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import logging
import os
import time
import matplotlib.pyplot as plt
import numpy as np
import torch
import torchaudio
import hubert_model
import utils
from models import SynthesizerTrn
from preprocess_wave import FeatureInput
logging.getLogger('matplotlib').setLevel(logging.WARNING)
dev = torch.device("cuda" if torch.cuda.is_available() else "cpu")
def timeit(func):
def run(*args, **kwargs):
t = time.time()
res = func(*args, **kwargs)
print('executing \'%s\' costed %.3fs' % (func.__name__, time.time() - t))
return res
return run
def get_end_file(dir_path, end):
file_lists = []
for root, dirs, files in os.walk(dir_path):
files = [f for f in files if f[0] != '.']
dirs[:] = [d for d in dirs if d[0] != '.']
for f_file in files:
if f_file.endswith(end):
file_lists.append(os.path.join(root, f_file).replace("\\", "/"))
return file_lists
def load_model(model_path, config_path):
# 获取模型配置
hps_ms = utils.get_hparams_from_file(config_path)
n_g_ms = SynthesizerTrn(
178,
hps_ms.data.filter_length // 2 + 1,
hps_ms.train.segment_size // hps_ms.data.hop_length,
n_speakers=hps_ms.data.n_speakers,
**hps_ms.model)
_ = utils.load_checkpoint(model_path, n_g_ms, None)
_ = n_g_ms.eval().to(dev)
# 加载hubert
hubert_soft = hubert_model.hubert_soft(get_end_file("./", "pt")[0])
feature_input = FeatureInput(hps_ms.data.sampling_rate, hps_ms.data.hop_length)
return n_g_ms, hubert_soft, feature_input, hps_ms
def resize2d_f0(x, target_len):
source = np.array(x)
source[source < 0.001] = np.nan
target = np.interp(np.arange(0, len(source) * target_len, len(source)) / target_len, np.arange(0, len(source)),
source)
res = np.nan_to_num(target)
return res
def get_units(in_path, hubert_soft):
source, sr = torchaudio.load(in_path)
source = torchaudio.functional.resample(source, sr, 16000)
if len(source.shape) == 2 and source.shape[1] >= 2:
source = torch.mean(source, dim=0).unsqueeze(0)
source = source.unsqueeze(0).to(dev)
with torch.inference_mode():
units = hubert_soft.units(source)
return units
def transcribe(source_path, length, transform, feature_input):
feature_pit = feature_input.compute_f0(source_path)
feature_pit = feature_pit * 2 ** (transform / 12)
feature_pit = resize2d_f0(feature_pit, length)
coarse_pit = feature_input.coarse_f0(feature_pit)
return coarse_pit
def get_unit_pitch(in_path, tran, hubert_soft, feature_input):
soft = get_units(in_path, hubert_soft).squeeze(0).cpu().numpy()
input_pitch = transcribe(in_path, soft.shape[0], tran, feature_input)
return soft, input_pitch
def clean_pitch(input_pitch):
num_nan = np.sum(input_pitch == 1)
if num_nan / len(input_pitch) > 0.9:
input_pitch[input_pitch != 1] = 1
return input_pitch
def plt_pitch(input_pitch):
input_pitch = input_pitch.astype(float)
input_pitch[input_pitch == 1] = np.nan
return input_pitch
def f0_to_pitch(ff):
f0_pitch = 69 + 12 * np.log2(ff / 440)
return f0_pitch
def f0_plt(in_path, out_path, tran, hubert_soft, feature_input):
s1, input_pitch = get_unit_pitch(in_path, tran, hubert_soft, feature_input)
s2, output_pitch = get_unit_pitch(out_path, 0, hubert_soft, feature_input)
plt.clf()
plt.plot(plt_pitch(input_pitch), color="#66ccff")
plt.plot(plt_pitch(output_pitch), color="orange")
plt.savefig("temp.jpg")
def calc_error(in_path, out_path, tran, feature_input):
input_pitch = feature_input.compute_f0(in_path)
output_pitch = feature_input.compute_f0(out_path)
sum_y = []
if np.sum(input_pitch == 0) / len(input_pitch) > 0.9:
mistake, var_take = 0, 0
else:
for i in range(min(len(input_pitch), len(output_pitch))):
if input_pitch[i] > 0 and output_pitch[i] > 0:
sum_y.append(abs(f0_to_pitch(output_pitch[i]) - (f0_to_pitch(input_pitch[i]) + tran)))
num_y = 0
for x in sum_y:
num_y += x
len_y = len(sum_y) if len(sum_y) else 1
mistake = round(float(num_y / len_y), 2)
var_take = round(float(np.std(sum_y, ddof=1)), 2)
return mistake, var_take
def infer(source_path, speaker_id, tran, net_g_ms, hubert_soft, feature_input):
sid = torch.LongTensor([int(speaker_id)]).to(dev)
soft, pitch = get_unit_pitch(source_path, tran, hubert_soft, feature_input)
pitch = torch.LongTensor(clean_pitch(pitch)).unsqueeze(0).to(dev)
stn_tst = torch.FloatTensor(soft)
with torch.no_grad():
x_tst = stn_tst.unsqueeze(0).to(dev)
x_tst_lengths = torch.LongTensor([stn_tst.size(0)]).to(dev)
audio = \
net_g_ms.infer(x_tst, x_tst_lengths, pitch, sid=sid, noise_scale=0.3, noise_scale_w=0.5,
length_scale=1)[0][
0, 0].data.float().cpu().numpy()
return audio, audio.shape[-1]
def del_temp_wav(path_data):
for i in get_end_file(path_data, "wav"): # os.listdir(path_data)#返回一个列表,里面是当前目录下面的所有东西的相对路径
os.remove(i)
def format_wav(audio_path, tar_sample):
raw_audio, raw_sample_rate = torchaudio.load(audio_path)
if len(raw_audio.shape) == 2 and raw_audio.shape[1] >= 2:
raw_audio = torch.mean(raw_audio, dim=0).unsqueeze(0)
tar_audio = torchaudio.functional.resample(raw_audio, raw_sample_rate, tar_sample)
torchaudio.save(audio_path[:-4] + ".wav", tar_audio, tar_sample)
return tar_audio, tar_sample
def fill_a_to_b(a, b):
if len(a) < len(b):
for _ in range(0, len(b) - len(a)):
a.append(a[0])
def mkdir(paths: list):
for path in paths:
if not os.path.exists(path):
os.mkdir(path)
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