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import onnxruntime |
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import librosa |
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import numpy as np |
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from lib.infer_pack.modules.F0Predictor.PMF0Predictor import PMF0Predictor |
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VEC_PATH = "pretrained/vec-768-layer-12.onnx" |
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MAX_LENGTH = 50.0 |
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F0_MIN = 50 |
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F0_MAX = 1100 |
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F0_MEL_MIN = 1127 * np.log(1 + F0_MIN / 700) |
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F0_MEL_MAX = 1127 * np.log(1 + F0_MAX / 700) |
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RESAMPLING_RATE = 16000 |
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class ContentVectorModel: |
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def __init__(self, vector_path=VEC_PATH): |
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providers = ["CPUExecutionProvider"] |
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self.model = onnxruntime.InferenceSession(vector_path, providers=providers) |
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def __call__(self, audio_wave): return self.process_audio(audio_wave) |
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def process_audio(self, audio_wave): |
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features = audio_wave.mean(-1) if audio_wave.ndim == 2 else audio_wave |
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features = np.expand_dims(np.expand_dims(features, 0), 0) |
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onnx_input = {self.model.get_inputs()[0].name: features} |
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logits = self.model.run(None, onnx_input)[0] |
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return logits.transpose(0, 2, 1) |
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class OnnxRVC: |
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def __init__(self, model_path, sampling_rate=40000, hop_size=512, vector_path=VEC_PATH): |
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self.vec_model = ContentVectorModel(f"{vector_path}.onnx") |
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providers = ["CPUExecutionProvider"] |
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self.model = onnxruntime.InferenceSession(model_path, providers=providers) |
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self.sampling_rate = sampling_rate |
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self.hop_size = hop_size |
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def forward(self, hubert, hubert_length, pitch, pitchf, ds, rnd): |
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onnx_input = {self.model.get_inputs()[0].name: hubert, self.model.get_inputs()[1].name: hubert_length, self.model.get_inputs()[2].name: pitch, self.model.get_inputs()[3].name: pitchf, self.model.get_inputs()[4].name: ds, self.model.get_inputs()[5].name: rnd} |
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return (self.model.run(None, onnx_input)[0] * 32767).astype(np.int16) |
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def inference(self, raw_path, sid, f0_method="pm", f0_up_key=0, pad_time=0.5, cr_threshold=0.02): |
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f0_predictor = PMF0Predictor(hop_length=self.hop_size, sampling_rate=self.sampling_rate, threshold=cr_threshold) |
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wav, sr = librosa.load(raw_path, sr=self.sampling_rate) |
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org_length = len(wav) |
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if org_length / sr > MAX_LENGTH: raise RuntimeError("Reached Max Length") |
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wav16k = librosa.resample(wav, orig_sr=sr, target_sr=RESAMPLING_RATE) |
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hubert = self.vec_model(wav16k) |
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hubert = np.repeat(hubert, 2, axis=2).transpose(0, 2, 1).astype(np.float32) |
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hubert_length = hubert.shape[1] |
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pitchf = f0_predictor.compute_f0(wav, hubert_length) |
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pitchf *= 2 ** (f0_up_key / 12) |
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pitch = pitchf.copy() |
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f0_mel = 1127 * np.log(1 + pitch / 700) |
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f0_mel = np.clip(f0_mel - F0_MEL_MIN, 0, None) * 254 / (F0_MEL_MAX - F0_MEL_MIN) + 1 |
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pitch = np.rint(f0_mel).astype(np.int64) |
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pitchf = pitchf.reshape(1, -1).astype(np.float32) |
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pitch = pitch.reshape(1, -1) |
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ds = np.array([sid]).astype(np.int64) |
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rnd = np.random.randn(1, 192, hubert_length).astype(np.float32) |
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hubert_length = np.array([hubert_length]).astype(np.int64) |
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out_wav = self.forward(hubert, hubert_length, pitch, pitchf, ds, rnd).squeeze() |
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out_wav = np.pad(out_wav, (0, 2 * self.hop_size), "constant") |
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return out_wav[:org_length] |