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import gradio as gr | |
import torch,pdb | |
import numpy as np | |
import soundfile as sf | |
from models import SynthesizerTrn256 | |
from scipy.io import wavfile | |
from fairseq import checkpoint_utils | |
import pyworld,librosa | |
import torch.nn.functional as F | |
device = torch.device("cuda" if torch.cuda.is_available() else "cpu") | |
model_path = "checkpoint_best_legacy_500.pt"#checkpoint_best_legacy_500.pt | |
print("load model(s) from {}".format(model_path)) | |
models, saved_cfg, task = checkpoint_utils.load_model_ensemble_and_task( | |
[model_path], | |
suffix="", | |
) | |
model = models[0] | |
model = model.to(device) | |
model.eval() | |
net_g = SynthesizerTrn256(513,40,192,192,768,2,6,3,0.1,"1", [3,7,11],[[1,3,5], [1,3,5], [1,3,5]],[10,4,2,2,2],512,[16,16,4,4,4],0) | |
weights=torch.load("qihai.pt", map_location=torch.device('cpu')) | |
net_g.load_state_dict(weights,strict=True) | |
net_g.eval().to(device) | |
def get_f0(x, f0_up_key=0): | |
f0_max = 1100.0 | |
f0_min = 50.0 | |
f0_mel_min = 1127 * np.log(1 + f0_min / 700) | |
f0_mel_max = 1127 * np.log(1 + f0_max / 700) | |
f0, t = pyworld.dio( | |
x.astype(np.double), | |
fs=16000, | |
f0_ceil=800, | |
frame_period=10, | |
) | |
f0 = pyworld.stonemask(x.astype(np.double), f0, t, 16000) | |
f0 *= pow(2, f0_up_key / 12) | |
f0_mel = 1127 * np.log(1 + f0 / 700) | |
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * 254 / (f0_mel_max - f0_mel_min) + 1 | |
f0_mel[f0_mel <= 1] = 1 | |
f0_mel[f0_mel > 255] = 255 | |
f0_coarse = np.rint(f0_mel).astype(np.int) | |
return f0_coarse | |
def vc_fn( input_audio,f0_up_key): | |
if input_audio is None: | |
return "You need to upload an audio", None | |
sampling_rate, audio = input_audio | |
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) | |
pitch = get_f0(audio, f0_up_key) | |
feats = torch.from_numpy(audio).float() | |
if feats.dim() == 2: # double channels | |
feats = feats.mean(-1) | |
assert feats.dim() == 1, feats.dim() | |
feats = feats.view(1, -1) | |
padding_mask = torch.BoolTensor(feats.shape).fill_(False) | |
inputs = { | |
"source": feats.to(device), | |
"padding_mask": padding_mask.to(device), | |
"output_layer": 9, # layer 9 | |
} | |
with torch.no_grad(): | |
logits = model.extract_features(**inputs) | |
feats = model.final_proj(logits[0]) | |
feats = F.interpolate(feats.permute(0, 2, 1), scale_factor=2).permute(0, 2, 1) | |
p_len = min(feats.shape[1], 10000, pitch.shape[0]) # 太大了爆显存 | |
feats = feats[:, :p_len, :] | |
pitch = pitch[:p_len] | |
p_len = torch.LongTensor([p_len]).to(device) | |
pitch = torch.LongTensor(pitch).unsqueeze(0).to(device) | |
with torch.no_grad(): | |
audio = net_g.infer(feats, p_len, pitch)[0][0, 0].data.cpu().float().numpy() | |
return "Success", (32000, audio) | |
app = gr.Blocks() | |
with app: | |
with gr.Tabs(): | |
with gr.TabItem("Basic"): | |
gr.Markdown(value="""""") | |
vc_input3 = gr.Audio(label="上传音频(长度小于45秒)") | |
f0_up_key = gr.Number(label="变调") | |
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, [ vc_input3, f0_up_key], [vc_output1, vc_output2]) | |
app.launch() |