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import csv
import datetime
import os
import re
import time
import uuid
from io import StringIO
import gradio as gr
import torch
import torchaudio
from huggingface_hub import HfApi, hf_hub_download, snapshot_download
from TTS.tts.configs.xtts_config import XttsConfig
from TTS.tts.models.xtts import Xtts
from vinorm import TTSnorm
# Initialize Hugging Face API
HF_TOKEN = os.environ.get("HF_TOKEN")
api = HfApi(token=HF_TOKEN)
PASSWORD = os.environ.get("KEY")
# Download model files if not already downloaded
print("Downloading viXTTS model files if not already present...")
checkpoint_dir = "model/"
repo_id = "capleaf/viXTTS"
use_deepspeed = False
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):
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,
)
# Load model configuration and initialize model
xtts_config = os.path.join(checkpoint_dir, "config.json")
config = XttsConfig()
config.load_json(xtts_config)
MODEL = Xtts.init_from_config(config)
MODEL.load_checkpoint(
config, checkpoint_dir=checkpoint_dir, use_deepspeed=use_deepspeed
)
if torch.cuda.is_available():
MODEL.cuda()
def authenticate(password):
if password == PASSWORD:
return gr.update(visible=True), gr.update(visible=False), gr.update(visible=False)
else:
return gr.update(visible=False), gr.update(visible=True, value="Invalid password"), gr.update(visible=True)
# Supported languages
supported_languages = config.languages
if "vi" not in supported_languages:
supported_languages.append("vi")
def normalize_vietnamese_text(text):
text = (
TTSnorm(text, unknown=False, lower=False, rule=True)
.replace("..", ".")
.replace("!.", "!")
.replace("?.", "?")
.replace(" .", ".")
.replace(" ,", ",")
.replace('"', "")
.replace("'", "")
.replace("AI", "Ây Ai")
.replace("A.I", "Ây Ai")
)
return text
def calculate_keep_len(text, lang):
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 predict(prompt, language, audio_file_pth, normalize_text=True):
if language not in supported_languages:
metrics_text = gr.Warning(
f"Language {language} is not supported. Please choose from the dropdown."
)
return None, metrics_text
if len(prompt) < 2:
metrics_text = gr.Warning("Please provide a longer prompt text.")
return None, metrics_text
try:
metrics_text = ""
t_latent = time.time()
try:
gpt_cond_latent, speaker_embedding = MODEL.get_conditioning_latents(
audio_path=audio_file_pth,
gpt_cond_len=30,
gpt_cond_chunk_len=4,
max_ref_length=60,
)
except Exception as e:
print("Speaker encoding error:", str(e))
metrics_text = gr.Warning("Error with reference audio.")
return None, metrics_text
prompt = re.sub("([^\x00-\x7F]|\w)(\.|\。|\?)", r"\1 \2", prompt)
if normalize_text and language == "vi":
prompt = normalize_vietnamese_text(prompt)
print("Generating new audio...")
t0 = time.time()
out = MODEL.inference(
prompt,
language,
gpt_cond_latent,
speaker_embedding,
repetition_penalty=5.0,
temperature=0.75,
enable_text_splitting=True,
)
inference_time = time.time() - t0
metrics_text += f"Time to generate audio: {round(inference_time * 1000)} ms\n"
real_time_factor = (time.time() - t0) / out["wav"].shape[-1] * 24000
metrics_text += f"Real-time factor (RTF): {real_time_factor:.2f}\n"
keep_len = calculate_keep_len(prompt, language)
out["wav"] = out["wav"][:keep_len]
torchaudio.save("output.wav", torch.tensor(out["wav"]).unsqueeze(0), 24000)
except RuntimeError as e:
print("RuntimeError:", str(e))
metrics_text = gr.Warning("An error occurred during processing.")
return None, metrics_text
return "output.wav", metrics_text
title = "Phòng Thu VMC"
with gr.Blocks(analytics_enabled=False) as demo:
with gr.Row():
with gr.Column():
gr.Markdown("## VMC LAB")
with gr.Column():
pass
with gr.Row():
with gr.Column():
input_text_gr = gr.Textbox(
label="Text Prompt",
info="One or two sentences at a time is better. Up to 200 text characters.",
value="Xin chào, hãy nhập nội dung cần thu âm vào đây",
)
language_gr = gr.Dropdown(
label="Language",
info="Select an output language for the synthesised speech",
choices=supported_languages,
value="vi",
)
normalize_text = gr.Checkbox(
label="Normalize Vietnamese Text",
info="Normalize Vietnamese Text",
value=True,
)
ref_gr = gr.Audio(
label="Reference Audio",
info="Click on the ✎ button to upload your own target speaker audio",
type="filepath",
value="model/samples/nu-luu-loat.wav",
)
tts_button = gr.Button("Send", elem_id="send-btn", visible=True)
with gr.Column():
audio_gr = gr.Audio(label="Synthesised Audio", autoplay=True)
out_text_gr = gr.Textbox(label="Metrics")
tts_button.click(
predict,
[input_text_gr, language_gr, ref_gr, normalize_text],
outputs=[audio_gr, out_text_gr],
api_name="predict",
)
login_btn.click(
fn=authenticate,
inputs=password_input,
outputs=[main_column, error_message, login_column]
)
submit_btn.click(
fn=pipe,
inputs=[text, voice, image_in],
outputs=[video_o],
concurrency_limit=3
)
demo.queue(max_size=10).launch(show_error=True, show_api=False)
demo.queue()
demo.launch(debug=True, show_api=True) |