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Browse files- app.py +297 -0
- audio.wav +0 -0
- requirements.txt +15 -0
app.py
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1 |
+
# This demo is adopted from https://github.com/coqui-ai/TTS/blob/dev/TTS/demos/xtts_ft_demo/xtts_demo.py
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# With some modifications to fit the viXTTS model
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import argparse
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import hashlib
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import logging
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import os
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import string
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import subprocess
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import sys
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import tempfile
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from datetime import datetime
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import gradio as gr
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import torch
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import torchaudio
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from huggingface_hub import hf_hub_download, snapshot_download
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+
from underthesea import sent_tokenize
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from unidecode import unidecode
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from TTS.tts.configs.xtts_config import XttsConfig
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from TTS.tts.models.xtts import Xtts
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+
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XTTS_MODEL = None
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SCRIPT_DIR = os.path.dirname(os.path.abspath(__file__))
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MODEL_DIR = os.path.join(SCRIPT_DIR, "model")
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OUTPUT_DIR = os.path.join(SCRIPT_DIR, "output")
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os.makedirs(OUTPUT_DIR, exist_ok=True)
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def clear_gpu_cache():
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if torch.cuda.is_available():
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torch.cuda.empty_cache()
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def load_model(checkpoint_dir="model/", repo_id="capleaf/viXTTS", use_deepspeed=False):
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global XTTS_MODEL
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clear_gpu_cache()
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os.makedirs(checkpoint_dir, exist_ok=True)
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required_files = ["model.pth", "config.json", "vocab.json", "speakers_xtts.pth"]
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files_in_dir = os.listdir(checkpoint_dir)
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if not all(file in files_in_dir for file in required_files):
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print(f"Missing model files! Downloading from {repo_id}...")
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snapshot_download(
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repo_id=repo_id,
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repo_type="model",
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local_dir=checkpoint_dir,
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)
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hf_hub_download(
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repo_id="coqui/XTTS-v2",
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filename="speakers_xtts.pth",
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local_dir=checkpoint_dir,
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)
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print( f"Model download finished...")
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xtts_config = os.path.join(checkpoint_dir, "config.json")
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config = XttsConfig()
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config.load_json(xtts_config)
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XTTS_MODEL = Xtts.init_from_config(config)
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print( "Loading model...")
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XTTS_MODEL.load_checkpoint(
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config, checkpoint_dir=checkpoint_dir, use_deepspeed=False
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)
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if torch.cuda.is_available():
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XTTS_MODEL.cuda()
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else:
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print("use cpu")
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XTTS_MODEL.cpu()
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print("Model Loaded!")
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return XTTS_MODEL
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def generate_hash(data):
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hash_object = hashlib.md5()
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hash_object.update(data)
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return hash_object.hexdigest()
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def get_file_name(text, max_char=50):
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filename = text[:max_char]
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filename = filename.lower()
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filename = filename.replace(" ", "_")
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filename = filename.translate(
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str.maketrans("", "", string.punctuation.replace("_", ""))
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)
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filename = unidecode(filename)
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current_datetime = datetime.now().strftime("%m%d%H%M%S")
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filename = f"{current_datetime}_{filename}"
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return filename
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def normalize_vietnamese_text(text):
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text = (
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text
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.replace("..", ".")
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.replace("!.", "!")
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.replace("?.", "?")
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.replace(" .", ".")
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.replace(" ,", ",")
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.replace('"', "")
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.replace("'", "")
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.replace("AI", "Ây Ai")
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.replace("A.I", "Ây Ai")
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)
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return text
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def calculate_keep_len(text, lang):
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"""Simple hack for short sentences"""
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if lang in ["ja", "zh-cn"]:
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return -1
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word_count = len(text.split())
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num_punct = text.count(".") + text.count("!") + text.count("?") + text.count(",")
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+
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if word_count < 5:
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return 15000 * word_count + 2000 * num_punct
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elif word_count < 10:
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return 13000 * word_count + 2000 * num_punct
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return -1
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def run_tts(lang, tts_text, speaker_audio_file, normalize_text):
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global XTTS_MODEL
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if XTTS_MODEL is None:
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return "You need to run the previous step to load the model !!", None, None
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if not speaker_audio_file:
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return "You need to provide reference audio!!!", None, None
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print("Computing conditioning latents...")
