visualizing and playing sound files where prediction fails

checkpointing model

pandas_parallel.py

@@ -1,25 +0,0 @@
-import multiprocessing
-import pandas as pd
-import numpy as np
-
-
-
-def _apply_df(args):
-    df, func, num, kwargs = args
-    return num, df.apply(func, **kwargs)
-
-def apply_by_multiprocessing(df,func,**kwargs):
-  cores = multiprocessing.cpu_count()
-  workers=kwargs.pop('workers') if 'workers' in kwargs else cores
-  pool = multiprocessing.Pool(processes=workers)
-  result = pool.map(_apply_df, [(d, func, i, kwargs) for i,d in enumerate(np.array_split(df, workers))])
-  pool.close()
-  result=sorted(result,key=lambda x:x[0])
-  return pd.concat([i[1] for i in result])
-
-def square(x):
-    return x**x
-
-if __name__ == '__main__':
-    df = pd.DataFrame({'a':range(10), 'b':range(10)})
-    apply_by_multiprocessing(df, square, axis=1, workers=4)

requirements-linux.txt

@@ -0,0 +1,76 @@
+bleach==1.5.0
+click==6.7
+cloudpickle==0.4.1
+cycler==0.10.0
+dask==0.15.4
+decorator==4.1.2
+distributed==1.19.3
+entrypoints==0.2.3
+enum34==1.1.6
+futures==3.1.1
+h5py==2.7.1
+HeapDict==1.0.0
+html5lib==0.9999999
+ipykernel==4.6.1
+ipython==6.2.1
+ipython-genutils==0.2.0
+ipywidgets==7.0.3
+jedi==0.11.0
+Jinja2==2.9.6
+jsonschema==2.6.0
+jupyter==1.0.0
+jupyter-client==5.1.0
+jupyter-console==5.2.0
+jupyter-core==4.3.0
+Keras==2.0.8
+locket==0.2.0
+Markdown==2.6.9
+MarkupSafe==1.0
+matplotlib==2.1.0
+mistune==0.7.4
+msgpack-python==0.4.8
+nbconvert==5.3.1
+nbformat==4.4.0
+notebook==5.2.0
+numexpr==2.6.4
+numpy==1.13.3
+pandas==0.20.3
+pandocfilters==1.4.2
+parso==0.1.0
+partd==0.3.8
+pexpect==4.2.1
+pickleshare==0.7.4
+progressbar2==3.34.3
+prompt-toolkit==1.0.15
+protobuf==3.4.0
+psutil==5.4.0
+ptyprocess==0.5.2
+PyAudio==0.2.11
+Pygments==2.2.0
+pyparsing==2.2.0
+pysndfile==1.0.0
+python-dateutil==2.6.1
+python-utils==2.2.0
+pytz==2017.2
+PyYAML==3.12
+pyzmq==16.0.2
+qtconsole==4.3.1
+scikit-learn==0.19.0
+scipy==0.19.1
+simplegeneric==0.8.1
+six==1.11.0
+sortedcontainers==1.5.7
+tables==3.4.2
+tblib==1.3.2
+tensorflow==1.3.0
+tensorflow-tensorboard==0.4.0rc1
+terminado==0.6
+testpath==0.3.1
+toolz==0.8.2
+tornado==4.5.2
+tqdm==4.19.4
+traitlets==4.3.2
+wcwidth==0.1.7
+Werkzeug==0.12.2
+widgetsnbextension==3.0.6
+zict==0.1.3

speech_data.py

@@ -1,114 +1,32 @@
 import pandas as pd
-from pandas_parallel import apply_by_multiprocessing
+from speech_utils import apply_by_multiprocessing
+from speech_utils import threadsafe_iter
 # import dask as dd
 # import dask.dataframe as ddf
 import tensorflow as tf
+from tensorflow.python.ops import data_flow_ops
 import numpy as np
-from spectro_gen import generate_aiff_spectrogram
+from speech_spectrum import generate_aiff_spectrogram
 from sklearn.model_selection import train_test_split
 import itertools
 import os
 import random
 import csv
 import gc
+import pickle
+from tqdm import tqdm
 
-def get_siamese_pairs(groupF1, groupF2):
-    group1 = [r for (i, r) in groupF1.iterrows()]
-    group2 = [r for (i, r) in groupF2.iterrows()]
-    diff = [(g1, g2) for g2 in group2 for g1 in group1]
-    same = [i for i in itertools.combinations(group1, 2)
-         ] + [i for i in itertools.combinations(group2, 2)]
-    random.shuffle(same)
-    random.shuffle(diff)
-    # return (random.sample(same,10), random.sample(diff,10))
-    return same[:10],diff[:10]
 
 def siamese_pairs(rightGroup, wrongGroup):
     group1 = [r for (i, r) in rightGroup.iterrows()]
     group2 = [r for (i, r) in wrongGroup.iterrows()]
     rightWrongPairs = [(g1, g2) for g2 in group2 for g1 in group1]
-    rightRightPairs = [i for i in itertools.combinations(group1, 2)]
-    random.shuffle(rightWrongPairs)
-    random.shuffle(rightRightPairs)
-    # return (random.sample(same,10), random.sample(diff,10))
-    return rightRightPairs[:10],rightWrongPairs[:10]
+    rightRightPairs = [i for i in itertools.combinations(group1, 2)]#+[i for i in itertools.combinations(group2, 2)]
+    # random.shuffle(rightWrongPairs)
+    # random.shuffle(rightRightPairs)
+    # return rightRightPairs[:10],rightWrongPairs[:10]
+    return rightRightPairs[:32],rightWrongPairs[:32]
 
