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| import os | |
| import torch | |
| import numpy as np | |
| import torchaudio | |
| import yaml | |
| from . import asteroid_test | |
| from huggingface_hub import hf_hub_download | |
| import logging | |
| logger = logging.getLogger(__name__) | |
| torchaudio.set_audio_backend("sox_io") | |
| def get_conf(): | |
| conf_filterbank = { | |
| 'n_filters': 64, | |
| 'kernel_size': 16, | |
| 'stride': 8 | |
| } | |
| conf_masknet = { | |
| 'in_chan': 64, | |
| 'n_src': 2, | |
| 'out_chan': 64, | |
| 'ff_hid': 256, | |
| 'ff_activation': "relu", | |
| 'norm_type': "gLN", | |
| 'chunk_size': 100, | |
| 'hop_size': 50, | |
| 'n_repeats': 2, | |
| 'mask_act': 'sigmoid', | |
| 'bidirectional': True, | |
| 'dropout': 0 | |
| } | |
| return conf_filterbank, conf_masknet | |
| def load_dpt_model(): | |
| print('Load Separation Model...') | |
| # 👇 從環境變數取得 HF Token | |
| from huggingface_hub import hf_hub_download | |
| speech_sep_token = os.getenv("SpeechSeparation") | |
| if not speech_sep_token: | |
| raise EnvironmentError("環境變數 SpeechSeparation 未設定!") | |
| # 👇 從 Hugging Face Hub 下載模型權重 | |
| model_path = hf_hub_download( | |
| repo_id="DeepLearning101/speech-separation", # 替換成你自己的 repo 名稱 | |
| filename="train_dptnet_aishell_partOverlap_B2_300epoch_quan-int8.p", | |
| token=speech_sep_token | |
| ) | |
| conf_filterbank, conf_masknet = get_conf() | |
| model_class = getattr(asteroid_test, "DPTNet") | |
| model = model_class(**conf_filterbank, **conf_masknet) | |
| model = torch.quantization.quantize_dynamic(model, {torch.nn.LSTM, torch.nn.Linear}, dtype=torch.qint8) | |
| try: | |
| state_dict = torch.load(model_path, map_location="cpu", weights_only=False) | |
| except pickle.UnpicklingError as e: | |
| raise RuntimeError( | |
| "模型載入失敗!請確認:\n" | |
| "1. 模型來源是否可信\n" | |
| "2. 是否為舊版 PyTorch 儲存的模型\n" | |
| "3. 嘗試鎖定 PyTorch 版本為 2.5.x" | |
| ) from e | |
| model.load_state_dict(state_dict) | |
| model.eval() | |
| return model | |
| import torchaudio | |
| import tempfile | |
| def dpt_sep_process(wav_path, model=None, outfilename=None): | |
| try: | |
| # 添加設備檢測 | |
| device = torch.device("cuda" if torch.cuda.is_available() else "cpu") | |
| model = model.to(device) | |
| # 強化音訊加載 | |
| x, sr = torchaudio.load(wav_path, format="wav") | |
| x = x.mean(dim=0, keepdim=True).to(device) | |
| # 自動重採樣 | |
| if sr != 16000: | |
| resampler = torchaudio.transforms.Resample(sr, 16000).to(device) | |
| x = resampler(x) | |
| sr = 16000 | |
| # 分塊處理避免OOM | |
| chunk_size = sr * 60 # 每次處理1分鐘 | |
| separated = [] | |
| for i in range(0, x.shape[1], chunk_size): | |
| chunk = x[:, i:i+chunk_size] | |
| with torch.no_grad(): | |
| est = model(chunk) | |
| separated.append(est.cpu()) | |
| est_sources = torch.cat(separated, dim=2) | |
| # 後處理修正 | |
| est_sources = est_sources.squeeze(0) | |
| sep_1, sep_2 = est_sources[0], est_sources[1] | |
| # 正規化增強 | |
| peak = 0.9 * torch.max(torch.abs(x)) | |
| sep_1 = peak * sep_1 / torch.max(torch.abs(sep_1)) | |
| sep_2 = peak * sep_2 / torch.max(torch.abs(sep_2)) | |
| # 使用臨時輸出目錄 | |
| with tempfile.TemporaryDirectory() as tmp_dir: | |
| sep1_path = os.path.join(tmp_dir, "sep1.wav") | |
| sep2_path = os.path.join(tmp_dir, "sep2.wav") | |
| torchaudio.save(sep1_path, sep_1.unsqueeze(0), sr) | |
| torchaudio.save(sep2_path, sep_2.unsqueeze(0), sr) | |
| # 移動檔案到最終位置 | |
| final_sep1 = outfilename.replace('.wav', '_sep1.wav') | |
| final_sep2 = outfilename.replace('.wav', '_sep2.wav') | |
| os.replace(sep1_path, final_sep1) | |
| os.replace(sep2_path, final_sep2) | |
| # 新增日誌 | |
| logger.info(f"💾 寫入輸出檔案至: {final_sep1}, {final_sep2}") | |
| return final_sep1, final_sep2 | |
| except RuntimeError as e: | |
| if "CUDA out of memory" in str(e): | |
| raise RuntimeError("記憶體不足,請縮短音訊長度") from e | |
| else: | |
| raise | |
| if __name__ == '__main__': | |
| print("This module should be used via Flask or Gradio.") |