├── .gitignore
├── Code
├── Local_Databases
│ └── AIR
│ │ └── ACE16.tar.gz
├── pythonsrc
│ ├── ace_discriminative_nets.py
│ ├── fe_utils.py
│ ├── gan_model_worker.sh
│ ├── run_ace_discriminative_nets.sh
│ ├── run_cnnrnn_net.sh
│ ├── utils_base.py
│ ├── utils_dnntrain.py
│ ├── utils_reverb.py
│ └── utils_spaudio.py
└── results_dir
│ └── ace_h5_info.h5
├── LICENSE
└── README.md
/.gitignore:
--------------------------------------------------------------------------------
1 | *..synctex.gz(busy)
2 | Code/results_dir/concWavs
3 | Code/results_dir/tensorlogs
4 | Code/results_dir/stNW_MultChan_all
5 | Code/examplestructs
6 | Code/Local_Databases
7 | Code/results_dir/feature_extractors_*.mat
8 | *.fls
9 | *.fdb_latexmk
10 | *.out
11 | *.swp
12 | *.aux
13 | *.bbl
14 | *.blg
15 | *.log
16 | *.synctex.gz
17 | *.mexmaci64
18 | *.mexa64
19 | *.mexa32
20 | *.m~
21 | *.asv
22 | *.glo
23 | *.xdy
24 | *.toc
25 | __pycache__
26 | .idea
27 | .spyproject
28 | *.pyc
29 | *.scp
30 | *.ark
31 | Code/results_dir/tensorlog*
32 | Code/results_dir/surface_model*.*
33 | Code/results_dir/airs_*
34 | Code/results_dir/boundary_ids_*
35 | Code/results_dir/names_*
36 | *.wav
37 | Code/matlabsrc/*.wav
38 | Code/pythonsrc/*.wav
39 | tmp.py
40 | Code/pythonsrc/train_file_tuples.py
41 | Code/results_dir/training_test_data.json
42 | Code/matlabsrc/ThirdParty/simonhenin-columnlegend-8883602/
43 | Code/results_dir/acenvgenmodel_cachedir/
44 | Code/results_dir/air_modeling_results/
45 | Code/results_dir/gan_h5_info.h5
46 | Code/results_dir/training_test_data_wav.h5
47 | Code/tmp.m
48 | Code/pythonsrc/requirements.txt
49 |
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/Code/Local_Databases/AIR/ACE16.tar.gz:
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https://raw.githubusercontent.com/papayiannis/reverberation_learning_python/81ec8e70bea614c5d8a38a8ece7a7a39ac1f50b9/Code/Local_Databases/AIR/ACE16.tar.gz
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/Code/pythonsrc/ace_discriminative_nets.py:
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1 | # Copyright 2018 Constantinos Papayiannis
2 | #
3 | # This file is part of Reverberation Learning Toolbox for Python.
4 | #
5 | # Reverberation Learning Toolbox for Python is free software: you can redistribute it and/or modify
6 | # it under the terms of the GNU General Public License as published by
7 | # the Free Software Foundation, either version 3 of the License, or
8 | # (at your option) any later version.
9 | #
10 | # Reverberation Learning Toolbox for Python is distributed in the hope that it will be useful,
11 | # but WITHOUT ANY WARRANTY; without even the implied warranty of
12 | # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 | # GNU General Public License for more details.
14 | #
15 | # You should have received a copy of the GNU General Public License
16 | # along with Reverberation Learning Toolbox for Python. If not, see .
17 |
18 | """
19 |
20 | This file is the main worker for training and evaluating the DNNs proposed in [1].
21 | The script /Code/pythonsrc/run_ace_discriminative_nets.sh offers usage examples for the experiments
22 | presented in the paper.
23 |
24 | This file was original distributed in the repository at:
25 | {repo}
26 |
27 | If you use this code in your work, then cite [1].
28 |
29 | [1] C. Papayiannis, C. Evers and P. A. Naylor,
30 | "End-to-End Classification of Reverberant Rooms Using DNNs,"
31 | in IEEE/ACM Transactions on Audio, Speech, and Language Processing,
32 | vol. 28, pp. 3010-3017, 2020, doi: 10.1109/TASLP.2020.3033628.
33 |
34 | """
35 |
36 | import argparse
37 | from os.path import exists
38 | from subprocess import call
39 | from time import time
40 |
41 | import numpy as np
42 | from keras import backend as K
43 | from keras.layers import Dense, InputLayer, Reshape, \
44 | Dropout, Conv2D, MaxPooling2D, GRU, Bidirectional, TimeDistributed, \
45 | Activation, BatchNormalization
46 | from keras.layers import Layer
47 | from keras.models import Sequential
48 | from keras.models import load_model
49 | from tabulate import tabulate
50 |
51 | from fe_utils import get_ace_xy
52 | from utils_base import float2str
53 | from utils_base import run_command
54 | from utils_dnntrain import model_trainer, get_scaler_descaler, PostEpochWorker, \
55 | accuracy_eval, batch_gen
56 |
57 | _TIMESTAMP = str(time()) + '_' + str(np.random.rand(1)[0])
58 | HOSTNAME = run_command('hostname')[0]
59 | FAST_TEST = False and not HOSTNAME == 'sapws'
60 | MODEL_BASENAME = 'ace_model'
61 |
62 | MAX_EPOCHS = 50 # 50
63 | UTT_PER_ENV_DEF = 20 # 20
64 | MAX_STEPS_PER_EPOCH = 1e100 # 1e100
65 |
66 | BATCH_SIZE_BASE_PER_CLASS_AIR = 1
67 | BATCH_SIZE_BASE_PER_CLASS_SPEECH = 1
68 | MODEL_FRAMESIZE_SPEECH = 320
69 | MAX_SPEECH_LEN = 5
70 | WAVFORM_LOGPOW = False
71 | GET_POW_SPEC = True
72 | START_AT_MAX = True
73 |
74 | MODEL_FS = 16000.
75 | SIM_DATA_FS = 16000.
76 |
77 | SCRATCHPAD_DEF = '/tmp/ace_models_unsup/'
78 | SCRATCHPAD = SCRATCHPAD_DEF
79 |
80 |
81 | class Attention(Layer):
82 | # https://www.analyticsvidhya.com/blog/2019/11/comprehensive-guide-attention-mechanism-deep-learning/
83 | def __init__(self, **kwargs):
84 | super(Attention, self).__init__(**kwargs)
85 |
86 | def build(self, input_shape):
87 | self.W = self.add_weight(name="att_weight", shape=(input_shape[-1], 1), initializer="normal")
88 | self.b = self.add_weight(name="att_bias", shape=(input_shape[1], 1), initializer="zeros")
89 | super(Attention, self).build(input_shape)
90 |
91 | def call(self, x):
92 | et = K.squeeze(K.tanh(K.dot(x, self.W) + self.b), axis=-1)
93 | at = K.softmax(et)
94 | at = K.expand_dims(at, axis=-1)
95 | output = x * at
96 | return K.sum(output, axis=1)
97 |
98 | def compute_output_shape(self, input_shape):
99 | return (input_shape[0], input_shape[-1])
100 |
101 | def get_config(self):
102 | return super(Attention, self).get_config()
103 |
104 |
105 | def get_model_speech(input_dims, n_outputs, dense_width=128, filters=(8, 16, 32), kernel_size=((3, 3),) * 3,
106 | strides=((1, 1),) * 3, pooling_size=((2, 3),) * 3, use_cnn=False, use_rnn=False,
107 | use_attention=False):
108 | """
109 | Constructs models for environment classification based on reverberant speech.
110 |
111 | Args:
112 | input_dims: Dimensionality of the input
113 | n_outputs: Number of output classes
114 | dense_width: Width of FF layers
115 | filters: Number of Conv filters
116 | kernel_size: Kernel size for Conv filters
117 | strides: Strides of Conv filters
118 | pooling_size: The pooling size for th Max Poolign layers.
119 | use_cnn: Enable the use of convolutional layers
120 | use_rnn: Enable the use of recurrent layers
121 | use_attention: Enable Attention
122 |
123 | Returns:
124 | A Keras Sequential model
125 |
126 | """
127 |
128 | activation_layer = lambda: Activation('relu')
129 |
130 | print(f'Generating model with inputs: {input_dims}')
131 |
132 | if use_rnn:
133 | n_recurrent = 2
134 | else:
135 | n_recurrent = 0
136 |
137 | model = Sequential()
138 | model.add(InputLayer(input_shape=tuple(list(input_dims))))
139 | model.add(BatchNormalization())
140 | if not use_cnn:
141 | for _, _ in enumerate(filters):
142 | model.add(TimeDistributed(
143 | Dense(dense_width, activation='linear', )
144 | ))
145 | model.add(activation_layer())
146 | else:
147 | model.add(Reshape((model.output_shape[1], model.output_shape[2], 1)))
148 | for i, nfilts in enumerate(filters):
149 | for _ in range(2):
150 | model.add(Conv2D(nfilts, kernel_size[i],
151 | activation='linear', padding='same',
152 | strides=strides[i]))
153 | model.add(activation_layer())
154 | model.add(MaxPooling2D(pooling_size[i]))
155 |
156 | if n_recurrent > 0:
157 | model.add(Reshape((-1, np.prod(model.output_shape[2:]).astype(int))))
158 | model.add(Bidirectional(GRU(dense_width, activation='linear',
159 | return_sequences=True if n_recurrent > 1 else use_attention)))
160 | model.add(activation_layer())
161 | for i in range(n_recurrent - 1):
162 | model.add(GRU(dense_width, activation='linear',
163 | return_sequences=True if i < n_recurrent - 2 else use_attention))
164 | if use_attention:
165 | model.add(Attention())
166 |
167 | model.add(Reshape((-1,)))
168 | else:
169 | model.add(Reshape((-1,)))
170 | model.add(Dropout(0.1))
171 | model.add(Dense(dense_width, activation='linear'))
172 | model.add(activation_layer())
173 | model.add(Dropout(0.1))
174 | model.add(Dense(dense_width, activation='linear'))
175 | model.add(activation_layer())
176 | model.add(Dense(dense_width, activation='linear'))
177 | model.add(activation_layer())
178 |
179 | model.add(Dense(n_outputs, activation='softmax'))
180 |
181 | return model
182 |
183 |
184 | def show_classification_results(all_preds, y, ids, class_names, fold=None, mark_wrongs=False):
185 | """
186 | Prints the results of the classification predictions in a way which allows for a comparison
187 | between the predictions and the true classes.
188 |
189 | Args:
190 | all_preds: A matrix of [ N_samples x N_classes ], with 1's on the predicted class
191 | y: A matrix of [ N_samples x N_classes ], with 1's on the true class
192 | ids: The id (a string) of each sample
193 | class_names: The unique classes in the classification problem (as strings)
194 | fold: The fold in which each sample belongs to (fold of cross validation)
195 | mark_wrongs: Put a marker next to misclassified samples
196 |
197 | Returns:
198 | Nothing
199 |
200 | """
201 |
202 | accuracy = np.sum(
203 | np.all(all_preds == y, axis=1)
204 | ) / float(y.shape[0])
205 | n_hots = np.sum(all_preds, axis=1)
206 | if ~np.all(n_hots == 1):
207 | too_hot = np.where(~(n_hots == 1))[-1]
208 | raise AssertionError(
209 | 'Predictions do not make sense because the following idxs had more than one hots ' +
210 | str(too_hot) + ' with the following hots ' + str(n_hots[too_hot]))
211 | n_hots = np.sum(y, axis=1)
212 | if ~np.all(n_hots == 1):
213 | too_hot = np.where(~(n_hots == 1))[-1]
214 | raise AssertionError(
215 | 'Ground truths do not make sense because the following idxs had more than one hots ' +
216 | str(too_hot) + ' with the following hots ' + str(n_hots[too_hot]))
217 | results = np.concatenate((
218 | np.atleast_2d(ids).T,
219 | np.atleast_2d(class_names[np.argmax(all_preds, axis=1)]).T
220 | ), axis=1)
221 | headers = ('AIR', 'Prediction')
222 | if fold is not None:
223 | results = np.concatenate((
224 | results,
225 | np.atleast_2d(fold).T
226 | ), axis=1)
227 | headers = tuple(list(headers) + ['Fold'])
228 | if mark_wrongs:
229 | correct = [i.replace('EE_lobby', 'EE-lobby').split('_')[1] for i in results[:, 0]
230 | ] == results[:, 1]
231 | results = results[~correct, :]
232 | print(f'Showing {(np.sum(~correct))} wrongs of ' + str(correct.size))
233 | print(tabulate(results, headers=headers))
234 |
235 | print(f'Overall Accuracy: {float2str(accuracy, 3)}')
236 |
237 |
238 | def train_eval(h5_loc, ace_base, timestamp,
239 | use_cnn=False, use_rnn=False,
240 | read_cache=True, cacheloc_master='/tmp/', split_type='position',
241 | speech_dir=None, use_attention=False):
242 | """
243 | Worker which trains and evaluates DNN solutions for room classification, based on the data
244 | provided with the ACE challenge database.
245 |
246 | Args:
247 | h5_loc: Location of HFD5 dataset file for the ACE database, which is provided with this
248 | repository at Code/results_dir/ace_h5_info.h5. Contains information about the filenames,
249 | number of channels and also ground truth acoustic parameter values.
250 | ace_base: The folder containing the ACE wav data.
251 | timestamp: A timestamp to use for file saving
252 | use_cnn: Use CNN layers
253 | use_rnn: Use RNN layers
254 | use_attention: Use Attention mechanism layers
255 | to the ACE database ones. The dataset has 2 fields, one is 'filenames', which contains
256 | the locations of the wav AIRs and the other is 'chan', which indicates the number of
257 | channels in the audio file.
258 | read_cache: Enable the reading of any cached data, if any.
259 | cacheloc_master: Location for saving and reading cached data.
260 | split_type: Choice between array and position. The cross validation folds
261 | speech_dir: Location of speech data. Given as a list of [location of train data,
262 | location of test data].
263 |
264 | Returns: Nothing
265 |
266 | """
267 |
268 | np.random.seed(601)
269 |
270 | experiment = 'room'
271 | cacheloc_train = cacheloc_master + '/train_set/'
272 | cacheloc_bs = cacheloc_master + '/bs_set_%d/'
273 | cacheloc_test = cacheloc_master + '/train_test/'
274 | print(f'Cache location train : {cacheloc_train}')
275 | print(f'Cache location test : {cacheloc_test}')
276 | print(f'Cache location bs : {cacheloc_bs}')
277 |
278 | call(["mkdir", "-p", SCRATCHPAD])
279 | model_filename = f'{SCRATCHPAD}/{MODEL_BASENAME}_{timestamp}.h5'
280 |
281 | feature_ex_config = {
282 | 'max_air_len': MAX_SPEECH_LEN,
283 | 'fs': SIM_DATA_FS, 'forced_fs': MODEL_FS,
284 | 'max_speech_read': MAX_SPEECH_LEN, 'drop_speech': True,
285 | 'as_hdf5_ds': True,
286 | 'framesize': MODEL_FRAMESIZE_SPEECH,
287 | 'keep_ids': None, 'utt_per_env': UTT_PER_ENV_DEF,
288 | 'write_cached_latest': True, 'wavform_logpow': WAVFORM_LOGPOW,
289 | 'read_cached_latest': read_cache, 'get_pow_spec': GET_POW_SPEC,
290 | 'start_at_max': False,
291 | }
292 |
293 | (x_out_train, (y_train, y_position_train)), ids_train, class_names_train, \
294 | (_, _, _), \
295 | ((group_names_array_train, group_names_position_train, group_names_room_train),
296 | (groups_array_train, groups_position_train, group_room_train)
297 | ) = get_ace_xy(h5_file=h5_loc, ace_base=ace_base,
298 | scratchpad=SCRATCHPAD + '/train/',
299 | speech_files=speech_dir[0],
300 | group_by=('array', 'position', 'room'),
301 | cacheloc=cacheloc_train,
302 | y_type=(experiment, 'position'),
303 | **feature_ex_config)
304 |
305 | groups_array_train = [groups_array_train[ii] for ii in group_names_array_train.argsort()]
306 | group_names_array_train = group_names_array_train[group_names_array_train.argsort()]
307 | groups_position_train = [groups_position_train[ii] for ii in
308 | group_names_position_train.argsort()]
309 | group_names_position_train = group_names_position_train[group_names_position_train.argsort()]
310 |
311 | (x_out_test, y_test), ids_test, class_names_test, \
312 | (_, _, _), \
313 | ((group_names_array_test, group_names_position_test),
314 | (groups_array_test, groups_position_test)
315 | ) = get_ace_xy(h5_file=h5_loc, ace_base=ace_base,
316 | scratchpad=SCRATCHPAD + '/test/',
317 | speech_files=speech_dir[
318 | 1] if speech_dir is not None else None,
319 | group_by=('array', 'position'), cacheloc=cacheloc_test,
320 | y_type=experiment, **feature_ex_config)
321 | groups_array_test = [groups_array_test[ii] for ii in group_names_array_test.argsort()]
322 | group_names_array_train = group_names_array_train[group_names_array_train.argsort()]
323 | groups_position_test = [groups_position_test[ii] for ii in
324 | group_names_position_test.argsort()]
325 | group_names_position_test = group_names_position_test[group_names_position_test.argsort()]
326 | if not group_names_array_test.size == group_names_array_train.size:
327 | raise AssertionError('Test and train sets do not match')
328 | if ~np.all(group_names_array_train == group_names_array_test):
329 | raise AssertionError('Test and train sets do not match')
330 | if not group_names_position_test.size == group_names_position_train.size:
331 | raise AssertionError('Test and train sets do not match')
332 | if ~np.all(group_names_position_test == group_names_position_train):
333 | raise AssertionError('Test and train sets do not match')
334 |
335 | scaler, _ = get_scaler_descaler(x_out_train)
336 | x_out_train = scaler(x_out_train)
337 | x_out_test = scaler(x_out_test)
338 | callback_eval_func = accuracy_eval
339 | all_preds = []
340 | all_folds = []
341 |
342 | if split_type == 'position':
343 | fold_set = range(len(groups_position_test))
344 | experiment_groups_train = groups_position_train
345 | experiment_groups_test = groups_position_test
346 | experiment_group_names = group_names_position_train
347 | elif split_type == 'array':
348 | fold_set = range(len(groups_array_test))
349 | experiment_groups_train = groups_array_train
350 | experiment_groups_test = groups_array_test
351 | experiment_group_names = group_names_array_train
352 | else:
353 | raise AssertionError(f'Bad split type {split_type}')
354 |
355 | print('Train set : ')
356 | for i in experiment_groups_train[0]:
357 | print(f'Train : {ids_train[i]}')
358 | print('Test set : ')
359 | for i in experiment_groups_test[0]:
360 | print(f'Test : {ids_test[i]}')
361 |
362 | for i, this_outgroup in enumerate(experiment_groups_train):
363 | eval_idxs = experiment_groups_test[i]
364 |
365 | val_idxs = []
366 | for i_g in range(len(groups_position_train)):
367 | possible_choices = [j for j in groups_position_train[i_g].tolist() if j not in this_outgroup.tolist()]
368 | if len(possible_choices) == 1:
369 | if np.random.rand() < .15:
370 | val_idxs += possible_choices
371 | elif len(possible_choices) > 0:
372 | val_idxs += np.random.choice(
373 | possible_choices, int(.15 / len(groups_position_train) * x_out_train.shape[0]),
374 | replace=False).tolist()
375 | val_idxs = np.sort(val_idxs).tolist()
376 |
377 | train_idxs = np.array([j for j in range(x_out_train.shape[0]) if j not in
378 | (this_outgroup.tolist() + val_idxs)]).astype(int)
379 | print(f'For fold (zero counting absolute) {fold_set[i]} (at step: {i + 1}'
380 | f' of {len(fold_set)}), at {experiment_group_names[i]}'
381 | f' with {train_idxs.size} train '
382 | f' and {eval_idxs.size} test samples')
383 |
384 | evaluation_eval_func = lambda cmodel: callback_eval_func(
385 | x_out_test[eval_idxs, :, :], y_test[eval_idxs, :], cmodel, prefix='Test')
386 |
387 | model_filename_fold = model_filename.replace('.h5', '_fold_' + str(i) + '.h5')
388 | if exists(model_filename_fold):
389 | print('Loading pre-trained model from {model_filename_fold}')
390 | model = load_model(model_filename_fold)
391 | print('Loaded pre-trained model from {model_filename_fold}')
392 | model.summary()
393 | else:
394 | print(f'File {model_filename_fold} does not exist so i will train the model')
395 | callbacks = [PostEpochWorker(
396 | (x_out_train[val_idxs, :, :],
397 | x_out_test[eval_idxs, :, :]),
398 | (y_train[val_idxs, :], y_test[eval_idxs, :]),
399 | model_filename,
400 | eval_fun=(
401 | lambda x, y, cmodel: callback_eval_func(x, y, cmodel, prefix='Val'),
402 | lambda x, y, cmodel: callback_eval_func(x, y, cmodel, prefix='Test')),
403 | eval_every_n_epochs=100),
404 | ]
405 |
406 | effective_batch_size = BATCH_SIZE_BASE_PER_CLASS_SPEECH * y_position_train.shape[1]
407 | this_batch_gen = batch_gen(
408 | x_out_train, y_train, BATCH_SIZE_BASE_PER_CLASS_SPEECH,
409 | y_to_balance=y_position_train, sub_idxs=train_idxs)
410 | steps_per_epoch = min(int(round(y_train.shape[0] / effective_batch_size)), MAX_STEPS_PER_EPOCH)
411 |
412 | print(f'For a batch size of {effective_batch_size} i will do {steps_per_epoch} steps per epoch.')