-def append_zeros(spgr, max_samples):
-    return np.lib.pad(spgr, [(0, max_samples - spgr.shape[0]), (0, 0)],
-                      'median')
-
-def padd_zeros(spgr, max_samples):
-    return np.lib.pad(spgr, [(0, max_samples - spgr.shape[0]), (0, 0)],
-                      'constant')
-
-def to_onehot(a,class_count=2):
-    # >>> a = np.array([1, 0, 3])
-    a_row_n = a.shape[0]
-    b = np.zeros((a_row_n, class_count))
-    b[np.arange(a_row_n), a] = 1
-    return b
-
-def create_pair(l, r, max_samples):
-    l_sample = padd_zeros(l, max_samples)
-    r_sample = padd_zeros(r, max_samples)
-    return np.asarray([l_sample, r_sample])
-
-
-def create_test_pair(l, r, max_samples):
-    l_sample = append_zeros(l, max_samples)
-    r_sample = append_zeros(r, max_samples)
-    return np.asarray([[l_sample, r_sample]])
-
-
-def create_X(sp, max_samples):
-    return create_pair(sp[0]['spectrogram'], sp[1]['spectrogram'], max_samples)
-
-
-# def get_word_pairs_data(word, max_samples):
-#     audio_samples = pd.read_csv(
-#         './outputs/audio.csv',
-#         names=['word', 'voice', 'rate', 'variant', 'file'])
-#     audio_samples = audio_samples.loc[audio_samples['word'] ==
-#                                       word].reset_index(drop=True)
-#     audio_samples.loc[:, 'spectrogram'] = audio_samples.loc[:, 'file'].apply(
-#         lambda x: 'outputs/audio/' + x).apply(generate_aiff_spectrogram)
-#     max_samples = audio_samples['spectrogram'].apply(
-#         lambda x: x.shape[0]).max()
-#     same_data, diff_data = [], []
-#     for (w, g) in audio_samples.groupby(audio_samples['word']):
-#         sample_norm = g.loc[audio_samples['variant'] == 'normal']
-#         sample_phon = g.loc[audio_samples['variant'] == 'phoneme']
-#         same, diff = get_siamese_pairs(sample_norm, sample_phon)
-#         same_data.extend([create_X(s, max_samples) for s in same])
-#         diff_data.extend([create_X(d, max_samples) for d in diff])
-#     Y = np.hstack([np.ones(len(same_data)), np.zeros(len(diff_data))])
-#     X = np.asarray(same_data + diff_data)
-#     # tr_pairs, te_pairs, tr_y, te_y = train_test_split(X, Y, test_size=0.1)
-#     return (X, Y)
-
-
-def create_spectrogram_data(audio_group='audio'):
-    audio_samples = pd.read_csv( './outputs/' + audio_group + '.csv'
-        , names=['word','phonemes', 'voice', 'language', 'rate', 'variant', 'file']
-        , quoting=csv.QUOTE_NONE)
-    # audio_samples = audio_samples.loc[audio_samples['word'] ==
-    #                                   'sunflowers'].reset_index(drop=True)
-    audio_samples['file_paths'] = audio_samples.loc[:, 'file'].apply(lambda x: 'outputs/' + audio_group + '/' + x)
-    audio_samples['file_exists'] = apply_by_multiprocessing(audio_samples['file_paths'], os.path.exists)
-    audio_samples = audio_samples[audio_samples['file_exists'] == True].reset_index()
-    audio_samples['spectrogram'] = apply_by_multiprocessing(audio_samples['file_paths'],generate_aiff_spectrogram)#.apply(
-    audio_samples['window_count'] = audio_samples.loc[:,'spectrogram'].apply(lambda x: x.shape[0])
-    audio_samples.to_pickle('outputs/{}-spectrogram.pkl'.format(audio_group))
-
-def create_spectrogram_tfrecords(audio_group='audio'):
-    audio_samples = pd.read_csv( './outputs/' + audio_group + '.csv'
-        , names=['word','phonemes', 'voice', 'language', 'rate', 'variant', 'file']
-        , quoting=csv.QUOTE_NONE)
-    # audio_samples = audio_samples.loc[audio_samples['word'] ==
-    #                                   'sunflowers'].reset_index(drop=True)
-    audio_samples['file_path'] = audio_samples.loc[:, 'file'].apply(lambda x: 'outputs/' + audio_group + '/' + x)
-    audio_samples['file_exists'] = apply_by_multiprocessing(audio_samples['file_path'], os.path.exists)
-    audio_samples = audio_samples[audio_samples['file_exists'] == True].reset_index()
 
 def _float_feature(value):
   return tf.train.Feature(float_list=tf.train.FloatList(value=value))
@@ -119,21 +37,48 @@ def create_spectrogram_tfrecords(audio_group='audio'):
 def _bytes_feature(value):
   return tf.train.Feature(bytes_list=tf.train.BytesList(value=value))
 
