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data_utils.py
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data_utils.py
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import string
import numpy as np
import librosa
import soundfile as sf
from textgrids import TextGrid
import torch
import matplotlib.pyplot as plt
from absl import flags
FLAGS = flags.FLAGS
flags.DEFINE_string('normalizers_file', 'normalizers.pkl', 'file with pickled feature normalizers')
phoneme_inventory = ['aa','ae','ah','ao','aw','ax','axr','ay','b','ch','d','dh','dx','eh','el','em','en','er','ey','f','g','hh','hv','ih','iy','jh','k','l','m','n','nx','ng','ow','oy','p','r','s','sh','t','th','uh','uw','v','w','y','z','zh','sil']
def normalize_volume(audio):
rms = librosa.feature.rms(audio)
max_rms = rms.max() + 0.01
target_rms = 0.2
audio = audio * (target_rms/max_rms)
max_val = np.abs(audio).max()
if max_val > 1.0: # this shouldn't happen too often with the target_rms of 0.2
audio = audio / max_val
return audio
def dynamic_range_compression_torch(x, C=1, clip_val=1e-5):
return torch.log(torch.clamp(x, min=clip_val) * C)
def spectral_normalize_torch(magnitudes):
output = dynamic_range_compression_torch(magnitudes)
return output
mel_basis = {}
hann_window = {}
def mel_spectrogram(y, n_fft, num_mels, sampling_rate, hop_size, win_size, fmin, fmax, center=False):
if torch.min(y) < -1.:
print('min value is ', torch.min(y))
if torch.max(y) > 1.:
print('max value is ', torch.max(y))
global mel_basis, hann_window
if fmax not in mel_basis:
mel = librosa.filters.mel(sampling_rate, n_fft, num_mels, fmin, fmax)
mel_basis[str(fmax)+'_'+str(y.device)] = torch.from_numpy(mel).float().to(y.device)
hann_window[str(y.device)] = torch.hann_window(win_size).to(y.device)
y = torch.nn.functional.pad(y.unsqueeze(1), (int((n_fft-hop_size)/2), int((n_fft-hop_size)/2)), mode='reflect')
y = y.squeeze(1)
spec = torch.stft(y, n_fft, hop_length=hop_size, win_length=win_size, window=hann_window[str(y.device)],
center=center, pad_mode='reflect', normalized=False, onesided=True)
spec = torch.sqrt(spec.pow(2).sum(-1)+(1e-9))
spec = torch.matmul(mel_basis[str(fmax)+'_'+str(y.device)], spec)
spec = spectral_normalize_torch(spec)
return spec
def load_audio(filename, start=None, end=None, max_frames=None, renormalize_volume=False):
audio, r = sf.read(filename)
if len(audio.shape) > 1:
audio = audio[:,0] # select first channel of stero audio
if start is not None or end is not None:
audio = audio[start:end]
if renormalize_volume:
audio = normalize_volume(audio)
if r == 16000:
audio = librosa.resample(audio, 16000, 22050)
else:
assert r == 22050
audio = np.clip(audio, -1, 1) # because resampling sometimes pushes things out of range
pytorch_mspec = mel_spectrogram(torch.tensor(audio, dtype=torch.float32).unsqueeze(0), 1024, 80, 22050, 256, 1024, 0, 8000, center=False)
mspec = pytorch_mspec.squeeze(0).T.numpy()
if max_frames is not None and mspec.shape[0] > max_frames:
mspec = mspec[:max_frames,:]
return mspec
def double_average(x):
assert len(x.shape) == 1
f = np.ones(9)/9.0
v = np.convolve(x, f, mode='same')
w = np.convolve(v, f, mode='same')
return w
def get_emg_features(emg_data, debug=False):
xs = emg_data - emg_data.mean(axis=0, keepdims=True)
frame_features = []
for i in range(emg_data.shape[1]):
x = xs[:,i]
w = double_average(x)
p = x - w
r = np.abs(p)
w_h = librosa.util.frame(w, frame_length=16, hop_length=6).mean(axis=0)
p_w = librosa.feature.rms(w, frame_length=16, hop_length=6, center=False)
p_w = np.squeeze(p_w, 0)
p_r = librosa.feature.rms(r, frame_length=16, hop_length=6, center=False)
p_r = np.squeeze(p_r, 0)
z_p = librosa.feature.zero_crossing_rate(p, frame_length=16, hop_length=6, center=False)
z_p = np.squeeze(z_p, 0)
r_h = librosa.util.frame(r, frame_length=16, hop_length=6).mean(axis=0)
s = abs(librosa.stft(np.ascontiguousarray(x), n_fft=16, hop_length=6, center=False))