413 | this_batch_gen_val = batch_gen(
414 | x_out_train, y_train, BATCH_SIZE_BASE_PER_CLASS_SPEECH,
415 | y_to_balance=y_position_train, sub_idxs=val_idxs)
416 | _, filename = model_trainer(
417 | this_batch_gen, x_out_train.shape[1:], y_train.shape[1],
418 | get_model_speech,
419 | tensorlog=True, callbacks=callbacks, epochs=MAX_EPOCHS,
420 | loss_patience=10, scratchpad=SCRATCHPAD,
421 | model_filename=model_filename_fold, use_attention=use_attention,
422 | print_summary=i == 0, use_cnn=use_cnn, use_rnn=use_rnn,
423 | val_gen=this_batch_gen_val, val_patience=15, steps_per_epoch=steps_per_epoch,
424 | )
425 | model = callbacks[0].best_val_model
426 |
427 | new_preds = evaluation_eval_func(model)
428 | all_preds.append(new_preds)
429 | all_folds.append(np.zeros(new_preds.shape[0], dtype=int) + fold_set[i])
430 | idxs_to_do = np.concatenate(experiment_groups_test[0:i + 1]).astype(int)
431 | show_classification_results(
432 | np.concatenate(all_preds, axis=0)[np.argsort(idxs_to_do), ...],
433 | y_test[np.sort(idxs_to_do), ...],
434 | ids_test[np.sort(idxs_to_do).tolist()],
435 | class_names_test, mark_wrongs=True,
436 | fold=np.concatenate(all_folds)[np.argsort(idxs_to_do)]
437 | )
438 | K.clear_session()
439 | print('Overall Test results for all folds:')
440 | idxs_to_do = np.concatenate(experiment_groups_test).astype(int)
441 | show_classification_results(
442 | np.concatenate(all_preds, axis=0)[np.argsort(idxs_to_do), ...],
443 | y_test[np.sort(idxs_to_do), ...],
444 | ids_test[np.sort(idxs_to_do).tolist()],
445 | class_names_test,
446 | fold=np.concatenate(all_folds)[np.argsort(idxs_to_do)]
447 | )
448 |
449 |
450 | if __name__ == '__main__':
451 | """
452 | This file is the main worker for training and evaluating the DNNs proposed in:
453 | C. Papayiannis, C. Evers and P. A. Naylor, "End-to-End Classification of Reverberant Rooms Using DNNs," in IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol. 28, pp. 3010-3017, 2020, doi: 10.1109/TASLP.2020.3033628.
454 |
455 | For usage help, run : python ace_discriminative_nets.py --help
456 | """
457 |
458 | parser = argparse.ArgumentParser(
459 | description='Arguments for training or ACE models and encodings')
460 | parser.add_argument('--ace', dest='ace_base', type=str,
461 | default='../Local_Databases/AIR/ACE16/',
462 | help='Location of the ACE database')
463 | parser.add_argument('--h5', dest='h5', type=str, default='../results_dir/ace_h5_info.h5',
464 | help='Location of HFD5 dataset file for the ACE database, which is '
465 | 'provided with this repository at Code/results_dir/ace_h5_info.h5. '
466 | 'Contains information about the filenames, number of channels and '
467 | 'also ground truth acoustic parameter values. If you want to create '
468 | 'a new one, then use fe_utils.compile_ace_h5')
469 | parser.add_argument('--speech', dest='speech', type=str, required=True, nargs='*',
470 | help='Locations where the wav files')
471 | parser.add_argument('--readcache', dest='readcache', action="store_true",
472 | default=False,
473 | help='Do not load new data, just read the last cached data')
474 | parser.add_argument('--attention', action="store_true",
475 | default=False, help='Use attention')
476 | parser.add_argument('--cnn', dest='cnn', action="store_true",
477 | default=False, help='Add CNN layers to the net')
478 | parser.add_argument('--rnn', dest='rnn', action="store_true",
479 | default=False, help='Add RNN layers to the net')
480 | parser.add_argument('--cacheloc', dest='cacheloc', type=str, default='/tmp/',
481 | help='Locations where the cache is located and/or will be stored')
482 | parser.add_argument('--split-type', type=str, default='position', choices=('position', 'array'),
483 | help='ACE Splits')
484 |
485 | args = parser.parse_args()
486 |
487 | run_command(f'mkdir -p {SCRATCHPAD}')
488 |
489 | train_eval(args.h5, args.ace_base, _TIMESTAMP, speech_dir=args.speech, read_cache=args.readcache,
490 | use_cnn=args.cnn, use_rnn=args.rnn, use_attention=args.attention,
491 | cacheloc_master=args.cacheloc, split_type=args.split_type)
492 |
--------------------------------------------------------------------------------
/Code/pythonsrc/fe_utils.py:
--------------------------------------------------------------------------------
1 | # Copyright 2018 Constantinos Papayiannis
2 | #
3 | # This file is part of Reverberation Learning Toolbox for Python.
4 | #
5 | # Reverberation Learning Toolbox for Python is free software: you can redistribute it and/or modify
6 | # it under the terms of the GNU General Public License as published by
7 | # the Free Software Foundation, either version 3 of the License, or
8 | # (at your option) any later version.
9 | #
10 | # Reverberation Learning Toolbox for Python is distributed in the hope that it will be useful,
11 | # but WITHOUT ANY WARRANTY; without even the implied warranty of
12 | # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 | # GNU General Public License for more details.
14 | #
15 | # You should have received a copy of the GNU General Public License
16 | # along with Reverberation Learning Toolbox for Python. If not, see .
17 |
18 | """
19 |
20 | This file contains a set of routines which make feature extraction and data handling easy for
21 | deep learning tasks around reverberation. It offeres some dedicated routines for the ACE
22 | challenge database (http://www.ee.ic.ac.uk/naylor/ACEweb/index.html)
23 |
24 | This file was original distributed in the repository at:
25 | {repo}
26 |
27 | If you use this code in your work, then cite:
28 | C. Papayiannis, C. Evers and P. A. Naylor,
29 | "End-to-End Classification of Reverberant Rooms Using DNNs,"
30 | in IEEE/ACM Transactions on Audio, Speech, and Language Processing,
31 | vol. 28, pp. 3010-3017, 2020, doi: 10.1109/TASLP.2020.3033628.
32 |
33 | """
34 |
35 | from os.path import abspath
36 | from os.path import basename
37 | from os.path import isfile
38 | from random import randint
39 | from random import sample
40 | from time import sleep
41 | from time import time
42 |
43 | import numpy as np
44 | import pandas as pd
45 | from h5py import File
46 | from numpy.fft import rfft
47 | from scipy.io import wavfile
48 | from scipy.io.wavfile import read
49 | from scipy.signal import fftconvolve
50 | from tabulate import tabulate
51 |
52 | from utils_base import find_all_ft, run_command
53 | from utils_base import flatten_list
54 | from utils_spaudio import enframe
55 | from utils_spaudio import my_resample
56 | from utils_spaudio import write_wav
57 |
58 |
59 | def print_split_report(y, split_idxs=(), split_names=()):
60 | """
61 |
62 | Prints the distribution of the labels of data across splits. If no splits exist,
63 | then a report on the global distribution is given. Splits are indicated with the set of
64 | indices in the provided tuple. Split names can be provided fro clarity.
65 |
66 | Args:
67 | y: The classes labels as a vector
68 | split_idxs: A list of lists of indices, indicating members of the vector which belong to
69 | each split.
70 | split_names: The name of each split as a list. Should have the same length as the
71 | index list.
72 |
73 | Returns:
74 | Nothing
75 |
76 | """
77 | if len(split_idxs) == 0:
78 | split_idxs = (np.arange(0, y.shape[0]),)
79 | if not len(split_idxs) == len(split_names):
80 | if len(split_names) > 0:
81 | raise AssertionError('Invalid Inputs')
82 | else:
83 | split_names = []
84 | for i in range(len(split_idxs)):
85 | split_names.append('Set ' + str(i))
86 |
87 | distributions = np.zeros((y.shape[1], len(split_idxs)), dtype=int)
88 | for i in range(len(split_idxs)):
89 | idxs = split_idxs[i]
90 | for j in range(y.shape[1]):
91 | distributions[j, i] = np.sum(y[idxs, j])
92 | print('Data Distributions:')
93 | print(tabulate(distributions, headers=split_names, showindex=True))
94 |
95 |
96 | def __make_start_at_max(x_in, fs=None, max_air_len=None, just_crop=False, leeway=0.0):
97 | x = np.array(x_in)
98 | max_samples = None
99 | if max_air_len is not None or leeway is not None:
100 | if fs is None:
101 | raise AssertionError('Max length and leeway should be given with fs')
102 | if max_air_len is not None:
103 | max_samples = int(np.ceil(max_air_len * fs))
104 |
105 | if leeway is None:
106 | leeway_samples = 0
107 | else:
108 | leeway_samples = int(np.ceil(leeway * fs))
109 | min_max = np.inf
110 | max_max = 0
111 | if not just_crop:
112 | for i in range(x.shape[0]):
113 | maxp = max(0, np.argmax(np.abs(x[i, :])) - leeway_samples)
114 | min_max = min(min_max, maxp)
115 | max_max = max(max_max, maxp)
116 | if maxp > 0:
117 | x[i, 0:-maxp] = np.array(x[i, maxp:])
118 | x[i, -maxp:] = 0
119 | # x = x[:, np.any(x > 0, axis=0)]
120 | print(f'Max shift : {max_max} min max : {min_max}')
121 | if max_air_len is not None:
122 | x = x[:, 0:max_samples]
123 | if x.shape[1] < max_samples:
124 | padding = np.zeros((x.shape[0], max_samples - x.shape[1]))
125 | x = np.concatenate((x, padding), axis=1)
126 | return x
127 |
128 |
129 | def __to_enframed(x, framesize=None, window=True):
130 | return np.stack(
131 | [enframe(x[i, :], framesize, int(np.ceil(framesize / 2)), hamming_window=window)
132 | for i in range(x.shape[0])])
133 |
134 |
135 | def __to_pow_spec(x, framesize=None, match_training_spectrum=False):
136 | print('Getting FFTs')
137 | oshape = x.shape
138 | if framesize is not None:
139 | if x.ndim < 3:
140 | x = __to_enframed(x, framesize=framesize)
141 | oshape = x.shape
142 | x = np.concatenate([x[i, :, :] for i in range(x.shape[0])], axis=0)
143 | x = (abs(rfft(x, axis=1)))
144 | if match_training_spectrum:
145 | av_spec = np.genfromtxt('../results_dir/surface_models/average_response_abs_fft.csv',
146 | delimiter=',')
147 | x *= np.atleast_2d(av_spec)
148 | x[x == 0] = np.min(x[x > 0]) * 0.01
149 | x = np.log(x)
150 | if framesize is not None:
151 | x = x.reshape(tuple(list(oshape)[0:2] + [-1]))
152 | return x
153 |
154 |
155 | def data_post_proc(x, fs, start_at_max, framesize, get_pow_spec, max_len, wavform_logpow):
156 | """
157 |
158 | Processes audio files, before they are passed to DNNs for training.
159 |
160 | Args:
161 | x: Audio signal samples a numpy array [N_signals X N_samples]
162 | fs: Sampling frequency
163 | start_at_max: Shift each signal so that they start at the maximum sample
164 | framesize: Set a framesize (in samples) for the signals to be enframed at. Turns the
165 | array of signals into a 3D array.
166 | get_pow_spec: Transform the signals into the log-power spectrum of the signals
167 | max_len: Maximum singal length in seconds (truncate or pad to this)
168 | wavform_logpow: Get the signals in the log-power time domain
169 |
170 | Returns:
171 | The processed signals
172 |
173 | """
174 |
175 | match_training_spectrum = False
176 | if get_pow_spec and wavform_logpow:
177 | raise AssertionError('Unexpected scenario')
178 |
179 | print('For this feature extraction i will:')
180 | if start_at_max:
181 | print('Make all inputs start at the maximum energy sample')
182 | if framesize is not None:
183 | print('Enframe the inputs')
184 | if get_pow_spec:
185 | print('Get the logpow spectrum')
186 | if max_len is not None:
187 | print('Truncate the maximum length of the input')
188 | if wavform_logpow:
189 | print('Convert the time domain samples to the logpow')
190 | print(f'And return the results for {x.shape[0]} inputs')
191 |
192 | if start_at_max or (max_len is not None):
193 | x = __make_start_at_max(x, fs, max_len, just_crop=(not start_at_max), leeway=0.0)
194 |
195 | if framesize is not None:
196 | x = __to_enframed(x, framesize=framesize, window=get_pow_spec)
197 |
198 | if get_pow_spec:
199 | x = __to_pow_spec(x, framesize=framesize, match_training_spectrum=match_training_spectrum)
200 |
201 | if wavform_logpow:
202 | x = x ** 2
203 | xmax = np.max(x)
204 | x[x < (xmax / 200)] = xmax / 200
205 | x = np.nan_to_num(np.log(x))
206 |
207 | return x
208 |
209 |
210 | def read_airs_from_wavs(wav_files, framesize=None, get_pow_spec=True,
211 | max_air_len=None, fs=None, forced_fs=None,
212 | keep_ids=None, cacheloc='/tmp/',
213 | start_at_max=True, read_cached_latest=False,
214 | wavform_logpow=False,
215 | write_cached_latest=True, max_speech_read=None,
216 | max_air_read=None, utt_per_env=1,
217 | parse_as_dirctories=True,
218 | speech_files=None, save_speech_associations=True,
219 | save_speech_examples=10, drop_speech=False, as_hdf5_ds=True,
220 | choose_channel=None, no_fex=False, scratchpad='/tmp/',
221 | copy_associations_to=None, given_associations=None):
222 | """
223 |
224 | Given a set of AIR files and additional inforamtion, data for the training of DNNs for
225 | environment classification are prepared.
226 |
227 | Args:
228 | wav_files: Location of AIR wav files
229 | framesize: The framesize to ues
230 | get_pow_spec: Convert audio to log-power spectrum domain
231 | max_air_len: The maximum length of the signals (truncate to or pad to)
232 | fs: The sampling frequency of the wav fiels to expect
233 | forced_fs: The sampling frequency to convert the data to
234 | keep_ids: None (not used)
235 | cacheloc: Location to use for cache reading and saving
236 | start_at_max: Modify the signals so that the maximum energy sample is at the begiing. (
237 | can be used to align AIRs)
238 | read_cached_latest: Read the data from the last saved cache (if nay)
239 | wavform_logpow: Get the signals in the log-power time domain
240 | write_cached_latest: Write the collected data in a cache for fast reuse
241 | max_speech_read: Maximum length of speech signal to read
242 | max_air_read: maximum aIR length to read up to
243 | utt_per_env: Number of utternaces to convolve with each AIR
244 | parse_as_dirctories: Parse the inputs as directiries and not as individual fiels
245 | speech_files: Speec files of locations
246 | save_speech_associations: Save the speech associations with the corresponding AIRs
247 | save_speech_examples: Enable the saving of examples of the reverberant speech created
248 | drop_speech: Do not include the speech samples in the saving of the cache or in the RAM.
