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import os |
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import sys |
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import time |
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import torch |
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import logging |
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import librosa |
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import argparse |
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import logging.handlers |
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import numpy as np |
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import soundfile as sf |
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import multiprocessing |
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import noisereduce as nr |
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from tqdm import tqdm |
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from scipy import signal |
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from scipy.io import wavfile |
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from distutils.util import strtobool |
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from concurrent.futures import ProcessPoolExecutor, as_completed |
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now_directory = os.getcwd() |
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sys.path.append(now_directory) |
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from main.configs.config import Config |
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logger = logging.getLogger(__name__) |
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logging.getLogger("numba.core.byteflow").setLevel(logging.ERROR) |
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logging.getLogger("numba.core.ssa").setLevel(logging.ERROR) |
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logging.getLogger("numba.core.interpreter").setLevel(logging.ERROR) |
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OVERLAP = 0.3 |
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MAX_AMPLITUDE = 0.9 |
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ALPHA = 0.75 |
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HIGH_PASS_CUTOFF = 48 |
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SAMPLE_RATE_16K = 16000 |
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config = Config() |
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per = 3.0 if config.is_half else 3.7 |
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translations = config.translations |
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def parse_arguments() -> tuple: |
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parser = argparse.ArgumentParser() |
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parser.add_argument("--model_name", type=str, required=True) |
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parser.add_argument("--dataset_path", type=str, default="./dataset") |
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parser.add_argument("--sample_rate", type=int, required=True) |
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parser.add_argument("--cpu_cores", type=int, default=2) |
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parser.add_argument("--cut_preprocess", type=lambda x: bool(strtobool(x)), default=True) |
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parser.add_argument("--process_effects", type=lambda x: bool(strtobool(x)), default=False) |
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parser.add_argument("--clean_dataset", type=lambda x: bool(strtobool(x)), default=False) |
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parser.add_argument("--clean_strength", type=float, default=0.7) |
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args = parser.parse_args() |
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return args |
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def load_audio(file, sample_rate): |
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try: |
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file = file.strip(" ").strip('"').strip("\n").strip('"').strip(" ") |
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audio, sr = sf.read(file) |
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if len(audio.shape) > 1: audio = librosa.to_mono(audio.T) |
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if sr != sample_rate: audio = librosa.resample(audio, orig_sr=sr, target_sr=sample_rate) |
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except Exception as e: |
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raise RuntimeError(f"{translations['errors_loading_audio']}: {e}") |
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return audio.flatten() |
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class Slicer: |
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def __init__(self, sr, threshold = -40.0, min_length = 5000, min_interval = 300, hop_size = 20, max_sil_kept = 5000): |
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if not min_length >= min_interval >= hop_size: raise ValueError(translations["min_length>=min_interval>=hop_size"]) |
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if not max_sil_kept >= hop_size: raise ValueError(translations["max_sil_kept>=hop_size"]) |
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min_interval = sr * min_interval / 1000 |
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self.threshold = 10 ** (threshold / 20.0) |
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self.hop_size = round(sr * hop_size / 1000) |
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self.win_size = min(round(min_interval), 4 * self.hop_size) |
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self.min_length = round(sr * min_length / 1000 / self.hop_size) |
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self.min_interval = round(min_interval / self.hop_size) |
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self.max_sil_kept = round(sr * max_sil_kept / 1000 / self.hop_size) |
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def _apply_slice(self, waveform, begin, end): |
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start_idx = begin * self.hop_size |
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if len(waveform.shape) > 1: |
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end_idx = min(waveform.shape[1], end * self.hop_size) |
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return waveform[:, start_idx:end_idx] |
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else: |