-    writer = tf.python_io.TFRecordWriter('./outputs/' + audio_group + '.tfrecords')
-    # audio_samples = audio_samples[:100]
-    for (w, word_group) in audio_samples.groupby(audio_samples['word']):
+def create_spectrogram_tfrecords(audio_group='audio',sample_count=0,train_test_ratio=0.1):
+    '''
+    http://warmspringwinds.github.io/tensorflow/tf-slim/2016/12/21/tfrecords-guide/
+    http://www.machinelearninguru.com/deep_learning/tensorflow/basics/tfrecord/tfrecord.html
+    '''
+    audio_samples = pd.read_csv( './outputs/' + audio_group + '.csv',index_col=0)
+    audio_samples['file_path'] = audio_samples.loc[:, 'file'].apply(lambda x: 'outputs/' + audio_group + '/' + x)
+    n_records,n_spec,n_features = 0,0,0
+
+    def write_samples(wg,sample_name):
+        word_group_prog = tqdm(wg,desc='Computing spectrogram')
+        record_file = './outputs/{}.{}.tfrecords'.format(audio_group,sample_name)
+        writer = tf.python_io.TFRecordWriter(record_file)
+        for (w, word_group) in word_group_prog:
+            word_group_prog.set_postfix(word=w,sample_name=sample_name)
             g = word_group.reset_index()
             g['spectrogram'] = apply_by_multiprocessing(g['file_path'],generate_aiff_spectrogram)
-        sample_right = g.loc[audio_samples['variant'] == 'low']
-        sample_wrong = g.loc[audio_samples['variant'] == 'medium']
+            sample_right = g.loc[g['variant'] == 'low']
+            sample_wrong = g.loc[g['variant'] == 'medium']
             same, diff = siamese_pairs(sample_right, sample_wrong)
             groups = [([0,1],same),([1,0],diff)]
             for (output,group) in groups:
-            for sample1,sample2 in group:
+                group_prog = tqdm(group,desc='Writing Spectrogram')
+                for sample1,sample2 in group_prog:
+                    same = sample1['variant'] == sample2['variant']
+                    phon_same = sample1['phonemes'] == sample2['phonemes']
+                    voice_diff = sample1['voice'] != sample2['voice']
+                    if not same and phon_same:
+                        continue
+                    if same and not voice_diff:
+                        continue
+                    group_prog.set_postfix(output=output
+                        ,var1=sample1['variant']
+                        ,var2=sample2['variant'])
                     spectro1,spectro2 = sample1['spectrogram'],sample2['spectrogram']
                     spec_n1,spec_n2 = spectro1.shape[0],spectro2.shape[0]
                     spec_w1,spec_w2 = spectro1.shape[1],spectro2.shape[1]
                     spec1,spec2 = spectro1.reshape(-1),spectro2.reshape(-1)
+                    nonlocal n_spec,n_records,n_features
+                    n_spec = max([n_spec,spec_n1,spec_n2])
+                    n_features = spec_w1
+                    n_records+=1
                     example = tf.train.Example(features=tf.train.Features(
                       feature={
                         'word': _bytes_feature([w.encode('utf-8')]),
@@ -158,63 +103,144 @@ def create_spectrogram_tfrecords(audio_group='audio'):
                       }
                     ))
                     writer.write(example.SerializeToString())
+                group_prog.close()
+        word_group_prog.close()
         writer.close()
 
-def create_tagged_data(audio_samples):
-    same_data, diff_data = [], []
-    for (w, g) in audio_samples.groupby(audio_samples['word']):
-        # sample_norm = g.loc[audio_samples['variant'] == 'low']
-        # sample_phon = g.loc[audio_samples['variant'] == 'medium']
-        sample_norm = g.loc[audio_samples['variant'] == 'normal']
-        sample_phon = g.loc[audio_samples['variant'] == 'phoneme']
-        same, diff = get_siamese_pairs(sample_norm, sample_phon)
-        same_data.extend([create_X(s) for s in same])
-        diff_data.extend([create_X(d) for d in diff])
-    print('creating all speech pairs')
-    Y_f = np.hstack([np.ones(len(same_data)), np.zeros(len(diff_data))])
-    Y = to_onehot(Y_f.astype(np.int8))
-    print('casting as array speech pairs')
-    X = np.asarray(same_data + diff_data)
-    return X,Y
+    word_groups = [i for i in audio_samples.groupby('word')]
+    wg_sampled = reservoir_sample(word_groups,sample_count) if sample_count > 0 else word_groups
+    tr_audio_samples,te_audio_samples = train_test_split(wg_sampled,test_size=train_test_ratio)
+    write_samples(tr_audio_samples,'train')
+    write_samples(te_audio_samples,'test')
+    const_file = os.path.join('./outputs',audio_group+'.constants')
+    pickle.dump((n_spec,n_features,n_records),open(const_file,'wb'))
 
-def create_speech_pairs_data(audio_group='audio'):
-    audio_samples = pd.read_pickle('outputs/{}-spectrogram.pkl'.format(audio_group))
-    # sample_size = audio_samples['spectrogram'][0].shape[1]
-    tr_audio_samples,te_audio_samples = train_test_split(audio_samples, test_size=0.1)
-    def save_samples_for(sample_name,samples):
-        print('generating {} siamese speech pairs'.format(sample_name))
-        X,Y = create_tagged_data(samples)
-        print('shuffling array speech pairs')
-        rng_state = np.random.get_state()
-        np.random.shuffle(X)
-        np.random.set_state(rng_state)
-        np.random.shuffle(Y)
-        print('pickling X/Y')
-        np.save('outputs/{}-train-X.npy'.format(audio_group), X)
-        np.save('outputs/{}-train-Y.npy'.format(audio_group), Y)
-    save_samples_for('train',tr_audio_samples)
-    save_samples_for('test',te_audio_samples)
+def padd_zeros(spgr, max_samples):
+    return np.lib.pad(spgr, [(0, max_samples - spgr.shape[0]), (0, 0)],
+                      'constant')
 