249 | Keep only the training data arrays
250 | as_hdf5_ds: Keep the data as HDF5 datasets on disk. (Reduces RAM usage a lot)
251 | choose_channel: Channels to use for each AIR
252 | no_fex: Skip the data processign phase and just return the raw singals
253 | scratchpad: Location to use for temporary saving
254 | copy_associations_to: Save a copy of the speech-aIR associations here
255 | given_associations: Provided associatiosn between speech files and AIRs. This can be used
256 | in the case where you want to use specific speech samples for specific AIRs
257 |
258 | Returns:
259 | (X, None), Sample_names, None,
260 | (AIRs, Speech, Reverberant_speech),
261 | (Group_name, Groups), Number_of_utternaces_convolved_with_each_AIR
262 |
263 | """
264 | run_command('mkdir -p ' + cacheloc)
265 | latest_file = cacheloc + '/training_test_data_wav.h5'
266 | timestamp = str(time())
267 | filename_associations = scratchpad + '/air_speech_associations_' + timestamp + '.csv'
268 | base_examples_dir = scratchpad + '/feature_extraction_examples/'
269 | if keep_ids is not None:
270 | raise AssertionError('No ids exist in this context')
271 | if speech_files is None:
272 | utt_per_env = 1
273 | if save_speech_associations:
274 | print('There is no speech to save in associations, setting to false')
275 | save_speech_associations = False
276 | if save_speech_examples:
277 | print('There is no speech to save audio for, setting to 0 examples')
278 | save_speech_examples = 0
279 |
280 | hf = None
281 | try:
282 | if isfile(latest_file) and read_cached_latest:
283 | print(f'Reading : {latest_file}')
284 | hf = File(latest_file, 'r')
285 | if as_hdf5_ds:
286 | x = hf['x']
287 | airs = hf['airs']
288 | utt_per_env = np.array(hf['utts'])
289 | rev_speech = hf['rev_names']
290 | clean_speech = hf['clean_speech']
291 | print(f'Done creating handles to : {latest_file}')
292 | else:
293 | utt_per_env = np.array(hf['utts'])
294 | x = np.array(hf.get('x'))
295 | ids = np.array(hf.get('ids'))
296 | airs = np.array(hf.get('airs'))
297 | rev_speech = np.array(hf.get('rev_names'))
298 | clean_speech = np.array(hf.get('clean_speech'))
299 | print(f'Done reading : {latest_file}')
300 | ids = np.array([x.decode() for x in hf['ids']])
301 | if given_associations is not None:
302 | print('! I read the cache so the given associations were not used')
303 | if copy_associations_to is not None:
304 | print(f'! I read the cache so the associations could not be saved at {copy_associations_to}')
305 | return (x, None), ids, None, (airs, clean_speech, rev_speech), utt_per_env
306 | except (ValueError, KeyError) as ME:
307 | print('Tried to read ' + latest_file + ' but failed with ' + ME.message)
308 | if hf is not None:
309 | hf.close()
310 |
311 | if given_associations is not None:
312 | print(f'You gave me speech associations, Speech: {len(given_associations["speech"])}'
313 | f' entries and Offsets: {len(given_associations["speech"])} entries')
314 |
315 | ids = None
316 | x = None
317 | x_speech = None
318 | x_rev_speech = None
319 |
320 | if forced_fs is None:
321 | forced_fs = fs
322 | resample_op = lambda x: x
323 | if not forced_fs == fs:
324 | resample_op = lambda x: np.array(
325 | my_resample(np.array(x.T, dtype=float), fs, forced_fs)
326 | ).T
327 |
328 | max_air_read_samples = None
329 | if max_air_read is not None:
330 | if fs is None:
331 | raise AssertionError('Cannot work with max_air_read without fs')
332 | max_air_read_samples = int(np.ceil(fs * max_air_read))
333 | if max_speech_read is not None:
334 | if fs is None:
335 | raise AssertionError('Cannot work with max_speech_read without fs')
336 | max_speech_read_samples = int(np.ceil(fs * max_speech_read))
337 | else:
338 | max_speech_read_samples = None
339 |
340 | if parse_as_dirctories:
341 | if not type(wav_files) is list:
342 | wav_files = [wav_files]
343 | wav_files = find_all_ft(wav_files, ft='.wav', find_iname=True)
344 | if speech_files is not None:
345 | if not type(speech_files) is list:
346 | speech_files = [speech_files]
347 | speech_files = find_all_ft(speech_files, ft='.wav', find_iname=True)
348 |
349 | if save_speech_examples:
350 | run_command('rm -r ' + base_examples_dir)
351 | run_command('mkdir -p ' + base_examples_dir)
352 |
353 | associations = []
354 | save_counter = 0
355 | all_names = [basename(i).replace('.wav', '') + '_' + str(j) for i in wav_files for j in
356 | range(utt_per_env)]
357 | if type(choose_channel) is list:
358 | choose_channel = [i for i in choose_channel for _ in range(utt_per_env)]
359 | wav_files = [i for i in wav_files for _ in range(utt_per_env)]
360 | offsets = []
361 | for i, this_wav_file in enumerate(wav_files):
362 | # if speech_files is not None:
363 | # print("Reading: " + this_wav_file + " @ " + str(i + 1) + " of " + str(len(wav_files)), end='')
364 | names = [all_names[i]]
365 | this_fs, airs = wavfile.read(this_wav_file)
366 | airs = airs.astype(float)
367 | if airs.ndim > 1:
368 | if choose_channel is not None:
369 | if type(choose_channel) is list:
370 | airs = airs[:, choose_channel[i]]
371 | names[0] += '_ch' + str(choose_channel[i])
372 | else:
373 | airs = airs[:, choose_channel]
374 | names[0] += '_ch' + str(choose_channel)
375 | else:
376 | names = [names[0] + '_' + str(ch_id) for ch_id in range(airs.shape[1])]
377 | airs = airs.T
378 | airs = np.atleast_2d(airs)
379 | airs /= np.repeat(np.atleast_2d(abs(airs).max()).T, airs.shape[1], 1).astype(float)
380 | if airs.shape[0] > 1 and given_associations is not None:
381 | raise AssertionError('Cannot work out given associations for multichannel airs')
382 | this_speech_all = []
383 | this_rev_speech_all = []
384 | if speech_files is not None:
385 | for ch_id in range(airs.shape[0]):
386 | if given_associations is None:
387 | chosen_file = sample(range(len(speech_files)), 1)[0]
388 | this_speech_file = speech_files[chosen_file]
389 | else:
390 | chosen_file = given_associations['speech'][i]
391 | this_speech_file = chosen_file
392 | associations.append(chosen_file)
393 | this_speech_fs, this_speech = wavfile.read(this_speech_file)
394 | if this_speech.ndim > 1:
395 | raise AssertionError('Can\'t deal with multichannel speech in this context')
396 | if not this_speech_fs == this_fs:
397 | this_speech = my_resample(this_speech, this_speech_fs, this_fs)
398 | max_offset_for_check = None
399 | if max_speech_read_samples is not None:
400 | max_offset_for_check = this_speech.size - max_speech_read_samples
401 | offset = randint(0, this_speech.size - max_speech_read_samples)
402 | this_speech = this_speech[offset:offset + max_speech_read_samples]
403 | else:
404 | offset = 0
405 | if given_associations is not None:
406 | offset = given_associations['offsets'][i]
407 | if max_speech_read_samples is not None:
408 | if offset >= max_offset_for_check:
409 | raise AssertionError(
410 | 'Invalid offset from given associations, got ' + str(
411 | offset) + ' expected max is ' + str(
412 | this_speech.size - max_speech_read_samples))
413 |
414 | conv_air = np.array(airs[ch_id, :])
415 | conv_air = conv_air[
416 | np.where(~(conv_air == 0))[-1][0]:np.where(~(conv_air == 0))[-1][-1]]
417 |
418 | # Making convolution
419 | this_rev_speech = fftconvolve(this_speech, conv_air, 'same')
420 | #
421 |
422 | dp_arival = np.argmax(abs(conv_air))
423 | this_rev_speech = this_rev_speech[dp_arival:]
424 | if dp_arival > 0:
425 | this_rev_speech = np.concatenate(
426 | (this_rev_speech, np.zeros(dp_arival, dtype=this_rev_speech.dtype)))
427 |
428 | this_speech = np.atleast_2d(this_speech)
429 | this_rev_speech = np.atleast_2d(this_rev_speech)
430 | this_speech_all.append(this_speech)
431 | this_rev_speech_all.append(this_rev_speech)
432 |
433 | offsets.append(offset)
434 | if save_speech_examples >= save_counter:
435 | save_names = [
436 | basename(this_wav_file).replace('.wav', '') + '_air_' + str(
437 | offset) + '.wav',
438 | basename(this_wav_file).replace('.wav', '') + '_clean_speech_' + str(
439 | offset) + '.wav',
440 | basename(this_wav_file).replace('.wav', '') + '_rev_speech_' + str(
441 | offset) + '.wav'
442 | ]
443 | for examples in range(len(save_names)):
444 | save_names[examples] = base_examples_dir + save_names[examples]
445 | write_wav(save_names[0], this_fs, airs[ch_id, :])
446 | write_wav(save_names[1], this_fs, this_speech.flatten())
447 | write_wav(save_names[2], this_fs, this_rev_speech.flatten())
448 | save_counter += 1
449 | this_speech = np.concatenate(this_speech_all, axis=0)
450 | this_rev_speech = np.concatenate(this_rev_speech_all, axis=0)
451 |
452 | if not this_fs == fs:
453 | raise AssertionError('Your sampling rates are not consistent')
454 | if i > 0:
455 | ids = np.concatenate((ids, names))
456 | else:
457 | ids = names
458 |
459 | if max_air_read is not None:
460 | airs = airs[:, 0:max_air_read_samples]
461 | if False and speech_files is not None:
462 | print(f"Got {airs.shape}")
463 | airs = resample_op(airs)
464 | if airs.ndim < 2:
465 | airs = np.atleast_2d(airs)
466 | # print('Done resampling')
467 | if i > 0:
468 | if x.shape[1] < airs.shape[1]:
469 | npads = -x.shape[1] + airs.shape[1]
470 | x = np.concatenate((x, np.zeros((x.shape[0], npads)).astype(x.dtype)), axis=1)
471 | x = np.concatenate((x, airs), axis=0)
472 | else:
473 | if x.shape[1] > airs.shape[1]:
474 | npads = x.shape[1] - airs.shape[1]
475 | airs = np.concatenate((airs, np.zeros((airs.shape[0], npads)).astype(
476 | airs.dtype)), axis=1)
477 | x.resize((x.shape[0] + airs.shape[0], x.shape[1]), refcheck=False)
478 | x[-airs.shape[0]:, :] = np.array(airs)
479 |
480 | if speech_files is not None:
481 | if x_speech.shape[1] < this_speech.shape[1]:
482 | npads = -x_speech.shape[1] + this_speech.shape[1]
483 | x_speech = np.concatenate(
484 | (x_speech, np.zeros((x_speech.shape[0], npads)).astype(x_speech.dtype)),
485 | axis=1)
486 | x_speech = np.concatenate((x_speech, this_speech), axis=0)
487 | else:
488 | if x_speech.shape[1] > this_speech.shape[1]:
489 | npads = x_speech.shape[1] - this_speech.shape[1]
490 | this_speech = np.concatenate(
491 | (this_speech, np.zeros((this_speech.shape[0],
492 | npads)).astype(
493 | this_speech.dtype)), axis=1)
494 | x_speech.resize(
495 | (x_speech.shape[0] + this_speech.shape[0], x_speech.shape[1]),
496 | refcheck=False)
497 | x_speech[-this_speech.shape[0]:, :] = this_speech
498 |
499 | if x_rev_speech.shape[1] < this_rev_speech.shape[1]:
500 | npads = -x_rev_speech.shape[1] + this_rev_speech.shape[1]
501 | x_rev_speech = np.concatenate(
502 | (x_rev_speech, np.zeros((x_rev_speech.shape[0], npads)
503 | ).astype(x_rev_speech.dtype)),
504 | axis=1)
505 | x_rev_speech = np.concatenate((x_rev_speech, this_rev_speech), axis=0)
506 | else:
507 | if x_rev_speech.shape[1] > this_rev_speech.shape[1]:
508 | npads = x_rev_speech.shape[1] - this_rev_speech.shape[1]
509 | this_rev_speech = np.concatenate(
510 | (this_rev_speech, np.zeros((this_rev_speech.shape[0], npads)
511 | ).astype(
512 | this_rev_speech.dtype)), axis=1)
513 | x_rev_speech.resize(
514 | (x_rev_speech.shape[0] + this_rev_speech.shape[0],
515 | x_rev_speech.shape[1]), refcheck=False)
516 | x_rev_speech[-this_rev_speech.shape[0]:, :] = this_rev_speech
517 | else:
518 | x = np.array(airs)
519 | if speech_files is not None:
520 | x_speech = np.array(this_speech)
521 | x_rev_speech = np.array(this_rev_speech)
522 |
523 | if save_speech_associations:
524 | df = pd.DataFrame(
525 | {'air': wav_files,
526 | 'speech': np.array(speech_files)[associations]
527 | if given_associations is None else
528 | given_associations['speech'],
529 | 'offsets': offsets
530 | if given_associations is None else
531 | given_associations['offsets']})
532 |
533 | df.to_csv(filename_associations, index=False)
534 | print(f'Saved: {filename_associations} ')
535 | if copy_associations_to is not None:
536 | run_command('cp ' + filename_associations + ' ' + copy_associations_to)
537 | print(f'Saved: {copy_associations_to}')
538 |
539 | if fs is not None:
540 | print(f'Got {x.shape[0]} AIRs of duration {x.shape[1] / float(fs)}')
541 | else:
542 | print(f'Got {x.shape[0]} AIRs of length {x.shape[1]}')
543 |
544 | if speech_files is not None:
545 | proc_data = x_rev_speech
546 | else:
547 | proc_data = x
548 |
549 | if drop_speech:
550 | x_rev_speech = []
551 | x_speech = []
552 | x = []
553 |
554 | if no_fex:
555 | x_out = None
556 | print('Skipping feature extraction')
557 | else:
558 | x_out = data_post_proc(np.array(proc_data), forced_fs, start_at_max, framesize,
559 | get_pow_spec, max_air_len, wavform_logpow)
560 |
561 | print(f'Left with {x_out.shape} AIR features data ')
562 |
563 | ids = ids.astype(str)
564 |
565 | wrote_h5 = False
566 | if write_cached_latest:
567 | try:
568 | hf = File(latest_file, 'w')
569 | if no_fex:
570 | hf.create_dataset('x', data=[])
571 | else:
572 | hf.create_dataset('x', data=x_out)
573 | hf.create_dataset('y', data=[])
574 | hf.create_dataset('ids', data=[x.encode() for x in ids])
575 | hf.create_dataset('class_names', data=[])
576 | hf.create_dataset('airs', data=x)
577 | hf.create_dataset('utts', data=utt_per_env)
578 | if speech_files is not None:
579 | hf.create_dataset('clean_speech', data=x_speech)
580 | hf.create_dataset('rev_names', data=x_rev_speech)
581 | else:
582 | hf.create_dataset('clean_speech', data=[])
583 | hf.create_dataset('rev_names', data=[])
584 | hf.close()
585 | wrote_h5 = True
586 | print(f'Wrote: {latest_file}')
587 | except IOError as ME:
588 | print(f'Cache writing failed with {ME}')
589 |
590 | if (not wrote_h5) and as_hdf5_ds:
591 | raise AssertionError('Could not provide data in correct format')
592 | if as_hdf5_ds:
593 | hf = File(latest_file, 'r')
594 | x_out = hf['x']
595 | ids = hf['ids']
596 | x = hf['airs']
597 | x_speech = hf['clean_speech']
598 | x_rev_speech = hf['rev_names']
599 | # hf.close()
600 | ids = np.array([x.decode() for x in ids])
601 |
602 | return (x_out, None), ids, None, (x, x_speech, x_rev_speech), utt_per_env
603 |
604 |
605 | def compile_ace_h5(wav_loc, saveloc, ft='.wav', all_single_channel=False):
606 | """
607 |
608 | Create an HDF5 dataset which contains information about a set of files which describe AIRs of
609 | acoustic environments. This file can be used to train DNNs using ace_discriminative_nets.py
610 |
611 | Args:
612 | wav_loc: The location of the wav files as a list
613 | saveloc: The location to save to the HDF5 file
614 | ft: The file type to look for
615 | all_single_channel: Assume that all responses are single channel (faster and does not
616 | require soxi)
617 |
618 | Returns:
619 | Nothing
620 |
621 | """
622 |
623 | all_wavs = find_all_ft(wav_loc, ft=ft, use_find=True)
624 | channels = []
625 | for i in range(len(all_wavs)):
626 | print(f'Reading : all_wavs[i]')
627 | all_wavs[i] = abspath(all_wavs[i])
628 | if all_single_channel:
629 | channels.append('1')
630 | else:
631 | try:
632 | channels.append(run_command('soxi -c ' + all_wavs[i])[0])
633 | except OSError as ME:
634 | print('I think that soxi is not installed because when i tried to use it to get '
635 | 'the number of channels, i got this ' + str(ME))
636 | raise
637 |
638 | hf = File(saveloc, 'w')
639 | hf.create_dataset('filenames', data=all_wavs)
640 | hf.create_dataset('chan', data=channels)
641 | hf.close()
642 | print(f'Done with : {saveloc}')
643 |
644 |
645 | def get_ace_xy(h5_file='../results_dir/ace_h5_info.h5', ace_base='../Local_Databases/AIR/ACE/',
646 | y_type='room', group_by=None, utt_per_env=1, speech_files=None,
647 | print_distributions=False,
648 | parse_as_dirctories=False,
649 | choose_channel=None,
650 | **kwargs):
651 | """
652 |
653 | Collects training data and labels for traiing of DNNs using ace_discriminative_nets,
654 | based on the ACE Challenge data[1].
655 |
656 | Args:
657 | h5_file: Location of HFD5 dataset file for the ACE database, which is provided with this
658 | repository at Code/results_dir/ace_h5_info.h5. Contains information about the filenames,
659 | number of channels and also ground truth acoustic parameter values. If you want to create a
660 | new one, then use fe_utils.compile_ace_h5
661 | ace_base: The location of the ACE database data
662 | y_type: Creating labels from the data using specific information. This
663 | can be either of:
664 | 'room', 'recording', 'array', 'recording', 'position', 'air'
665 | group_by: Creating grouping information from the data using specific information. This
666 | can be either of:
667 | 'room', 'recording', 'array', 'recording', 'position', 'air'
668 | utt_per_env: Number of speech utterances to convolve with each AIR
669 | speech_files: Speech directory to pick up speech from and convolve it with the AIRs
670 | print_distributions: Print data information with regards to class distributions
671 | parse_as_dirctories: (ignored)
672 | choose_channel: (ignored)
673 | **kwargs: Additional arguments to be passed to read_airs_from_wavs
674 |
675 | Returns:
676 | (X, Y), Sample_names, Class_names,
677 | (AIRs, Speech, Reverberant_speech),
678 | (Group_name, Groups)
679 |
680 | """
681 | parse_as_dirctories = False
682 |
683 | hf = File(h5_file, 'r')
684 | wav_files = list((np.array(hf.get('filenames')).astype(str)).tolist())
685 | chan = list((np.array(hf.get('chan')).astype(int) - 1).tolist())
686 |
687 | type_dict = {'502': 'Office', '803': 'Office', '503': 'Meeting_Room', '611': 'Meeting_Room',
688 | '403a': 'Lecture_Room', '508': 'Lecture_Room', 'EE-lobby': 'Building_Lobby'}
689 | basenames = [thename.split('/')[-1].replace('EE_lobby', 'EE-lobby') for thename in wav_files]
690 | room = [thename.split('_')[1] for thename in basenames]
691 | array = [thename.split('_')[0] for thename in basenames]
692 | room_type = [type_dict[thename.split('_')[1]] for thename in basenames]
693 | recording = basenames
694 |
695 | if ace_base is None:
696 | x_out = None
697 | x = None
698 | x_speech = None
699 | x_rev_speech = None
700 | ids = flatten_list([[basename(this_file).replace('.wav', '') + '_' + str(j) + '_ch' + str(k)
701 | for k in range(chan[i])]
702 | for i, this_file in enumerate(wav_files)
703 | for j in range(utt_per_env)])
704 | else:
705 | for i in range(len(wav_files)):
706 | wav_files[i] = ace_base + '/' + wav_files[i]
707 | (x_out, _), \
708 | ids, _, \
709 | (x, x_speech, x_rev_speech), \
710 | utt_per_env = read_airs_from_wavs(
711 | wav_files, utt_per_env=utt_per_env, speech_files=speech_files,
712 | parse_as_dirctories=parse_as_dirctories,
713 | choose_channel=chan,
714 | **kwargs)
715 | if 'ch' not in ids[0]:
716 | if np.sum(['ch' in ids[i] for i in range(len(ids))]) > 0:
717 | raise AssertionError('Unexpected condition')
718 | ch = [0 for _ in range(len(ids))]
719 | else:
720 | ch = [int(i.split('ch')[1]) for i in ids]
721 |
722 | y = []
723 | class_names = []
724 |
725 | flat_back_y = False
726 | if not (isinstance(y_type, list) or isinstance(y_type, tuple)):
727 | flat_back_y = True
728 | y_type = (y_type,)
729 |
730 | for this_y_type in y_type:
731 | if this_y_type == 'room':
732 | new_y, new_class_names, _ = categorical_to_mat(room)
733 | def_group_by = 'recording'
734 | elif this_y_type == 'type':
735 | new_y, new_class_names, _ = categorical_to_mat(room_type)
736 | def_group_by = 'room'
737 | elif this_y_type == 'array':
738 | new_y, new_class_names, _ = categorical_to_mat(array)
739 | def_group_by = 'recording'
740 | elif this_y_type == 'position' or y_type == 'position':
741 | new_y, new_class_names, _ = categorical_to_mat(recording)
742 | def_group_by = 'air'
743 | elif this_y_type == 'channel':
744 | new_y, new_class_names, _ = categorical_to_mat(ch)
745 | def_group_by = 'position'
746 | else:
747 | raise AssertionError('Invalid y_type')
748 | y.append(new_y)
749 | class_names.append(new_class_names)
750 |
751 | flat_back_groups = False
752 | if group_by is None:
753 | group_by = (def_group_by,)
754 | elif not (isinstance(group_by, list) or isinstance(group_by, tuple)):
755 | flat_back_groups = True
756 | group_by = (group_by,)
757 |
758 | group_name, groups = ([], [])
759 | for this_group_by in group_by:
760 | if this_group_by == 'recording' or this_group_by == 'position':
761 | _, new_group_name, new_groups = categorical_to_mat(recording)
762 | elif this_group_by == 'room':
763 | _, new_group_name, new_groups = categorical_to_mat(room)
764 | elif this_group_by == 'array':
765 | _, new_group_name, new_groups = categorical_to_mat(array)
766 | elif this_group_by == 'air':
767 | new_groups = np.atleast_2d(np.arange(len(y))).T
768 | new_group_name = np.array(ids)
769 | elif this_group_by == 'channel':
770 | max_ch = max(ch) + 1
771 | ch_array = np.zeros((len(ch), max_ch), dtype=bool)
772 | for i in range(len(ch)):
773 | ch_array[i, ch[i]] = True
774 | new_groups = np.array(ch_array)
775 | new_group_name = np.array(['ch_' + str(i) for i in range(max_ch)])
776 | else:
777 | raise AssertionError('Invalid group_by ' + this_group_by)
778 | group_name.append(new_group_name)
779 | groups.append(new_groups)
780 |
781 | for i in range(len(y)):
782 | if print_distributions:
783 | print_split_report(y[i])
784 | if np.any(~(np.sum(y[i], axis=1) == 1)):
785 | raise AssertionError('Invalid y outputs')
786 | y[i] = np.concatenate([y[i][ii:ii + 1, :] for ii in range(y[i].shape[0])
787 | for _ in range(utt_per_env)],
788 | axis=0)
789 | for ii in range(len(groups)):
790 | groups[ii] = [np.concatenate([list(range(i * utt_per_env, (i + 1) * utt_per_env))
791 | for i in groups[ii][j]]).astype(int)
792 | for j in range(len(groups[ii]))]
793 |
794 | y = tuple(y)
795 | class_names = tuple(class_names)
796 | groups = tuple(groups)
797 | group_name = tuple(group_name)
798 | if flat_back_groups:
799 | groups = groups[0]
800 | group_name = group_name[0]
801 | if flat_back_y:
802 | y = y[0]
803 | class_names = class_names[0]
804 |
805 | return (x_out, y), ids, class_names, (x, x_speech, x_rev_speech), (group_name, groups)
806 |
807 |
808 | def categorical_to_mat(categorical):
809 | """
810 |
811 | Converts a categorical variable vector into a one-hot class matrix
812 |
813 | Args:
814 | categorical: The categorical variable vector
815 |
816 | Returns:
817 | The class matrix as an array
818 | The name of each class corresponding to each column of the array
819 | The list of indices of 'categorical' which belong to the labels in 'unique_vals'
820 |
821 | """
822 | categorical = np.array(categorical)
823 | unique_vals = np.unique(categorical)
824 | y = np.zeros((categorical.size, unique_vals.size), dtype=bool)
825 | groups = []
826 | for i, val in enumerate(unique_vals):
827 | y[categorical == val, i] = True
828 | groups.append(np.where(categorical == val)[-1])
829 | # groups=np.array(groups)
830 | return y, unique_vals, groups
831 |
832 |
833 | def collect_wavs(filenames, dest_fs=None):
834 | """
835 |
836 | Collects and packages a set of wav files to an array of samples
837 |
838 | Args:
839 | filenames: File locations as a list
840 | dest_fs: Sampling frequency to use
841 |
842 | Returns:
843 | An array of the samples of the files as [N_files x N_samples]
844 |
845 | """
846 | if not (isinstance(filenames, list) or isinstance(filenames, tuple)):
847 | filenames = [filenames]
848 | samples = []
849 | max_len = 0
850 | for the_filename in filenames:
851 | fs, new_samples = read(the_filename)
852 | if dest_fs:
853 | new_samples = my_resample(new_samples, fs, dest_fs)
854 | if new_samples.ndim == 1:
855 | new_samples = np.atleast_2d(new_samples).T
856 | max_len = max(max_len, new_samples.shape[0])
857 | samples.append(new_samples)
858 | for i in range(len(samples)):
859 | this_len = samples[i].shape[0]
860 | missing = max_len - this_len
861 | if missing > 0:
862 | samples[i] = np.concatenate(
863 | (samples[i], np.zeros((missing, samples[i].shape[1]), dtype=samples[i].dtype)))
864 |
865 | out = np.concatenate([samples[i].T for i in range(len(samples))], axis=0)
866 |
867 | return out
868 |
--------------------------------------------------------------------------------
/Code/pythonsrc/gan_model_worker.sh:
--------------------------------------------------------------------------------
1 | #!/bin/bash
2 |
3 | # Copyright 2018 Constantinos Papayiannis
4 | #
5 | # This file is part of Reverberation Learning Toolbox for Python.