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end_idx = min(waveform.shape[0], end * self.hop_size) |
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return waveform[start_idx:end_idx] |
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def slice(self, waveform): |
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samples = waveform.mean(axis=0) if len(waveform.shape) > 1 else waveform |
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if samples.shape[0] <= self.min_length: return [waveform] |
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rms_list = get_rms(y=samples, frame_length=self.win_size, hop_length=self.hop_size).squeeze(0) |
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sil_tags = [] |
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silence_start, clip_start = None, 0 |
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for i, rms in enumerate(rms_list): |
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if rms < self.threshold: |
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if silence_start is None: silence_start = i |
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continue |
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if silence_start is None: continue |
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is_leading_silence = silence_start == 0 and i > self.max_sil_kept |
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need_slice_middle = (i - silence_start >= self.min_interval and i - clip_start >= self.min_length) |
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if not is_leading_silence and not need_slice_middle: |
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silence_start = None |
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continue |
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if i - silence_start <= self.max_sil_kept: |
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pos = rms_list[silence_start : i + 1].argmin() + silence_start |
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if silence_start == 0: sil_tags.append((0, pos)) |
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else: sil_tags.append((pos, pos)) |
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clip_start = pos |
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elif i - silence_start <= self.max_sil_kept * 2: |
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pos = rms_list[i - self.max_sil_kept : silence_start + self.max_sil_kept + 1].argmin() |
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pos += i - self.max_sil_kept |
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pos_l = (rms_list[silence_start : silence_start + self.max_sil_kept + 1].argmin() + silence_start) |
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pos_r = (rms_list[i - self.max_sil_kept : i + 1].argmin() + i - self.max_sil_kept) |
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if silence_start == 0: |
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sil_tags.append((0, pos_r)) |
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clip_start = pos_r |
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else: |
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sil_tags.append((min(pos_l, pos), max(pos_r, pos))) |
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clip_start = max(pos_r, pos) |
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else: |
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pos_l = (rms_list[silence_start : silence_start + self.max_sil_kept + 1].argmin() + silence_start) |
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pos_r = (rms_list[i - self.max_sil_kept : i + 1].argmin() + i - self.max_sil_kept) |
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if silence_start == 0: sil_tags.append((0, pos_r)) |
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else: sil_tags.append((pos_l, pos_r)) |
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clip_start = pos_r |
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silence_start = None |
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total_frames = rms_list.shape[0] |
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if (silence_start is not None and total_frames - silence_start >= self.min_interval): |
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silence_end = min(total_frames, silence_start + self.max_sil_kept) |
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pos = rms_list[silence_start : silence_end + 1].argmin() + silence_start |
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sil_tags.append((pos, total_frames + 1)) |
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if not sil_tags: return [waveform] |
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else: |
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chunks = [] |
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if sil_tags[0][0] > 0: chunks.append(self._apply_slice(waveform, 0, sil_tags[0][0])) |
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for i in range(len(sil_tags) - 1): |
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chunks.append(self._apply_slice(waveform, sil_tags[i][1], sil_tags[i + 1][0])) |
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if sil_tags[-1][1] < total_frames: chunks.append(self._apply_slice(waveform, sil_tags[-1][1], total_frames)) |
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return chunks |
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def get_rms(y, frame_length=2048, hop_length=512, pad_mode="constant"): |
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padding = (int(frame_length // 2), int(frame_length // 2)) |
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y = np.pad(y, padding, mode=pad_mode) |
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axis = -1 |
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out_strides = y.strides + tuple([y.strides[axis]]) |
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x_shape_trimmed = list(y.shape) |
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x_shape_trimmed[axis] -= frame_length - 1 |
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out_shape = tuple(x_shape_trimmed) + tuple([frame_length]) |
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xw = np.lib.stride_tricks.as_strided(y, shape=out_shape, strides=out_strides) |
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target_axis = axis - 1 if axis < 0 else axis + 1 |
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xw = np.moveaxis(xw, -1, target_axis) |
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slices = [slice(None)] * xw.ndim |
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slices[axis] = slice(0, None, hop_length) |
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x = xw[tuple(slices)] |
