def prepare_input(filename):
from tensorflow.contrib.framework.python.ops import audio_ops
from tensorflow.python.ops import io_ops
with tf.Session(graph=tf.Graph()) as sess:
wav_filename_placeholder = tf.placeholder(tf.string, [])
wav_loader = io_ops.read_file(wav_filename_placeholder)
wav_decoder = tf.audio.decode_wav(wav_loader,
desired_channels=1,
desired_samples=16000,
name='decoded_sample_data')
spectrum = audio_ops.audio_spectrogram(input=wav_decoder[0],
window_size=640,
stride=320,
magnitude_squared=True,
name='AudioSpectrogram')
final = audio_ops.mfcc(spectrogram=spectrum,
sample_rate=wav_decoder[1],
upper_frequency_limit=4000.0,
lower_frequency_limit=20.0,
filterbank_channel_count=40,
dct_coefficient_count=10,
name='Mfcc')
data = sess.run(final,
feed_dict={wav_filename_placeholder: filename})
print(f'Data shape: {data.shape}')
return data
def get_test_data(self, how_many, offset, model_settings, sess, features='mfcc'):
candidates = self.data_index
if how_many == -1:
sample_count = len(candidates)
else:
sample_count = max(0, min(how_many, len(candidates) - offset))
desired_samples = model_settings['desired_samples']
data = np.zeros((sample_count, model_settings['fingerprint_size']))
wav_filename_placeholder = tf.placeholder(tf.string, [], name='wav_file_names')
wav_loader = io_ops.read_file(wav_filename_placeholder)
wav_decoder = contrib_audio.decode_wav(
wav_loader, desired_channels=1, desired_samples=desired_samples)
spectrogram = contrib_audio.audio_spectrogram(
wav_decoder.audio,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
mfcc = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=model_settings['dct_coefficient_count'])
for i in range(offset, offset + sample_count):
input_dict = {wav_filename_placeholder : candidates[i]}
if features == "spectrogram":
data[i - offset, :] = sess.run(spectrogram, feed_dict=input_dict).flatten()
elif features == "raw":
data[i - offset, :] = sess.run(wav_decoder.audio, feed_dict=input_dict).flatten()
else:
data[i - offset, :] = sess.run(mfcc, feed_dict=input_dict).flatten()
return data
def _build_processing_graph(self):
"""Builds a TensorFlow graph to apply the input distortions.
Creates a graph that loads a WAVE file, decodes it, scales the volume,
shifts it in time, adds in background noise, calculates a spectrogram, and
then builds an MFCC fingerprint from that.
This must be called with an active TensorFlow session running, and it
creates multiple placeholder inputs, and one output:
- wav_filename_placeholder_: Filename of the WAV to load.
- mfcc_: Output 2D fingerprint of processed audio.
"""
with tf.name_scope('audio_processing'):
desired_samples = self._model_settings['desired_samples']
self.wav_filename_placeholder_ = tf.placeholder(tf.string, [])
wav_loader = io_ops.read_file(self.wav_filename_placeholder_)
wav_decoder = contrib_audio.decode_wav(
wav_loader,
desired_channels=1,
desired_samples=desired_samples)
background_clamp = tf.clip_by_value(wav_decoder.audio, -1.0, 1.0)
# Run the spectrogram and MFCC ops to get a 2D 'fingerprint' of the audio.
spectrogram = contrib_audio.audio_spectrogram(
background_clamp,
window_size=self._model_settings['window_size_samples'],
stride=self._model_settings['window_stride_samples'],
magnitude_squared=True)
self.mfcc_ = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=self.