6 | #
7 | # Reverberation Learning Toolbox for Python is free software: you can redistribute it and/or modify
8 | # it under the terms of the GNU General Public License as published by
9 | # the Free Software Foundation, either version 3 of the License, or
10 | # (at your option) any later version.
11 | #
12 | # Reverberation Learning Toolbox for Python is distributed in the hope that it will be useful,
13 | # but WITHOUT ANY WARRANTY; without even the implied warranty of
14 | # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
15 | # GNU General Public License for more details.
16 | #
17 | # You should have received a copy of the GNU General Public License
18 | # along with Reverberation Learning Toolbox for Python. If not, see .
19 |
20 | # Give the following arguments
21 | # Location of saved modeling results for ace which contains the result in the form the-location/the-name-of-the-environment-containing-the-room-name/log.txt
22 | # Scratchpad location
23 |
24 | ##################################################################################################################
25 | ##################################################################################################################
26 | #
27 | # Description:
28 | #
29 | # This script collects the results of modeling ACE AIRs [2] using the proposed model in [1]. Then these results are
30 | # used to train GANs for the rooms in ACE. This will result in 1 GAN for each of the 7 rooms, part of the ACE
31 | # challenge measurements. Each GAN is trained using data for each room and then produces 100 acoustic environment
32 | # instances from each model. These instances are to be used for training DNNs, providing a method for data
33 | # augmentation.
34 | #
35 | # To collect the results prior to running this step, run:
36 | # bash ace_acenvgenmodeling.sh /tmp/modeling_results
37 | #
38 | # Usage:
39 | # bash gan_model_worker.sh
40 | #
41 | # Example:
42 | # bash gan_model_worker.sh /tmp/modeling_results/ /tmp/gan_results/
43 | #
44 | #
45 | # This file was original distributed in the repository at:
46 | # {repo}
47 | # If you use this code in your work, then cite [1].
48 | #
49 | # [1] C. Papayiannis, C. Evers and P. A. Naylor,
50 | # "End-to-End Classification of Reverberant Rooms Using DNNs,"
51 | # in IEEE/ACM Transactions on Audio, Speech, and Language Processing,
52 | # vol. 28, pp. 3010-3017, 2020, doi: 10.1109/TASLP.2020.3033628.
53 | # [2] http://www.ee.ic.ac.uk/naylor/ACEweb/index.html
54 | #
55 | #################################################################################################################
56 | #################################################################################################################
57 |
58 |
59 | set -e
60 |
61 | if [ "$#" -lt 2 ]; then
62 | echo 'Illegal number of parameters, expected >= 2.
63 | 1) The location where the logs of ace_acenvgenmodeling.sh results were saved
64 | 2) The location where the results and the data augmentation h5 dataset will be saved
65 | 3+) Arguments passed to gan_model.py
66 | Example:
67 | bash gan_model_worker.sh /tmp/modeling_results/ /tmp/gan_results/
68 |
69 | '
70 | exit 1
71 | fi
72 |
73 | ace_results=$1
74 | saveloc=$2
75 | test_array='Mobile'
76 |
77 | shift
78 | shift
79 |
80 | mkdir -p $saveloc
81 |
82 | for i in 611 403a 803 503 502 508 EE_lobby; do
83 | $HOME/anaconda2/bin/python -u acenvgenmodel_collect_results.py ` find $ace_results/*${i}* -name log.txt | grep -v "$test_array"` --saveloc $saveloc/ref_rep_$i/
84 | log_dest=$saveloc/log_${i}.txt
85 | echo Working for room $i and saving log at $log_dest
86 | $HOME/anaconda2/bin/python -u gan_model.py --h5 $saveloc/ref_rep_$i/reflection_y_data.h5 --saveloc $saveloc/gan_$i/ --airname GAN_${i}_%d_RIR.wav --nodisplay $* | tee $log_dest
87 | done
88 |
89 | aug_h5=$saveloc/gan_aug_data.h5
90 | python -c "from fe_utils import compile_ace_h5; compile_ace_h5('$saveloc','$aug_h5',ft='RIR.wav');"
91 | echo Saved data augmentation AIR dataset at $aug_h5
92 |
93 | echo All done
94 |
--------------------------------------------------------------------------------
/Code/pythonsrc/run_ace_discriminative_nets.sh:
--------------------------------------------------------------------------------
1 | #!/bin/bash
2 |
3 | # Copyright 2018 Constantinos Papayiannis
4 | #
5 | # This file is part of Reverberation Learning Toolbox for Python.
6 | #
7 | # Reverberation Learning Toolbox for Python is free software: you can redistribute it and/or modify
8 | # it under the terms of the GNU General Public License as published by
9 | # the Free Software Foundation, either version 3 of the License, or
10 | # (at your option) any later version.
11 | #
12 | # Reverberation Learning Toolbox for Python is distributed in the hope that it will be useful,
13 | # but WITHOUT ANY WARRANTY; without even the implied warranty of
14 | # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
15 | # GNU General Public License for more details.
16 | #
17 | # You should have received a copy of the GNU General Public License
18 | # along with Reverberation Learning Toolbox for Python. If not, see .
19 |
20 | ##################################################################################################################
21 | ##################################################################################################################
22 | #
23 | # Description:
24 | # This script uses the code provided with this repository to run the experiments described in [1].
25 | # The code is able to train and evaluate DNNs for the task of 'Room Classification' using the data provided with
26 | # the ACE challenge database [2]. The following DNNs and configurations can be trained and evaluated:
27 | # Using AIRs only
28 | # 1) RNN
29 | # 2) CNN
30 | # 3) CRNN
31 | # 4) Att RNN
32 | # 5) Att-CRNN
33 | #
34 | # A copy of the necessary files from the ACE database is provided with this repo. To unpack it do
35 | # Code/Local_Databases/AIR$ tar zxf ACE16.tar.gz
36 | # The corpus was published under 'Creative Commons Attribution-NoDerivatives 4.0 International Public License' and in the package you can find a copy of the license
37 | #
38 | # Usage:
39 | # bash run_ace_discriminative_nets.sh
40 | #
41 | # : Location of ACE challenge data
42 | # : Must include a TRAIN and a TEST subdirectory, with the corresponding speech files included. The provided script wav_concatenator.sh can be used to create longer speech utterances, which are used in this experiment. The experiment will use 5s of speech per AIR and will assume that speech utterances are longer. It uses offsetting of longer utterances as a primitive data augmentation method. The script has been successfully trailed with the TIMIT database. It creates an concatenation of all the wav files in a directory, for each directory. Since TIMIT has one directory per speaker it created a long utterance per speaker, ideal for the task.
43 | # : Location of HFD5 dataset file for the ACE database, which is provided with this repository at Code/results_dir/ace_h5_info.h5. Contains information about the filenames, number of channels and also ground truth acoustic parameter values. If you want to create a new one, then use fe_utils.compile_ace_h5
44 | # : 0 or 1 to read any available caches
45 | # : List of indices between 1 to 8, see above
46 | #
47 | # Example:
48 | # bash run_ace_discriminative_nets.sh ../Local_Databases/AIR/ACE16 ../results_dir/concWavs/concWavs/Local_Databases/speech/TIMIT/TIMIT/ ../results_dir/ace_h5_info.h5 0 4 8
49 | #
50 | #
51 | # This file was original distributed in the repository at:
52 | # {repo}
53 | # If you use this code in your work, then cite [1].
54 | #
55 | # [1] C. Papayiannis, C. Evers and P. A. Naylor,
56 | # "End-to-End Classification of Reverberant Rooms Using DNNs,"
57 | # in IEEE/ACM Transactions on Audio, Speech, and Language Processing,
58 | # vol. 28, pp. 3010-3017, 2020, doi: 10.1109/TASLP.2020.3033628.
59 | # [2] http://www.ee.ic.ac.uk/naylor/ACEweb/index.html
60 | #
61 | #################################################################################################################
62 | #################################################################################################################
63 |
64 | set -e
65 |
66 | if [ "$#" -lt 5 ]; then
67 | echo 'Illegal number of parameters, expected >= 5.
68 | 1) ACE dir (/media/cp510/ExtraGiaWindows/db_exp_data/Local_Databases/AIR/ACE16)
69 | 2) speech dir (/home/cp510/GitHub/base_git_repo/Code/results_dir/concWavs)
70 | 3) h5 loc (/home/cp510/GitHub/base_git_repo/Code/results_dir/ace_h5_info.h5)
71 | 4) 0/1 whether cache should be read or not.
72 | 5) Modes to run 1-5
73 | Example:
74 | run_ace_discriminative_nets.sh ../Local_Databases/AIR/ACE16 ../results_dir/concWavs/ ../results_dir/ace_h5_info.h5 1 4 5 8
75 | '
76 | exit 1
77 | fi
78 |
79 | ace_dir=$1
80 | speech_dir=$2
81 | h5_loc=$3
82 | force_readcache=$4
83 | shift
84 | shift
85 | shift
86 | shift
87 | mode=$@
88 | base_scrap=/tmp/
89 | mkdir -p $base_scrap/ace_discriminative_nets_eval/
90 | logloc=$base_scrap/ace_discriminative_nets_eval/log.txt
91 |
92 | args_speech=(--ace "$ace_dir" --h5 "$h5_loc" --speech "$speech_dir/TRAIN" "$speech_dir/TEST" --cacheloc "$base_scrap" --split array)
93 |
94 | echo "Speech Args : ${args_speech[*]}"
95 |
96 | if [ $force_readcache == 1 ]; then
97 | readcache_arg='--readcache'
98 | else
99 | readcache_arg=''
100 | fi
101 |
102 | for this_mode in ${mode[@]}; do
103 |
104 | case $this_mode in
105 | 1) # RNN
106 | mode_args="--rnn"
107 | ;;
108 | 2) # CNN
109 | mode_args="--cnn"
110 | ;;
111 | 3) # CRNN
112 | mode_args="--rnn --cnn"
113 | ;;
114 | 4) # Att-RNN
115 | mode_args="--rnn --att"
116 | ;;
117 | 5) # Att-CRNN
118 | mode_args="--rnn --cnn --att"
119 | ;;
120 | *)
121 | echo Invalid mode $this_mode
122 | continue
123 | ;;
124 | esac
125 |
126 | cmd=(python -u ace_discriminative_nets.py ${mode_args[@]} ${args_speech[*]} $readcache_arg)
127 | echo Running ${cmd[@]}
128 | ${cmd[@]} | tee $logloc
129 |
130 | readcache_arg='--readcache'
131 | done
132 |
133 | echo Logs at $logloc
134 |
--------------------------------------------------------------------------------
/Code/pythonsrc/run_cnnrnn_net.sh:
--------------------------------------------------------------------------------
1 | #!/bin/bash
2 |
3 | # Copyright 2018 Constantinos Papayiannis
4 | #
5 | # This file is part of Reverberation Learning Toolbox for Python.
6 | #
7 | # Reverberation Learning Toolbox for Python is free software: you can redistribute it and/or modify
8 | # it under the terms of the GNU General Public License as published by
9 | # the Free Software Foundation, either version 3 of the License, or
10 | # (at your option) any later version.
11 | #
12 | # Reverberation Learning Toolbox for Python is distributed in the hope that it will be useful,
13 | # but WITHOUT ANY WARRANTY; without even the implied warranty of
14 | # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
15 | # GNU General Public License for more details.
16 | #
17 | # You should have received a copy of the GNU General Public License
18 | # along with Reverberation Learning Toolbox for Python. If not, see .
19 |
20 | ##################################################################################################################
21 | ##################################################################################################################
22 | #
23 | # Description:
24 | # This script uses the code provided with this repository to run the experiments described in [1].
25 | # The code is able to train and evaluate DNNs for the task of 'Room Classification' using the data provided with
26 | # the ACE challenge database [2]. The trained models are CNN-RNNs
27 | #
28 | # A copy of the necessary files from the ACE database is provided with this repo. To unpack it do
29 | # Code/Local_Databases/AIR$ tar zxf ACE16.tar.gz
30 | # The corpus was published under 'Creative Commons Attribution-NoDerivatives 4.0 International Public License' and in the package you can find a copy of the license
31 | #
32 | # Usage:
33 | # bash run_cnnrnn_net.sh.sh <>
34 | #
35 | # : Location of ACE challenge data
36 | # : Must include a TRAIN and a TEST subdirectory, with the corresponding speech files included. The provided script wav_concatenator.sh can be used to create longer speech utterances, which are used in this experiment. The experiment will use 5s of speech per AIR and will assume that speech utterances are longer. It uses offsetting of longer utterances as a primitive data augmentation method. The script has been successfully trailed with the TIMIT database. It creates an concatenation of all the wav files in a directory, for each directory. Since TIMIT has one directory per speaker it created a long utterance per speaker, ideal for the task.
37 | # : Location of HFD5 dataset file for the ACE database, which is provided with this repository at Code/results_dir/ace_h5_info.h5. Contains information about the filenames, number of channels and also ground truth acoustic parameter values. If you want to create a new one, then use fe_utils.compile_ace_h5
38 | # : 0 or 1 to read any available caches
39 | #
40 | # Example:
41 | # bash run_cnnrnn_net.sh.sh ../Local_Databases/AIR/ACE16 ../results_dir/concWavs/concWavs/Local_Databases/speech/TIMIT/TIMIT/ ../results_dir/ace_h5_info.h5 0
42 | #
43 | #
44 | # This file was original distributed in the repository at:
45 | # {repo}
46 | # If you use this code in your work, then cite [1].
47 | #
48 | # [1] C. Papayiannis, C. Evers and P. A. Naylor,
49 | # "End-to-End Classification of Reverberant Rooms Using DNNs,"
50 | # in IEEE/ACM Transactions on Audio, Speech, and Language Processing,
51 | # vol. 28, pp. 3010-3017, 2020, doi: 10.1109/TASLP.2020.3033628.
52 | # [2] http://www.ee.ic.ac.uk/naylor/ACEweb/index.html
53 | #
54 | #################################################################################################################
55 | #################################################################################################################
56 |
57 | set -e
58 |
59 | if [ "$#" -lt 5 ]; then
60 | echo 'Illegal number of parameters, expected >= 5.
61 | 1) ACE dir (/media/cp510/ExtraGiaWindows/db_exp_data/Local_Databases/AIR/ACE16)
62 | 2) speech dir (/home/cp510/GitHub/base_git_repo/Code/results_dir/concWavs)
63 | 3) h5 loc (/home/cp510/GitHub/base_git_repo/Code/results_dir/ace_h5_info.h5)
64 | 4) 0/1 whether cache should be read or not.
65 | 5+)Passed to ace_discriminative_nets.py
66 | Example:
67 | run_cnnrnn_net.sh.sh ../Local_Databases/AIR/ACE16 ../results_dir/concWavs/ ../results_dir/ace_h5_info.h5 1
68 | '
69 | exit 1
70 | fi
71 |
72 | python_loc=$HOME/anaconda2/bin/python
73 | $python_loc utils_base.py
74 |
75 | ace_dir=$1
76 | speech_dir=$2
77 | h5_loc=$3
78 | force_readcache=$4
79 | shift
80 | shift
81 | shift
82 | shift
83 | extras=$@
84 |
85 | base_scrap=/tmp/
86 | if [ `hostname | cut -d'-' -f1` == 'login' ]; then
87 | base_scrap=$WORK
88 | fi
89 |
90 | save_loc_base=$base_scrap/ace_discriminative_nets_eval/
91 |
92 | tmp_args=( ${extras[@]} )
93 | args_speech=( --utts 20 --experiment room --ace $ace_dir --h5 $h5_loc --speech $speech_dir/TRAIN $speech_dir/TEST --cacheloc $base_scrap ${tmp_args[@]})
94 | args_air=( --experiment room --ace $ace_dir --h5 $h5_loc --cacheloc $base_scrap ${tmp_args[@]})
95 | # Change this to speech if you want to do speech
96 | final_args=${args_air[@]}
97 |
98 | echo Args : ${args_final[*]}
99 |
100 |
101 | echo Running Speech CNN RNN
102 | this_saveloc=$save_loc_base/speech/speech_cnn_rnn
103 | mkdir -p $this_saveloc
104 | $python_loc -u ace_discriminative_nets.py --saveloc $this_saveloc ${final_args[*]} --cnn --rnn $readcache_arg | tee $this_saveloc/log.txt
105 |
106 |
107 |
108 |
--------------------------------------------------------------------------------
/Code/pythonsrc/utils_base.py:
--------------------------------------------------------------------------------
1 | # Copyright 2018 Constantinos Papayiannis
2 | #
3 | # This file is part of Reverberation Learning Toolbox for Python.