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power = np.mean(np.abs(x) ** 2, axis=-2, keepdims=True) |
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return np.sqrt(power) |
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class PreProcess: |
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def __init__(self, sr, exp_dir, per): |
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self.slicer = Slicer(sr=sr, threshold=-42, min_length=1500, min_interval=400, hop_size=15, max_sil_kept=500) |
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self.sr = sr |
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self.b_high, self.a_high = signal.butter(N=5, Wn=HIGH_PASS_CUTOFF, btype="high", fs=self.sr) |
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self.per = per |
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self.exp_dir = exp_dir |
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self.device = "cpu" |
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self.gt_wavs_dir = os.path.join(exp_dir, "sliced_audios") |
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self.wavs16k_dir = os.path.join(exp_dir, "sliced_audios_16k") |
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os.makedirs(self.gt_wavs_dir, exist_ok=True) |
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os.makedirs(self.wavs16k_dir, exist_ok=True) |
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def _normalize_audio(self, audio: torch.Tensor): |
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tmp_max = torch.abs(audio).max() |
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if tmp_max > 2.5: return None |
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return (audio / tmp_max * (MAX_AMPLITUDE * ALPHA)) + (1 - ALPHA) * audio |
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def process_audio_segment(self, normalized_audio: np.ndarray, sid, idx0, idx1): |
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if normalized_audio is None: |
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logs(f"{sid}-{idx0}-{idx1}-filtered") |
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return |
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wavfile.write(os.path.join(self.gt_wavs_dir, f"{sid}_{idx0}_{idx1}.wav"), self.sr, normalized_audio.astype(np.float32)) |
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audio_16k = librosa.resample(normalized_audio, orig_sr=self.sr, target_sr=SAMPLE_RATE_16K) |
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wavfile.write(os.path.join(self.wavs16k_dir, f"{sid}_{idx0}_{idx1}.wav"), SAMPLE_RATE_16K, audio_16k.astype(np.float32)) |
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def process_audio(self, path, idx0, sid, cut_preprocess, process_effects, clean_dataset, clean_strength): |
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try: |
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audio = load_audio(path, self.sr) |
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if process_effects: |
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audio = signal.lfilter(self.b_high, self.a_high, audio) |
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audio = self._normalize_audio(audio) |
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if clean_dataset: audio = nr.reduce_noise(y=audio, sr=self.sr, prop_decrease=clean_strength) |
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idx1 = 0 |
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if cut_preprocess: |
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for audio_segment in self.slicer.slice(audio): |
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i = 0 |
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while 1: |
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start = int(self.sr * (self.per - OVERLAP) * i) |
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i += 1 |
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if len(audio_segment[start:]) > (self.per + OVERLAP) * self.sr: |
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tmp_audio = audio_segment[start : start + int(self.per * self.sr)] |
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self.process_audio_segment(tmp_audio, sid, idx0, idx1) |
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idx1 += 1 |
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else: |
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tmp_audio = audio_segment[start:] |
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self.process_audio_segment(tmp_audio, sid, idx0, idx1) |
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idx1 += 1 |
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break |
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else: self.process_audio_segment(audio, sid, idx0, idx1) |
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except Exception as e: |
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raise RuntimeError(f"{translations['process_audio_error']}: {e}") |
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def process_file(args): |
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pp, file, cut_preprocess, process_effects, clean_dataset, clean_strength = (args) |
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file_path, idx0, sid = file |
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pp.process_audio(file_path, idx0, sid, cut_preprocess, process_effects, clean_dataset, clean_strength) |
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def preprocess_training_set(input_root, sr, num_processes, exp_dir, per, cut_preprocess, process_effects, clean_dataset, clean_strength): |
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start_time = time.time() |
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pp = PreProcess(sr, exp_dir, per) |
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logger.info(translations["start_preprocess"].format(num_processes=num_processes)) |
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files = [] |
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idx = 0 |
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for root, _, filenames in os.walk(input_root): |
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try: |
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sid = 0 if root == input_root else int(os.path.basename(root)) |
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for f in filenames: |
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if f.lower().endswith((".wav", ".mp3", ".flac", ".ogg")): |
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files.append((os.path.join(root, f), idx, sid)) |
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idx += 1 |
