import io import logging import time from pathlib import Path import librosa import matplotlib.pyplot as plt import numpy as np import soundfile from inference import infer_tool from inference import slicer from inference.infer_tool import Svc logging.getLogger('numba').setLevel(logging.WARNING) chunks_dict = infer_tool.read_temp("inference/chunks_temp.json") def main(): import argparse parser = argparse.ArgumentParser(description='sovits4 inference') # 一定要设置的部分 parser.add_argument('-m', '--model_path', type=str, default="logs/44k/G_0.pth", help='模型路径') parser.add_argument('-c', '--config_path', type=str, default="configs/config.json", help='配置文件路径') parser.add_argument('-cl', '--clip', type=float, default=0, help='音频强制切片，默认0为自动切片，单位为秒/s') parser.add_argument('-n', '--clean_names', type=str, nargs='+', default=["君の知らない物語-src.wav"], help='wav文件名列表，放在raw文件夹下') parser.add_argument('-t', '--trans', type=int, nargs='+', default=[0], help='音高调整，支持正负（半音）') parser.add_argument('-s', '--spk_list', type=str, nargs='+', default=['nen'], help='合成目标说话人名称') # 可选项部分 parser.add_argument('-a', '--auto_predict_f0', action='store_true', default=False,help='语音转换自动预测音高，转换歌声时不要打开这个会严重跑调') parser.add_argument('-cm', '--cluster_model_path', type=str, default="logs/44k/kmeans_10000.pt", help='聚类模型路径，如果没有训练聚类则随便填') parser.add_argument('-cr', '--cluster_infer_ratio', type=float, default=0, help='聚类方案占比，范围0-1，若没有训练聚类模型则默认0即可') parser.add_argument('-lg', '--linear_gradient', type=float, default=0, help='两段音频切片的交叉淡入长度，如果强制切片后出现人声不连贯可调整该数值，如果连贯建议采用默认值0，单位为秒') parser.add_argument('-fmp', '--f0_mean_pooling', type=bool, default=False, help='是否对F0使用均值滤波器(池化)，对部分哑音有改善。注意，启动该选项会导致推理速度下降，默认关闭') parser.add_argument('-eh', '--enhance', type=bool, default=False, help='是否使用NSF_HIFIGAN增强器,该选项对部分训练集少的模型有一定的音质增强效果，但是对训练好的模型有反面效果，默认关闭') # 不用动的部分 parser.add_argument('-sd', '--slice_db', type=int, default=-40, help='默认-40，嘈杂的音频可以-30，干声保留呼吸可以-50') parser.add_argument('-d', '--device', type=str, default=None, help='推理设备，None则为自动选择cpu和gpu') parser.add_argument('-ns', '--noice_scale', type=float, default=0.4, help='噪音级别，会影响咬字和音质，较为玄学') parser.add_argument('-p', '--pad_seconds', type=float, default=0.5, help='推理音频pad秒数，由于未知原因开头结尾会有异响，pad一小段静音段后就不会出现') parser.add_argument('-wf', '--wav_format', type=str, default='flac', help='音频输出格式') parser.add_argument('-lgr', '--linear_gradient_retain', type=float, default=0.75, help='自动音频切片后，需要舍弃每段切片的头尾。该参数设置交叉长度保留的比例，范围0-1,左开右闭') parser.add_argument('-eak', '--enhancer_adaptive_key', type=int, default=0, help='使增强器适应更高的音域(单位为半音数)|默认为0') args = parser.parse_args() clean_names = args.clean_names trans = args.trans spk_list = args.spk_list slice_db = args.slice_db wav_format = args.wav_format auto_predict_f0 = args.auto_predict_f0 cluster_infer_ratio = args.cluster_infer_ratio noice_scale = args.noice_scale pad_seconds = args.pad_seconds clip = args.clip lg = args.linear_gradient lgr = args.linear_gradient_retain F0_mean_pooling = args.f0_mean_pooling enhance = args.enhance enhancer_adaptive_key = args.enhancer_adaptive_key svc_model = Svc(args.model_path, args.config_path, args.device, args.cluster_model_path,enhance) infer_tool.mkdir(["raw", "results"]) infer_tool.fill_a_to_b(trans, clean_names) for clean_name, tran in zip(clean_names, trans): raw_audio_path = f"raw/{clean_name}" if "." not in raw_audio_path: raw_audio_path += ".wav" infer_tool.format_wav(raw_audio_path) wav_path = Path(raw_audio_path).with_suffix('.wav') chunks = slicer.cut(wav_path, db_thresh=slice_db) audio_data, audio_sr = slicer.chunks2audio(wav_path, chunks) per_size = int(clip*audio_sr) lg_size = int(lg*audio_sr) lg_size_r = int(lg_size*lgr) lg_size_c_l = (lg_size-lg_size_r)//2 lg_size_c_r = lg_size-lg_size_r-lg_size_c_l lg = np.linspace(0,1,lg_size_r) if lg_size!=0 else 0 for spk in spk_list: audio = [] for (slice_tag, data) in audio_data: print(f'#=====segment start, {round(len(data) / audio_sr, 3)}s======') length = int(np.ceil(len(data) / audio_sr * svc_model.target_sample)) if slice_tag: print('jump empty segment') _audio = np.zeros(length) audio.extend(list(infer_tool.pad_array(_audio, length))) continue if per_size != 0: datas = infer_tool.split_list_by_n(data, per_size,lg_size) else: datas = [data] for k,dat in enumerate(datas): per_length = int(np.ceil(len(dat) / audio_sr * svc_model.target_sample)) if clip!=0 else length if clip!=0: print(f'###=====segment clip start, {round(len(dat) / audio_sr, 3)}s======') # padd pad_len = int(audio_sr * pad_seconds) dat = np.concatenate([np.zeros([pad_len]), dat, np.zeros([pad_len])]) raw_path = io.BytesIO() soundfile.write(raw_path, dat, audio_sr, format="wav") raw_path.seek(0) out_audio, out_sr = svc_model.infer(spk, tran, raw_path, cluster_infer_ratio=cluster_infer_ratio, auto_predict_f0=auto_predict_f0, noice_scale=noice_scale, F0_mean_pooling = F0_mean_pooling, enhancer_adaptive_key = enhancer_adaptive_key ) _audio = out_audio.cpu().numpy() pad_len = int(svc_model.target_sample * pad_seconds) _audio = _audio[pad_len:-pad_len] _audio = infer_tool.pad_array(_audio, per_length) if lg_size!=0 and k!=0: lg1 = audio[-(lg_size_r+lg_size_c_r):-lg_size_c_r] if lgr != 1 else audio[-lg_size:] lg2 = _audio[lg_size_c_l:lg_size_c_l+lg_size_r] if lgr != 1 else _audio[0:lg_size] lg_pre = lg1*(1-lg)+lg2*lg audio = audio[0:-(lg_size_r+lg_size_c_r)] if lgr != 1 else audio[0:-lg_size] audio.extend(lg_pre) _audio = _audio[lg_size_c_l+lg_size_r:] if lgr != 1 else _audio[lg_size:] audio.extend(list(_audio)) key = "auto" if auto_predict_f0 else f"{tran}key" cluster_name = "" if cluster_infer_ratio == 0 else f"_{cluster_infer_ratio}" res_path = f'./results/{clean_name}_{key}_{spk}{cluster_name}.{wav_format}' soundfile.write(res_path, audio, svc_model.target_sample, format=wav_format) svc_model.clear_empty() if __name__ == '__main__': main()