_model_settings['dct_coefficient_count'])
def mfcc_tensorflow(wavfile, _sr, frame_size, frame_shift, order=13):
sess = tf.InteractiveSession()
wav_filename_placeholder = tf.placeholder(tf.string, [])
wav_loader = io_ops.read_file(wav_filename_placeholder)
wav_decoder = contrib_audio.decode_wav(wav_loader, desired_channels=1)
wav_data = wav_decoder.audio
wav_sample_rate = sess.run(wav_decoder,
feed_dict={
wav_filename_placeholder: wavfile
}).sample_rate
check_sample_rate(wavfile, _sr, wav_sample_rate)
spectrogram = contrib_audio.audio_spectrogram(wav_data,
window_size=frame_size,
stride=frame_shift,
magnitude_squared=True)
mfcc_ = contrib_audio.mfcc(spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=order)
mfcc_data = sess.run(mfcc_, feed_dict={wav_filename_placeholder: wavfile})
return mfcc_data
def load_mfcc_file(sess, filename):
filename_ph = tf.placeholder(tf.string)
loader = io_ops.read_file(filename_ph)
decoder = contrib_audio.decode_wav(loader, desired_channels=1, desired_samples=16000)
spectrogram = contrib_audio.audio_spectrogram( decoder.audio, window_size=480, stride=160, magnitude_squared=True)
mfcc = contrib_audio.mfcc( spectrogram, decoder.sample_rate, dct_coefficient_count=40)
return sess.run(mfcc, feed_dict={filename_ph: filename})
def prepare_processing_graph(self, model_settings):
"""Builds a TensorFlow graph to apply the input distortions.
Creates a graph that loads a WAVE file, decodes it, scales the volume,
shifts it in time, adds in background noise, calculates a spectrogram, and
then builds an MFCC fingerprint from that.
This must be called with an active TensorFlow session running, and it
creates multiple placeholder inputs, and one output:
- wav_filename_placeholder_: Filename of the WAV to load.
- foreground_volume_placeholder_: How loud the main clip should be.
- time_shift_padding_placeholder_: Where to pad the clip.
- time_shift_offset_placeholder_: How much to move the clip in time.
- background_data_placeholder_: PCM sample data for background noise.
- background_volume_placeholder_: Loudness of mixed-in background.
- mfcc_: Output 2D fingerprint of processed audio.
Args:
model_settings: Information about the current model being trained.
"""
desired_samples = model_settings['desired_samples']
self.wav_filename_placeholder_ = tf.placeholder(tf.string, [])
wav_loader = io_ops.read_file(self.wav_filename_placeholder_)
wav_decoder = contrib_audio.decode_wav(
# wav_loader, desired_channels=1, desired_samples=desired_samples)
wav_loader, desired_channels=1, desired_samples=16000)
# Allow the audio sample's volume to be adjusted.
self.foreground_volume_placeholder_ = tf.placeholder(tf.float32, [])
scaled_foreground = tf.multiply(wav_decoder.audio,
self.foreground_volume_placeholder_)
# Shift the sample's start position, and pad any gaps with zeros.
self.time_shift_padding_placeholder_ = tf.placeholder(tf.int32, [2, 2])
self.time_shift_offset_placeholder_ = tf.placeholder(tf.int32, [2])
padded_foreground = tf.pad(
scaled_foreground,
self.time_shift_padding_placeholder_,
mode='CONSTANT')
sliced_foreground = tf.slice(padded_foreground,
self.time_shift_offset_placeholder_,
[desired_samples, -1])
# Mix in background noise.
self.background_data_placeholder_ = tf.placeholder(tf.float32,
[desired_samples, 1])
self.background_volume_placeholder_ = tf.placeholder(tf.float32, [])
background_mul = tf.multiply(self.background_data_placeholder_,
self.background_volume_placeholder_)
background_add = tf.add(background_mul, sliced_foreground)
background_clamp = tf.clip_by_value(background_add, -1.0, 1.0)
###################### M F C C #################################
# Run the spectrogram and MFCC ops to get a 2D 'fingerprint' of the audio.
spectrogram = contrib_audio.audio_spectrogram(
background_clamp,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
self.mfcc_ = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=model_settings['dct_coefficient_count'])
def wav_to_features(filenames_dataset, hparams, feature_count):
dataset = filenames_dataset.map(lambda filename: io_ops.read_file(filename))
dataset = dataset.map(lambda wav_loader: contrib_audio.decode_wav(wav_loader, desired_channels=1))
dataset = dataset.map(lambda wav_decoder:
(contrib_audio.audio_spectrogram(
wav_decoder.audio,
window_size=int(hparams.sample_rate * hparams.window_size_ms / 1000),
stride=int(hparams.sample_rate * hparams.window_stride_ms / 1000),
magnitude_squared=True), wav_decoder.sample_rate))
dataset = dataset.map(lambda spectrogram, sample_rate: contrib_audio.mfcc(
spectrogram, sample_rate,
dct_coefficient_count=feature_count))
dataset = dataset.map(lambda inputs: (
inputs,
tf.nn.moments(inputs, axes=[1])
))
dataset = dataset.map(lambda inputs, moments: (
tf.divide(tf.subtract(inputs, moments[0]), moments[1]),
tf.shape(inputs)[1]
))
dataset = dataset.map(lambda inputs, seq_len: (
inputs[0],
seq_len
))
return dataset
def create_inference_graph(wanted_words, sample_rate, clip_duration_ms,
clip_stride_ms, window_size_ms, window_stride_ms,
dct_coefficient_count, model_architecture):
"""Creates an audio model with the nodes needed for inference.