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except ValueError: |
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raise ValueError(f"{translations['not_integer']} '{os.path.basename(root)}'.") |
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with tqdm(total=len(files), desc=translations["preprocess"]) as pbar: |
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with ProcessPoolExecutor(max_workers=num_processes) as executor: |
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futures = [ |
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executor.submit( |
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process_file, |
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( |
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pp, |
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file, |
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cut_preprocess, |
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process_effects, |
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clean_dataset, |
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clean_strength, |
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), |
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) |
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for file in files |
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] |
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for future in as_completed(futures): |
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try: |
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future.result() |
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except Exception as e: |
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raise RuntimeError(f"{translations['process_error']}: {e}") |
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pbar.update(1) |
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elapsed_time = time.time() - start_time |
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logger.info(translations["preprocess_success"].format(elapsed_time=f"{elapsed_time:.2f}")) |
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if __name__ == "__main__": |
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args = parse_arguments() |
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experiment_directory = os.path.join("assets", "logs", args.model_name) |
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num_processes = args.cpu_cores |
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num_processes = multiprocessing.cpu_count() if num_processes is None else int(num_processes) |
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dataset = args.dataset_path |
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sample_rate = args.sample_rate |
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cut_preprocess = args.cut_preprocess |
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preprocess_effects = args.process_effects |
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clean_dataset = args.clean_dataset |
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clean_strength = args.clean_strength |
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os.makedirs(experiment_directory, exist_ok=True) |
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if len([f for f in os.listdir(os.path.join(dataset)) if os.path.isfile(os.path.join(dataset, f)) and f.lower().endswith((".wav", ".mp3", ".flac", ".ogg"))]) < 1: raise FileNotFoundError("Không tìm thấy dữ liệu") |
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log_file = os.path.join(experiment_directory, "preprocess.log") |
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if logger.hasHandlers(): logger.handlers.clear() |
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else: |
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console_handler = logging.StreamHandler() |
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console_formatter = logging.Formatter(fmt="\n%(asctime)s.%(msecs)03d | %(levelname)s | %(module)s | %(message)s", datefmt="%Y-%m-%d %H:%M:%S") |
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console_handler.setFormatter(console_formatter) |
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console_handler.setLevel(logging.INFO) |
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file_handler = logging.handlers.RotatingFileHandler(log_file, maxBytes=5*1024*1024, backupCount=3, encoding='utf-8') |
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file_formatter = logging.Formatter(fmt="\n%(asctime)s.%(msecs)03d | %(levelname)s | %(module)s | %(message)s", datefmt="%Y-%m-%d %H:%M:%S") |
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file_handler.setFormatter(file_formatter) |
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file_handler.setLevel(logging.DEBUG) |
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logger.addHandler(console_handler) |
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logger.addHandler(file_handler) |
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logger.setLevel(logging.DEBUG) |
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logger.debug(f"{translations['modelname']}: {args.model_name}") |
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logger.debug(f"{translations['export_process']}: {experiment_directory}") |
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logger.debug(f"{translations['dataset_folder']}: {dataset}") |
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logger.debug(f"{translations['pretrain_sr']}: {sample_rate}") |
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logger.debug(f"{translations['cpu_core']}: {num_processes}") |
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logger.debug(f"{translations['split_audio']}: {cut_preprocess}") |
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logger.debug(f"{translations['preprocess_effect']}: {preprocess_effects}") |
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logger.debug(f"{translations['clear_audio']}: {clean_dataset}") |
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if clean_dataset: logger.debug(f"{translations['clean_strength']}: {clean_strength}") |
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try: |
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preprocess_training_set(dataset, sample_rate, num_processes, experiment_directory, per, cut_preprocess, preprocess_effects, clean_dataset, clean_strength) |
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except Exception as e: |
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logger.error(f"{translations['process_audio_error']} {e}") |
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logger.info(f"{translations['preprocess_model_success']} {args.model_name}") |