-def speech_data(audio_group='audio'):
-    X = np.load('outputs/{}-X.npy'.format(audio_group)) / 255.0
-    Y = np.load('outputs/{}-Y.npy'.format(audio_group))
-    return (X,Y)
+def reservoir_sample(iterable, k):
+    it = iter(iterable)
+    if not (k > 0):
+        raise ValueError("sample size must be positive")
 
-def speech_model_data():
-    tr_pairs = np.load('outputs/tr_pairs.npy') / 255.0
-    te_pairs = np.load('outputs/te_pairs.npy') / 255.0
-    tr_pairs[tr_pairs < 0] = 0
-    te_pairs[te_pairs < 0] = 0
-    tr_y = np.load('outputs/tr_y.npy')
-    te_y = np.load('outputs/te_y.npy')
-    return tr_pairs, te_pairs, tr_y, te_y
+    sample = list(itertools.islice(it, k)) # fill the reservoir
+    random.shuffle(sample) # if number of items less then *k* then
+                           #   return all items in random order.
+    for i, item in enumerate(it, start=k+1):
+        j = random.randrange(i) # random [0..i)
+        if j < k:
+            sample[j] = item # replace item with gradually decreasing probability
+    return sample
 
 
+def read_siamese_tfrecords_generator(audio_group='audio',batch_size=32,test_size=0):
+    records_file  = os.path.join('./outputs',audio_group+'.train.tfrecords')
+    input_pairs = []
+    output_class = []
+    const_file = os.path.join('./outputs',audio_group+'.constants')
+    (n_spec,n_features,n_records) = pickle.load(open(const_file,'rb'))
+    print('reading tfrecords({}-train)...'.format(audio_group))
+
+    # @threadsafe_iter
+    def record_generator():
+        input_data = []
+        output_data = []
+        while True:
+            record_iterator = tf.python_io.tf_record_iterator(path=records_file)
+            #tqdm(enumerate(record_iterator),total=n_records)
+            for (i,string_record) in enumerate(record_iterator):
+                example = tf.train.Example()
+                example.ParseFromString(string_record)
+                spec_n1 = example.features.feature['spec_n1'].int64_list.value[0]
+                spec_n2 = example.features.feature['spec_n2'].int64_list.value[0]
+                spec_w1 = example.features.feature['spec_w1'].int64_list.value[0]
+                spec_w2 = example.features.feature['spec_w2'].int64_list.value[0]
+                spec1 = np.array(example.features.feature['spec1'].float_list.value).reshape(spec_n1,spec_w1)
+                spec2 = np.array(example.features.feature['spec2'].float_list.value).reshape(spec_n2,spec_w2)
+                p_spec1,p_spec2 = padd_zeros(spec1,n_spec),padd_zeros(spec2,n_spec)
+                input_data.append(np.asarray([p_spec1,p_spec2]))
+                output = example.features.feature['output'].int64_list.value
+                output_data.append(np.asarray(output))
+                if len(input_data) == batch_size:
+                    input_arr = np.asarray(input_data)
+                    output_arr = np.asarray(output_data)
+                    yield ([input_arr[:, 0], input_arr[:, 1]],output_arr)
+                    input_data = []
+                    output_data = []
+
+    # Read test in one-shot
+    te_records_file  = os.path.join('./outputs',audio_group+'.test.tfrecords')
+    te_re_iterator = tf.python_io.tf_record_iterator(path=records_file)
+    te_n_records = len([i for i in te_re_iterator])
+    te_re_iterator = tf.python_io.tf_record_iterator(path=records_file)
+    print('reading tfrecords({}-test)...'.format(audio_group))
+    test_size = min([test_size,te_n_records]) if test_size > 0 else te_n_records
+    input_data = np.zeros((test_size,2,n_spec,n_features))
+    output_data = np.zeros((test_size,2))
+    random_samples = enumerate(reservoir_sample(te_re_iterator,test_size))
+    for (i,string_record) in tqdm(random_samples,total=test_size):
+        example = tf.train.Example()
+        example.ParseFromString(string_record)
+        spec_n1 = example.features.feature['spec_n1'].int64_list.value[0]
+        spec_n2 = example.features.feature['spec_n2'].int64_list.value[0]
+        spec_w1 = example.features.feature['spec_w1'].int64_list.value[0]
+        spec_w2 = example.features.feature['spec_w2'].int64_list.value[0]
+        spec1 = np.array(example.features.feature['spec1'].float_list.value).reshape(spec_n1,spec_w1)
+        spec2 = np.array(example.features.feature['spec2'].float_list.value).reshape(spec_n2,spec_w2)
+        p_spec1,p_spec2 = padd_zeros(spec1,n_spec),padd_zeros(spec2,n_spec)
+        input_data[i] = np.asarray([p_spec1,p_spec2])
+        output = example.features.feature['output'].int64_list.value
+        output_data[i] = np.asarray(output)
+
+    return record_generator,input_data,output_data,n_spec,n_features,n_records
+
+def audio_samples_word_count(audio_group='audio'):
+    audio_samples = pd.read_csv( './outputs/' + audio_group + '.csv')
+    return len(audio_samples.groupby(audio_samples['word']))
+
+def fix_csv(audio_group='audio'):
+    audio_csv_lines = open('./outputs/' + audio_group + '.csv.orig','r').readlines()
+    audio_csv_data = [i.strip().split(',') for i in audio_csv_lines]
+    proper_rows = [i for i in audio_csv_data if len(i) == 7]
+    with open('./outputs/' + audio_group + '.csv','w') as fixed_csv:
+        fixed_csv_w = csv.writer(fixed_csv, quoting=csv.QUOTE_MINIMAL)
+        fixed_csv_w.writerows(proper_rows)
+    audio_samples = pd.read_csv( './outputs/' + audio_group + '.csv'
+        , names=['word','phonemes', 'voice', 'language', 'rate', 'variant', 'file'])
+    audio_samples['file_path'] = audio_samples.loc[:, 'file'].apply(lambda x: 'outputs/' + audio_group + '/' + x)
+    audio_samples['file_exists'] = apply_by_multiprocessing(audio_samples['file_path'], os.path.exists)
+    audio_samples = audio_samples[audio_samples['file_exists'] == True]
+    audio_samples = audio_samples.drop(['file_path','file_exists'],axis=1).reset_index(drop=True)
+    audio_samples.to_csv('./outputs/' + audio_group + '.csv')
+
+def convert_old_audio():
+    audio_samples = pd.read_csv( './outputs/audio.csv.old'
+        , names=['word', 'voice', 'rate', 'variant', 'file'])
+    audio_samples['phonemes'] = 'unknown'
+    audio_samples['language'] = 'en-US'
+    audio_samples.loc[audio_samples['variant'] == 'normal','variant'] = 'low'
+    audio_samples.loc[audio_samples['variant'] == 'phoneme','variant'] = 'medium'
+    audio_samples = audio_samples[['word','phonemes', 'voice', 'language', 'rate', 'variant', 'file']]
+    audio_samples.to_csv('./outputs/audio_new.csv',index=False,header=False)
+
 if __name__ == '__main__':
     # sunflower_pairs_data()
     # create_spectrogram_data()
     # create_spectrogram_data('story_words')
-    create_spectrogram_tfrecords('story_words')
+    # create_spectrogram_tfrecords('story_words')
+    # create_spectrogram_tfrecords('story_words_test')
+    # read_siamese_tfrecords('story_all')
+    # read_siamese_tfrecords('story_words_test')
+    # padd_zeros_siamese_tfrecords('story_words')
+    # fix_csv('story_words')
+    # pickle_constants('story_words')
+    # create_spectrogram_tfrecords('audio',sample_count=100)
+    # create_spectrogram_tfrecords('story_all',sample_count=25)
+    create_spectrogram_tfrecords('story_words',sample_count=10,train_test_ratio=0.2)
+    # create_spectrogram_tfrecords('audio',sample_count=50)
+    # read_siamese_tfrecords_generator('audio')
+    # padd_zeros_siamese_tfrecords('audio')
     # create_padded_spectrogram()
     # create_speech_pairs_data()
     # print(speech_model_data())