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+
gpt_cond_latent, speaker_embedding = XTTS_MODEL.get_conditioning_latents(
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audio_path=speaker_audio_file,
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gpt_cond_len=XTTS_MODEL.config.gpt_cond_len,
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max_ref_length=XTTS_MODEL.config.max_ref_len,
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sound_norm_refs=XTTS_MODEL.config.sound_norm_refs,
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)
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if normalize_text and lang == "vi":
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tts_text = normalize_vietnamese_text(tts_text)
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+
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147 |
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# Split text by sentence
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148 |
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if lang in ["ja", "zh-cn"]:
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sentences = tts_text.split("。")
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else:
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sentences = sent_tokenize(tts_text)
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+
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from pprint import pprint
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+
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pprint(sentences)
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+
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wav_chunks = []
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158 |
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for sentence in sentences:
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if sentence.strip() == "":
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continue
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wav_chunk = XTTS_MODEL.inference(
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+
text=sentence,
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+
language=lang,
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gpt_cond_latent=gpt_cond_latent,
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speaker_embedding=speaker_embedding,
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166 |
+
# The following values are carefully chosen for viXTTS
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temperature=0.3,
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+
length_penalty=1.0,
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repetition_penalty=10.0,
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top_k=30,
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top_p=0.85,
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enable_text_splitting=True,
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)
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174 |
+
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175 |
+
keep_len = calculate_keep_len(sentence, lang)
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176 |
+
wav_chunk["wav"] = wav_chunk["wav"][:keep_len]
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177 |
+
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178 |
+
wav_chunks.append(torch.tensor(wav_chunk["wav"]))
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179 |
+
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180 |
+
out_wav = torch.cat(wav_chunks, dim=0).unsqueeze(0)
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181 |
+
gr_audio_id = os.path.basename(os.path.dirname(speaker_audio_file))
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182 |
+
out_path = os.path.join(OUTPUT_DIR, f"{get_file_name(tts_text)}_{gr_audio_id}.wav")
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183 |
+
print("Saving output to ", out_path)
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184 |
+
torchaudio.save(out_path, out_wav, 24000)
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185 |
+
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186 |
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return "Speech generated !", out_path
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187 |
+
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188 |
+
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189 |
+
# Define a logger to redirect
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190 |
+
class Logger:
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191 |
+
def __init__(self, filename="log.out"):
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192 |
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self.log_file = filename
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+
self.terminal = sys.stdout
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194 |
+
self.log = open(self.log_file, "w")
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195 |
+
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196 |
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def write(self, message):
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197 |
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self.terminal.write(message)
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198 |
+
self.log.write(message)
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199 |
+
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200 |
+
def flush(self):
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201 |
+
self.terminal.flush()
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202 |
+
self.log.flush()
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203 |
+
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204 |
+
def isatty(self):
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return False
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+
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# Redirect stdout and stderr to a file
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209 |
+
sys.stdout = Logger()
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210 |
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sys.stderr = sys.stdout
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211 |
+
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212 |
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213 |
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logging.basicConfig(
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214 |
+
level=logging.ERROR,
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215 |
+
format="%(asctime)s [%(levelname)s] %(message)s",
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handlers=[logging.StreamHandler(sys.stdout)],
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)
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218 |
+
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219 |
+
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220 |
+
def read_logs():
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221 |
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sys.stdout.flush()
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222 |
+
with open(sys.stdout.log_file, "r") as f:
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return f.read()
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+
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+
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if __name__ == "__main__":
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+
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REFERENCE_AUDIO = os.path.join(SCRIPT_DIR, "audio.wav")
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229 |
+
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230 |
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print("start loading model")
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231 |
+
XTTS_MODEL = load_model()
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232 |
+
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233 |
+
with gr.Blocks() as demo:
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234 |
+
intro = """
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235 |
+
# Fake giọng Demo
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236 |
+
Customize from HuggingFace: [viXTTS](https://huggingface.co/capleaf/viXTTS)
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237 |
+
"""
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238 |
+
gr.Markdown(intro)
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239 |
+
with gr.Row():
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240 |
+
with gr.Column() as col2:
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241 |
+
speaker_reference_audio = gr.Audio(
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242 |
+
label="Giọng đọc mẫu:",
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243 |
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value=REFERENCE_AUDIO,
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244 |
+
type="filepath",
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)
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246 |
+
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247 |
+
tts_language = gr.Dropdown(
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248 |
+
label="Language",
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249 |
+
value="vi",
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250 |
+
choices=[
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"vi",
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+
"en",
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"es",
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+
"fr",
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"de",
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"it",
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"pt",
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"pl",
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"tr",
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260 |
+
"ru",
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"nl",
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262 |
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"cs",
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263 |
+
"ar",
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264 |
+
"zh",
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265 |
+
"hu",
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266 |
+
"ko",
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267 |
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"ja",
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268 |
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],
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+
)
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270 |
+
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271 |
+
normalize_text = gr.Checkbox(
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272 |
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label="Normalize Input Text",
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273 |
+
value=True,
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274 |
+
)
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275 |
+
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276 |
+
tts_text = gr.Textbox(
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277 |
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label="Input Text.",
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278 |
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value="Xin chào, tôi là một công cụ chuyển đổi văn bản thành giọng nói tiếng Việt được phát triển bởi nhóm Nón lá.",
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279 |
+
)
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280 |
+
tts_btn = gr.Button(value="Inference", variant="primary")
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281 |
+
|
282 |
+
with gr.Column() as col3:
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283 |
+
progress_gen = gr.Label(label="Progress:")
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284 |
+
tts_output_audio = gr.Audio(label="Kết quả.")
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285 |
+
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286 |
+
tts_btn.click(
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287 |
+
fn=run_tts,
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288 |
+
inputs=[
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+
tts_language,
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290 |
+
tts_text,
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291 |
+
speaker_reference_audio,
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292 |
+
normalize_text,
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293 |
+
],
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294 |
+
outputs=[progress_gen, tts_output_audio],
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295 |
+
)
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296 |
+
|
297 |
+
demo.launch()
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audio.wav
ADDED
Binary file (459 kB). View file
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requirements.txt
ADDED
@@ -0,0 +1,15 @@
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1 |
+
TTS @ git+https://github.com/thinhlpg/TTS.git@ff217b3f27b294de194cc59c5119d1e08b06413c
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2 |
+
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3 |
+
gradio
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4 |
+
deepfilternet==0.5.6
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5 |
+
vinorm==2.0.7
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6 |
+
underthesea==6.8.0
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7 |
+
deepspeed
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8 |
+
cutlet
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9 |
+
unidic
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10 |
+
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11 |
+
huggingface-hub~=0.27.0
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12 |
+
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13 |
+
torch~=2.2.2
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+
torchaudio~=2.2.2
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15 |
+
Unidecode~=1.3.8
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