_{filename}"
return filename
def normalize_vietnamese_text(text):
text = text.encode('utf-8', 'ignore').decode('utf-8')
text = (
TTSnorm(text, unknown=False, lower=False, rule=True)
.replace("..", ".")
.replace("!.", "!")
.replace("?.", "?")
.replace(" .", ".")
.replace(" ,", ",")
.replace('"', "")
.replace("'", "")
)
return text
def calculate_keep_len(text, lang):
"""Simple hack for short sentences"""
if lang in ["ja", "zh-cn"]:
return -1
word_count = len(text.split())
num_punct = text.count(".") + text.count("!") + text.count("?") + text.count(",")
if word_count < 5:
return 15000 * word_count + 2000 * num_punct
elif word_count < 10:
return 13000 * word_count + 2000 * num_punct
return -1
def run_tts(lang, tts_text, speaker_audio_file, use_deepfilter, normalize_text):
global filter_cache, conditioning_latents_cache, cache_queue
if XTTS_MODEL is None:
return "You need to run the previous step to load the model !!", None, None
if not speaker_audio_file:
return "You need to provide reference audio!!!", None, None
# Use the file name as the key, since it's suppose to be unique 💀
speaker_audio_key = speaker_audio_file
if not speaker_audio_key in cache_queue:
cache_queue.append(speaker_audio_key)
invalidate_cache()
# Check if filtered reference is cached
if use_deepfilter and speaker_audio_key in filter_cache:
print("Using filter cache...")
speaker_audio_file = filter_cache[speaker_audio_key]
elif use_deepfilter:
print("Running filter...")
subprocess.run(
[
"deepFilter",
speaker_audio_file,
"-o",
os.path.dirname(speaker_audio_file),
]
)
filter_cache[speaker_audio_key] = speaker_audio_file.replace(
".wav", FILTER_SUFFIX
)
speaker_audio_file = filter_cache[speaker_audio_key]
# Check if conditioning latents are cached
cache_key = (
speaker_audio_key,
XTTS_MODEL.config.gpt_cond_len,
XTTS_MODEL.config.max_ref_len,
XTTS_MODEL.config.sound_norm_refs,
)
if cache_key in conditioning_latents_cache:
print("Using conditioning latents cache...")
gpt_cond_latent, speaker_embedding = conditioning_latents_cache[cache_key]
else:
print("Computing conditioning latents...")
gpt_cond_latent, speaker_embedding = XTTS_MODEL.get_conditioning_latents(
audio_path=speaker_audio_file,
gpt_cond_len=XTTS_MODEL.config.gpt_cond_len,
max_ref_length=XTTS_MODEL.config.max_ref_len,
sound_norm_refs=XTTS_MODEL.config.sound_norm_refs,
)
conditioning_latents_cache[cache_key] = (gpt_cond_latent, speaker_embedding)
if normalize_text and lang == "vi":
tts_text = normalize_vietnamese_text(tts_text)
# Split text by sentence
if lang in ["ja", "zh-cn"]:
sentences = tts_text.split("。")
else:
sentences = sent_tokenize(tts_text)
from pprint import pprint
pprint(sentences)
wav_chunks = []
for sentence in sentences:
if sentence.strip() == "":
continue
wav_chunk = XTTS_MODEL.inference(
text=sentence,
language=lang,
gpt_cond_latent=gpt_cond_latent,
speaker_embedding=speaker_embedding,
# The following values are carefully chosen for viXTTS
temperature=0.3,
length_penalty=1.0,
repetition_penalty=10.0,
top_k=30,
top_p=0.85,
enable_text_splitting=True,
)
keep_len = calculate_keep_len(sentence, lang)
wav_chunk["wav"] = wav_chunk["wav"][:keep_len]
wav_chunks.append(torch.tensor(wav_chunk["wav"]))
out_wav = torch.cat(wav_chunks, dim=0).unsqueeze(0)
gr_audio_id = os.path.basename(os.path.dirname(speaker_audio_file))