4 | #
5 | # Reverberation Learning Toolbox for Python is free software: you can redistribute it and/or modify
6 | # it under the terms of the GNU General Public License as published by
7 | # the Free Software Foundation, either version 3 of the License, or
8 | # (at your option) any later version.
9 | #
10 | # Reverberation Learning Toolbox for Python is distributed in the hope that it will be useful,
11 | # but WITHOUT ANY WARRANTY; without even the implied warranty of
12 | # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 | # GNU General Public License for more details.
14 | #
15 | # You should have received a copy of the GNU General Public License
16 | # along with Reverberation Learning Toolbox for Python. If not, see .
17 |
18 | """
19 | This file defines a set of basic functions to be used across a variety of applications
20 |
21 | This file was original distributed in the repository at:
22 | {repo}
23 |
24 | If you use this code in your work, then cite:
25 | C. Papayiannis, C. Evers and P. A. Naylor,
26 | "End-to-End Classification of Reverberant Rooms Using DNNs,"
27 | in IEEE/ACM Transactions on Audio, Speech, and Language Processing,
28 | vol. 28, pp. 3010-3017, 2020, doi: 10.1109/TASLP.2020.3033628.
29 |
30 | """
31 |
32 | import inspect
33 | import sys
34 | from os import walk
35 | from os.path import join
36 | from subprocess import Popen, PIPE
37 | from time import time
38 |
39 | import numpy as np
40 |
41 |
42 | def matrix_stats(x, flat=False):
43 | """
44 | Prints statistics for the data in a given matrix
45 |
46 | Args:
47 | x: The input data (np.array)
48 | flat: Flatten the data and provide global statistics instead of per column
49 |
50 | Returns:
51 | Nothing
52 |
53 | """
54 | args = {'axis': 0} if not flat else {}
55 | print_func = lambda xx: float2str(xx, num_decimals=5)
56 | print('For data with shape ' + str(x.shape))
57 | print('Mean : ' + print_func(np.mean(x, **args)))
58 | print('Median : ' + print_func(np.median(x, **args)))
59 | print('Min : ' + print_func(np.min(x, **args)))
60 | print('Max : ' + print_func(np.max(x, **args)))
61 | print('aMin : ' + print_func(np.min(np.abs(x), **args)))
62 | print('aMax : ' + print_func(np.max(np.abs(x), **args)))
63 | print('StD : ' + print_func(np.std(x, **args)))
64 |
65 |
66 | def isiter(x):
67 | """
68 | Checks if x is iterable
69 |
70 | Args:
71 | x: Check this
72 |
73 | Returns:
74 | Iterable or not
75 |
76 | """
77 | try:
78 | if x[0] > 0:
79 | pass
80 | except TypeError:
81 | return False
82 | except IndexError:
83 | pass
84 | return True
85 |
86 |
87 | def flatten_list(x):
88 | """
89 | Flattens lists of lists (of lists, of lists...)
90 |
91 | Args:
92 | x: The list of lists
93 |
94 | Returns:
95 | The flat list
96 |
97 | """
98 | if not isiter(x):
99 | raise TypeError('You did not given me something which makes sense')
100 | for i in range(len(x)):
101 | if isiter(x[i]):
102 | x[i] = flatten_list(x[i])
103 | else:
104 | x[i] = [x[i]]
105 |
106 | xx = np.concatenate([i for i in x]).tolist()
107 | return xx
108 |
109 |
110 | def add_axis_back(x, times=1, make_copy=False):
111 | """
112 | Add an axis to as the last dimension of a numpy array with dimensionality 1. useful for
113 | adding channels to DNN training data when they natively are not present in the data.
114 |
115 | Args:
116 | x: The original data
117 | times: Number of axis to add
118 | make_copy: Make a copy of the array before changing it
119 |
120 | Returns:
121 | The array
122 |
123 | """
124 | if make_copy:
125 | x = np.array(x)
126 | for i in range(times):
127 | x.shape = tuple(list(x.shape) + [1])
128 | return x
129 |
130 |
131 | def add_axis_front(x, times=1, make_copy=False):
132 | """
133 | Add an axis to as the first dimension of a numpy array with dimensionality 1. useful for
134 | adding channels to DNN training data when they natively are not present in the data.
135 |
136 | Args:
137 | x: The original data
138 | times: Number of axis to add
139 | make_copy: Make a copy of the array before changing it
140 |
141 | Returns:
142 | The new array
143 |
144 | """
145 | if make_copy:
146 | x = np.array(x)
147 | for i in range(times):
148 | x.shape = tuple([1] + list(x.shape))
149 | return x
150 |
151 |
152 | def repack_array_list(array_in, shapes=None, orientation='portrait'):
153 | """
154 | Pack an array into a list of vectors. The vectors can be either the rows or the columns.
155 | This operation is the opposite of flatten_array_list.
156 |
157 | Args:
158 | array_in: The array
159 | shapes: The original shapes of the vectors. This assumes that you had a list of vectors
160 | and each one had its own size. You had to put them into a 2D array so you made some of
161 | them longer or shorter. This list will contain the shapes of the original vectors.
162 | orientation: Setting this to 'portrait'means that you stuck the original arrays so that
163 | they are the columns of the array. Anything else means that they are the rows.
164 |
165 | Returns:
166 | The list of vectors
167 |
168 | """
169 | doing_landscape = not orientation == 'portrait'
170 | outlist = []
171 | array_in = np.atleast_2d(array_in)
172 | if shapes is None:
173 | if doing_landscape:
174 | shapes = (array_in.shape[1:], 1) * array_in.shape[0]
175 | else:
176 | shapes = (0, array_in.shape[1:]) * array_in.shape[0]
177 | row_counter = 0
178 | for i in shapes:
179 | if doing_landscape:
180 | if len(i) == 1:
181 | i = [i[0], 1]
182 | new_row_counter = row_counter + i[0]
183 | outlist.append(array_in[row_counter:new_row_counter, :][:, 0:i[1]])
184 | else:
185 | if len(i) == 1:
186 | i = [1, i[0]]
187 | new_row_counter = row_counter + i[1]
188 | outlist.append(array_in[:, row_counter:new_row_counter][0:i[0], :])
189 | return outlist
190 |
191 |
192 | def flatten_array_list(list_in, orientation='portrait'):
193 | """
194 | PAcks a list of vectors into an array. The vectors can be either the rows or the columns of
195 | the new array. This operation is the opposite of repack_array_list.
196 |
197 | Args:
198 | list_in: The lsit of vectors
199 | orientation: Setting this to 'portrait'means that you will stick the original arrays so
200 | that they are the columns of the array. Anything else means that they are the rows.
201 |
202 | Returns:
203 | The new array
204 | The shapes of the vectors in the given list
205 | """
206 |
207 | doing_landscape = not orientation == 'portrait'
208 | if type(list_in) is np.ndarray:
209 | out_mat = np.atleast_2d(list_in)
210 | return out_mat, (out_mat.shape,)
211 | if len(list_in) < 2:
212 | out_mat = np.atleast_2d(list_in)
213 | return out_mat, (out_mat.shape,)
214 | max_y = 0
215 | for i in list_in:
216 | max_y = max(max_y, np.atleast_2d(i).shape[1 - doing_landscape])
217 | if doing_landscape:
218 | out_array = np.zeros((max_y, np.sum([np.atleast_2d(i).shape[1] for i in list_in])))
219 | else:
220 | out_array = np.zeros((np.sum([np.atleast_2d(i).shape[0] for i in list_in]), max_y))
221 | counter = 0
222 | or_shapes = []
223 | for i in range(len(list_in)):
224 | or_shapes.append(np.array(list_in[i]).shape)
225 | twoddlist = np.atleast_2d(list_in[i])
226 | n_padding = max_y - twoddlist.shape[1 - doing_landscape]
227 | next_counter = counter + twoddlist.shape[0 + doing_landscape]
228 | if not doing_landscape:
229 | newis = np.concatenate(
230 | (twoddlist, np.zeros((twoddlist.shape[0], n_padding),
231 | dtype=list_in[i].dtype)), axis=1)
232 | else:
233 | newis = np.concatenate(
234 | (np.zeros((twoddlist.shape[0], n_padding),
235 | dtype=list_in[i].dtype), twoddlist), axis=1)
236 | if not doing_landscape:
237 | out_array[counter:next_counter, :] = newis
238 | else:
239 | out_array[:, counter:next_counter] = newis
240 | counter = next_counter
241 | return out_array, tuple(or_shapes)
242 |
243 |
244 | def get_git_hash():
245 | """
246 |
247 | Get the Git has of the current commit of the repo in this directory
248 |
249 | Returns:
250 | The hash
251 |
252 | """
253 | return run_command('git rev-parse HEAD')[0]
254 |
255 |
256 | def eprint(*args, **kwargs):
257 | """
258 | Prints to stderr
259 |
260 | Args:
261 | *args: Passed to print
262 | **kwargs: Passed to print
263 |
264 | Returns:
265 |
266 | """
267 | print(*args, file=sys.stderr, **kwargs)
268 |
269 |
270 | def run_command_list_stdout(command):
271 | """
272 | String to be run in bash
273 |
274 | Args:
275 | command: The command.
276 |
277 | Returns:
278 | The stdout as a list of strings. Each string element is a returned line
279 |
280 | """
281 | std_out = run_command(command)[0].decode()
282 | return std_out.rstrip().split("\n")
283 |
284 |
285 | def run_command(command):
286 | """
287 | String to be run in bash
288 |
289 | Args:
290 | command: The command.
291 |
292 | Returns:
293 | The stdout
294 |
295 | """
296 | proc = Popen(command.split(' '), stdout=PIPE, stderr=PIPE)
297 | std_out, std_err = [x.decode() for x in proc.communicate()]
298 | if len(std_err) > 0:
299 | print('stderr: ' + std_err)
300 | return std_out.rstrip(), std_err.rstrip()
301 |
302 |
303 | def join_strings(str_iter, delim=', '):
304 | """
305 | Joins elements of ant iterable as a string delimited by delim
306 |
307 | Args:
308 | str_iter: An iterable
309 | delim: The delimiter
310 |
311 | Returns:
312 | The concatenated string
313 |
314 | """
315 | out = ''
316 | for i in str_iter:
317 | out += delim + str(i)
318 | return out[len(delim):]
319 |
320 |
321 | def find_all_ft(directory_location, ft=".Wav", use_find=True, find_iname=False):
322 | """
323 | Finds all files of a specific filetype in a given set of directories (and subdirectories of
324 | them)
325 |
326 | Args:
327 | directory_location: The list of directories to look into
328 | ft: The extension of the file to look for
329 | use_find: Use the unix `find` command to do this
330 | find_iname: Use case insensitive search
331 |
332 | Returns:
333 | The list of files found
334 |
335 | """
336 | if isinstance(directory_location, str):
337 | directory_location = [directory_location]
338 | if use_find:
339 | name = 'name'
340 | if find_iname:
341 | name = 'iname'
342 | directories = ''
343 | for i in directory_location:
344 | directories += i + ' '
345 | directories = directories[0:-1]
346 | all_files = run_command(f'find -L {directories} -type f -{name} *{ft}')[0].rstrip().split("\n")
347 | else:
348 | if find_iname:
349 | raise AssertionError(
350 | 'You are expecting case insensitive searching but you are not using \'use_find\' '
351 | 'which allows this')
352 | print(f'Finding all {ft} in {directory_location}')
353 | all_files = []
354 | if not isinstance(directory_location, list):
355 | directory_location = [directory_location]
356 | for this_dir in directory_location:
357 | for root, dirs, files in walk(this_dir):
358 | for file in files:
359 | if file.endswith(ft):
360 | all_files.append(join(root, file))
361 | print(f'Found {len(all_files)} of {ft} in {directory_location}')
362 | return all_files
363 |
364 |
365 | def float2str(floatval, num_decimals=2):
366 | """
367 | Convert a float (or a numy array) to a string with a given precision
368 |
369 | Args:
370 | floatval: The valueof the float
371 | num_decimals: Decimal points to use
372 |
373 | Returns:
374 | The string
375 |
376 | """
377 | floatval = np.atleast_1d(np.array(floatval, dtype=float)).flatten()
378 | conv_rule = lambda x: (
379 | "{0:." + str(num_decimals if x >= 0 else num_decimals - 1) + "f}").format(x)
380 | if floatval.size == 0:
381 | return ""
382 | elif floatval.size == 1:
383 | return conv_rule(floatval[0])
384 | else:
385 | stris = ''
386 | for i in floatval:
387 | stris += ', ' + conv_rule(i)
388 | return stris[1:]
389 |
390 |
391 | def getfname():
392 | """
393 | Get the name of the calling function
394 |
395 | Returns:
396 | The name
397 |
398 | """
399 | curframe = inspect.currentframe()
400 | calframe = inspect.getouterframes(curframe, 2)
401 | return calframe[1][3]
402 |
403 |
404 | def matmax(alike):
405 | """
406 | Finds the maximum value and the index of it
407 |
408 | Args:
409 | alike: An iterable
410 |
411 | Returns:
412 | The maximum value
413 | The index of the maximum value
414 |
415 | """
416 | maxi = np.argmax(alike)
417 | maxv = alike[maxi]
418 | return [maxv, maxi]
419 |
420 |
421 | def matmin(alike):
422 | """
423 | Finds the minimum value and the index of it
424 |
425 | Args:
426 | alike: An iterable
427 |
428 | Returns:
429 | The minimum value
430 | The index of the minimum value
431 |
432 | """
433 | mini = np.argmin(alike)
434 | minv = alike[mini]
435 | return [minv, mini]
436 |
437 |
438 | def column_vector(alike):
439 | """
440 | Converts the input to a column vector (numpy array)
441 |
442 | Args:
443 | alike: Input
444 |
445 | Returns:
446 | The column vector
447 |
448 | """
449 | alike = np.atleast_1d(np.array(alike)).flatten()
450 | nelements = alike.size
451 | outmat = np.array(alike)
452 | outmat.shape = (nelements, 1)
453 | return outmat
454 |
455 |
456 | def row_vector(alike):
457 | """
458 | Converts the input to a row vector (numpy array)
459 |
460 | Args:
461 | alike: Input
462 |
463 | Returns:
464 | The row vector
465 |
466 | """
467 | npa = np.atleast_1d(np.array(alike)).flatten()
468 | nelements = npa.size
469 | npa.shape = (1, nelements)
470 | return npa
471 |
472 |
473 | def get_timestamp():
474 | """
475 | Generate a timestamp from the current time
476 |
477 | Returns:
478 | The timestamp as a string
479 |
480 | """
481 | timestamp = str(time())
482 | return timestamp
483 |
484 |
485 | def isclose(a, b, rel_tol=1e-09, abs_tol=0.0):
486 | """
487 | Check if floats are equal to a given precision
488 |
489 | Args:
490 | a: Float to compare
491 | b: Float to compare
492 | rel_tol: Relative tolerance
493 | abs_tol: Absolute tolerance
494 |
495 | Returns:
496 | Yes/no
497 |
498 | """
499 | res = np.abs(a - b) <= np.maximum(rel_tol * np.maximum(np.abs(a), np.abs(b)), abs_tol)
500 | return res
501 |
502 |
503 | if __name__ == '__main__':
504 | print(f'Your repo git hash: {get_git_hash()}')
505 |
--------------------------------------------------------------------------------
/Code/pythonsrc/utils_dnntrain.py:
--------------------------------------------------------------------------------
1 | # Copyright 2018 Constantinos Papayiannis
2 | #
3 | # This file is part of Reverberation Learning Toolbox for Python.
4 | #
5 | # Reverberation Learning Toolbox for Python is free software: you can redistribute it and/or modify
6 | # it under the terms of the GNU General Public License as published by
7 | # the Free Software Foundation, either version 3 of the License, or
8 | # (at your option) any later version.
9 | #
10 | # Reverberation Learning Toolbox for Python is distributed in the hope that it will be useful,
11 | # but WITHOUT ANY WARRANTY; without even the implied warranty of
12 | # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 | # GNU General Public License for more details.
14 | #
15 | # You should have received a copy of the GNU General Public License
16 | # along with Reverberation Learning Toolbox for Python. If not, see .
17 |
18 | """
19 |
20 | This file contains a number of routines useful in the training and evaluation of DNNs using Keras.
21 |
22 | This file was original distributed in the repository at:
23 | {repo}
24 |
25 | If you use this code in your work, then cite:
26 | C. Papayiannis, C. Evers and P. A. Naylor,
27 | "End-to-End Classification of Reverberant Rooms Using DNNs,"
28 | in IEEE/ACM Transactions on Audio, Speech, and Language Processing,
29 | vol. 28, pp. 3010-3017, 2020, doi: 10.1109/TASLP.2020.3033628.
30 |
31 | """
32 |
33 | from random import shuffle
34 | from subprocess import call
35 | from time import time
36 |
37 | import numpy as np
38 | from keras.callbacks import Callback
39 | from keras.callbacks import EarlyStopping, TensorBoard
40 | from keras.utils import plot_model
41 |
42 | from fe_utils import print_split_report
43 | from utils_base import float2str
44 |
45 |
46 | def multi_batch_gen(x_data_list, y_out_list, samples_per_class, y_to_balance=None, sub_idxs=None,
47 | **kwargs):
48 | """
49 | Batch generator for keras model training. Based on the function in this file 'batch_gen', this
50 | function allows for batch generators to be created which fuse together a number of datasets.
51 |
52 | In addition to the documentation of 'batch_gen', this function allows for a batch generator
53 | to be created which accepts a list arrays X, which contain training data and their
54 | corresponding labels Y. Then the batches will be constructed as if the data were part of only
55 | one dataset, using samples from all of them during hte batch construction. It is useful if
56 | the arrays are too big to be loaded onto RAM and they are stored in HDF5 datasets for instance.
57 |
58 | Args:
59 | x_data_list: List of X data as
60 | [[N_samples_1 X data_dimensionality],
61 | [N_samples_2 X data_dimensionality],...]
62 | y_out_list: List of labels or Y data as:
63 | [[N_samples_1 X out_data_dim],...
64 | [N_samples_2 X out_data_dim],...]
65 | samples_per_class: List of number of samples per class to take, per set
66 | y_to_balance: List of labels or Y data to be used to balance selections from each batch
67 | as:
68 | [[N_samples_1 X 1],...
69 | [N_samples_2 X 1],...]
70 | sub_idxs:
71 | **kwargs: List of lists of indices of samples to consider, indicating their locations in
72 | the X and Y array.
73 |
74 | Returns: The generator.