speech_siamese.py

@@ -1,7 +1,8 @@
 from __future__ import absolute_import
 from __future__ import print_function
 import numpy as np
-from speech_data import speech_model_data
+# from speech_data import speech_model_data
+from speech_data import read_siamese_tfrecords_generator
 from keras.models import Model,load_model
 from keras.layers import Input, Dense, Dropout, LSTM, Lambda, Concatenate
 from keras.losses import categorical_crossentropy
@@ -11,41 +12,44 @@ from keras.utils import to_categorical
 from keras.optimizers import RMSprop
 from keras.callbacks import TensorBoard, ModelCheckpoint
 from keras import backend as K
+from speech_utils import create_dir
 
-
-def euclidean_distance(vects):
-    x, y = vects
-    return K.sqrt(
-        K.maximum(K.sum(K.square(x - y), axis=1, keepdims=True), K.epsilon()))
-
-
-def eucl_dist_output_shape(shapes):
-    shape1, shape2 = shapes
-    return (shape1[0], 1)
-
-
-def contrastive_loss(y_true, y_pred):
-    '''Contrastive loss from Hadsell-et-al.'06
-    http://yann.lecun.com/exdb/publis/pdf/hadsell-chopra-lecun-06.pdf
-    '''
-    return K.mean(y_true * K.square(y_pred) +
-                  (1 - y_true) * K.square(K.maximum(1 - y_pred, 0)))
+# def euclidean_distance(vects):
+#     x, y = vects
+#     return K.sqrt(
+#         K.maximum(K.sum(K.square(x - y), axis=1, keepdims=True), K.epsilon()))
+#
+#
+# def eucl_dist_output_shape(shapes):
+#     shape1, shape2 = shapes
+#     return (shape1[0], 1)
+#
+#
+# def contrastive_loss(y_true, y_pred):
+#     '''Contrastive loss from Hadsell-et-al.'06
+#     http://yann.lecun.com/exdb/publis/pdf/hadsell-chopra-lecun-06.pdf
+#     '''
+#     return K.mean(y_true * K.square(y_pred) +
+#                   (1 - y_true) * K.square(K.maximum(1 - y_pred, 0)))
 
 def create_base_rnn_network(input_dim):
     '''Base network to be shared (eq. to feature extraction).
     '''
     inp = Input(shape=input_dim)
-    ls1 = LSTM(256, return_sequences=True)(inp)
+    ls0 = LSTM(512, return_sequences=True)(inp)
+    ls1 = LSTM(256, return_sequences=True)(ls0)
     ls2 = LSTM(128, return_sequences=True)(ls1)
     # ls3 = LSTM(32, return_sequences=True)(ls2)
     ls4 = LSTM(64)(ls2)
+    d1 = Dense(128, activation='relu')(ls4)
+    d2 = Dense(64, activation='relu')(d1)
     return Model(inp, ls4)
 
 
 def compute_accuracy(y_true, y_pred):
     '''Compute classification accuracy with a fixed threshold on distances.
     '''
-    pred = y_pred.ravel() < 0.5
+    pred = y_pred.ravel() > 0.5
     return np.mean(pred == y_true)
 
