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@@ -14,7 +14,7 @@ from models_infer import SynthesizerTrn
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from text import text_to_sequence
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import gradio as gr
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import torchaudio
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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def get_text(text, hps):
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text_norm = text_to_sequence(text, hps.symbols, hps.data.text_cleaners)
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if hps.data.add_blank:
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@@ -37,14 +37,14 @@ def create_vc_fn(model, hps, speaker_ids):
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if sampling_rate != hps.data.sampling_rate:
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audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=hps.data.sampling_rate)
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with no_grad():
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y = torch.FloatTensor(audio)
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y = torch.FloatTensor(audio).to(device)
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y = y.unsqueeze(0)
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spec = spectrogram_torch(y, hps.data.filter_length,
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hps.data.sampling_rate, hps.data.hop_length, hps.data.win_length,
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center=False)
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spec_lengths = LongTensor([spec.size(-1)])
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sid_src = LongTensor([original_speaker_id])
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sid_tgt = LongTensor([target_speaker_id])
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center=False).to(device)
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spec_lengths = LongTensor([spec.size(-1)]).to(device)
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sid_src = LongTensor([original_speaker_id]).to(device)
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sid_tgt = LongTensor([target_speaker_id]).to(device)
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audio = model.voice_conversion(spec, spec_lengths, sid_src=sid_src, sid_tgt=sid_tgt)[0][
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0, 0].data.cpu().float().numpy()
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del y, spec, spec_lengths, sid_src, sid_tgt
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@@ -54,11 +54,11 @@ def create_vc_fn(model, hps, speaker_ids):
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if __name__ == "__main__":
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parser = argparse.ArgumentParser()
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parser.add_argument("--model_dir", default="./G_latest.pth", help="directory to your fine-tuned model")
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parser.add_argument("--share", default=True, help="make link public (used in colab)")
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parser.add_argument("--share", default=False, help="make link public (used in colab)")
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args = parser.parse_args()
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hps = utils.get_hparams_from_file("./configs/finetune_speaker.json")
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device = "cpu"
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net_g = SynthesizerTrn(
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len(hps.symbols),
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