GPT-SoVITS/playground/freerun.py

103 lines
2.9 KiB
Python

import onnxruntime as ort
import numpy as np
import onnx
from tqdm import tqdm
import torchaudio
import torch
MODEL_PATH = "playground/v2proplus_export/v2proplus"
def audio_postprocess(
audios,
fragment_interval: float = 0.3,
):
zero_wav = np.zeros((int(32000 * fragment_interval),)).astype(np.float32)
for i, audio in enumerate(audios):
max_audio = np.abs(audio).max() # 简单防止16bit爆音
if max_audio > 1:
audio /= max_audio
audio = np.concatenate([audio, zero_wav], axis=0)
audios[i] = audio
audio = np.concatenate(audios, axis=0)
# audio = (audio * 32768).astype(np.int16)
audio_tensor = torch.from_numpy(audio).unsqueeze(0)
torchaudio.save('playground/output.wav', audio_tensor, 32000)
return audio
input_phones = np.load("playground/input_phones.npy")
input_bert = np.load("playground/input_bert.npy").T.astype(np.float32)
ref_phones = np.load("playground/ref_phones.npy")
ref_bert = np.load("playground/ref_bert.npy").T.astype(np.float32)
audio_prompt_hubert = np.load("playground/audio_prompt_hubert.npy").astype(np.float32)
encoder = ort.InferenceSession(MODEL_PATH+"_export_t2s_encoder.onnx")
outputs = encoder.run(None, {
"text_seq": input_phones,
"text_bert": input_bert,
"ref_seq": ref_phones,
"ref_bert": ref_bert,
"ssl_content": audio_prompt_hubert
})
print(outputs[0].shape, outputs[1].shape)
x = outputs[0]
prompts = outputs[1]
fsdec = ort.InferenceSession(MODEL_PATH+"_export_t2s_fsdec.onnx")
sdec = ort.InferenceSession(MODEL_PATH+"_export_t2s_sdec.onnx")
# for i in tqdm(range(10000)):
[y, k, v, y_emb, x_example] = fsdec.run(None, {
"x": x,
"prompts": prompts
})
early_stop_num = -1
prefix_len = prompts.shape[1]
stop = False
for idx in tqdm(range(1, 1500)):
# [1, N] [N_layer, N, 1, 512] [N_layer, N, 1, 512] [1, N, 512] [1] [1, N, 512] [1, N]
[y, k, v, y_emb, logits, samples] = sdec.run(None, {
"iy": y,
"ik": k,
"iv": v,
"iy_emb": y_emb,
"ix_example": x_example
})
if early_stop_num != -1 and (y.shape[1] - prefix_len) > early_stop_num:
stop = True
if np.argmax(logits, axis=-1)[0] == 1024 or samples[0, 0] == 1024:
stop = True
if stop:
break
y[0, -1] = 0
pred_semantic = np.expand_dims(y[:, -idx:], axis=0)
# Read and resample reference audio
waveform, sample_rate = torchaudio.load("playground/ref/audio.wav")
if sample_rate != 32000:
resampler = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=32000)
waveform = resampler(waveform)
ref_audio = waveform.numpy().astype(np.float32)
vtis = ort.InferenceSession(MODEL_PATH+"_export_vits.onnx")
[audio] = vtis.run(None, {
"text_seq": input_phones,
"pred_semantic": pred_semantic,
"ref_audio": ref_audio
})
print(audio.shape, audio.dtype, audio.min(), audio.max())
audio_postprocess([audio])