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Update app.py
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app.py
CHANGED
@@ -9,6 +9,17 @@ from models import SynthesizerTrn
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from text.symbols import symbols
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from text import text_to_sequence
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def get_text(text, hps):
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text_norm = text_to_sequence(text, hps.data.text_cleaners)
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if hps.data.add_blank:
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@@ -16,7 +27,6 @@ def get_text(text, hps):
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text_norm = torch.LongTensor(text_norm)
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return text_norm
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-
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hps = utils.get_hparams_from_file("configs/biaobei_base.json")
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net_g = SynthesizerTrn(
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@@ -26,7 +36,6 @@ net_g = SynthesizerTrn(
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**hps.model)
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_ = net_g.eval()
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# _ = utils.load_checkpoint("logs/woman_csmsc/G_100000.pth", net_g, None)
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_ = utils.load_checkpoint("G_aatrox.pth", net_g, None)
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def vc_fn(input):
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@@ -34,10 +43,6 @@ def vc_fn(input):
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with torch.no_grad():
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x_tst = stn_tst.unsqueeze(0)
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x_tst_lengths = torch.LongTensor([stn_tst.size(0)])
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# x_tst = stn_tst.cpu().unsqueeze(0)
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# x_tst_lengths = torch.LongTensor([stn_tst.size(0)]).cpu()
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audio = net_g.infer(x_tst, x_tst_lengths, noise_scale=.667, noise_scale_w=0.8, length_scale=1)[0][0,0].data.cpu().float().numpy()
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sampling_rate = 22050
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return (sampling_rate, audio)
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from text.symbols import symbols
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from text import text_to_sequence
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%matplotlib inline
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import matplotlib.pyplot as plt
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import json
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import math
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from torch import nn
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from torch.nn import functional as F
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from torch.utils.data import DataLoader
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from data_utils import TextAudioLoader, TextAudioCollate, TextAudioSpeakerLoader, TextAudioSpeakerCollate
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from scipy.io.wavfile import write
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def get_text(text, hps):
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text_norm = text_to_sequence(text, hps.data.text_cleaners)
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if hps.data.add_blank:
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text_norm = torch.LongTensor(text_norm)
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return text_norm
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hps = utils.get_hparams_from_file("configs/biaobei_base.json")
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net_g = SynthesizerTrn(
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**hps.model)
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_ = net_g.eval()
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_ = utils.load_checkpoint("G_aatrox.pth", net_g, None)
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def vc_fn(input):
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with torch.no_grad():
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x_tst = stn_tst.unsqueeze(0)
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x_tst_lengths = torch.LongTensor([stn_tst.size(0)])
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audio = net_g.infer(x_tst, x_tst_lengths, noise_scale=.667, noise_scale_w=0.8, length_scale=1)[0][0,0].data.cpu().float().numpy()
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sampling_rate = 22050
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return (sampling_rate, audio)
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