app.py

# 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 torch
import torchaudio
from huggingface_hub import hf_hub_download, snapshot_download
from underthesea import sent_tokenize
from unidecode import unidecode

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")
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="capleaf/viXTTS", 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):
        print(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,
        )
        print( 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)
    print( "Loading model...")
    XTTS_MODEL.load_checkpoint(
        config, checkpoint_dir=checkpoint_dir, use_deepspeed=False
    )
    if torch.cuda.is_available():
        XTTS_MODEL.cuda()
    else:
        print("use cpu")
        XTTS_MODEL.cpu()

    print("Model Loaded!")

    return XTTS_MODEL
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
        .replace("..", ".")
        .replace("!.", "!")
        .replace("?.", "?")
        .replace(" .", ".")
        .replace(" ,", ",")
        .replace('"', "")
        .replace("'", "")
        .replace("AI", "Ây Ai")
        .replace("A.I", "Ây Ai")
    )
    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, normalize_text):
    global XTTS_MODEL

    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

    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,
    )

    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__":

    REFERENCE_AUDIO = os.path.join(SCRIPT_DIR, "audio.wav")

    print("start loading model")
    XTTS_MODEL = load_model()

    with gr.Blocks() as demo:
        intro = """
        # Fake giọng Demo
        Customize from HuggingFace: [viXTTS](https://huggingface.co./capleaf/viXTTS)
        """
        gr.Markdown(intro)
        with gr.Row():
            with gr.Column() as col2:
                speaker_reference_audio = gr.Audio(
                    label="Giọng đọc mẫu:",
                    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",
                    ],
                )

                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 phát triển bởi nhóm Nón lá.",
                )
                tts_btn = gr.Button(value="Inference", variant="primary")

            with gr.Column() as col3:
                progress_gen = gr.Label(label="Progress:")
                tts_output_audio = gr.Audio(label="Kết quả.")

        tts_btn.click(
            fn=run_tts,
            inputs=[
                tts_language,
                tts_text,
                speaker_reference_audio,
                normalize_text,
            ],
            outputs=[progress_gen, tts_output_audio],
        )

    demo.launch()