slice.py

import argparse
import shutil
from pathlib import Path
from queue import Queue
from threading import Thread
from typing import Any, Optional

import soundfile as sf
import torch
from tqdm import tqdm

from config import get_path_config
from style_bert_vits2.logging import logger
from style_bert_vits2.utils.stdout_wrapper import SAFE_STDOUT


def is_audio_file(file: Path) -> bool:
    supported_extensions = [".wav", ".flac", ".mp3", ".ogg", ".opus", ".m4a"]
    return file.suffix.lower() in supported_extensions


def get_stamps(

    vad_model: Any,

    utils: Any,

    audio_file: Path,

    min_silence_dur_ms: int = 700,

    min_sec: float = 2,

    max_sec: float = 12,

):
    """

    min_silence_dur_ms: int (ミリ秒):

        このミリ秒数以上を無音だと判断する。

        逆に、この秒数以下の無音区間では区切られない。

        小さくすると、音声がぶつ切りに小さくなりすぎ、

        大きくすると音声一つ一つが長くなりすぎる。

        データセットによってたぶん要調整。

    min_sec: float (秒):

        この秒数より小さい発話は無視する。

    max_sec: float (秒):

        この秒数より大きい発話は無視する。

    """

    (get_speech_timestamps, _, read_audio, *_) = utils
    sampling_rate = 16000  # 16kHzか8kHzのみ対応

    min_ms = int(min_sec * 1000)

    wav = read_audio(str(audio_file), sampling_rate=sampling_rate)
    speech_timestamps = get_speech_timestamps(
        wav,
        vad_model,
        sampling_rate=sampling_rate,
        min_silence_duration_ms=min_silence_dur_ms,
        min_speech_duration_ms=min_ms,
        max_speech_duration_s=max_sec,
    )

    return speech_timestamps


def split_wav(

    vad_model: Any,

    utils: Any,

    audio_file: Path,

    target_dir: Path,

    min_sec: float = 2,

    max_sec: float = 12,

    min_silence_dur_ms: int = 700,

    time_suffix: bool = False,

) -> tuple[float, int]:
    margin: int = 200  # ミリ秒単位で、音声の前後に余裕を持たせる
    speech_timestamps = get_stamps(
        vad_model=vad_model,
        utils=utils,
        audio_file=audio_file,
        min_silence_dur_ms=min_silence_dur_ms,
        min_sec=min_sec,
        max_sec=max_sec,
    )

    data, sr = sf.read(audio_file)

    total_ms = len(data) / sr * 1000

    file_name = audio_file.stem
    target_dir.mkdir(parents=True, exist_ok=True)

    total_time_ms: float = 0
    count = 0

    # タイムスタンプに従って分割し、ファイルに保存
    for i, ts in enumerate(speech_timestamps):
        start_ms = max(ts["start"] / 16 - margin, 0)
        end_ms = min(ts["end"] / 16 + margin, total_ms)

        start_sample = int(start_ms / 1000 * sr)
        end_sample = int(end_ms / 1000 * sr)
        segment = data[start_sample:end_sample]

        if time_suffix:
            file = f"{file_name}-{int(start_ms)}-{int(end_ms)}.wav"
        else:
            file = f"{file_name}-{i}.wav"
        sf.write(str(target_dir / file), segment, sr)
        total_time_ms += end_ms - start_ms
        count += 1

    return total_time_ms / 1000, count


if __name__ == "__main__":
    parser = argparse.ArgumentParser()
    parser.add_argument(
        "--min_sec", "-m", type=float, default=2, help="Minimum seconds of a slice"
    )
    parser.add_argument(
        "--max_sec", "-M", type=float, default=12, help="Maximum seconds of a slice"
    )
    parser.add_argument(
        "--input_dir",
        "-i",
        type=str,
        default="inputs",
        help="Directory of input wav files",
    )
    parser.add_argument(
        "--model_name",
        type=str,
        required=True,
        help="The result will be in Data/{model_name}/raw/ (if Data is dataset_root in configs/paths.yml)",
    )
    parser.add_argument(
        "--min_silence_dur_ms",
        "-s",
        type=int,
        default=700,
        help="Silence above this duration (ms) is considered as a split point.",
    )
    parser.add_argument(
        "--time_suffix",
        "-t",
        action="store_true",
        help="Make the filename end with -start_ms-end_ms when saving wav.",
    )
    parser.add_argument(
        "--num_processes",
        type=int,
        default=3,
        help="Number of processes to use. Default 3 seems to be the best.",
    )
    args = parser.parse_args()