75 |
76 | """
77 | nsets = len(x_data_list)
78 | if y_to_balance is None:
79 | y_to_balance = [None] * nsets
80 | if sub_idxs is None:
81 | sub_idxs = [None] * nsets
82 |
83 | if not nsets == len(y_out_list):
84 | raise AssertionError('Invalid inputs')
85 | if not nsets == len(samples_per_class):
86 | raise AssertionError('Invalid inputs')
87 | if y_to_balance is None:
88 | y_to_balance = [None for _ in range(nsets)]
89 | if sub_idxs is None:
90 | sub_idxs = [None for _ in range(nsets)]
91 | if not nsets == len(samples_per_class):
92 | raise AssertionError('Invalid inputs')
93 |
94 | gens = []
95 | for i in range(nsets):
96 | gens.append(
97 | batch_gen(x_data_list[i], y_out_list[i], samples_per_class[i], sub_idxs=sub_idxs[i],
98 | y_to_balance=y_to_balance[i], **kwargs))
99 |
100 | while True:
101 | xs = [None for _ in range(nsets)]
102 | ys = [None for _ in range(nsets)]
103 | for i in range(nsets):
104 | xs[i], ys[i] = gens[i].next()
105 | yield np.concatenate(xs, axis=0), np.concatenate(ys, axis=0)
106 |
107 |
108 | def batch_gen(x_data, y_out, samples_per_class, y_to_balance=None, no_check=True, verbose=False,
109 | sub_idxs=None, augmentation_func=None, aug_prob=1.):
110 | """
111 | Batch generator for keras model training. Features:
112 | * It is capable of handling simultaneously multiple Inputs, useful for Functional API models.
113 | * Performs batch balancing in terms of the classes in Y. It accounts for class imbalance
114 | in the data. It can do that based on the give Y labels or balancing data can be given.
115 | * A smaller subset of the data can be made visible to batch generator and the rest will be
116 | ignore and not used for the training.
117 | * A data augmentation function can be given, which will modify a percentage of the batch
118 | samples. The percentage is configurable with an augmentation probability.
119 | * Able to directly work on HDF5 datasets.
120 |
121 | (Array refers to numpy array or HDF5 datasets)
122 | Args:
123 | x_data: Array of training input data as [N_samples X data_dimensionality], or a list of
124 | such arrays, which would correspond to the set of inputs to the network for Functional
125 | API nets with multiple inputs.
126 | y_out: Labels of the data, or y data for regression as an array
127 | [N_samples X out_data_dimensionality].
128 | samples_per_class: The number of samples to include in each batch from each class
129 | y_to_balance: An array of labels for the data as [N_samples X 1], used to balance the data.
130 | no_check: Skip any checks
131 | verbose: Verbose output
132 | sub_idxs: Indices of samples to consider, indicating their locations in the X and Y
133 | array. Passed as an iterable.
134 | augmentation_func: Function used to modify samples as a data augmentation strategy.
135 | aug_prob: Probability of a sample in the batch to be modified, using the augmentation_func.
136 |
137 | Note: If you do not want the balancing operation then pass y_to_balance=[0]*N_samples
138 | and samples_per_class=desired_batch_size
139 |
140 | Returns: The generator
141 |
142 | """
143 | turn_to_list = False
144 | if not isinstance(x_data, list) or isinstance(x_data, tuple):
145 | if isinstance(augmentation_func, list) or isinstance(augmentation_func, tuple):
146 | raise AssertionError('Unexpected condition')
147 | x_data = [x_data]
148 | turn_to_list = True
149 | nsets = len(x_data)
150 | if len(x_data) > 1:
151 | for i in range(1, nsets):
152 | if not x_data[i].shape[0] == x_data[i].shape[i]:
153 | raise AssertionError('Input error')
154 |
155 | if sub_idxs is None:
156 | sub_idxs = np.arange(0, x_data[0].shape[0]).astype(int)
157 | else:
158 | sub_idxs = np.sort(sub_idxs)
159 |
160 | if augmentation_func is None:
161 | augmentation_func = [lambda x: x for _ in range(nsets)]
162 | elif turn_to_list:
163 | augmentation_func = [augmentation_func]
164 | if not (isinstance(augmentation_func, list) or isinstance(augmentation_func, tuple)):
165 | raise AssertionError('Invalid Input')
166 | for i in range(nsets):
167 | if augmentation_func[i] is None:
168 | augmentation_func[i] = lambda x: x
169 |
170 | if not len(augmentation_func) == nsets:
171 | raise AssertionError('Input error')
172 |
173 | if y_to_balance is None:
174 | if y_out is None:
175 | raise AssertionError('Cannot work without any y\'s')
176 | y_to_balance = y_out
177 |
178 | index_pools = []
179 | for i in range(y_to_balance.shape[1]):
180 | index_pools.append(np.where(y_to_balance[sub_idxs, i])[-1].tolist())
181 | shuffle(index_pools[-1])
182 | else:
183 | samples_per_pool = samples_per_class
184 | counter = [0 for _ in range(len(index_pools))]
185 | while True:
186 | these_idxs = []
187 | for i in range(len(index_pools)):
188 | if len(index_pools[i]) > 0:
189 | subsamples = min(samples_per_pool, len(index_pools[i]))
190 | repeats = int(np.ceil(samples_per_pool / float(subsamples)))
191 | for _ in range(repeats):
192 | these_idxs += (index_pools[i][counter[i]:counter[i] + samples_per_pool])
193 | remove = subsamples * repeats - samples_per_pool
194 | if remove > 0:
195 | these_idxs = these_idxs[0:-remove]
196 | counter[i] += len(these_idxs)
197 | for i in range(len(index_pools)):
198 | if counter[i] >= len(index_pools[i]):
199 | if verbose:
200 | print('Reshuffling batch gen pool ' + str(i) + ' because i gave you ' +
201 | str(counter[i]) + ' samples already')
202 | shuffle(index_pools[i])
203 | counter[i] = 0
204 | if not no_check and (not len(these_idxs) == samples_per_pool * y_out.shape[1]):
205 | raise AssertionError('Generator failure')
206 | if verbose:
207 | print('New batch ready: ')
208 | print_split_report(y_to_balance[sub_idxs[these_idxs], ...])
209 | these_idxs = np.sort(these_idxs).tolist()
210 |
211 | if aug_prob == 0:
212 | if isinstance(x_data, list) or isinstance(x_data, tuple):
213 | out_x_aug = [x_data[k][sub_idxs[these_idxs], ...] for k in range(len(x_data))]
214 | else:
215 | out_x_aug = x_data[sub_idxs[these_idxs], ...]
216 | else:
217 | out_x_aug = [np.concatenate([
218 | augmentation_func[k](x_data[k][i:i + 1, ...])
219 | if np.random.rand() < aug_prob
220 | else
221 | x_data[k][i:i + 1, ...]
222 | for i in sub_idxs[these_idxs]
223 | ],
224 | axis=0) for k in range(nsets)]
225 | returner = lambda x: x
226 | if turn_to_list:
227 | returner = lambda x: x[0]
228 | if y_out is None:
229 | out_y = None
230 | elif isinstance(y_out, list) or isinstance(y_out, tuple):
231 | out_y = [y_out[k][sub_idxs[these_idxs], ...] for k in range(len(y_out))]
232 | else:
233 | out_y = y_out[sub_idxs[these_idxs], ...]
234 | yield returner(out_x_aug), out_y
235 |
236 |
237 | def model_trainer(the_batch_gen, in_shape, out_shape, get_model, val_patience=15,
238 | loss_patience=10, val_gen=None,
239 | tensorlog=False, callbacks=[], model_filename=None, epochs=1000,
240 | save_model_image=True, print_summary=True, steps_per_epoch=10,
241 | scratchpad='/tmp/', **kwargs):
242 | """
243 | A function which trains a Keras model for classification. It handles callbacks
244 | and puts together the training strategy, given a batch generator.
245 |
246 | Args:
247 | the_batch_gen: Batch generator, able to be used with Keras fit_generator
248 | in_shape: Input dimensionality for the model
249 | out_shape: Number of classes
250 | get_model: A function which constructs the model as
251 | get_model(input_shape, out_shape, **kwargs)
252 | and returns only a Keras Sequential model
253 | val_patience: Validation loss patience for Early Stopping.
254 | loss_patience: Training loss patience for Early Stopping.
255 | val_gen: Batch generator, able to be used with Keras fit_generator. Used for generating
256 | validation data.
257 | tensorlog: Location to save Tensorboard logs.
258 | callbacks: A list of callbacks to append to the network.
259 | model_filename: The filename for the model to be saved in.
260 | epochs: Number of training epochs
261 | save_model_image: Ask for the model diagram to be saved.
262 | print_summary: Print a summary of the model structure.
263 | steps_per_epoch: Number of generator batches to be used per epoch.
264 | scratchpad: Location for any data saving
265 | **kwargs: Passed to get_model
266 |
267 | Returns:
268 |
269 | """
270 |
271 | timestamp = str(time())
272 | if model_filename is None:
273 | call(["mkdir", "-p", scratchpad])
274 | model_filename = scratchpad + '/ace_model' + timestamp + '.h5'
275 | input_shape = in_shape
276 | model = get_model(input_shape, out_shape, **kwargs)
277 | if print_summary:
278 | model.summary()
279 |
280 | model.compile(loss='categorical_crossentropy', optimizer='adam')
281 |
282 | if loss_patience > 0:
283 | callbacks.append(
284 | EarlyStopping(
285 | monitor='loss', min_delta=0, patience=loss_patience, verbose=1,
286 | mode='auto'))
287 | if val_patience is not None:
288 | if val_patience > 0:
289 | callbacks.append(
290 | EarlyStopping(
291 | monitor='val_loss', min_delta=0, patience=val_patience, verbose=1,
292 | mode='auto'))
293 | if tensorlog:
294 | tensordir = model_filename.replace('.h5', '_tensorlog')
295 | callbacks.append(
296 | TensorBoard(log_dir=tensordir, histogram_freq=0,
297 | batch_size=16,
298 | write_graph=False, write_grads=False, write_images=True,
299 | embeddings_freq=0, embeddings_layer_names=None,
300 | embeddings_metadata=None))
301 | print('Will save Tensorboard Logs at : ' + tensordir)
302 | if save_model_image:
303 | imgdir = model_filename.replace('.h5', '.pdf')
304 | try:
305 | plot_model(model, to_file=imgdir, show_shapes=True)
306 | except (ImportError, ValueError):
307 | print('Could not save model image')
308 | print('Saved model image at: ' + imgdir)
309 |
310 | print('Training...')
311 | model.fit_generator(the_batch_gen, epochs=epochs,
312 | validation_data=val_gen, validation_steps=int(np.ceil(.15 * steps_per_epoch)),
313 | verbose=0, callbacks=callbacks, steps_per_epoch=steps_per_epoch, )
314 | for i in callbacks:
315 | if type(i) is PostEpochWorker:
316 | best_val_model = i.best_val_model
317 | if best_val_model is not None:
318 | model = best_val_model
319 | try:
320 | model.save(model_filename)
321 | print('Saved : ' + model_filename)
322 | except IOError as ME:
323 | print('Could not save model ' + model_filename + ' because ' + ME.message)
324 | return model, model_filename
325 |
326 |
327 | def accuracy_eval(x, y, cmodel, prefix=None):
328 | """
329 | Accepts input data, labels and a model to predict the labels, which are then evaluated in
330 | terms of their accuracy.
331 |
332 | Can be combined with PostEpochWorker, to provide an evaluation of the accuracy in a custom
333 | way during the training of DNNs as:
334 |
335 | PostEpochWorker(
336 | (x_out_train[val_idxs, :, :],
337 | x_out_test[test_idxs, :, :]),
338 | (y_train[val_idxs, :],
339 | y_test[test_idxs, :]),
340 | model_filename,
341 | eval_fun=(
342 | lambda x, y, cmodel: accuracy_eval(x, y, cmodel, prefix='Val'),
343 | lambda x, y, cmodel: accuracy_eval(x, y, cmodel, prefix='Test')),
344 | eval_every_n_epochs=100)
345 |
346 | Args:
347 | x: Input data
348 | y: Labels
349 | cmodel: Trained model for inference
350 | prefix: Prefix for the reporting
351 |
352 | Returns: The predictions
353 |
354 | """
355 | y_pred = np.argmax(cmodel.predict(x), axis=1).flatten()
356 | acc = np.sum(y_pred == np.argmax(y, axis=1)).flatten() / float(x.shape[0])
357 | print(((prefix + ' ') if prefix is not None else '') + 'Accuracy: ' + float2str(acc, 4))
358 | y_pred_out = np.zeros_like(y)
359 | for i in range(y_pred_out.shape[0]):
360 | y_pred_out[i, y_pred[i]] = True
361 | n_hots = np.sum(y_pred_out, axis=1)
362 | if ~np.all(n_hots == 1):
363 | too_hot = np.where(~(n_hots == 1))[-1]
364 | raise AssertionError(
365 | 'Predictions do not make sense because the following idxs had more than one hots ' +
366 | str(too_hot) + ' with the following hots ' + str(n_hots[too_hot]))
367 | return y_pred_out
368 |
369 |
370 | def get_scaler_descaler(x, verbose=False):
371 | """
372 | Creates scaling and descaling functions for preparation of training data and reconstruction
373 | from DNNs
374 |
375 | Args:
376 | x: Input data
377 | verbose: Verbose reporting
378 |
379 | Returns:
380 | Scaler function object
381 | Descaler function object
382 |
383 | """
384 | if x.ndim > 2:
385 | conced_x = np.concatenate(x, axis=0)
386 | else:
387 | conced_x = np.array(x)
388 | subval = np.min(conced_x, axis=0)
389 | scale_val = np.max(conced_x, axis=0) - subval
390 | scale_val[scale_val == 0] = 1
391 |
392 | subval.shape = tuple([1, 1] + list(subval.shape))
393 | scale_val.shape = tuple([1, 1] + list(scale_val.shape))
394 |
395 | if verbose:
396 | print('Will construct scaler with subs: ' + float2str(
397 | subval) + "\n" + '... and scalers ' + float2str(
398 | scale_val))
399 |
400 | def scaler(y):
401 | return (y - subval) / scale_val
402 |
403 | def descaler(y):
404 | return y * scale_val + subval
405 |
406 | return scaler, descaler
407 |
408 |
409 | class PostEpochWorker(Callback):
410 | def __init__(self, x_data, y_data, model_filename, eval_fun=None, eval_every_n_epochs=1,
411 | save_best=True):
412 | """
413 | Produces an instance of a keras Callback. It allows for running a set of routines when a
414 | number of training epochs has elapsed.
415 |
416 | Each routine accepts the X and Y data and the best performing model up to the current
417 | epoch. The best performing model is evaluated by the validation accuracy (or the train
418 | accuracy if no validation is done). It allows for you to evaluate you model during
419 | training and print performance reports of the model. it also allows you to have snapshots
420 | of your model during training.
421 |
422 | Args:
423 | x_data: The list of X data
424 | y_data: The list of Y data
425 | model_filename: The filename to use for saving the model snapshots
426 | eval_fun: The list of functions to run
427 | eval_every_n_epochs: The number of epochs after each to run the routines
428 | save_best: Set True to save the model snapshots
429 | """
430 | Callback.__init__(self)
431 | self.save_best = save_best
432 | self.eval_every_n_epochs = eval_every_n_epochs
433 | self.eval_fun = eval_fun
434 | if eval_fun is None:
435 | self.eval_fun = []
436 |
437 | elif not isinstance(eval_fun, list) and not isinstance(eval_fun, tuple):
438 | self.eval_fun = [eval_fun]
439 | else:
440 | self.eval_fun = eval_fun
441 | if not isinstance(x_data, list) and not isinstance(x_data, tuple):
442 | self.x_test = [x_data]
443 | else:
444 | self.x_test = x_data
445 | if not isinstance(y_data, list) and not isinstance(y_data, tuple):
446 | self.y_test = [y_data]
447 | else:
448 | self.y_test = y_data
449 | if not isinstance(self.y_test, type(self.x_test)) or not isinstance(self.eval_fun,
450 | type(self.x_test)):
451 | assert AssertionError('Given types for x_y or incorrect')
452 | self.best_val_loss = None
453 | self.model_filename = model_filename
454 | self.best_val_model = None
455 | self.update_since_last = False
456 |
457 | def on_train_begin(self, logs={}):
458 | self.losses = []
459 | if self.save_best:
460 | print('Will save best model as ' + self.model_filename)
461 |
462 | def run_eval(self, epoch, logs={}):
463 | if epoch is None:
464 | eval_go = True
465 | else:
466 | eval_go = ((epoch + 1) % self.eval_every_n_epochs == 0) or self.eval_every_n_epochs == 1
467 | if eval_go:
468 | if len(self.eval_fun) > 0:
469 | if self.update_since_last:
470 | print('Running eval at epoch ' + (
471 | str(epoch) if epoch is not None else '*last*') +
472 | ' ')
473 | for i in range(len(self.eval_fun)):
474 | if self.eval_every_n_epochs > 1:
475 | self.eval_fun[i](np.array(self.x_test[i]), np.array(self.y_test[i]),
476 | self.best_val_model)
477 | else:
478 | self.eval_fun[i](np.array(self.x_test[i]), np.array(self.y_test[i]),
479 | self.model)
480 | else:
481 | print(f'Skipping eval of epoch {epoch} since this is a dead season {" " * 25}', end='\r')
482 | self.update_since_last = False
483 |
484 | def on_epoch_end(self, epoch, logs={}):
485 |
486 | cval_loss = logs.get('val_loss')
487 | loss_name = 'val_loss'
488 | if cval_loss is None:
489 | loss_name = 'loss'
490 | cval_loss = logs.get('loss')
491 | if self.best_val_loss is None or cval_loss <= self.best_val_loss:
492 | self.update_since_last = True
493 | self.best_val_loss = cval_loss
494 | self.best_val_model = self.model
495 | if self.save_best:
496 | try:
497 | self.model.save(self.model_filename)
498 | except TypeError:
499 | print(f'Could not save model {self.model_filename}')
500 | print(f'At epoch : {epoch} found new best {loss_name} model with {loss_name} '
501 | f'{float2str(self.best_val_loss, 12)}{" " * 25}', end='\r')
502 |
503 | self.run_eval(epoch)
504 |
505 | def on_train_end(self, logs=None):
506 | print(' ' * 74, end='\r')
507 | self.run_eval(None)
508 |
--------------------------------------------------------------------------------
/Code/pythonsrc/utils_reverb.py:
--------------------------------------------------------------------------------
1 | # Copyright 2018 Constantinos Papayiannis
2 | #
3 | # This file is part of Reverberation Learning Toolbox for Python.
4 | #
5 | # Reverberation Learning Toolbox for Python is free software: you can redistribute it and/or modify
6 | # it under the terms of the GNU General Public License as published by
7 | # the Free Software Foundation, either version 3 of the License, or
8 | # (at your option) any later version.
9 | #
10 | # Reverberation Learning Toolbox for Python is distributed in the hope that it will be useful,
11 | # but WITHOUT ANY WARRANTY; without even the implied warranty of
12 | # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 | # GNU General Public License for more details.
14 | #
15 | # You should have received a copy of the GNU General Public License
16 | # along with Reverberation Learning Toolbox for Python. If not, see .
17 |
18 | """
19 |
20 | This is a collection of functions relevant to room acoustics and reverberation
21 |
22 | This file was original distributed in the repository at:
23 | {repo}
24 |
25 | If you use this code in your work, then cite:
26 | C. Papayiannis, C. Evers and P. A. Naylor,
27 | "End-to-End Classification of Reverberant Rooms Using DNNs,"
28 | in IEEE/ACM Transactions on Audio, Speech, and Language Processing,
29 | vol. 28, pp. 3010-3017, 2020, doi: 10.1109/TASLP.2020.3033628.
30 |
31 | """
32 | import numpy as np
33 | from scipy.optimize import curve_fit
34 |
35 | from utils_base import matmax, getfname, column_vector
36 | from utils_spaudio import enframe, overlapadd
37 | from utils_spaudio import get_array_energy
38 |
39 |
40 | def npm(h, hhat):
41 | """
42 |
43 | Estimates the Normalized Projection Misalignment (NPM) from [1]
44 |
45 | Args:
46 | h: AIR to compare
47 | hhat: AIR to compare
48 |
49 | Returns: The NPM
50 |
51 | [1] Morgan, D.R., Benesty, J. and Sondhi, M.M., 1998. On the evaluation of estimated impulse
52 | responses. IEEE Signal processing letters, 5(7), pp.174-176.
