Voice-Clone / app.py
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Update app.py
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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 spaces
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
from fastapi import FastAPI, File, UploadFile
from fastapi.responses import FileResponse
app = FastAPI()
os.system("python -m unidic download")
HF_TOKEN = os.environ.get("HF_TOKEN")
api = HfApi(token=HF_TOKEN)
print("Downloading if not downloaded viXTTS")
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,
)
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()
supported_languages = config.languages
if not "vi" 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
@spaces.GPU(queue=False)
def predict(prompt, language, audio_file_pth, normalize_text=True):
if language not in supported_languages:
metrics_text = gr.Warning(f"Language you put {language} in is not in is not in our Supported Languages, please choose from dropdown")
return (None, metrics_text)
speaker_wav = audio_file_pth
if len(prompt) < 2:
metrics_text = gr.Warning("Please give a longer prompt text")
return (None, metrics_text)
if len(prompt) > 250:
metrics_text = gr.Warning(str(len(prompt)) + " characters.\n" + "Your prompt is too long, please keep it under 250 characters\n" + "Văn bản quá dài, vui lòng giữ dưới 250 ký tự.")
return (None, metrics_text)
try:
metrics_text = ""
t_latent = time.time()
try:
(gpt_cond_latent, speaker_embedding) = MODEL.get_conditioning_latents(audio_path=speaker_wav, 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("It appears something wrong with reference, did you unmute your microphone?")
return (None, metrics_text)
prompt = re.sub("([^\x00-\x7F]|\w)(\.|\。|\?)", r"\1 \2\2", prompt)
if normalize_text and language == "vi":
prompt = normalize_vietnamese_text(prompt)
print("I: 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
print(f"I: Time to generate audio: {round(inference_time*1000)} milliseconds")
metrics_text += f"Time to generate audio: {round(inference_time*1000)} milliseconds\n"
real_time_factor = (time.time() - t0) / out["wav"].shape[-1] * 24000
print(f"Real-time factor (RTF): {real_time_factor}")
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:
if "device-side assert" in str(e):
print(f"Exit due to: Unrecoverable exception caused by language:{language} prompt:{prompt}", flush=True)
gr.Warning("Unhandled Exception encounter, please retry in a minute")
print("Cuda device-assert Runtime encountered need restart")
error_time = datetime.datetime.now().strftime("%d-%m-%Y-%H:%M:%S")
error_data = [error_time, prompt, language, audio_file_pth]
error_data = [str(e) if type(e) != str else e for e in error_data]
print(error_data)
print(speaker_wav)
write_io = StringIO()
csv.writer(write_io).writerows([error_data])
csv_upload = write_io.getvalue().encode()
filename = error_time + "_" + str(uuid.uuid4()) + ".csv"
print("Writing error csv")
error_api = HfApi()
error_api.upload_file(path_or_fileobj=csv_upload, path_in_repo=filename, repo_id="coqui/xtts-flagged-dataset", repo_type="dataset")
print("Writing error reference audio")
speaker_filename = error_time + "_reference_" + str(uuid.uuid4()) + ".wav"
error_api = HfApi()
error_api.upload_file(path_or_fileobj=speaker_wav, path_in_repo=speaker_filename, repo_id="coqui/xtts-flagged-dataset", repo_type="dataset")
space = api.get_space_runtime(repo_id=repo_id)
if space.stage != "BUILDING":
api.restart_space(repo_id=repo_id)
else:
print("TRIED TO RESTART but space is building")
else:
if "Failed to decode" in str(e):
print("Speaker encoding error", str(e))
metrics_text = gr.Warning(metrics_text="It appears something wrong with reference, did you unmute your microphone?")
else:
print("RuntimeError: non device-side assert error:", str(e))
metrics_text = gr.Warning("Something unexpected happened please retry again.")
return (None, metrics_text)
return ("output.wav", metrics_text)
@app.post("/synthesize")
async def api_synthesize(prompt: str, language: str = "vi", audio_file: UploadFile = File(...)):
audio_file_path = f"temp_{uuid.uuid4()}.wav"
with open(audio_file_path, "wb") as f:
f.write(await audio_file.read())
audio_output_path, metrics_text = predict(prompt, language, audio_file_path)
return FileResponse(audio_output_path, media_type="audio/wav")
with gr.Blocks(analytics_enabled=False) as demo:
with gr.Row():
with gr.Column():
gr.Markdown("""
# viXTTS Demo ✨
- Github: https://github.com/thinhlpg/vixtts-demo/
- viVoice: https://github.com/thinhlpg/viVoice
""")
with gr.Column():
pass
with gr.Row():
with gr.Column():
input_text_gr = gr.Textbox(label="Text Prompt (Văn bản cần đọc)", info="Mỗi câu nên từ 10 từ trở lên. Tối đa 250 ký tự (khoảng 2 - 3 câu).", value="Xin chào, tôi là một mô hình chuyển đổi văn bản thành giọng nói tiếng Việt.")
language_gr = gr.Dropdown(label="Language (Ngôn ngữ)", choices=["vi", "en", "es", "fr", "de", "it", "pt", "pl", "tr", "ru", "nl", "cs", "ar", "zh-cn", "ja", "ko", "hu", "hi"], max_choices=1, value="vi")
normalize_text = gr.Checkbox(label="Chuẩn hóa văn bản tiếng Việt", info="Normalize Vietnamese text", value=True)
ref_gr = gr.Audio(label="Reference Audio (Giọng mẫu)", type="filepath", value="model/samples/nu-luu-loat.wav")
tts_button = gr.Button("Đọc 🗣️🔥", elem_id="send-btn", visible=True, variant="primary")
with gr.Column():
audio_gr = gr.Audio(label="Synthesised Audio", autoplay=True)
out_text_gr = gr.Text(label="Metrics")
tts_button.click(
predict,
[input_text_gr, language_gr, ref_gr, normalize_text],
outputs=[audio_gr, out_text_gr],
api_name="predict",
)
demo.queue()
demo.launch(debug=True, show_api=True, share=True, server_name="0.0.0.0", server_port=7860)