import os import logging logger = logging.getLogger(__name__) import librosa import numpy as np import soundfile as sf import torch from infer.lib.uvr5_pack.lib_v5 import nets_61968KB as Nets from infer.lib.uvr5_pack.lib_v5 import spec_utils from infer.lib.uvr5_pack.lib_v5.model_param_init import ModelParameters from infer.lib.uvr5_pack.lib_v5.nets_new import CascadedNet from infer.lib.uvr5_pack.utils import inference # 导入彩色日志 try: from lib.logger import log except ImportError: log = None class AudioPre: def __init__(self, agg, model_path, device, is_half, tta=False): self.model_path = model_path self.device = device self.data = { # Processing Options "postprocess": False, "tta": tta, # Constants "window_size": 512, "agg": agg, "high_end_process": "mirroring", } if log: log.model(f"加载UVR5模型: {os.path.basename(model_path)}") log.detail(f"设备: {device}, 半精度: {is_half}") log.config(f"激进度: {agg}, 窗口大小: 512") mp = ModelParameters("infer/lib/uvr5_pack/lib_v5/modelparams/4band_v2.json") model = Nets.CascadedASPPNet(mp.param["bins"] * 2) cpk = torch.load(model_path, map_location="cpu", weights_only=False) model.load_state_dict(cpk) model.eval() if is_half: model = model.half().to(device) else: model = model.to(device) self.mp = mp self.model = model if log: log.success("UVR5模型加载完成") def _path_audio_( self, music_file, ins_root=None, vocal_root=None, format="flac", is_hp3=False ): if ins_root is None and vocal_root is None: return "No save root." name = os.path.basename(music_file) if log: log.audio(f"处理音频文件: {name}") log.detail(f"输入路径: {music_file}") log.detail(f"伴奏输出: {ins_root}") log.detail(f"人声输出: {vocal_root}") log.config(f"输出格式: {format}, HP3模式: {is_hp3}") if ins_root is not None: os.makedirs(ins_root, exist_ok=True) if vocal_root is not None: os.makedirs(vocal_root, exist_ok=True) X_wave, y_wave, X_spec_s, y_spec_s = {}, {}, {}, {} bands_n = len(self.mp.param["band"]) if log: log.detail(f"频段数量: {bands_n}") # print(bands_n) for d in range(bands_n, 0, -1): bp = self.mp.param["band"][d] if log: log.detail(f"处理频段 {d}: 采样率={bp['sr']}, 重采样类型={bp['res_type']}") if d == bands_n: # high-end band ( X_wave[d], _, ) = librosa.load( # 理论上librosa读取可能对某些音频有bug,应该上ffmpeg读取,但是太麻烦了弃坑 music_file, sr=bp["sr"], mono=False, dtype=np.float32, res_type=bp["res_type"], ) if X_wave[d].ndim == 1: X_wave[d] = np.asfortranarray([X_wave[d], X_wave[d]]) if log: log.detail(f"加载高频段: shape={X_wave[d].shape}") else: # lower bands X_wave[d] = librosa.resample( X_wave[d + 1], orig_sr=self.mp.param["band"][d + 1]["sr"], target_sr=bp["sr"], res_type=bp["res_type"], ) if log: log.detail(f"重采样频段 {d}: shape={X_wave[d].shape}") # Stft of wave source X_spec_s[d] = spec_utils.wave_to_spectrogram_mt( X_wave[d], bp["hl"], bp["n_fft"], self.mp.param["mid_side"], self.mp.param["mid_side_b2"], self.mp.param["reverse"], ) # pdb.set_trace() if d == bands_n and self.data["high_end_process"] != "none": input_high_end_h = (bp["n_fft"] // 2 - bp["crop_stop"]) + ( self.mp.param["pre_filter_stop"] - self.mp.param["pre_filter_start"] ) input_high_end = X_spec_s[d][ :, bp["n_fft"] // 2 - input_high_end_h : bp["n_fft"] // 2, : ] if log: log.progress("合并频谱图...") X_spec_m = spec_utils.combine_spectrograms(X_spec_s, self.mp) aggresive_set = float(self.data["agg"] / 100) aggressiveness = { "value": aggresive_set, "split_bin": self.mp.param["band"][1]["crop_stop"], } if log: log.detail(f"激进度设置: {aggresive_set}") log.progress("执行模型推理...") with torch.no_grad(): pred, X_mag, X_phase = inference( X_spec_m, self.device, self.model, aggressiveness, self.data ) # Postprocess if self.data["postprocess"]: if log: log.detail("执行后处理...") pred_inv = np.clip(X_mag - pred, 0, np.inf) pred = spec_utils.mask_silence(pred, pred_inv) y_spec_m = pred * X_phase v_spec_m = X_spec_m - y_spec_m if ins_root is not None: if log: log.progress("生成伴奏音频...") if self.data["high_end_process"].startswith("mirroring"): input_high_end_ = spec_utils.mirroring( self.data["high_end_process"], y_spec_m, input_high_end, self.mp ) wav_instrument = spec_utils.cmb_spectrogram_to_wave( y_spec_m, self.mp, input_high_end_h, input_high_end_ ) else: wav_instrument = spec_utils.cmb_spectrogram_to_wave(y_spec_m, self.mp) logger.info("%s instruments done" % name) if log: log.success(f"{name} 伴奏分离完成") if is_hp3 == True: head = "vocal_" else: head = "instrument_" if format in ["wav", "flac"]: output_path = os.path.join( ins_root, head + "{}_{}.