53 |
54 | """
55 |
56 | hhat = np.array(hhat)
57 | h = np.array(h)
58 |
59 | if hhat.ndim > 1 or h.ndim > 1:
60 | raise AssertionError('Expecting single channel responses')
61 |
62 | h = h.flatten()
63 | hhat = hhat.flatten()
64 |
65 | epsilon = np.sum(h * hhat) / (
66 | np.sqrt(np.sum(hhat * hhat) * np.sum(h * h))
67 | )
68 | npm_val = 1 - epsilon ** 2
69 |
70 | return npm_val
71 |
72 |
73 | def scale_with_absorption_coefs(x, fs, freqs, abs_coef, framesize=0.020, times=(1,)):
74 | """
75 |
76 | Filters a given input, based on the sound energy abosrption coefficients provided,
77 | in a frame-based mode
78 |
79 | Args:
80 | x: Audio signal
81 | fs: Sampling frequency
82 | freqs: Frequency points
83 | abs_coef: Sound energy absorption coefficient at given frquency point
84 | framesize: Framesize in samples
85 | times: Number of times to pass the signal through the filering
86 |
87 | Returns:
88 | The filtered signal
89 |
90 | """
91 |
92 | times = np.array(times).round().astype(int)
93 | abs_coef = np.atleast_2d(abs_coef)
94 |
95 | if np.sum(times) == 0:
96 | return x
97 |
98 | framelength = int(np.ceil(framesize * fs))
99 | window = np.hanning(framelength)
100 | original_length = x.size
101 | if x.size < framelength:
102 | missing = framelength - x.size
103 | x = np.concatenate((x.flatten(), np.zeros((missing,)).astype(x.dtype)))
104 | x = np.atleast_2d(x)
105 | else:
106 | x = enframe(x, framelength, int(np.ceil(framelength / 2)))
107 | if x.shape[0] > 1:
108 | for i in range(x.shape[0]):
109 | x[i, :] = x[i, :] * window
110 | xft = np.fft.rfft(x, axis=1)
111 | dreqs = np.arange(0, xft.shape[1], 1) / float(xft.shape[1]) * fs / 2.
112 |
113 | if not times.size == abs_coef.shape[0]:
114 | raise AssertionError('times for application should match filters')
115 |
116 | def get_scale(f, freqs_local, abs_coef_local):
117 | previous = np.where(f > freqs_local)[-1]
118 | if previous.size == 0:
119 | previous = 0
120 | else:
121 | previous = previous[-1]
122 | scale = 1
123 | for i, this_time in enumerate(times):
124 | if this_time > 0:
125 | if previous == freqs_local.size - 1:
126 | this_absorption = abs_coef_local[i, -1]
127 | else:
128 | this_absorption = abs_coef_local[i, previous] * (f - freqs_local[previous]) / (
129 | freqs_local[previous + 1] - freqs_local[previous]) + \
130 | abs_coef_local[i, previous + 1] * (
131 | -f + freqs_local[previous + 1]) / (
132 | freqs_local[previous + 1] - freqs_local[previous])
133 |
134 | scale *= np.sqrt(1 - this_absorption) ** this_time
135 | return scale
136 |
137 | freqs = np.array(freqs)
138 | scale = []
139 | for i in range(dreqs.size):
140 | scale.append(get_scale(dreqs[i], freqs, abs_coef))
141 | xft[:, i] = xft[:, i] * scale[-1]
142 |
143 | y = np.fft.irfft(xft, axis=1)
144 | if y.shape[0] > 1:
145 | y = overlapadd(y, inc=int(np.ceil(framelength / 2)))[0]
146 | else:
147 | y = y.flatten()
148 |
149 | y = y[0:original_length]
150 | return y
151 |
152 |
153 | def get_drr_linscale(air_fir_taps, sampling_freq, direct_window_length_secs=0.0008,
154 | ignore_reflections_up_to=None):
155 | """
156 | Estimates the Direct to Reverberant ration given an Acoustic Impulse Response (AIR)
157 |
158 | Args:
159 | air_fir_taps: The taps of the AIR
160 | sampling_freq: The sampling frequency
161 | direct_window_length_secs: The length of the window that is estimated ot contain the
162 | direct sound
163 | ignore_reflections_up_to: The reflections up to this point (in seconds) are ignored and
164 | not considered to be part of either the early or the late part.
165 |
166 | Returns: The DRR in linear scale
167 |
168 | """
169 | air_fir_taps = np.array(air_fir_taps)
170 | if air_fir_taps.ndim > 1:
171 | raise NameError(getfname() + "AIR_Not1D")
172 | nsidesamples = int(np.ceil(direct_window_length_secs / 2. * sampling_freq))
173 | dpathcenter = abs(air_fir_taps).argmax()
174 | dstartsample = max(dpathcenter, dpathcenter - nsidesamples)
175 | dendsample = min(dpathcenter + nsidesamples, air_fir_taps.size)
176 | if ignore_reflections_up_to is not None:
177 | ignore_until_sample = int(np.ceil(ignore_reflections_up_to * sampling_freq))
178 | if ignore_until_sample > dendsample:
179 | air_fir_taps[dendsample:ignore_until_sample] = 0
180 | else:
181 | print('You gave an ignore range for DRR calculation but it was invalid')
182 | return get_array_energy(air_fir_taps[dstartsample:dendsample]) \
183 | / get_array_energy(air_fir_taps[dendsample:])
184 |
185 |
186 | def get_t60_decaymodel(air_fir_taps, sampling_freq):
187 | """ Estimates the Reverberation Time, given an Acoustic Impulse Response.
188 |
189 | The mode of operation is defined by the reference below.
190 | This is a wrapper of a python translation of the code made available by
191 | the authors of: Karjalainen, Antsalo, and Peltonen,
192 | Estimation of Modal Decay Parameters from Noisy Response Measurements.
193 | at : http://www.acoustics.hut.fi/software/decay
194 |
195 | Examples:
196 | When involving an AIR that is sampled at 48 kHz for example.
197 |
198 | reverb_time = get_t60_decaymodel(air, 48000)
199 |
200 | Args:
201 | air_fir_taps : Acoustic Impulse Response to process
202 | sampling_freq : The sampling frequency at which air was recorded at
203 |
204 | Returns:
205 | An estimate of the reverbration time in seconds
206 |
207 | """
208 |
209 | def decay_model(x_points, param0, param1, param2):
210 | """ The function used bo the non-linear least squares fitting method to
211 | estimate the decay parameters"""
212 | expf = 0.4
213 | y1_dm = np.multiply(param0, np.exp(np.multiply(param1, x_points)))
214 | y2_dm = param2
215 | fit_res = np.multiply(weights, np.power(
216 | np.add(np.power(y1_dm, 2), np.power(y2_dm, 2)), 0.5 * expf))
217 | return fit_res
218 |
219 | # air is a 1D list
220 | # Set up things. Move to dB domain and scale
221 | leny = len(air_fir_taps)
222 | air = np.multiply(20, np.log10(abs(air_fir_taps) + np.finfo(float).eps))
223 | _, ymaxi = matmax(air)
224 | air = air - air[ymaxi]
225 | weights = [1] * leny
226 | weights[0:max(1, ymaxi)] = [0] * max(1, ymaxi)
227 | time_points = np.linspace(0, leny / float(sampling_freq), leny)
228 | # Lin fit
229 | leny2 = leny // 2
230 | leny10 = leny // 10
231 | ydata = np.power(np.power(10, air / 20.), 0.4)
232 | start_of_range = np.nonzero(weights)[0][0]
233 | meanval1 = np.mean(ydata[start_of_range:leny10 + start_of_range + 1])
234 | meanvaln = np.mean(ydata[leny - leny10:leny])
235 | tmat = np.concatenate((np.ones((leny2, 1)),
236 | column_vector(time_points[start_of_range:leny2 + start_of_range])),
237 | axis=1)
238 | tau0 = np.linalg.lstsq(tmat, air[start_of_range:leny2 + start_of_range], rcond=None)
239 | tau0 = tau0[0][1] / 8.7
240 | ydata = np.multiply(weights, np.array(ydata))
241 | fit_bounds = ([0, -2000, 0], [200., -0.1, 200.])
242 | if tau0 > -0.1: # to satisfy the bounds
243 | tau0 = -0.1
244 |
245 | sol_final = curve_fit(decay_model, time_points, ydata, p0=(meanval1, tau0, meanvaln),
246 | bounds=fit_bounds)
247 |
248 | reverb_time = np.log(1 / 1000.) / float(sol_final[0][1])
249 | if reverb_time <= 0:
250 | raise NameError(getfname() + ':NegativeRT')
251 |
252 | return reverb_time
253 |
254 |
255 | def get_edc(air_fir_taps):
256 | """
257 | Calculates and returns the Energy Decay Curve (EDC) (Schroeder Integral) of the supplied
258 | AIR.
259 |
260 | Args:
261 | air_fir_taps: Taps of FIR filter representation of AIR
262 |
263 | Returns:
264 | The EDC of the supplied AIR as a numpy.array
265 |
266 | Examples:
267 | edcurve = get_edc(air_fir_taps)
268 |
269 | """
270 | edcurve = np.flip(np.square(np.cumsum(np.array(air_fir_taps[::-1])) ** 2).flatten(), 0)
271 | return edcurve
272 |
273 |
274 | def air_up_to_db(air_fir_taps, up_to_db):
275 | """
276 |
277 | Args:
278 | air_fir_taps: Input AIR
279 | up_to_db: The energy cutoff point in dB
280 |
281 | Returns: The AIR truncated from tap 0 to tap N, the point at which the energy remaining is
282 | less than 'up_to_db' of the energy of the entire AIR
283 | than up_to_db of the
284 |
285 | """
286 | up_to = 10 ** (up_to_db / 10.)
287 | air_edc = get_edc(air_fir_taps)
288 | cutoff_sample = np.where(air_edc < up_to)[0]
289 | if cutoff_sample.size > 0:
290 | cutoff_sample = min(air_fir_taps.size, np.where(air_edc < up_to)[0][0])
291 | else:
292 | cutoff_sample = air_fir_taps.size
293 |
294 | return air_fir_taps[0:cutoff_sample]
295 |
--------------------------------------------------------------------------------
/Code/pythonsrc/utils_spaudio.py:
--------------------------------------------------------------------------------
1 | # Copyright 2018 Constantinos Papayiannis
2 | #
3 | # This file is part of Reverberation Learning Toolbox for Python.
4 | #
5 | # Reverberation Learning Toolbox for Python is free software: you can redistribute it and/or modify
6 | # it under the terms of the GNU General Public License as published by
7 | # the Free Software Foundation, either version 3 of the License, or
8 | # (at your option) any later version.
9 | #
10 | # Reverberation Learning Toolbox for Python is distributed in the hope that it will be useful,
11 | # but WITHOUT ANY WARRANTY; without even the implied warranty of
12 | # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 | # GNU General Public License for more details.
14 | #
15 | # You should have received a copy of the GNU General Public License
16 | # along with Reverberation Learning Toolbox for Python. If not, see .
17 |
18 | """
19 |
20 | This is a collection of functions relevant to speech and audio processing.
21 |
22 | This file was original distributed in the repository at:
23 | {repo}
24 |
25 | If you use this code in your work, then cite:
26 | C. Papayiannis, C. Evers and P. A. Naylor,
27 | "End-to-End Classification of Reverberant Rooms Using DNNs,"
28 | in IEEE/ACM Transactions on Audio, Speech, and Language Processing,
29 | vol. 28, pp. 3010-3017, 2020, doi: 10.1109/TASLP.2020.3033628.
30 |
31 | """
32 | import numpy as np
33 | from scipy.signal import lfilter
34 |
35 | from utils_base import getfname, column_vector, row_vector
36 |
37 | resample_eng = None
38 |
39 | from utils_base import repack_array_list
40 |
41 | import matplotlib.pyplot as mplot
42 |
43 | from utils_base import flatten_array_list
44 | from scipy.io import wavfile
45 | from scipy.signal import welch
46 |
47 | try:
48 | import matlab
49 | import matlab.engine
50 | except ImportError:
51 | print('Could not import matlab libraries')
52 |
53 |
54 | def my_resample(x, fs_old, fs_new, matlab_eng=None, verbose=False,
55 | close_after=False):
56 | """
57 |
58 | Uses the Matlab engine to resample audio files. Produces much better results than other
59 | alternatives.
60 |
61 | If you do not have matlab, then you can :
62 | bash conda install -c conda-forge resampy
63 | then replace this function with
64 | def my_resample(x, fs_old, fs_new, matlab_eng=None, verbose=False,
65 | close_after=False):
66 | from resampy import resample
67 | return resample(x, fs_old, fs_new)
68 | You can use any other library other than resampy of course.
69 |
70 | Args:
71 | x: Signal to resample
72 | fs_old: Old sampling rate
73 | fs_new: New sampling rate
74 | matlab_eng: Matlab engine object if pre-initialized
75 | verbose: Verbose reporting
76 | close_after: Close the matlab engine when done
77 |
78 | Returns: The resampled audio signal
79 |
80 | """
81 | global resample_eng
82 | was_int16 = False
83 |
84 | if fs_old == fs_new:
85 | return x
86 | if x.dtype == 'int16':
87 | was_int16 = True
88 | x = (x / np.iinfo('int16').max).astype('float')
89 | if verbose:
90 | print('Your input for resampling is int16, will go to float for '
91 | 'calculations then put it back to int16')
92 |
93 | if resample_eng is not None:
94 | eng = resample_eng
95 | print('Using static engine')
96 | elif matlab_eng is not None:
97 | print('Using provided Matlab engine')
98 | eng = matlab_eng
99 | my_resample.ext_eng = eng
100 | else:
101 | print('Creating Matlab engine')
102 | eng = matlab.engine.start_matlab()
103 | resample_eng = eng
104 |
105 | x, shapes = flatten_array_list(x, orientation='landscape')
106 |
107 | print('Resampling ' + str(x.shape) + ' from ' + str(fs_old) + ' to ' + str(fs_new))
108 |
109 | (up, down) = (float(fs_new) / fs_old).as_integer_ratio()
110 | x_tmp = matlab.double(x.tolist())
111 | x_out = eng.resample(x_tmp, matlab.double([up]), matlab.double([down]), nargout=1)
112 |
113 | if close_after:
114 | eng.exit()
115 | resample_eng = None
116 |
117 | x_out = np.array(x_out)
118 | print('Got ' + str(x_out.shape) + ' output samples')
119 | x_out = repack_array_list(x_out, shapes=shapes, orientation='landscape')
120 | if len(x_out) == 1:
121 | x_out = x_out[0]
122 |
123 | if was_int16:
124 | if type(x_out) is list:
125 | for i in range(len(x_out)):
126 | x_out[i] = (x_out[i].astype('float64') * np.iinfo('int16').max).astype('int16')
127 | else:
128 | x_out = (x_out.astype('float64') * np.iinfo('int16').max).astype('int16')
129 |
130 | return x_out
131 |
132 |
133 | def get_psd(x, fs, window_seconds, ):
134 | """
135 |
136 | Estimates the Power Spectral Density and Power spectrum of a given signal, using :
137 | P. Welch, "The use of the fast Fourier transform for the
138 | estimation of power spectra: A method based on time averaging
139 | over short, modified periodograms", IEEE Trans. Audio
140 | Electroacoust. vol. 15, pp. 70-73, 1967.
141 |
142 | Args:
143 | x: Signal
144 | fs: Sampling frequency
145 | window_seconds: Window length in seconds
146 |
147 | Returns: frequency_points, PSD, Power Spectrum
148 |
149 | """
150 |
151 | nperseg = int(np.ceil(fs * window_seconds))
152 | _, psd = welch(x, fs=fs, window='hamming', nperseg=nperseg, noverlap=None, nfft=None,
153 | detrend='constant', return_onesided=True, scaling='density', axis=-1)
154 | f, pspec = welch(x, fs=fs, window='hamming', nperseg=nperseg, noverlap=None, nfft=None,
155 | detrend='constant', return_onesided=True, scaling='spectrum', axis=-1)
156 |
157 | return f, psd, pspec
158 |
159 |
160 | def distitpf(pf1, pf2, mode='0'):
161 | """
162 |
163 | Adaptation of the Itakura distance estimation method from Voicebox[1]
164 |
165 | Args:
166 | pf1: Power spectrum to compare
167 | pf2: Power spectrum to compare
168 | mode: Character string selecting the following options:
169 | 'x' Calculate the full distance matrix from every row of PF1 to every row of PF2
170 | 'd' Calculate only the distance between corresponding rows of PF1 and PF2
171 | The default is 'd' if PF1 and PF2 have the same number of rows otherwise 'x'.
172 |
173 | Returns: Itakura distance
174 |
175 | [1] http://www.ee.ic.ac.uk/hp/staff/dmb/voicebox/doc/voicebox/distitpf.html
176 |
177 | """
178 | #
179 | pf1 = np.atleast_2d(pf1)
180 | pf2 = np.atleast_2d(pf2)
181 | (nf1, p2) = pf1.shape
182 | p1 = p2 - 1
183 | nf2 = pf2.shape[0]
184 |
185 | if mode == 'd' or (not mode == 'x' and nf1 == nf2):
186 | nx = min(nf1, nf2);
187 | r = pf1[0:nx, :] / pf2[0:nx, :]
188 | q = np.log(r);
189 | d = (np.log((np.sum(r[:, 1:p1], 1) + 0.5 * (r[:, 0] + r[:, p2 - 1])) / p1) -
190 | (np.sum(q[:, 1:p1], 1) + 0.5 * (q[:, 0] + q[:, p2 - 1])) / p1)[0]
191 | else:
192 | r = np.transpose(pf1[:, :, np.ones((nf2,), dtype=int)], axes=[0, 2, 1]) / \
193 | np.transpose(pf2[:, :, np.ones((nf1,), dtype=int)], axes=[2, 0, 1])
194 | q = np.log(r)
195 | d = np.log((np.sum(r[:, :, 1:p1], 2) + 0.5 * (r[:, :, 0] + r[:, :, p2 - 1])) / p1) - \
196 | (np.sum(q[:, :, 1:p1], 2) + 0.5 * (q[:, :, 0] + q[:, :, p2 - 1])) / p1
197 | return d
198 |
199 |
200 | def write_wav(filename, fs, ss):
201 | """
202 |
203 | Writes audio samples as wav files to disk.
204 |
205 | Args:
206 | filename: Name to save file as
207 | fs: Sampling frequency
208 | ss: Audio samples as a numpy array
209 |
210 | Returns: Nothing
211 |
212 | """
213 | wavfile.write(filename, fs, (ss.astype('float64') / abs(ss).max() * np.iinfo('int16').max
214 | ).astype('int16'))
215 | print('Wrote : ' + filename)
216 |
217 |
218 | def ar_to_cepstrum(ar_coef, cep_order=None):
219 | """
220 |
221 | Converts Autoregressive (AR) coeficients to cepstral coefficients using the method discussed
222 | in [1]
223 |
224 | Args:
225 | ar_coef: AR coefficients
226 | cep_order: Order up to which to estimate cepstral coefficients
227 |
228 | Returns:
229 | The cepstral coefficients
230 |
231 | [1] K. Kalpakis, D. Gada, and V. Puttagunta, 'Distance measures for effective clustering of
232 | ARIMA time-series,' in ICDM, San Jose, California, USA, 2001, pp. 273-280.