Uses the supplied arguments to create a model, and inserts the input and
output nodes that are needed to use the graph for inference.
Args:
wanted_words: Comma-separated list of the words we're trying to recognize.
sample_rate: How many samples per second are in the input audio files.
clip_duration_ms: How many samples to analyze for the audio pattern.
clip_stride_ms: How often to run recognition. Useful for models with cache.
window_size_ms: Time slice duration to estimate frequencies from.
window_stride_ms: How far apart time slices should be.
dct_coefficient_count: Number of frequency bands to analyze.
model_architecture: Name of the kind of model to generate.
"""
words_list = input_data.prepare_words_list(wanted_words.split(','))
model_settings = models.prepare_model_settings(
len(words_list),
sample_rate,
clip_duration_ms,
window_size_ms,
window_stride_ms,
dct_coefficient_count,
)
runtime_settings = {'clip_stride_ms': clip_stride_ms}
wav_data_placeholder = tf.placeholder(tf.string, [], name='wav_data')
decoded_sample_data = contrib_audio.decode_wav(
wav_data_placeholder,
desired_channels=1,
desired_samples=model_settings['desired_samples'],
name='decoded_sample_data')
spectrogram = contrib_audio.audio_spectrogram(
decoded_sample_data.audio,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
fingerprint_input = contrib_audio.mfcc(
spectrogram,
decoded_sample_data.sample_rate,
dct_coefficient_count=dct_coefficient_count)
fingerprint_frequency_size = model_settings['dct_coefficient_count']
fingerprint_time_size = model_settings['spectrogram_length']
reshaped_input = tf.reshape(
fingerprint_input,
[-1, fingerprint_time_size * fingerprint_frequency_size])
logits = models.create_model(reshaped_input,
model_settings,
model_architecture,
is_training=False,
runtime_settings=runtime_settings)
# Create an output to use for inference.
tf.nn.softmax(logits, name='labels_softmax')
def prepare_processing_graph(self, model_settings):
"""Builds a TensorFlow graph to apply the input distortions.
Creates a graph that loads a WAVE file, decodes it, scales the volume,
shifts it in time, adds in background noise, calculates a spectrogram, and
then builds an MFCC fingerprint from that.
This must be called with an active TensorFlow session running, and it
creates multiple placeholder inputs, and one output:
- wav_filename_placeholder_: Filename of the WAV to load.
- foreground_volume_placeholder_: How loud the main clip should be.
- time_shift_padding_placeholder_: Where to pad the clip.
- time_shift_offset_placeholder_: How much to move the clip in time.
- background_data_placeholder_: PCM sample data for background noise.
- background_volume_placeholder_: Loudness of mixed-in background.
- mfcc_: Output 2D fingerprint of processed audio.
Args:
model_settings: Information about the current model being trained.
"""
desired_samples = model_settings['desired_samples']
self.wav_filename_placeholder_ = tf.placeholder(tf.string, [])
wav_loader = io_ops.read_file(self.wav_filename_placeholder_)
wav_decoder = contrib_audio.decode_wav(
wav_loader, desired_channels=1, desired_samples=desired_samples)
# Allow the audio sample's volume to be adjusted.
self.foreground_volume_placeholder_ = tf.placeholder(tf.float32, [])
scaled_foreground = tf.multiply(wav_decoder.audio,
self.foreground_volume_placeholder_)
# Shift the sample's start position, and pad any gaps with zeros.
self.time_shift_padding_placeholder_ = tf.placeholder(tf.int32, [2, 2])
self.time_shift_offset_placeholder_ = tf.placeholder(tf.int32, [2])
padded_foreground = tf.pad(
scaled_foreground,
self.time_shift_padding_placeholder_,
mode='CONSTANT')
sliced_foreground = tf.slice(padded_foreground,
self.time_shift_offset_placeholder_,
[desired_samples, -1])