{}".format(name, self.data["agg"], format), ) sf.write( output_path, (np.array(wav_instrument) * 32768).astype("int16"), self.mp.param["sr"], ) if log: log.audio(f"保存伴奏: {os.path.basename(output_path)}") else: path = os.path.join( ins_root, head + "{}_{}.wav".format(name, self.data["agg"]) ) sf.write( path, (np.array(wav_instrument) * 32768).astype("int16"), self.mp.param["sr"], ) if os.path.exists(path): opt_format_path = path[:-4] + ".%s" % format if log: log.detail(f"转换格式: {format}") os.system('ffmpeg -i "%s" -vn "%s" -q:a 2 -y' % (path, opt_format_path)) if os.path.exists(opt_format_path): try: os.remove(path) except: pass if log: log.audio(f"保存伴奏: {os.path.basename(opt_format_path)}") if vocal_root is not None: if log: log.progress("生成人声音频...") if is_hp3 == True: head = "instrument_" else: head = "vocal_" if self.data["high_end_process"].startswith("mirroring"): input_high_end_ = spec_utils.mirroring( self.data["high_end_process"], v_spec_m, input_high_end, self.mp ) wav_vocals = spec_utils.cmb_spectrogram_to_wave( v_spec_m, self.mp, input_high_end_h, input_high_end_ ) else: wav_vocals = spec_utils.cmb_spectrogram_to_wave(v_spec_m, self.mp) logger.info("%s vocals done" % name) if log: log.success(f"{name} 人声分离完成") if format in ["wav", "flac"]: output_path = os.path.join( vocal_root, head + "{}_{}.{}".format(name, self.data["agg"], format), ) sf.write( output_path, (np.array(wav_vocals) * 32768).astype("int16"), self.mp.param["sr"], ) if log: log.audio(f"保存人声: {os.path.basename(output_path)}") else: path = os.path.join( vocal_root, head + "{}_{}.wav".format(name, self.data["agg"]) ) sf.write( path, (np.array(wav_vocals) * 32768).astype("int16"), self.mp.param["sr"], ) if os.path.exists(path): opt_format_path = path[:-4] + ".%s" % format if log: log.detail(f"转换格式: {format}") os.system('ffmpeg -i "%s" -vn "%s" -q:a 2 -y' % (path, opt_format_path)) if os.path.exists(opt_format_path): try: os.remove(path) except: pass if log: log.audio(f"保存人声: {os.path.basename(opt_format_path)}") class AudioPreDeEcho: def __init__(self, agg, model_path, device, is_half, tta=False): self.model_path = model_path self.device = device self.data = { # Processing Options "postprocess": False, "tta": tta, # Constants "window_size": 512, "agg": agg, "high_end_process": "mirroring", } if log: log.model(f"加载UVR5 DeEcho模型: {os.path.basename(model_path)}") log.detail(f"设备: {device}, 半精度: {is_half}") log.config(f"激进度: {agg}, 窗口大小: 512") mp = ModelParameters("infer/lib/uvr5_pack/lib_v5/modelparams/4band_v3.json") nout = 64 if "DeReverb" in model_path else 48 if log: log.detail(f"模型输出通道: {nout}") model = CascadedNet(mp.param["bins"] * 2, nout) cpk = torch.load(model_path, map_location="cpu", weights_only=False) model.load_state_dict(cpk) model.eval() if is_half: model = model.half().to(device) else: model = model.to(device) self.mp = mp self.model = model if log: log.success("UVR5 DeEcho模型加载完成") def _path_audio_( self, music_file, vocal_root=None, ins_root=None, format="flac", is_hp3=False ): # 3个VR模型vocal和ins是反的 if ins_root is None and vocal_root is None: return "No save root." name = os.path.basename(music_file) if log: log.audio(f"DeEcho处理音频: {name}") log.detail(f"输入路径: {music_file}") if ins_root is not None: os.makedirs(ins_root, exist_ok=True) if vocal_root is not None: os.makedirs(vocal_root, exist_ok=True) X_wave, y_wave, X_spec_s, y_spec_s = {}, {}, {}, {} bands_n = len(self.mp.param["band"]) if log: log.detail(f"频段数量: {bands_n}") for d in range(bands_n, 0, -1): bp = self.mp.param["band"][d] if log: log.detail(f"处理频段 {d}: 采样率={bp['sr']}") if d == bands_n: # high-end band ( X_wave[d], _, ) = librosa.load( # 理论上librosa读取可能对某些音频有bug,应该上ffmpeg读取,但是太麻烦了弃坑 music_file, sr=bp["sr"], mono=False, dtype=np.float32, res_type=bp["res_type"], ) if X_wave[d].ndim == 1: X_wave[d] = np.asfortranarray([X_wave[d], X_wave[d]]) else: # lower bands X_wave[d] = librosa.resample( X_wave[d + 1], orig_sr=self.mp.param["band"][d + 1]["sr"], target_sr=bp["sr"], res_type=bp["res_type"], ) # Stft of wave source X_spec_s[d] = spec_utils.wave_to_spectrogram_mt( X_wave[d], bp["hl"], bp["n_fft"], self.mp.param["mid_side"], self.mp.param["mid_side_b2"], self.mp.param["reverse"], ) # pdb.set_trace() if d == bands_n and self.data["high_end_process"] != "none": input_high_end_h = (bp["n_fft"] // 2 - bp["crop_stop"]) + ( self.mp.param["pre_filter_stop"] - self.mp.param["pre_filter_start"] ) input_high_end = X_spec_s[d][ :, bp["n_fft"] // 2 - input_high_end_h : bp["n_fft"] // 2, : ] if log: log.progress("合并频谱图并执行推理...") X_spec_m = spec_utils.combine_spectrograms(X_spec_s, self.mp) aggresive_set = float(self.data["agg"] / 100) aggressiveness = { "value": aggresive_set, "split_bin": self.mp.param["band"][1]["crop_stop"], } with torch.no_grad(): pred, X_mag, X_phase = inference( X_spec_m, self.device, self.model, aggressiveness, self.data ) # Postprocess if self.data["postprocess"]: pred_inv = np.clip(X_mag - pred, 0, np.inf) pred = spec_utils.mask_silence(pred, pred_inv) y_spec_m = pred * X_phase v_spec_m = X_spec_m - y_spec_m if ins_root is not None: if log: log.progress("生成伴奏音频...") if self.data["high_end_process"].startswith("mirroring"): input_high_end_ = spec_utils.mirroring( self.data["high_end_process"], y_spec_m, input_high_end, self.mp ) wav_instrument = spec_utils.cmb_spectrogram_to_wave( y_spec_m, self.mp, input_high_end_h, input_high_end_ ) else: wav_instrument = spec_utils.cmb_spectrogram_to_wave(y_spec_m, self.mp) logger.info("%s instruments done" % name) if log: log.success(f"{name} 伴奏分离完成") if format in ["wav", "flac"]: output_path = os.path.join( ins_root, "vocal_{}_{}.{}".format(name, self.data["agg"], format), ) sf.write( output_path, (np.array(wav_instrument) * 32768).astype("int16"), self.mp.param["sr"], ) if log: log.audio(f"保存伴奏: {os.path.basename(output_path)}") else: path = os.path.join( ins_root, "vocal_{}_{}.wav".format(name, self.data["agg"]) ) sf.write( path, (np.array(wav_instrument) * 32768).astype("int16"), self.mp.param["sr"], ) if os.path.exists(path): opt_format_path = path[:-4] + ".%s" % format if log: log.detail(f"转换格式: {format}") os.system('ffmpeg -i "%s" -vn "%s" -q:a 2 -y' % (path, opt_format_path)) if os.path.exists(opt_format_path): try: os.remove(path) except: pass if vocal_root is not None: if log: log.progress("生成人声音频...") if self.data["high_end_process"].startswith("mirroring"): input_high_end_ = spec_utils.mirroring( self.data["high_end_process"], v_spec_m, input_high_end, self.mp ) wav_vocals = spec_utils.cmb_spectrogram_to_wave( v_spec_m, self.mp, input_high_end_h, input_high_end_ ) else: wav_vocals = spec_utils.cmb_spectrogram_to_wave(v_spec_m, self.mp) logger.info("%s vocals done" % name) if log: log.success(f"{name} 人声分离完成") if format in ["wav", "flac"]: output_path = os.path.join( vocal_root, "instrument_{}_{}.{}".format(name, self.data["agg"], format), ) sf.write( output_path, (np.array(wav_vocals) * 32768).astype("int16"), self.mp.param["sr"], ) if log: log.audio(f"保存人声: {os.path.basename(output_path)}") else: path = os.path.join( vocal_root, "instrument_{}_{}.wav".format(name, self.data["agg"]) ) sf.write( path, (np.array(wav_vocals) * 32768).astype("int16"), self.mp.param["sr"], ) if os.path.exists(path): opt_format_path = path[:-4] + ".%s" % format if log: log.detail(f"转换格式: {format}") os.system('ffmpeg -i "%s" -vn "%s" -q:a 2 -y' % (path, opt_format_path)) if os.path.exists(opt_format_path): try: os.remove(path) except: pass