 
@@ -56,11 +60,12 @@ def accuracy(y_true, y_pred):
 
 def dense_classifier(processed):
     conc_proc = Concatenate()(processed)
-    d1 = Dense(16, activation='relu')(conc_proc)
+    d1 = Dense(64, activation='relu')(conc_proc)
     # dr1 = Dropout(0.1)(d1)
-    d2 = Dense(8, activation='relu')(d1)
+    d2 = Dense(128, activation='relu')(d1)
+    d3 = Dense(8, activation='relu')(d2)
     # dr2 = Dropout(0.1)(d2)
-    return Dense(2, activation='softmax')(d2)
+    return Dense(2, activation='softmax')(d3)
 
 def siamese_model(input_dim):
     # input_dim = (15, 1654)
@@ -78,17 +83,24 @@ def siamese_model(input_dim):
     return model
 
 
-def train_siamese():
+def train_siamese(audio_group = 'audio'):
     # the data, shuffled and split between train and test sets
-    tr_pairs, te_pairs, tr_y_e, te_y_e = speech_model_data()
-    tr_y = to_categorical(tr_y_e, num_classes=2)
-    te_y = to_categorical(te_y_e, num_classes=2)
-    input_dim = (tr_pairs.shape[2], tr_pairs.shape[3])
+    # tr_pairs, te_pairs, tr_y_e, te_y_e = speech_model_data()
+    batch_size = 128
+    model_dir = './models/'+audio_group
+    create_dir(model_dir)
+    log_dir = './logs/'+audio_group
+    create_dir(log_dir)
+    tr_gen_fn,te_pairs,te_y,n_step,n_features,n_records = read_siamese_tfrecords_generator(audio_group,batch_size=batch_size)
+    tr_gen = tr_gen_fn()
+    # tr_y = to_categorical(tr_y_e, num_classes=2)
+    # te_y = to_categorical(te_y_e, num_classes=2)
+    input_dim = (n_step, n_features)
 
     model = siamese_model(input_dim)
 
     tb_cb = TensorBoard(
-        log_dir='./logs/siamese_logs',
+        log_dir=log_dir,
         histogram_freq=1,
         batch_size=32,
         write_graph=True,
@@ -97,39 +109,45 @@ def train_siamese():
         embeddings_freq=0,
         embeddings_layer_names=None,
         embeddings_metadata=None)
-    cp_file_fmt = './models/siamese_speech_model-{epoch:02d}-epoch-{val_loss:0.2f}\
+    cp_file_fmt = model_dir+'/siamese_speech_model-{epoch:02d}-epoch-{val_loss:0.2f}\
 -acc.h5'
 
     cp_cb = ModelCheckpoint(
         cp_file_fmt,
         monitor='val_loss',
         verbose=0,
-        save_best_only=False,
-        save_weights_only=False,
+        save_best_only=True,
+        save_weights_only=True,
         mode='auto',
         period=1)
     # train
-    rms = RMSprop(lr=0.001)
+    rms = RMSprop()#lr=0.001
     model.compile(loss=categorical_crossentropy, optimizer=rms, metrics=[accuracy])
-    model.fit(
-        [tr_pairs[:, 0], tr_pairs[:, 1]],
-        tr_y,
-        batch_size=128,
-        epochs=50,
-        validation_data=([te_pairs[:, 0], te_pairs[:, 1]], te_y),
-        callbacks=[tb_cb, cp_cb])
-
-    model.save('./models/siamese_speech_model-final.h5')
+    # model.fit(
+    #     [tr_pairs[:, 0], tr_pairs[:, 1]],
+    #     tr_y,
+    #     batch_size=128,
+    #     epochs=100,
+    #     validation_data=([te_pairs[:, 0], te_pairs[:, 1]], te_y),
+    #     callbacks=[tb_cb, cp_cb])
+    model.fit_generator(tr_gen
+                        ,epochs=1000
+                        ,steps_per_epoch=n_records//batch_size
+                        ,validation_data=([te_pairs[:, 0], te_pairs[:, 1]], te_y)
+                        ,use_multiprocessing=True, workers=1
+                        ,callbacks=[tb_cb, cp_cb])
+    model.save(model_dir+'/siamese_speech_model-final.h5')
     # compute final accuracy on training and test sets
-    y_pred = model.predict([tr_pairs[:, 0], tr_pairs[:, 1]])
-    tr_acc = compute_accuracy(tr_y, y_pred)
+    # y_pred = model.predict([tr_pairs[:, 0], tr_pairs[:, 1]])
+    # tr_acc = compute_accuracy(tr_y, y_pred)
+    # print('* Accuracy on training set: %0.2f%%' % (100 * tr_acc))
+
     y_pred = model.predict([te_pairs[:, 0], te_pairs[:, 1]])
     te_acc = compute_accuracy(te_y, y_pred)
-
-    print('* Accuracy on training set: %0.2f%%' % (100 * tr_acc))
     print('* Accuracy on test set: %0.2f%%' % (100 * te_acc))
 
 
 
 if __name__ == '__main__':
-    train_siamese()
+    train_siamese('story_words')
+    # train_siamese('audio')

speech_tools.py

@@ -1,15 +1,36 @@
 import pyaudio
+from pysndfile import sndio as snd
 import numpy as np
 # from matplotlib import pyplot as plt
-from spectro_gen import plot_stft, generate_spec_frec
+from speech_spectrum import plot_stft, generate_spec_frec
 