    path_config = get_path_config()
    dataset_root = path_config.dataset_root

    model_name = str(args.model_name)
    input_dir = Path(args.input_dir)
    output_dir = dataset_root / model_name / "raw"
    min_sec: float = args.min_sec
    max_sec: float = args.max_sec
    min_silence_dur_ms: int = args.min_silence_dur_ms
    time_suffix: bool = args.time_suffix
    num_processes: int = args.num_processes

    audio_files = [file for file in input_dir.rglob("*") if is_audio_file(file)]

    logger.info(f"Found {len(audio_files)} audio files.")
    if output_dir.exists():
        logger.warning(f"Output directory {output_dir} already exists, deleting...")
        shutil.rmtree(output_dir)

    # モデルをダウンロードしておく
    _ = torch.hub.load(
        repo_or_dir="litagin02/silero-vad",
        model="silero_vad",
        onnx=True,
        trust_repo=True,
    )

    # Silero VADのモデルは、同じインスタンスで並列処理するとおかしくなるらしい
    # ワーカーごとにモデルをロードするようにするため、Queueを使って処理する
    def process_queue(

        q: Queue[Optional[Path]],

        result_queue: Queue[tuple[float, int]],

        error_queue: Queue[tuple[Path, Exception]],

    ):
        # logger.debug("Worker started.")
        vad_model, utils = torch.hub.load(
            repo_or_dir="litagin02/silero-vad",
            model="silero_vad",
            onnx=True,
            trust_repo=True,
        )
        while True:
            file = q.get()
            if file is None:  # 終了シグナルを確認
                q.task_done()
                break
            try:
                rel_path = file.relative_to(input_dir)
                time_sec, count = split_wav(
                    vad_model=vad_model,
                    utils=utils,
                    audio_file=file,
                    target_dir=output_dir / rel_path.parent,
                    min_sec=min_sec,
                    max_sec=max_sec,
                    min_silence_dur_ms=min_silence_dur_ms,
                    time_suffix=time_suffix,
                )
                result_queue.put((time_sec, count))
            except Exception as e:
                logger.error(f"Error processing {file}: {e}")
                error_queue.put((file, e))
                result_queue.put((0, 0))
            finally:
                q.task_done()

    q: Queue[Optional[Path]] = Queue()
    result_queue: Queue[tuple[float, int]] = Queue()
    error_queue: Queue[tuple[Path, Exception]] = Queue()

    # ファイル数が少ない場合は、ワーカー数をファイル数に合わせる
    num_processes = min(num_processes, len(audio_files))

    threads = [
        Thread(target=process_queue, args=(q, result_queue, error_queue))
        for _ in range(num_processes)
    ]
    for t in threads:
        t.start()

    pbar = tqdm(total=len(audio_files), file=SAFE_STDOUT)
    for file in audio_files:
        q.put(file)

    # result_queueを監視し、要素が追加されるごとに結果を加算しプログレスバーを更新
    total_sec = 0
    total_count = 0
    for _ in range(len(audio_files)):
        time, count = result_queue.get()
        total_sec += time
        total_count += count
        pbar.update(1)

    # 全ての処理が終わるまで待つ
    q.join()

    # 終了シグナル None を送る
    for _ in range(num_processes):
        q.put(None)

    for t in threads:
        t.join()

    pbar.close()

    if not error_queue.empty():
        error_str = "Error slicing some files:"
        while not error_queue.empty():
            file, e = error_queue.get()
            error_str += f"\n{file}: {e}"
        raise RuntimeError(error_str)

    logger.info(
        f"Slice done! Total time: {total_sec / 60:.2f} min, {total_count} files."
    )