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import paddle
import paddle.nn as nn
import paddle
import os
import numpy as np
import math
import paddle.nn as nn
import ffmpeg
from scipy.signal.windows import hann
from librosa.core import stft, istft
class UNet(nn.Layer):
def __init__(self, use_elu=False):
super(UNet, self).__init__()
self.use_elu = use_elu
self.pad = nn.Pad2D(padding=[1, 2, 1, 2])
### Encoder ###
# First Layer
self.conv1 = nn.Conv2D(2, 16, kernel_size=5, stride=2) ## padding
self.encoder1 = self.encoder_block(16)
# Second Layer
self.conv2 = nn.Conv2D(16, 32, kernel_size=5, stride=2)
self.encoder2 = self.encoder_block(32)
# Third Layer
self.conv3 = nn.Conv2D(32, 64, kernel_size=5, stride=2)
self.encoder3 = self.encoder_block(64)
# Fourth Layer
self.conv4 = nn.Conv2D(64, 128, kernel_size=5, stride=2)
self.encoder4 = self.encoder_block(128)
# Fifth Layer
self.conv5 = nn.Conv2D(128, 256, kernel_size=5, stride=2)
self.encoder5 = self.encoder_block(256)
# Sixth Layer
self.conv6 = nn.Conv2D(256, 512, kernel_size=5, stride=2)
self.encoder6 = self.encoder_block(512)
### Decoder ###
# First Layer
self.decoder1 = self.decoder_block(512, 256, dropout=True)
# Second Layer
self.decoder2 = self.decoder_block(512, 128, dropout=True)
# Third Layer
self.decoder3 = self.decoder_block(256, 64, dropout=True)
# Fourth Layer
self.decoder4 = self.decoder_block(128, 32)
# Fifth Layer
self.decoder5 = self.decoder_block(64, 16)
# Sixth Layer
self.decoder6 = self.decoder_block(32, 1)
# Last Layer
self.mask = nn.Conv2D(1, 2, kernel_size=4, dilation=2, padding=3)
self.sig = nn.Sigmoid()
def encoder_block(self, out_channel):
if not self.use_elu:
return nn.Sequential(
nn.BatchNorm2D(out_channel, epsilon=1e-3, momentum=0.01),
nn.LeakyReLU(0.2)
)
else:
return nn.Sequential(
nn.BatchNorm2D(out_channel, epsilon=1e-3, momentum=0.01),
nn.ELU()
)
def decoder_block(self, in_channel, out_channel, dropout=False):
layers = [
nn.Conv2DTranspose(in_channel, out_channel, kernel_size=5, stride=2)
]
if not self.use_elu:
layers.append(nn.ReLU())
else:
layers.append(nn.ELU())
layers.append(nn.BatchNorm2D(out_channel, epsilon=1e-3, momentum=0.01))
if dropout:
layers.append(nn.Dropout(0.5))
return nn.Sequential(*layers)
def forward(self, x):
### Encoder ###
skip1 = self.pad(x)
skip1 = self.conv1(skip1)
down1 = self.encoder1(skip1)
skip2 = self.pad(down1)
skip2 = self.conv2(skip2)
down2 = self.encoder2(skip2)
skip3 = self.pad(down2)
skip3 = self.conv3(skip3)
down3 = self.encoder3(skip3)
skip4 = self.pad(down3)
skip4 = self.conv4(skip4)
down4 = self.encoder4(skip4)
skip5 = self.pad(down4)
skip5 = self.conv5(skip5)
down5 = self.encoder5(skip5)
skip6 = self.pad(down5)
skip6 = self.conv6(skip6)
down6 = self.encoder6(skip6)
### Decoder ###
up1 = self.decoder1(skip6)
up1 = up1[:, :, 1: -2, 1: -2]
merge1 = paddle.concat((skip5, up1), 1)
up2 = self.decoder2(merge1)
up2 = up2[:, :, 1: -2, 1: -2]
merge2 = paddle.concat((skip4, up2), 1)
up3 = self.decoder3(merge2)
up3 = up3[:, :, 1: -2, 1: -2]
merge3 = paddle.concat((skip3, up3), 1)
up4 = self.decoder4(merge3)
up4 = up4[:, :, 1: -2, 1: -2]
merge4 = paddle.concat((skip2, up4), 1)
up5 = self.decoder5(merge4)
up5 = up5[:, :, 1: -2, 1: -2]
merge5 = paddle.concat((skip1, up5), 1)
up6 = self.decoder6(merge5)
up6 = up6[:, :, 1: -2, 1: -2]
m = self.mask(up6)
m = self.sig(m)
return m * x
class Separator(object):
def __init__(self, params):