# Mix in background noise.
self.background_data_placeholder_ = tf.placeholder(tf.float32,
[desired_samples, 1])
self.background_volume_placeholder_ = tf.placeholder(tf.float32, [])
background_mul = tf.multiply(self.background_data_placeholder_,
self.background_volume_placeholder_)
background_add = tf.add(background_mul, sliced_foreground)
background_clamp = tf.clip_by_value(background_add, -1.0, 1.0)
# Run the spectrogram and MFCC ops to get a 2D 'fingerprint' of the audio.
spectrogram = contrib_audio.audio_spectrogram(
background_clamp,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
self.mfcc_ = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=model_settings['dct_coefficient_count'])
def samples_to_mfccs(samples, sample_rate):
spectrogram = contrib_audio.audio_spectrogram(samples,
window_size=Config.audio_window_samples,
stride=Config.audio_step_samples,
magnitude_squared=True)
mfccs = contrib_audio.mfcc(spectrogram, sample_rate, dct_coefficient_count=Config.n_input)
mfccs = tf.reshape(mfccs, [-1, Config.n_input])
return mfccs, tf.shape(mfccs)[0]
def samples_to_mfccs(samples, sample_rate):
spectrogram = contrib_audio.audio_spectrogram(samples,
window_size=Config.audio_window_samples,
stride=Config.audio_step_samples,
magnitude_squared=True)
mfccs = contrib_audio.mfcc(spectrogram, sample_rate, dct_coefficient_count=Config.n_input)
mfccs = tf.reshape(mfccs, [-1, Config.n_input])
return mfccs, tf.shape(mfccs)[0]
def _load_sample(
wav_filename,
model_settings):
"""Creates an audio model with the nodes needed for inference.
Uses the supplied arguments to create a model, and inserts the input and
output nodes that are needed to use the graph for inference.
Args:
wanted_words: Comma-separated list of the words we're trying to recognize.
sample_rate: How many samples per second are in the input audio files.
clip_duration_ms: How many samples to analyze for the audio pattern.
clip_stride_ms: How often to run recognition. Useful for models with cache.
window_size_ms: Time slice duration to estimate frequencies from.
window_stride_ms: How far apart time slices should be.
dct_coefficient_count: Number of frequency bands to analyze.
model: Name of the kind of model to generate.
"""
wav_loader = io_ops.read_file(wav_filename)
decoded_sample_data = contrib_audio.decode_wav(
wav_loader,
desired_channels=1,
desired_samples=model_settings['desired_samples'],
name='decoded_sample_data')
if model_settings['input_format'] == 'raw':
print(decoded_sample_data.audio.shape)
reshaped_input = tf.reshape(decoded_sample_data.audio, [
-1, model_settings['desired_samples']
])
print(reshaped_input.shape)
else:
spectrogram = contrib_audio.audio_spectrogram(
decoded_sample_data.audio,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
fingerprint_input = contrib_audio.mfcc(
spectrogram,
decoded_sample_data.sample_rate,
lower_frequency_limit=model_settings['lower_frequency_limit'],
upper_frequency_limit=model_settings['upper_frequency_limit'],
filterbank_channel_count=model_settings['filterbank_channel_count'],
dct_coefficient_count=model_settings['dct_coefficient_count'])
fingerprint_frequency_size = model_settings['dct_coefficient_count']
fingerprint_time_size = model_settings['spectrogram_length']
reshaped_input = tf.reshape(fingerprint_input, [
-1, fingerprint_time_size * fingerprint_frequency_size
])
return reshaped_input
def create_inference_graph(wanted_words, sample_rate, clip_duration_ms,
clip_stride_ms, window_size_ms, window_stride_ms,
dct_coefficient_count, model_architecture):
"""Creates an audio model with the nodes needed for inference.
Uses the supplied arguments to create a model, and inserts the input and
output nodes that are needed to use the graph for inference.
Args:
wanted_words: Comma-separated list of the words we're trying to recognize.
sample_rate: How many samples per second are in the input audio files.
clip_duration_ms: How many samples to analyze for the audio pattern.
clip_stride_ms: How often to run recognition. Useful for models with cache.
window_size_ms: Time slice duration to estimate frequencies from.
window_stride_ms: How far apart time slices should be.
dct_coefficient_count: Number of frequency bands to analyze.
model_architecture: Name of the kind of model to generate.