233 |
234 | """
235 | back_to_flat = False
236 | if ar_coef.ndim == 1:
237 | back_to_flat = True
238 | ar_coef = np.atleast_2d(ar_coef)
239 | cep_order = ar_coef.shape[1] if cep_order is None else cep_order
240 | cep_coef = np.zeros((ar_coef.shape[0], cep_order), dtype=ar_coef.dtype)
241 | if not np.all(cep_coef[:, 0] == 1):
242 | raise AssertionError('Expected first AR coef ot be 1')
243 | for i in range(cep_coef.shape[0]):
244 | for k in range(cep_order):
245 | if k == 1:
246 | cep_coef[i, k] = -ar_coef[i, k]
247 | elif k <= cep_coef.shape[1]:
248 | cep_coef[i, k] = -ar_coef[i, k]
249 | for m in range(k):
250 | cep_coef[i, k] = cep_coef[i, k] - (1 - (m + 1) / k) * ar_coef[i, m] * cep_coef[
251 | i, k - m]
252 | else:
253 | cep_coef[i, k] = 0
254 | for m in range(k):
255 | cep_coef[i, k] = cep_coef[i, k] - (1 - (m + 1) / k) * ar_coef[i, m] * cep_coef[
256 | i, k - m]
257 | if back_to_flat:
258 | cep_coef.flatten()
259 |
260 | return cep_coef
261 |
262 |
263 | def align_max_samples(yin, scan_range=None):
264 | """
265 |
266 | Aligns input signals so that the maximum energy samples are at the same index.
267 |
268 | Args:
269 | yin: List of input singals
270 | scan_range: Range of indicesto scan for in order to find the maximum energy sample
271 |
272 | Returns:
273 | List of aligned signals
274 | List of delay introduces in each signal during alignment
275 |
276 | """
277 |
278 | yin = flatten_array_list(yin)[0]
279 | if yin.ndim == 1:
280 | raise ValueError('Expected 2D array as input')
281 |
282 | if yin.shape[0] == 1:
283 | yout = yin
284 | delays = np.array([0]).astype(float)
285 | return yout, delays
286 |
287 | if scan_range is None:
288 | scan_range = range(yin.shape[1])
289 | delays = abs(yin[:, scan_range]).argmax(axis=1)
290 |
291 | delays = abs(delays - delays.max())
292 | padding = delays.max()
293 |
294 | yout = np.concatenate(
295 | (np.zeros_like(yin),
296 | np.zeros((yin.shape[0], padding), dtype=yin.dtype)),
297 | axis=1).astype(yin.dtype)
298 |
299 | for idx, this_delay in enumerate(delays):
300 | tmp = np.zeros_like(yout[idx, :])
301 | tmp[this_delay:this_delay + yin.shape[1]] = yin[idx, :]
302 | yout[idx, :] = tmp
303 | if padding > 0:
304 | yout = yout[:, 0:-padding]
305 |
306 | return yout, delays
307 |
308 |
309 | def scale_x_to_y(x, y):
310 | """
311 |
312 | Estimates a scaling for signal X, in order to Least Squares match the amplitude of samples in
313 | X and Y
314 |
315 | Args:
316 | x: X
317 | y: Y
318 |
319 | Returns:
320 | Scale (scalar)
321 |
322 | """
323 | scale = np.linalg.lstsq(np.atleast_2d(x).T, np.atleast_2d(y).T, rcond=None)[0][0][0]
324 | return scale
325 |
326 |
327 | def fractional_alignment(yin, resolution=0.01, ls_scale=False, take_base_as=0):
328 | """
329 |
330 | Fractionally aligns signals (between (-1,+1) sample shifts) based on a least squares method
331 | of the mismatch of the samples, operating on a fixed resolution grid.
332 |
333 | Args:
334 | yin: List of input signals
335 | resolution: Resolution of search grid for fractional alignment
336 | ls_scale: Enable the scaling of the signals to a least squares mach of their samples in
337 | addition to the delaying
338 | take_base_as: Signal to take as the reference signal in the matching process. The default
339 | is to take the first signal in the list. This signal will remain unchanged and the rest
340 | will be matched to it.
341 |
342 | Returns:
343 | List of aligned signals
344 | List of delay introduces in each signal during alignment
345 | List of scale introduces in each signal
346 |
347 | """
348 |
349 | def get_mse(frac_kernels, idx, x, valid_range, y):
350 | x = np.convolve(x, frac_kernels[:, idx])
351 | x = x[valid_range]
352 | if ls_scale:
353 | scale = scale_x_to_y(x, y)
354 | else:
355 | scale = 1.
356 | mse = np.sum((y - x * scale) ** 2)
357 | return mse, scale
358 |
359 | if yin.ndim == 1:
360 | raise ValueError('Expected 2D array as input')
361 |
362 | resolution = float(resolution)
363 | npoints = int(np.ceil(1. / resolution))
364 | yin = np.array(yin)
365 | if yin.shape[0] == 1:
366 | yout = yin
367 | delays = np.array([0]).astype(float)
368 | return yout, delays
369 |
370 | context = 4
371 | valid_range = np.arange(context, context + yin.shape[1], 1).astype(int)
372 | _, frac_kernels = gm_frac_delayed_copies(np.ones((npoints,)),
373 | np.arange(context, context + 1, resolution) - .5,
374 | 2 * context + 1)
375 | yout = np.zeros_like(yin)
376 | yout[take_base_as, :] = yin[take_base_as, :]
377 | delays = np.zeros((yin.shape[0],))
378 | scales = np.zeros((yin.shape[0],))
379 | scales[take_base_as] = 1.
380 | all_delays = np.arange(0, 1, resolution) - .5
381 | for i in range(yin.shape[0]):
382 | if take_base_as == i:
383 | continue
384 | mse_scales = np.array(
385 | [get_mse(frac_kernels, idx, yin[i, :], valid_range, yin[take_base_as, :])
386 | for idx in range(0, npoints)])
387 | min_point = mse_scales[:, 0].argmin()
388 | delays[i] = all_delays[min_point]
389 | scales[i] = mse_scales[min_point, 1]
390 | thisyout = scales[i] * np.convolve(yin[i, :], frac_kernels[:, min_point])
391 | yout[i, :] = thisyout[valid_range]
392 |
393 | return yout, delays, scales
394 |
395 |
396 | def gm_frac_delayed_copies(amplitudes, delays, tot_length, excitation_signal=np.array([]),
397 | center_filter_peaks=True):
398 | """
399 |
400 | The function when called implements the model defined as:
401 | (1) h(n) = \sum_{i=1}^{D}\left[{\beta_i}h_e(n){\ast}\frac{\sin~\pi(n-k_i)}{\pi(n-k_i)}\right]
402 | This is the model proposed in [1]
403 |
404 | This is effectively the summation on D copies of the signal excitation_signal.
405 | This copies are placed at sample locations 'delays' (which are not bound to integers) and
406 | their scaling is defined by 'amplitudes'. The length of the signal is defined as 'tot_length'.
407 |
408 | Args:
409 | amplitudes: Vector containing the scaling of each copy
410 | delays: Sample index of occurence each copy
411 | tot_length: Total length of the output vector
412 | excitation_signal: The signal to be copied at each location
413 | center_filter_peaks: Center the signal in 'excitation_signal', so that the samples of
414 | maximum energy occur at samples 'delays'
415 |
416 | Returns:
417 | y : h(n) from the equation (1)
418 | Y : A matrix containing as vectors the components of the summation in equation (1)
419 |
420 | [1] Papayiannis, C., Evers, C. and Naylor, P.A., 2017, August. Sparse parametric modeling of
421 | the early part of acoustic impulse responses. In Signal Processing Conference (EUSIPCO),
422 | 2017 25th European (pp. 678-682). IEEE.
423 |
424 | """
425 | excitation_signal = np.atleast_2d(excitation_signal)
426 | if excitation_signal.shape[1] == 1:
427 | excitation_signal = excitation_signal.T
428 |
429 | if excitation_signal.size > 0:
430 | nfilters = excitation_signal.shape[0]
431 | else:
432 | nfilters = 0
433 | sincspan = 9
434 |
435 | amplitudes = np.atleast_2d(amplitudes)
436 | if amplitudes.shape[1] == 1:
437 | amplitudes = amplitudes.T
438 | delays = np.atleast_2d(delays)
439 | if delays.shape[1] == 1:
440 | delays = delays.T
441 |
442 | tot_components = amplitudes.size
443 | if delays.size != tot_components:
444 | raise NameError(getfname() + ':InputsMissmatch tot components are ' + str(
445 | tot_components) + ' and got delays ' + str(delays.size))
446 | if nfilters > 1:
447 | raise NameError(getfname() + ':InputsMissmatch')
448 | if tot_components < 1:
449 | yindiv = np.array([])
450 | ysignal = np.zeros((tot_length,))
451 | return [ysignal, yindiv]
452 | sample_indices = np.repeat(column_vector(np.arange(0, tot_length, 1, dtype=np.float64)),
453 | tot_components, axis=1)
454 | sample_indices_offsetting = np.repeat(row_vector(delays), tot_length, axis=0)
455 | sample_indices -= sample_indices_offsetting
456 | yindiv = np.sinc(sample_indices) * np.repeat(row_vector(amplitudes), tot_length, axis=0)
457 | if ~np.isinf(sincspan):
458 | spanscale_numer = sincspan * np.sin(np.pi * sample_indices / float(sincspan))
459 | spanscale_denom = (np.pi * sample_indices) # Lanczos kernel
460 | limiting_case_idx = np.where(spanscale_denom == 0)
461 | spanscale_denom[limiting_case_idx] = 1
462 | spanscale = spanscale_numer / spanscale_denom
463 | spanscale[limiting_case_idx] = 1
464 | yindiv *= spanscale
465 | yindiv[np.where(abs(sample_indices > sincspan))] = 0
466 |
467 | excitation_signal = excitation_signal.flatten()
468 | if nfilters > 0:
469 | if not center_filter_peaks:
470 | yindiv = lfilter(excitation_signal, [1], yindiv, axis=0)
471 | else:
472 | fcenter = np.argmax(abs(excitation_signal), axis=0)
473 | if fcenter == 0:
474 | yindiv = lfilter(excitation_signal, [1], yindiv, axis=0)
475 | else:
476 | futuresamples = fcenter
477 | tmpconc = np.concatenate((yindiv, np.zeros((futuresamples, yindiv.shape[1]))),
478 | axis=0)
479 | tmpconc = lfilter(excitation_signal, [1], tmpconc, axis=0)
480 | yindiv = tmpconc[futuresamples:, :]
481 |
482 | ysignal = np.sum(yindiv, axis=1)
483 | return [ysignal, yindiv]
484 |
485 |
486 | def enframe(alike, flength, fincr, hamming_window=False):
487 | """
488 | Breaks the input into frames of length flength, with an increment of fincr samples per frame
489 | Args:
490 | alike: Input array like description of a vector
491 | flength: Frame Length in samples
492 | fincr: Frame Increment in Samples
493 | hamming_window: Apply hamming window on frames
494 |
495 | Returns: The signal broken into frames
496 |
497 | """
498 | npa = np.array(alike)
499 | flength = int(flength)
500 | fincr = int(fincr)
501 | if npa.ndim > 1:
502 | raise NameError(getfname() + ':Non1DInput')
503 | if not (flength % fincr) == 0:
504 | raise NameError(getfname() + ':SizeOfFrameNotMultipleOfIncrement')
505 | nshifts = int(flength / float(fincr))
506 | totnframes = int(np.ceil(npa.size / float(fincr)))
507 | noframes_per_shift = int(np.ceil(totnframes / float(nshifts)))
508 | discardlastframes = (totnframes % nshifts) > 0
509 | xout = np.zeros((noframes_per_shift * nshifts, flength))
510 | tnsamples = int(noframes_per_shift * flength)
511 | fidxs = np.arange(0, noframes_per_shift, dtype=np.int) * nshifts
512 | npadding = (flength - npa.size % flength) % flength + flength
513 | npa = np.append(npa, np.zeros(npadding))
514 | for i in range(nshifts):
515 | fidxs += i
516 | xout[fidxs, :] = np.array(npa[i * fincr:i * fincr + tnsamples]).reshape(fidxs.size, flength)
517 | if hamming_window:
518 | xout[fidxs, :] = xout[fidxs, :] * np.hamming(flength)
519 | if discardlastframes:
520 | xout = xout[0:-1, :]
521 | return xout
522 |
523 |
524 | def overlapadd(input_frames, window_samples=None, inc=None,
525 | previous_partial_output=None):
526 | """
527 | Performs overlap-add using the frames in input_samples. This is a Python implementation of
528 | the MATLAB code available in the VOICEBOX toolbox.
529 | (http://www.ee.ic.ac.uk/hp/staff/dmb/voicebox/doc/voicebox/overlapadd.html)
530 |
531 | Args:
532 | input_frames: The array of input frames of size M X window_samples
533 | window_samples: The window to be used for the frames
534 | inc: The increment in samples between frames
535 | previous_partial_output: Provide the partial output returned from a previous call to
536 | this function
537 |
538 | Returns:
539 | The overlap-add result and the partial output at the end
540 |
541 | """
542 |
543 | if window_samples is None:
544 | window_samples = np.ones((input_frames.shape[1],))
545 | elif window_samples.size != input_frames.shape[1]:
546 | raise NameError(getfname() + ":WindowSizeDoesNotMatchFrameSize")
547 | if inc is None:
548 | inc = input_frames.shape[1]
549 | elif inc > input_frames.shape[1]:
550 | raise NameError(getfname() + ":SampleIncrementTooLarge")
551 | nr = input_frames.shape[0]
552 | nf = input_frames.shape[1]
553 |
554 | nb = int(np.ceil(nf / float(inc)))
555 | no = int(nf + (nr - 1) * inc)
556 | overlapped_output_shape = (no, nb)
557 |
558 | z = np.zeros((int(no * nb),))
559 | # input_frames = np.asfortranarray(input_frames)
560 |
561 | zidx = (
562 | np.repeat(row_vector(np.arange(0, nf, dtype=np.int)), nr, axis=0) +
563 | np.repeat(column_vector(np.arange(0, nr, dtype=np.int) * inc +
564 | (np.arange(0, nr, dtype=np.int) % nb) * no), nf, axis=1))
565 | # input_frames_windowed = input_frames * np.repeat(row_vector(window_samples), n_frames, axis=0)
566 | input_frames *= np.repeat(row_vector(window_samples), nr, axis=0)
567 | z[zidx.flatten(order='F').astype(np.int32)] = input_frames.flatten(order='F')
568 | z = z.reshape(overlapped_output_shape, order='F')
569 | if z.ndim > 1:
570 | z = np.sum(z, axis=1)
571 | if previous_partial_output is not None:
572 | if previous_partial_output.ndim > 1:
573 | raise NameError(getfname() + "PrevPartialOutDimError")
574 | else:
575 | z[0:previous_partial_output.size] += previous_partial_output
576 | out_samples = int(inc * nr)
577 | if no < out_samples:
578 | z[out_samples] = 0
579 | current_partial_output = np.array([])
580 | else:
581 | current_partial_output = z[out_samples:]
582 | z = z[0:out_samples]
583 |
584 | return z, current_partial_output
585 |
586 |
587 | def get_array_energy(alike):
588 | """
589 | Getthe total energy of the elements in an array
590 |
591 | Args:
592 | alike: The input array
593 |
594 | Returns:
595 | The total energy
596 |
597 | """
598 | return np.sum(np.array(alike, dtype='float128') ** 2)
599 |
600 |
601 | def plotnorm(x=None, y=None, title=None, interactive=False, clf=False, savelocation=None,
602 | no_rescaling=False, **mplotargs):
603 | """
604 |
605 | A useful and flexible plotting tool for signal processing.
606 | It allows you to plot a number of signals on the same normalised scale. It can plot signals
607 | in the time domain when provided with a sampling frequency.
608 |
609 | Args:
610 | x: The x axis points or the sampling frequency as a scalar
611 | y: The list of vectros or the array to plot. the vectors can have a different number of
612 | elements each
613 | title: The string to use as the plot title
614 | interactive: Wait for the user to close the plot before continuing
615 | clf: Clear the plot before plotting
616 | savelocation: Save hte plot as this file
617 | no_rescaling: Do not normalize the scale of the signals
618 | **mplotargs: Arguments to be passed to matplotlib.pyplot.plot
619 |
620 | Returns:
621 | The plot
622 |
623 | """
624 |
625 | hasfs = False
626 | if (x is None) & (y is not None):
627 | x = np.arange(y.size)
628 | elif (not isinstance(x, np.ndarray)) & (y is not None):
629 | sampling_freq = float(x)
630 | x = np.arange(y.size) / sampling_freq
631 | hasfs = True
632 | elif x.size != y.size:
633 | raise NameError(getfname() + 'XYSizeMismatch')
634 | elif x.size == 0:
635 | return
636 | if not no_rescaling:
637 | y = y / abs(y).max()
638 |
639 | if clf:
640 | mplot.clf()
641 | res = mplot.plot(x, y, linewidth=0.5, **mplotargs)
642 | gen_title = 'Normalised Amplitude'
643 | if hasfs:
644 | gen_title += ' at Fs=' + repr(sampling_freq) + 'Hz'
645 | mplot.xlabel('Time (s)')
646 | else:
647 | mplot.xlabel('Sample')
648 | if title is None:
649 | mplot.title(gen_title)
650 | elif not title == '':
651 | mplot.title(title)
652 | mplot.ylabel('Normalised Amplitude')
653 | mplot.grid(True)
654 | if savelocation is not None:
655 | mplot.savefig(savelocation)
656 | print('Saved: ' + savelocation)
657 | if interactive:
658 | mplot.show()
659 |
660 | return res
661 |
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/README.md:
--------------------------------------------------------------------------------
1 |
2 | # Reverberation Learning Toolbox for Python
3 |
4 | Copyright 2019 [Constantinos Papayiannis](https://www.linkedin.com/in/papayiannis/)
5 |
6 | # Introduction
7 |
8 | The work in [1] has been one of the first steps in using deep learning to classify reverberant environments based on their acoustics. DNNs were provided with reverberant speech signals and the signals were classified in terms of the room where the recording was made in. This repository includes the code that was used in [1]. It is intended to be used by other researchers that aim to build on the work and follow the many promising research paths that stem from it. It also contains useful code for DSP, speech processing and deep learning using Keras.
9 |
10 |
11 | # Setup
12 |
13 | To use the repository start by setting up your environment. I assume you have Anaconda and you are working with Python 3.
14 |
15 | ```bash
16 | # Get the repository
17 | git clone https://github.com/papayiannis/reverberation_learning_python
18 | cd reverberation_learning_python
19 | # Get dependencies
20 | conda install numpy keras scipy tabulate matplotlib pandas seaborn h5py scikit-learn
21 | # Unpack the AIR data
22 | cd Code/Local_Databases/AIR
23 | tar zxf ACE16.tar.gz
24 | ```
25 |
26 | # Room classification
27 |
28 |
29 | To train a DNN for room classification from reverberant speech, do the following
30 |
31 | ```bash
32 | # Unpack the AIR data
33 | cd Code/Local_Databases/AIR
34 | tar zxf ACE16.tar.gz
35 | cd ../../pythonsrc
36 | mkdir -p /tmp/train_test_speech
37 | ln -s $TRAIN_SPEECH_LOC /tmp/train_test_speech/TRAIN
38 | ln -s $TEST_SPEECH_LOC /tmp/train_test_speech/TEST
39 | # Run the training example for a CNN-RNN room classifier using ACE AIRs and your speech files
40 | bash run_ace_discriminative_nets.sh ../Local_Databases/AIR/ACE16 \
41 | /tmp/train_test_speech/ ../results_dir/ace_h5_info.h5 0 5
42 | ```
43 |
44 | The index 8 choses an Attention-CRNN. The locations ```$TRAIN_SPEECH_LOC``` and ```$TEST_SPEECH_LOC``` contain respectively locations where speech wav files are included, for training and for testing of the trained DNNs. The experiments have used TIMIT but any other dataset can be used in practice.
45 |
46 | # Bibliography
47 |
48 | [1]: C. Papayiannis, C. Evers and P. A. Naylor, "End-to-End Classification of Reverberant Rooms Using DNNs," in IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol. 28, pp. 3010-3017, 2020, doi: 10.1109/TASLP.2020.3033628.
49 |
50 |
51 | _Reverberation Learning Toolbox for Python is free software: you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation, either version 3 of the License, or (at your option) any later version._
52 |
53 |
54 |
55 |
56 |
57 |
58 |
59 |
60 |
61 |
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