# s has feature dimension first and time second
if debug:
plt.subplot(7,1,1)
plt.plot(x)
plt.subplot(7,1,2)
plt.plot(w_h)
plt.subplot(7,1,3)
plt.plot(p_w)
plt.subplot(7,1,4)
plt.plot(p_r)
plt.subplot(7,1,5)
plt.plot(z_p)
plt.subplot(7,1,6)
plt.plot(r_h)
plt.subplot(7,1,7)
plt.imshow(s, origin='lower', aspect='auto', interpolation='nearest')
plt.show()
frame_features.append(np.stack([w_h, p_w, p_r, z_p, r_h], axis=1))
frame_features.append(s.T)
frame_features = np.concatenate(frame_features, axis=1)
return frame_features.astype(np.float32)
class FeatureNormalizer(object):
def __init__(self, feature_samples, share_scale=False):
""" features_samples should be list of 2d matrices with dimension (time, feature) """
feature_samples = np.concatenate(feature_samples, axis=0)
self.feature_means = feature_samples.mean(axis=0, keepdims=True)
if share_scale:
self.feature_stddevs = feature_samples.std()
else:
self.feature_stddevs = feature_samples.std(axis=0, keepdims=True)
def normalize(self, sample):
sample -= self.feature_means
sample /= self.feature_stddevs
return sample
def inverse(self, sample):
sample = sample * self.feature_stddevs
sample = sample + self.feature_means
return sample
def combine_fixed_length(tensor_list, length):
total_length = sum(t.size(0) for t in tensor_list)
if total_length % length != 0:
pad_length = length - (total_length % length)
tensor_list = list(tensor_list) # copy
tensor_list.append(torch.zeros(pad_length,*tensor_list[0].size()[1:], dtype=tensor_list[0].dtype, device=tensor_list[0].device))
total_length += pad_length
tensor = torch.cat(tensor_list, 0)
n = total_length // length
return tensor.view(n, length, *tensor.size()[1:])
def decollate_tensor(tensor, lengths):
b, s, d = tensor.size()
tensor = tensor.view(b*s, d)
results = []
idx = 0
for length in lengths:
assert idx + length <= b * s
results.append(tensor[idx:idx+length])
idx += length
return results
def splice_audio(chunks, overlap):
chunks = [c.copy() for c in chunks] # copy so we can modify in place
assert np.all([c.shape[0]>=overlap for c in chunks])
result_len = sum(c.shape[0] for c in chunks) - overlap*(len(chunks)-1)
result = np.zeros(result_len, dtype=chunks[0].dtype)
ramp_up = np.linspace(0,1,overlap)
ramp_down = np.linspace(1,0,overlap)
i = 0
for chunk in chunks:
l = chunk.shape[0]
# note: this will also fade the beginning and end of the result
chunk[:overlap] *= ramp_up
chunk[-overlap:] *= ramp_down
result[i:i+l] += chunk
i += l-overlap
return result
def print_confusion(confusion_mat, n=10):
# axes are (pred, target)
target_counts = confusion_mat.sum(0) + 1e-4
aslist = []
for p1 in range(len(phoneme_inventory)):
for p2 in range(p1):
if p1 != p2:
aslist.append(((confusion_mat[p1,p2]+confusion_mat[p2,p1])/(target_counts[p1]+target_counts[p2]), p1, p2))
aslist.sort()
aslist = aslist[-n:]
max_val = aslist[-1][0]
min_val = aslist[0][0]
val_range = max_val - min_val
print('Common confusions (confusion, accuracy)')
for v, p1, p2 in aslist:
p1s = phoneme_inventory[p1]
p2s = phoneme_inventory[p2]
print(f'{p1s} {p2s} {v*100:.1f} {(confusion_mat[p1,p1]+confusion_mat[p2,p2])/(target_counts[p1]+target_counts[p2])*100:.1f}')
def read_phonemes(textgrid_fname, max_len=None):
tg = TextGrid(textgrid_fname)
phone_ids = np.zeros(int(tg['phones'][-1].xmax*86.133)+1, dtype=np.int64)
phone_ids[:] = -1
phone_ids[-1] = phoneme_inventory.index('sil') # make sure list is long enough to cover full length of original sequence
for interval in tg['phones']:
phone = interval.text.lower()
if phone in ['', 'sp', 'spn']:
phone = 'sil'
if phone[-1] in string.digits:
phone = phone[:-1]
ph_id = phoneme_inventory.index(phone)
phone_ids[int(interval.xmin*86.133):int(interval.xmax*86.133)] = ph_id
assert (phone_ids >= 0).all(), 'missing aligned phones'
if max_len is not None:
phone_ids = phone_ids[:max_len]
assert phone_ids.shape[0] == max_len
return phone_ids