"""
words_list = input_data.prepare_words_list(wanted_words.split(','))
model_settings = models.prepare_model_settings(
len(words_list), sample_rate, clip_duration_ms, window_size_ms,
window_stride_ms, dct_coefficient_count)
runtime_settings = {'clip_stride_ms': clip_stride_ms}
wav_data_placeholder = tf.placeholder(tf.string, [], name='wav_data')
decoded_sample_data = contrib_audio.decode_wav(
wav_data_placeholder,
desired_channels=1,
desired_samples=model_settings['desired_samples'],
name='decoded_sample_data')
spectrogram = contrib_audio.audio_spectrogram(
decoded_sample_data.audio,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
fingerprint_input = contrib_audio.mfcc(
spectrogram,
decoded_sample_data.sample_rate,
dct_coefficient_count=dct_coefficient_count)
fingerprint_frequency_size = model_settings['dct_coefficient_count']
fingerprint_time_size = model_settings['spectrogram_length']
reshaped_input = tf.reshape(fingerprint_input, [
-1, fingerprint_time_size * fingerprint_frequency_size
])
logits = models.create_model(
reshaped_input, model_settings, model_architecture, is_training=False,
runtime_settings=runtime_settings)
# Create an output to use for inference.
tf.nn.softmax(logits, name='labels_softmax')
def AudioToMfcc(sample_rate, audio, window_size_ms, window_stride_ms,
num_coefficients):
window_size_samples = sample_rate * window_size_ms // 1000
window_stride_samples = sample_rate * window_stride_ms // 1000
spectrogram = contrib_audio.audio_spectrogram(
audio,
window_size=window_size_samples,
stride=window_stride_samples,
magnitude_squared=True)
mfcc = contrib_audio.mfcc(spectrogram,
sample_rate,
dct_coefficient_count=num_coefficients)
return mfcc
def __init__(self, sample_rate: int, dct_coef_count: int=-1):
'''
suppose the channel number is 1.
'''
assert sample_rate == 16_000
if dct_coef_count == -1:
dct_coef_count = DataGraphMFCC.max_mfcc_num
else:
assert dct_coef_count <= DataGraphMFCC.max_mfcc_num
self._sample_rate = sample_rate
samples_per_second = sample_rate / 1000
window = int(DataGraphMFCC.window_duration * samples_per_second)
stride = int(DataGraphMFCC.stride_duration * samples_per_second)
self._graph = tf.Graph()
with self._graph.as_default():
self._in_wav_file = tf.placeholder(tf.string, [], name='wav_filename')
self._in_frame_num = tf.placeholder(tf.int32, [])
wav_loader = io_ops.read_file(self._in_wav_file)
wav_decoder = contrib_audio.decode_wav(wav_loader, desired_channels=1)
self._out_audio = tf.squeeze(wav_decoder.audio)
self._out_sample_rate = wav_decoder.sample_rate
self._in_audio = tf.placeholder(tf.float32, [None])
in_audio = tf.expand_dims(self._in_audio, -1)
audio_clamp = tf.clip_by_value(in_audio, -1.0, 1.0)
spectrogram = contrib_audio.audio_spectrogram(
audio_clamp,
window_size=window,
stride=stride,
magnitude_squared=True)
self._out_spectrogram = spectrogram
feat_ts = contrib_audio.mfcc(
spectrogram=spectrogram,
sample_rate=sample_rate,
dct_coefficient_count=dct_coef_count,
)
self._out_mfcc = feat_ts[0]
self._out_real_mfcc_len = tf.shape(self._out_mfcc)[0]
diff = tf.maximum(0, self._in_frame_num - self._out_real_mfcc_len)
self._out_expanded_mfcc = tf.pad(
self._out_mfcc,
[[0, diff], [0, 0]],
)[: self._in_frame_num]
self._sess = tf.Session(graph=self._graph)
print(f"DataGgraphMFCC graph is created!")
def build_graph(self):
""" Graph to extract mfcc fingerprint given wav file
Here we add the necessary input & output tensors, to decode wav,
serialize mfcc fingerprint, restore from checkpoint etc.
Returns:
input_wav_filename: A tensor containing wav filename as the input layer.
mfcc_fingerprint: The MFCC fingerprint tensor, that will be materialized later.