-
-def record_spectrogram(n_sec, plot=False, playback=False):
 SAMPLE_RATE = 22050
 N_CHANNELS = 2
+
+def file_player():
+    p_oup = pyaudio.PyAudio()
+    def play_file(audiopath,plot=False):
+        print('playing',audiopath)
+        samples, samplerate, form = snd.read(audiopath)
+        stream = p_oup.open(
+            format=pyaudio.paFloat32,
+            channels=2,
+            rate=samplerate,
+            output=True)
+        one_channel = np.asarray([samples, samples]).T.reshape(-1)
+        audio_data = one_channel.astype(np.float32).tobytes()
+        stream.write(audio_data)
+        stream.close()
+        if plot:
+            plot_stft(samples, SAMPLE_RATE)
+    def close_player():
+        p_oup.terminate()
+    return play_file,close_player
+
+def record_spectrogram(n_sec, plot=False, playback=False):
+    # show_record_prompt()
     N_SEC = n_sec
     CHUNKSIZE = int(SAMPLE_RATE * N_SEC / N_CHANNELS)  # fixed chunk size
-    # show_record_prompt()
     input('Press [Enter] to start recording sample... ')
     p_inp = pyaudio.PyAudio()
     stream = p_inp.open(

speech_utils.py

@@ -0,0 +1,74 @@
+import os
+import threading
+
+import multiprocessing
+import pandas as pd
+import numpy as np
+
+
+
+def _apply_df(args):
+    df, func, num, kwargs = args
+    return num, df.apply(func, **kwargs)
+
+def apply_by_multiprocessing(df,func,**kwargs):
+  cores = multiprocessing.cpu_count()
+  workers=kwargs.pop('workers') if 'workers' in kwargs else cores
+  pool = multiprocessing.Pool(processes=workers)
+  result = pool.map(_apply_df, [(d, func, i, kwargs) for i,d in enumerate(np.array_split(df, workers))])
+  pool.close()
+  result=sorted(result,key=lambda x:x[0])
+  return pd.concat([i[1] for i in result])
+
+def square(x):
+    return x**x
+
+if __name__ == '__main__':
+    df = pd.DataFrame({'a':range(10), 'b':range(10)})
+    apply_by_multiprocessing(df, square, axis=1, workers=4)
+
+
+def rm_rf(d):
+    for path in (os.path.join(d,f) for f in os.listdir(d)):
+        if os.path.isdir(path):
+            rm_rf(path)
+        else:
+            os.unlink(path)
+    os.rmdir(d)
+
+def create_dir(direc):
+    if not os.path.exists(direc):
+        os.makedirs(direc)
+    else:
+        rm_rf(direc)
+        create_dir(direc)
+
+
+#################### Now make the data generator threadsafe ####################
+
+class threadsafe_iter:
+    """Takes an iterator/generator and makes it thread-safe by
+    serializing call to the `next` method of given iterator/generator.
+    """
+    def __init__(self, it):
+        self.it = it
+        self.lock = threading.Lock()
+
+    def __iter__(self):
+        return self
+
+    def __next__(self): # Py3
+        with self.lock:
+            return next(self.it)
+
+    def next(self):     # Py2
+        with self.lock:
+            return self.it.next()
+
+
+def threadsafe_generator(f):
+    """A decorator that takes a generator function and makes it thread-safe.
+    """
+    def g(*a, **kw):
+        return threadsafe_iter(f(*a, **kw))
+    return g

test_siamese.py

@@ -1,13 +1,15 @@
-from speech_siamese import siamese_model
-from record_mic_speech import record_spectrogram
-from importlib import reload
+# from speech_siamese import siamese_model
+from speech_tools import record_spectrogram, file_player
+# from importlib import reload
 # import speech_data
 # reload(speech_data)
-from speech_data import create_test_pair,get_word_pairs_data,speech_data
 import numpy as np
-
-model = siamese_model((15, 1654))
-model.load_weights('./models/siamese_speech_model-final.h5')
+import pandas as pd
+import os
+import pickle
+import tensorflow as tf
+import csv
+from speech_data import padd_zeros
 
 def predict_recording_with(m,sample_size=15):
     spec1 = record_spectrogram(n_sec=1.4)
@@ -24,7 +26,85 @@ def test_with(audio_group):
     print(np.argmax(model.predict([X[:, 0], X[:, 1]]),axis=1))
     print(Y.astype(np.int8))
 