self.num_instruments = params['num_instruments']
self.output_dir = params['output_dir']
self.model_list = nn.LayerList()
for i, name in enumerate(self.num_instruments):
print('Loading model for instrumment {}'.format(i))
net = UNet(use_elu=params['use_elu'])
net.eval()
state_dict = paddle.load(os.path.join(params['checkpoint_path'], '%dstems_%s.pdparams' % (len(self.num_instruments), name)))
net.set_dict(state_dict)
self.model_list.append(net)
self.T = params['T']
self.F = params['F']
self.frame_length = params['frame_length']
self.frame_step = params['frame_step']
self.samplerate = params['sample_rate']
def _load_audio(
self, path, offset=None, duration=None,
sample_rate=None, dtype=np.float32):
""" Loads the audio file denoted by the given path
and returns it data as a waveform.
:param path: Path of the audio file to load data from.
:param offset: (Optional) Start offset to load from in seconds.
:param duration: (Optional) Duration to load in seconds.
:param sample_rate: (Optional) Sample rate to load audio with.
:param dtype: (Optional) Numpy data type to use, default to float32.
:returns: Loaded data a (waveform, sample_rate) tuple.
:raise SpleeterError: If any error occurs while loading audio.
"""
if not isinstance(path, str):
path = path.decode()
probe = ffmpeg.probe(path)
metadata = next(
stream
for stream in probe['streams']
if stream['codec_type'] == 'audio')
n_channels = metadata['channels']
if sample_rate is None:
sample_rate = metadata['sample_rate']
output_kwargs = {'format': 'f32le', 'ar': sample_rate}
process = (
ffmpeg
.input(path)
.output('pipe:', **output_kwargs)
.run_async(pipe_stdout=True, pipe_stderr=True))
buffer, _ = process.communicate()
waveform = np.frombuffer(buffer, dtype='<f4').reshape(-1, n_channels)
if not waveform.dtype == np.dtype(dtype):
waveform = waveform.astype(dtype)
return waveform, sample_rate
def _to_ffmpeg_codec(codec):
ffmpeg_codecs = {
'm4a': 'aac',
'ogg': 'libvorbis',
'wma': 'wmav2',
}
return ffmpeg_codecs.get(codec) or codec
def _save_to_file(
self, path, data, sample_rate,
codec=None, bitrate=None):
""" Write waveform data to the file denoted by the given path
using FFMPEG process.
:param path: Path of the audio file to save data in.
:param data: Waveform data to write.
:param sample_rate: Sample rate to write file in.
:param codec: (Optional) Writing codec to use.
:param bitrate: (Optional) Bitrate of the written audio file.
:raise IOError: If any error occurs while using FFMPEG to write data.
"""
directory = os.path.dirname(path)
#get_logger().debug('Writing file %s', path)
input_kwargs = {'ar': sample_rate, 'ac': data.shape[1]}
output_kwargs = {'ar': sample_rate, 'strict': '-2'}
if bitrate:
output_kwargs['audio_bitrate'] = bitrate
if codec is not None and codec != 'wav':
output_kwargs['codec'] = _to_ffmpeg_codec(codec)
process = (
ffmpeg
.input('pipe:', format='f32le', **input_kwargs)
.output(path, **output_kwargs)
.overwrite_output()
.run_async(pipe_stdin=True, pipe_stderr=True, quiet=True))
process.stdin.write(data.astype('<f4').tobytes())
process.stdin.close()
process.wait()
def _stft(self, data, inverse=False, length=None):
"""
Single entrypoint for both stft and istft. This computes stft and istft with librosa on stereo data. The two
channels are processed separately and are concatenated together in the result. The expected input formats are:
(n_samples, 2) for stft and (T, F, 2) for istft.