"""
self.wav_filename_placeholder_ = tf.placeholder(tf.string, [])
wav_loader = io_ops.read_file(self.wav_filename_placeholder_)
wav_decoder = contrib_audio.decode_wav(
wav_loader,
desired_channels=1,
desired_samples=self.desired_samples)
# Allow the audio sample's volume to be adjusted.
self.foreground_volume_placeholder_ = tf.placeholder(tf.float32, [])
scaled_foreground = tf.multiply(wav_decoder.audio,
self.foreground_volume_placeholder_)
# Shift the sample's start position, and pad any gaps with zeros.
self.time_shift_padding_placeholder_ = tf.placeholder(tf.int32, [2, 2])
self.time_shift_offset_placeholder_ = tf.placeholder(tf.int32, [2])
padded_foreground = tf.pad(scaled_foreground,
self.time_shift_padding_placeholder_,
mode='CONSTANT')
sliced_foreground = tf.slice(padded_foreground,
self.time_shift_offset_placeholder_,
[self.desired_samples, -1])
# Mix in background noise.
self.background_data_placeholder_ = tf.placeholder(
tf.float32, [self.desired_samples, 1])
self.background_volume_placeholder_ = tf.placeholder(tf.float32, [])
background_mul = tf.multiply(self.background_data_placeholder_,
self.background_volume_placeholder_)
background_add = tf.add(background_mul, sliced_foreground)
background_clamp = tf.clip_by_value(background_add, -1.0, 1.0)
spectrogram = contrib_audio.audio_spectrogram(
background_clamp,
window_size=self.window_size_samples,
stride=self.window_stride_samples,
magnitude_squared=True)
self.mfcc_fingerprint_ = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=self.opt.dct_coefficient_count)
def _single_mfcc(self, audio, window_size_samples, window_stride_samples,
magnitude_squared, **kwargs):
spectrogram = self._single_spectrogram(audio, window_size_samples,
window_stride_samples,
magnitude_squared)
mfcc = contrib_audio.mfcc(
spectrogram,
kwargs["sample_rate_const"],
upper_frequency_limit=kwargs["upper_edge_hertz"],
lower_frequency_limit=kwargs["lower_edge_hertz"],
filterbank_channel_count=kwargs["num_mel_bins"],
dct_coefficient_count=kwargs["num_mfccs"],
)
return mfcc
def encode_data(audio_data, sample_rate):
spectrogram = contrib_audio.audio_spectrogram(
audio_data,
window_size_samples,
window_stride_samples
)
print(spectrogram.shape)
mfcc = contrib_audio.mfcc(
spectrogram,
sample_rate=sample_rate,
dct_coefficient_count=dct_coefficient_count
)
return mfcc
def wav_to_mfcc(self, raw_data):
spectrogram = audio_ops.audio_spectrogram(
raw_data,
window_size=self.parameters['spectogram_window_size'],
stride=self.parameters['spectogram_stride'],
magnitude_squared=True)
mfcc = audio_ops.mfcc(
spectrogram,
self.parameters['audio_sample_rate'],
dct_coefficient_count=self.parameters['dtc_coefficient_count'])
mfcc = tf.expand_dims(mfcc, -1)
self.input_dimensions = (
self.input_dimensions[0] / self.parameters['spectogram_stride'] -
2, self.parameters['dtc_coefficient_count'], 1)
mfcc = tf.squeeze(mfcc, 0)
return mfcc
def prepare_processing_graph(self, model_settings):
"""
Build a tensorflow graph
creates a graph that loads a wave file, decodes it, scales the volume, shifts it in time
calculates a spectrogram and builds MFCC fingerprint from that
input:
model_settings: info about model being trained
"""
desired_samples = model_settings['desired_samples']
self.wav_filename_placeholder_ = tf.placeholder(tf.string, [])
wav_loader = io_ops.read_file(self.wav_filename_placeholder_)
wav_decoder = contrib_audio.decode_wav(wav_loader,
desired_channels=1,
desired_samples=desired_samples)
# Allow the audio sample's volume to be adjusted.
self.foreground_volume_placeholder_ = tf.placeholder(tf.float32, [])
scaled_foreground = tf.multiply(wav_decoder.audio,
self.foreground_volume_placeholder_)
