def _extract_audio_features(self, audio_path):
if self.feature_type == "MFCC":
return tools.calculate_mfcc(audio_path)
if self.feature_type == "Pros":
return tools.extract_prosodic_features(audio_path)
if self.feature_type == "MFCC+Pros":
mfcc_vectors = tools.calculate_mfcc(audio_path)
pros_vectors = tools.extract_prosodic_features(audio_path)
mfcc_vectors, pros_vectors = tools.shorten(mfcc_vectors, pros_vectors)
return np.concatenate((mfcc_vectors, pros_vectors), axis=1)
if self.feature_type =="Spectro":
return tools.calculate_spectrogram(audio_path)
if self.feature_type == "Spectro+Pros":
spectr_vectors = tools.calculate_spectrogram(audio_path)
pros_vectors = tools.extract_prosodic_features(audio_path)
spectr_vectors, pros_vectors = tools.shorten(spectr_vectors, pros_vectors)
return np.concatenate((spectr_vectors, pros_vectors), axis=1)
# Unknown feature type
print(f"ERROR: unknown feature type '{self.feature_type}' in the 'extract_audio_features' call!")
print(f"Possible values: {self.supported_features}.")
exit(-1)
def _encode_vectors(audio_filename, gesture_filename, text_filename,
embedding_model, mode, args, augment_with_context):
"""
Extract features from a given pair of audio and motion files.
To be used by "_save_data_as_sequences" and "_save_dataset" functions.
Args:
audio_filename: file name for an audio file (.wav)
gesture_filename: file name for a motion file (.bvh)
text_filename: file name with the text transcript (.json)
embedding_model: the embedding model to encode the text with
mode: dataset type ('train', 'dev' or 'test')
args: see the 'create_dataset' function for details
augment_with_context: if True, the data sequences will be augmented with future/past context
intended use: True if the data will be used for training,
False if it will be used for validation/testing
Returns:
input_vectors [N, T, D] : speech features
text_vectors : text features
output_vectors [N, T, D] : motion features
"""
debug = False
if mode == 'test':
seq_length = 0
elif mode == 'train' or mode == "train_mirrored":
seq_length = args.seq_len
elif mode == 'dev':
seq_length = 5 * args.seq_len
else:
print(
f"ERROR: Unknown dataset type '{mode}'! Possible values: 'train', 'train_mirrored', 'dev' and 'test'."
)
exit(-1)
# Step 1: Vectorizing speech, with features of 'n_inputs' dimension, time steps of 0.01s
# and window length with 0.025s => results in an array of 100 x 'n_inputs'
if args.feature_type == "MFCC":
input_vectors = tools.calculate_mfcc(audio_filename)
elif args.feature_type == "Pros":
input_vectors = tools.extract_prosodic_features(audio_filename)
elif args.feature_type == "GeMAPS":
input_vectors = tools.extract_gemaps_features(audio_filename)
elif args.feature_type == "MFCC+Pros":
mfcc_vectors = tools.calculate_mfcc(audio_filename)
pros_vectors = tools.extract_prosodic_features(audio_filename)
mfcc_vectors, pros_vectors = tools.shorten(mfcc_vectors, pros_vectors)
input_vectors = np.concatenate((mfcc_vectors, pros_vectors), axis=1)
elif args.feature_type == "Spectro":
input_vectors = tools.calculate_spectrogram(audio_filename)
elif args.feature_type == "Spectro+Pros":
spectr_vectors = tools.calculate_spectrogram(audio_filename)
pros_vectors = tools.extract_prosodic_features(audio_filename)
spectr_vectors, pros_vectors = tools.shorten(spectr_vectors,
pros_vectors)
input_vectors = np.concatenate((spectr_vectors, pros_vectors), axis=1)
# Step 2: Read BVH
ges_str = np.load(gesture_filename)
output_vectors = ges_str['clips']
# Subsample motion (from 60 fsp to 20 fps)
output_vectors = output_vectors[0::3]
# Step 3: Obtain text transcription:
if isinstance(embedding_model, BertEmbedding):
text_encoding = encode_json_transcript_with_bert(
text_filename, embedding_model)
elif isinstance(embedding_model, FastText):
text_encoding = encode_json_transcript_with_fasttext(
text_filename, embedding_model)
if debug:
print(input_vectors.shape)
print(output_vectors.shape)
print(text_encoding.shape)
# Step 4: Align vector length
min_len = min(len(input_vectors), len(output_vectors),
2 * len(text_encoding))
# make sure the length is even
if min_len % 2 == 1:
min_len -= 1
input_vectors, output_vectors = tools.shorten(input_vectors,
output_vectors, min_len)
text_encoding = text_encoding[:int(min_len / 2)]
if debug:
print(input_vectors.shape)
print(output_vectors.shape)
print(text_encoding.shape)
if not augment_with_context:
return input_vectors, text_encoding, output_vectors
# create a list of sequences with a fixed past and future context length ( overlap them to use data more efficiently)
# ToDo: make sure the allignment holds
start_ind = args.past_context
seq_step = 10 # overlap of sequences: 0.5s
# Test if the context length is appropriate
assert args.past_context % 2 == 0
assert args.future_context % 2 == 0
assert seq_step % 2 == 0
n_reserved_inds = seq_length + args.future_context
stop_ind = input_vectors.shape[0] - n_reserved_inds
input_vectors_final = np.array([
input_vectors[i - args.past_context:i + n_reserved_inds]
for i in range(start_ind, stop_ind, seq_step)
])
stop_ind = output_vectors.shape[0] - n_reserved_inds
output_vectors_final = np.array([
output_vectors[i - args.past_context:i + n_reserved_inds]
for i in range(start_ind, stop_ind, seq_step)
])
# The text was sampled at half the sampling rate compared to audio
# So the 1 frame of text corresponds to 2 frames of audio
stop_ind = text_encoding.shape[0] - n_reserved_inds // 2
text_vectors_final = np.array([
text_encoding[i - args.past_context // 2:i + n_reserved_inds // 2]
for i in range(start_ind // 2, stop_ind, seq_step // 2)
])
if debug:
print(input_vectors_final.shape)
print(output_vectors_final.shape)
print(text_vectors_final.shape)
return input_vectors_final, text_vectors_final, output_vectors_final