:param data: np.array with either the waveform or the complex spectrogram depending on the parameter inverse
:param inverse: should a stft or an istft be computed.
:return: Stereo data as numpy array for the transform. The channels are stored in the last dimension
"""
assert not (inverse and length is None)
data = np.asfortranarray(data)
N = self.frame_length
H = self.frame_step
win = hann(N, sym=False)
fstft = istft if inverse else stft
win_len_arg = {"win_length": None,
"length": length} if inverse else {"n_fft": N}
n_channels = data.shape[-1]
out = []
for c in range(n_channels):
d = data[:, :, c].T if inverse else data[:, c]
s = fstft(d, hop_length=H, window=win, center=False, **win_len_arg)
s = np.expand_dims(s.T, 2-inverse)
out.append(s)
if len(out) == 1:
return out[0]
return np.concatenate(out, axis=2-inverse)
def _pad_and_partition(self, tensor, T):
old_size = tensor.shape[3]
new_size = math.ceil(old_size/T) * T
tensor = nn.functional.pad(tensor, [0, new_size - old_size, 0, 0])
split_size = new_size // T
return paddle.concat(paddle.split(tensor, split_size, axis=3), axis=0)
def separate(self, input_wav):
wav_name = input_wav.split('/')[-1].split('.')[0]
output_dir = self.output_dir
if not os.path.exists(output_dir):
os.makedirs(output_dir)
source_audio, samplerate = self._load_audio(input_wav) # Length * 2
# assert int(samplerate) == 44100
if source_audio.shape[1] == 1:
source_audio = paddle.concat((source_audio, source_audio), axis=1)
elif source_audio.shape[1] > 2:
source_audio = source_audio[:, :2]
stft = self._stft(source_audio) # L * F * 2
stft = stft[:, : self.F, :]
stft_mag = abs(stft) # L * F * 2
stft_mag = paddle.to_tensor(stft_mag)
stft_mag = stft_mag.unsqueeze(0).transpose([0, 3, 2, 1]) # 1 * 2 * F * L
L = stft.shape[0]
stft_mag = self._pad_and_partition(
stft_mag, self.T) # [(L + T) / T] * 2 * F * T
stft_mag = stft_mag.transpose((0, 1, 3, 2))
# stft_mag : B * 2 * T * F
B = stft_mag.shape[0]
masks = []
stft_mag = stft_mag
for model, name in zip(self.model_list, self.num_instruments):
mask = model(stft_mag)
masks.append(mask)
paddle.save(model.state_dict(), '2stems_%s.pdparams' % name)
mask_sum = sum([m ** 2 for m in masks])
mask_sum += 1e-10
for i in range(len(self.num_instruments)):
mask = masks[i]
mask = (mask ** 2 + 1e-10/2) / (mask_sum)
mask = mask.transpose((0, 1, 3, 2)) # B x 2 X F x T
mask = paddle.concat(paddle.split(mask, mask.shape[0], axis=0), axis=3)
mask = mask.squeeze(0)[:, :, :L] # 2 x F x L
mask = mask.transpose([2, 1, 0])
# End using GPU
mask = mask.detach().numpy()
stft_masked = stft * mask
stft_masked = np.pad(
stft_masked, ((0, 0), (0, 1025), (0, 0)), 'constant')
wav_masked = self._stft(
stft_masked, inverse=True, length=source_audio.shape[0])
save_path = os.path.join(
output_dir, (wav_name + '-' + self.num_instruments[i] + '.wav'))
self._save_to_file(save_path, wav_masked,
samplerate, 'wav', '128k')
print('Audio {} separated'.format(wav_name)) |