# Shift the sample's start position, and pad any gaps with zeros.
self.time_shift_padding_placeholder_ = tf.placeholder(tf.int32, [2, 2])
self.time_shift_offset_placeholder_ = tf.placeholder(tf.int32, [2])
padded_foreground = tf.pad(scaled_foreground,
self.time_shift_padding_placeholder_,
mode='CONSTANT')
sliced_foreground = tf.slice(padded_foreground,
self.time_shift_offset_placeholder_,
[desired_samples, -1])
self.background_data_placeholder_ = tf.placeholder(
np.float32, [desired_samples, 1])
self.background_volume_placeholder_ = tf.placeholder(np.float32, [])
background_mul = tf.multiply(self.background_data_placeholder_,
self.background_volume_placeholder_)
background_add = tf.add(background_mul, sliced_foreground)
background_clamp = tf.clip_by_value(background_add, -1.0, 1.0)
spectrogram = contrib_audio.audio_spectrogram(
background_clamp,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
self.mfcc_ = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=model_settings['dct_coefficient_count'])
def create_inference_graph_and_load_variables(sess, FLAGS):
"""Creates an audio model with the nodes needed for inference.
Uses the supplied arguments to create a model, and inserts the input and
output the trained model graph.
"""
model_settings = data_utils.prepare_settings(FLAGS.num_classes,
FLAGS.sample_rate,
FLAGS.clip_duration_ms,
FLAGS.window_size_ms,
FLAGS.window_stride_ms,
FLAGS.dct_coefficient_count)
runtime_settings = {'clip_stride_ms': FLAGS.clip_stride_ms}
wav_data_placeholder = tf.placeholder(tf.string, [], name='wav_data')
decoded_sample_data = contrib_audio.decode_wav(
wav_data_placeholder,
desired_channels=1,
desired_samples=model_settings['desired_samples'],
name='decoded_sample_data')
spectrogram = contrib_audio.audio_spectrogram(
decoded_sample_data.audio,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
fingerprint_input = contrib_audio.mfcc(
spectrogram,
decoded_sample_data.sample_rate,
dct_coefficient_count=FLAGS.dct_coefficient_count)
fingerprint_frequency_size = model_settings['dct_coefficient_count']
fingerprint_time_size = model_settings['spectrogram_length']
reshaped_input = tf.reshape(
fingerprint_input,
[-1, fingerprint_time_size, fingerprint_frequency_size, 1],
name="model_input")
# Init model and load variables
model = models.create_model(FLAGS)
fw = framework.Framework(sess,
model,
None,
FLAGS,
input_tensor=reshaped_input)
# Create an output to use for inference
logits = tf.nn.softmax(model.get_raw_scores(), name='labels_softmax')
def save_my_test_file(self, model_settings):
desired_samples = model_settings['desired_samples']
self.wav_filename_placeholder_ = tf.placeholder(tf.string, [])
wav_loader = io_ops.read_file(self.wav_filename_placeholder_)
wav_decoder = contrib_audio.decode_wav(
# wav_loader, desired_channels=1, desired_samples=desired_samples)
wav_loader, desired_channels=1, desired_samples=16000)
# Allow the audio sample's volume to be adjusted.
self.foreground_volume_placeholder_ = tf.placeholder(tf.float32, [])
scaled_foreground = tf.multiply(wav_decoder.audio,
self.foreground_volume_placeholder_)
# Shift the sample's start position, and pad any gaps with zeros.
self.time_shift_padding_placeholder_ = tf.placeholder(tf.int32, [2, 2])
self.time_shift_offset_placeholder_ = tf.placeholder(tf.int32, [2])
padded_foreground = tf.pad(
scaled_foreground,
self.time_shift_padding_placeholder_,
mode='CONSTANT')
sliced_foreground = tf.slice(padded_foreground,
self.time_shift_offset_placeholder_,
[desired_samples, -1])
# Mix in background noise.
self.background_data_placeholder_ = tf.placeholder(tf.float32,
[desired_samples, 1])
self.background_volume_placeholder_ = tf.placeholder(tf.float32, [])
background_mul = tf.multiply(self.background_data_placeholder_,
self.background_volume_placeholder_)
background_add = tf.add(background_mul, sliced_foreground)
background_clamp = tf.clip_by_value(background_add, -1.0, 1.0)
self.filename_ = tf.placeholder(tf.string)
# save_wav_file(self.filename_, np.array(background_clamp), 16000)
wav_encoder = contrib_audio.encode_wav(background_clamp,
16000)
self.wav_saver = io_ops.write_file(self.filename_, wav_encoder)
# with tf.Session(graph=tf.Graph()) as sess:
# sess.run(wav_saver)
spectrogram = contrib_audio.audio_spectrogram(
background_clamp,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
self.test_mfcc_ = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=model_settings['dct_coefficient_count'])
def get_mfcc_graph(model_settings):
g = tf.Graph()
with g.as_default():
input_file_placeholder = tf.compat.v1.placeholder(tf.string, [],
name='wav_filename')
wav_loader = io_ops.read_file(input_file_placeholder)
wav_decoder = audio_ops.decode_wav(
wav_loader,
desired_channels=1,
desired_samples=model_settings['desired_samples'])
