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| import gradio as gr | |
| import os | |
| os.system('cd monotonic_align && python setup.py build_ext --inplace && cd ..') | |
| import logging | |
| numba_logger = logging.getLogger('numba') | |
| numba_logger.setLevel(logging.WARNING) | |
| import librosa | |
| import torch | |
| import commons | |
| import utils | |
| from models import SynthesizerTrn | |
| from text.symbols import symbols | |
| from text import text_to_sequence | |
| import numpy as np | |
| import soundfile as sf | |
| from preprocess_wave import FeatureInput | |
| def resize2d(x, target_len): | |
| source = np.array(x) | |
| source[source<0.001] = np.nan | |
| target = np.interp(np.arange(0, len(source)*target_len, len(source))/ target_len, np.arange(0, len(source)), source) | |
| res = np.nan_to_num(target) | |
| return res | |
| def transcribe(path, length, transform): | |
| featur_pit = featureInput.compute_f0(path) | |
| featur_pit = featur_pit * 2**(transform/12) | |
| featur_pit = resize2d(featur_pit, length) | |
| coarse_pit = featureInput.coarse_f0(featur_pit) | |
| return coarse_pit | |
| def get_text(text, hps): | |
| text_norm = text_to_sequence(text, hps.data.text_cleaners) | |
| if hps.data.add_blank: | |
| text_norm = commons.intersperse(text_norm, 0) | |
| text_norm = torch.LongTensor(text_norm) | |
| print(text_norm.shape) | |
| return text_norm | |
| convert_cnt = [0] | |
| hps_ms = utils.get_hparams_from_file("configs/nyarumul.json") | |
| net_g_ms = SynthesizerTrn( | |
| len(symbols), | |
| hps_ms.data.filter_length // 2 + 1, | |
| hps_ms.train.segment_size // hps_ms.data.hop_length, | |
| n_speakers=hps_ms.data.n_speakers, | |
| **hps_ms.model) | |
| featureInput = FeatureInput(hps_ms.data.sampling_rate, hps_ms.data.hop_length) | |
| hubert = torch.hub.load("bshall/hubert:main", "hubert_soft") | |
| _ = utils.load_checkpoint("nyarumodel.pth", net_g_ms, None) | |
| def vc_fn(sid,random1, input_audio,vc_transform): | |
| if input_audio is None: | |
| return "You need to upload an audio", None | |
| sampling_rate, audio = input_audio | |
| # print(audio.shape,sampling_rate) | |
| duration = audio.shape[0] / sampling_rate | |
| if duration > 45: | |
| return "请上传小于45s的音频，需要转换长音频请使用colab", None | |
| audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32) | |
| if len(audio.shape) > 1: | |
| audio = librosa.to_mono(audio.transpose(1, 0)) | |
| if sampling_rate != 16000: | |
| audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000) | |
| source = torch.FloatTensor(audio).unsqueeze(0).unsqueeze(0) | |
| print(source.shape) | |
| with torch.inference_mode(): | |
| units = hubert.units(source) | |
| soft = units.squeeze(0).numpy() | |
| audio22050 = librosa.resample(audio, orig_sr=16000, target_sr=22050) | |
| sf.write("temp.wav", audio22050, 22050) | |
| pitch = transcribe("temp.wav", soft.shape[0], vc_transform) | |
| pitch = torch.LongTensor(pitch).unsqueeze(0) | |
| sid = torch.LongTensor([0]) if sid == "猫雷" else torch.LongTensor([1]) | |
| stn_tst = torch.FloatTensor(soft) | |
| with torch.no_grad(): | |
| x_tst = stn_tst.unsqueeze(0) | |
| x_tst_lengths = torch.LongTensor([stn_tst.size(0)]) | |
| audio = net_g_ms.infer(x_tst, x_tst_lengths, pitch=pitch,sid=sid, noise_scale=float(random1), | |
| noise_scale_w=0.1, length_scale=1)[0][0, 0].data.float().numpy() | |
| convert_cnt[0] += 1 | |
| print(convert_cnt[0]) | |
| return "Success", (hps_ms.data.sampling_rate, audio) | |
| app = gr.Blocks() | |
| with app: | |
| with gr.Tabs(): | |
| with gr.TabItem("Basic"): | |
| gr.Markdown(value="""**new!!!!!** 更新了一个训练更多epoch的版本:链接[在这里](https://huggingface.co/spaces/innnky/nyaru-svc2.0-advanced) | |
| ，增加了3个音色，增加了f0音高曲线等方便查看转换后是否跑调 | |
| old: 本模型相比与前一个模型，音质和音准方面有一定的提升，但是低音音域目前存在较大问题。 | |
| 目前猫雷模型能够唱的最低音为#G3(207hz) 低于该音会当场爆炸（之前的模型只是会跑调）， | |
| 因此请不要让这个模型唱男声的音高，请使用变调功能将音域移动至207hz以上。 | |
| 该模型的 [github仓库链接](https://github.com/innnky/so-vits-svc) | |
| 如果想自己制作并训练模型可以访问这个 [github仓库](https://github.com/IceKyrin/sovits_guide) | |
| ps: 更新了一下模型，现在和视频中不是一个同一个模型，b站视频中的模型在git历史中（因为之前数据集中似乎混入了一些杂项导致音色有些偏离猫雷音色） | |
| """) | |
| sid = gr.Dropdown(label="音色",choices=['猫雷'], value="猫雷") | |
| vc_input3 = gr.Audio(label="上传音频（长度小于45秒）") | |
| vc_transform = gr.Number(label="变调（整数，可以正负，半音数量，升高八度就是12）",value=0) | |
| random1 = gr.Number(label="随机化程度，似乎会影响音质，建议保持默认",value=0.4) | |
| vc_submit = gr.Button("转换", variant="primary") | |
| vc_output1 = gr.Textbox(label="Output Message") | |
| vc_output2 = gr.Audio(label="Output Audio") | |
| vc_submit.click(vc_fn, [sid,random1, vc_input3, vc_transform], [vc_output1, vc_output2]) | |
| app.launch() |