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# 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")