-test_with('rand_edu')
+def evaluate_siamese(audio_group='audio',model_file = 'siamese_speech_model-305-epoch-0.20-acc.h5'):
+    # audio_group='audio';model_file = 'siamese_speech_model-305-epoch-0.20-acc.h5'
+    records_file  = os.path.join('./outputs',audio_group+'.train.tfrecords')
+    const_file = os.path.join('./outputs',audio_group+'.constants')
+    model_weights_path =os.path.join('./models/story_words/',model_file)
+    (n_spec,n_features,n_records) = pickle.load(open(const_file,'rb'))
+    print('evaluating tfrecords({}-train)...'.format(audio_group))
+
+    model = siamese_model((n_spec, n_features))
+    model.load_weights(model_weights_path)
+    record_iterator = tf.python_io.tf_record_iterator(path=records_file)
+    #tqdm(enumerate(record_iterator),total=n_records)
+    result_csv = open('./outputs/' + audio_group + '.results.csv','w')
+    result_csv_w = csv.writer(result_csv, quoting=csv.QUOTE_MINIMAL)
+    result_csv_w.writerow(["phoneme1","phoneme2","voice1","voice2","rate1","rate2","variant1","variant2","file1","file2"])
+    for (i,string_record) in enumerate(record_iterator):
+        # string_record = next(record_iterator)
+        example = tf.train.Example()
+        example.ParseFromString(string_record)
+        spec_n1 = example.features.feature['spec_n1'].int64_list.value[0]
+        spec_n2 = example.features.feature['spec_n2'].int64_list.value[0]
+        spec_w1 = example.features.feature['spec_w1'].int64_list.value[0]
+        spec_w2 = example.features.feature['spec_w2'].int64_list.value[0]
+        spec1 = np.array(example.features.feature['spec1'].float_list.value).reshape(spec_n1,spec_w1)
+        spec2 = np.array(example.features.feature['spec2'].float_list.value).reshape(spec_n2,spec_w2)
+        p_spec1,p_spec2 = padd_zeros(spec1,n_spec),padd_zeros(spec2,n_spec)
+        input_arr = np.asarray([[p_spec1,p_spec2]])
+        output_arr = np.asarray([example.features.feature['output'].int64_list.value])
+        y_pred = model.predict([input_arr[:, 0], input_arr[:, 1]])
+        predicted = np.asarray(y_pred[0]>0.5).astype(output_arr.dtype)
+        expected = output_arr[0]
+        if np.all(predicted == expected):
+            continue
+        word = example.features.feature['word'].bytes_list.value[0].decode()
+        phoneme1 = example.features.feature['phoneme1'].bytes_list.value[0].decode()
+        phoneme2 = example.features.feature['phoneme2'].bytes_list.value[0].decode()
+        voice1 = example.features.feature['voice1'].bytes_list.value[0].decode()
+        voice2 = example.features.feature['voice2'].bytes_list.value[0].decode()
+        language = example.features.feature['language'].bytes_list.value[0].decode()
+        rate1 = example.features.feature['rate1'].int64_list.value[0]
+        rate2 = example.features.feature['rate2'].int64_list.value[0]
+        variant1 = example.features.feature['variant1'].bytes_list.value[0].decode()
+        variant2 = example.features.feature['variant2'].bytes_list.value[0].decode()
+        file1 = example.features.feature['file1'].bytes_list.value[0].decode()
+        file2 = example.features.feature['file2'].bytes_list.value[0].decode()
+        print(phoneme1,phoneme2,voice1,voice2,rate1,rate2,variant1,variant2,file1,file2)
+        result_csv_w.writerow([phoneme1,phoneme2,voice1,voice2,rate1,rate2,variant1,variant2,file1,file2])
+    result_csv.close()
+
+
+def play_results(audio_group='audio'):
+    result_data = pd.read_csv('./outputs/' + audio_group + '.results.csv')
+    play_file,close_player = file_player()
+    quit = False
+    for (i,r) in result_data.iterrows():
+        if quit:
+            break
+        keys = ["phoneme1","phoneme2","voice1","voice2","rate1","rate2","variant1","variant2"]
+        row_vals = [str(r[k]) for k in keys]
+        h_str = '\t'.join(keys)
+        row_str = '\t'.join(row_vals)
+        while True:
+            print(h_str)
+            print(row_str)
+            play_file('./outputs/'+audio_group+'/'+r['file1'],True)
+            play_file('./outputs/'+audio_group+'/'+r['file2'],True)
+            a = input("press 'r/q/[Enter]' to replay/quit/continue:\t")
+            if a == 'r':
+                continue
+            if a == 'q':
+                quit = True
+                break
+            else:
+                break
+    close_player()
+
+# evaluate_siamese('story_words',model_file='siamese_speech_model-305-epoch-0.20-acc.h5')
+play_results('story_words')
+# test_with('rand_edu')
 # sunflower_data,sunflower_result = get_word_pairs_data('sweater',15)
 # print(np.argmax(model.predict([sunflower_data[:, 0], sunflower_data[:, 1]]),axis=1))
 # print(sunflower_result)

tts_samplegen.py

@@ -3,6 +3,7 @@ from AppKit import NSSpeechSynthesizer, NSSpeechInputModeProperty
 from AppKit import NSSpeechModePhoneme
 from Foundation import NSURL
 import json
+import csv
 import random
 import os
 import re
@@ -81,6 +82,11 @@ class SynthFile(object):
 
         return ','.join([str(c) for c in cols])+'\n'
 
+    def get_values(self):
+        cols = [self.word, self.phoneme, self.voice,
+                self.voice_lang, self.rate, self.variant,
+                self.filename]
+        return [str(c) for c in cols]
 
 class SynthVariant(object):
     """docstring for SynthVariant."""
@@ -191,22 +197,11 @@ def synth_generator():
         print("It took {} to synthsize all variants.".format(time_str))
     return synth_for_words
 
-
-def write_synths(synth_list, fname, csv=False):
-    f = open(fname, 'w')
-    if csv:
-        for s in synth_list:
-            f.write(s.get_csv())
-    else:
-        json.dump([s.get_json() for s in synth_list], f)
-    f.close()
-
-
 def synth_logger(fname, csv=False):
     f = open(fname, 'w')
-
+    s_csv_w = csv.writer(f, quoting=csv.QUOTE_MINIMAL)
     def csv_writer(s):
-        f.write(s.get_csv())
+        s_csv_w.writerow(s.get_values())
     synth_list = []
 
     def json_writer(s):