out_path = os.path.join(OUTPUT_DIR, f"{get_file_name(tts_text)}_{gr_audio_id}.wav")
print("Saving output to ", out_path)
torchaudio.save(out_path, out_wav, 24000)
return "Speech generated !", out_path
# Define a logger to redirect
class Logger:
def __init__(self, filename="log.out"):
self.log_file = filename
self.terminal = sys.stdout
self.log = open(self.log_file, "w")
def write(self, message):
self.terminal.write(message)
self.log.write(message)
def flush(self):
self.terminal.flush()
self.log.flush()
def isatty(self):
return False
# Redirect stdout and stderr to a file
sys.stdout = Logger()
sys.stderr = sys.stdout
logging.basicConfig(
level=logging.ERROR,
format="%(asctime)s [%(levelname)s] %(message)s",
handlers=[logging.StreamHandler(sys.stdout)],
)
def read_logs():
sys.stdout.flush()
with open(sys.stdout.log_file, "r") as f:
return f.read()
if __name__ == "__main__":
parser = argparse.ArgumentParser(
description="""XTTS VI inference demo\n\n""",
formatter_class=argparse.RawTextHelpFormatter,
)
parser.add_argument(
"--port",
type=int,
help="Port to run the gradio demo. Default: 5003",
default=5003,
)
parser.add_argument(
"--model_dir",
type=str,
help="Path to the checkpoint directory. This directory must contain 04 files: model.pth, config.json, vocab.json and speakers_xtts.pth",
default=None,
)
parser.add_argument(
"--reference_audio",
type=str,
help="Path to the reference audio file.",
default=None,
)
args = parser.parse_args()
if args.model_dir:
MODEL_DIR = os.path.abspath(args.model_dir)
REFERENCE_AUDIO = os.path.join(SCRIPT_DIR, "assets", "vi-man_kien-thuc-quan-su.wav")
if args.reference_audio:
REFERENCE_AUDIO = os.abspath(args.reference_audio)
with gr.Blocks() as demo:
intro = """
# XTTS VI Inference Demo
"""
gr.Markdown(intro)
with gr.Row():
with gr.Column() as col1:
repo_id = gr.Textbox(
label="HuggingFace Repo ID",
value="tuandaodev/xtts-vi-vinai-100h-custom-dvae",
)
checkpoint_dir = gr.Textbox(
label="XTTS VI model directory",
value=MODEL_DIR,
)
use_deepspeed = gr.Checkbox(
value=True, label="Use DeepSpeed for faster inference"
)
progress_load = gr.Label(label="Progress:")
load_btn = gr.Button(
value="Step 1 - Load XTTS VI model", variant="primary"
)
with gr.Column() as col2:
speaker_reference_audio = gr.Audio(
label="Speaker reference audio:",
value=REFERENCE_AUDIO,
type="filepath",
)
tts_language = gr.Dropdown(
label="Language",
value="vi",
choices=[
"vi",
"en",
"es",
"fr",
"de",
"it",
"pt",
"pl",
"tr",
"ru",
"nl",
"cs",
"ar",
"zh",
"hu",
"ko",
"ja",
],
)
use_filter = gr.Checkbox(
label="Denoise Reference Audio",
value=True,
)
normalize_text = gr.Checkbox(
label="Normalize Input Text",
value=True,
)
tts_text = gr.Textbox(
label="Input Text.",
value="Xin chào, tôi là một công cụ chuyển đổi văn bản thành giọng nói tiếng Việt, được huấn luyện trong môn học xử lý giọng nói.",
)
tts_btn = gr.Button(value="Step 2 - Inference", variant="primary")
with gr.Column() as col3:
progress_gen = gr.Label(label="Progress:")
tts_output_audio = gr.Audio(label="Generated Audio.")
load_btn.click(
fn=load_model,
inputs=[checkpoint_dir, repo_id, use_deepspeed],
outputs=[progress_load],
)
tts_btn.click(
fn=run_tts,
inputs=[
tts_language,
tts_text,
speaker_reference_audio,
use_filter,
normalize_text,
],
outputs=[progress_gen, tts_output_audio],
)
demo.launch(share=True, debug=False, server_port=args.port, server_name="0.0.0.0")
|