# Run the spectrogram and MFCC ops to get a 2D 'fingerprint' of the audio.
spectrograms_power = audio_ops.audio_spectrogram(
wav_decoder.audio,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
USE_POWER = True
if USE_POWER:
# Warp the linear scale spectrograms into the mel-scale.
num_spectrogram_bins = spectrograms_power.shape[-1].value
lower_edge_hertz, upper_edge_hertz, num_mel_bins = 20.0, 4000.0, 40
linear_to_mel_weight_matrix = tf.signal.linear_to_mel_weight_matrix(
num_mel_bins, num_spectrogram_bins, 16000.0, lower_edge_hertz,
upper_edge_hertz)
mel_spectrograms = tf.tensordot(spectrograms_power,
linear_to_mel_weight_matrix, 1)
mel_spectrograms.set_shape(
spectrograms_power.shape[:-1].concatenate(
linear_to_mel_weight_matrix.shape[-1:]))
# Compute a stabilized log to get log-magnitude mel-scale spectrograms.
log_mel_spectrograms = tf.math.log(mel_spectrograms + 1e-6)
# Compute MFCCs from log_mel_spectrograms and take the first NDCT.
mfccs = tf.signal.mfccs_from_log_mel_spectrograms(
log_mel_spectrograms)[
..., :model_settings['dct_coefficient_count']]
#output = tf.expand_dims(mfccs, axis=0)
output = mfccs
else:
output = audio_ops.mfcc(
spectrograms_power,
wav_decoder.sample_rate,
dct_coefficient_count=model_settings['dct_coefficient_count'])
return g, input_file_placeholder, output, wav_decoder.audio
def prepare_processing_graph(self, model_settings):
desired_samples = model_settings['desired_samples']
self.wav_filename_placeholder_ = tf.placeholder(tf.string, [])
wav_loader = io_ops.read_file(self.wav_filename_placeholder_)
wav_decoder = contrib_audio.decode_wav(wav_loader,
desired_channels=1,
desired_samples=desired_samples)
# Allow the audio sample's volume to be adjusted.
self.foreground_volume_placeholder_ = tf.placeholder(tf.float32, [])
scaled_foreground = tf.multiply(wav_decoder.audio,
self.foreground_volume_placeholder_)
# Shift the sample's start position, and pad any gaps with zeros.
self.time_shift_padding_placeholder_ = tf.placeholder(tf.int32, [2, 2])
self.time_shift_offset_placeholder_ = tf.placeholder(tf.int32, [2])
padded_foreground = tf.pad(scaled_foreground,
self.time_shift_padding_placeholder_,
mode='CONSTANT')
sliced_foreground = tf.slice(padded_foreground,
self.time_shift_offset_placeholder_,
[desired_samples, -1])
# Mix in background noise.
self.background_data_placeholder_ = tf.placeholder(
tf.float32, [desired_samples, 1])
self.background_volume_placeholder_ = tf.placeholder(tf.float32, [])
background_mul = tf.multiply(self.background_data_placeholder_,
self.background_volume_placeholder_)
background_add = tf.add(background_mul, sliced_foreground)
background_clamp = tf.clip_by_value(background_add, -1.0, 1.0)
# Run the spectrogram and MFCC ops to get a 2D 'fingerprint' of the audio.
print 'window_size_samples', model_settings['window_size_samples']
print 'window_stride_samples', model_settings['window_stride_samples']
print 'background_clamp', background_clamp
spectrogram = contrib_audio.audio_spectrogram(
background_clamp,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
print 'spectrogram', spectrogram
print 'dct_coefficient_count', model_settings['dct_coefficient_count']
print 'wav_decoder.sample_rate', wav_decoder.sample_rate
self.mfcc_ = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=model_settings['dct_coefficient_count'])
print 'self.mfcc_', self.mfcc_
def __init__(self,
desired_samples=16000,
window_size_samples=480,
window_stride_samples=160):
self.wav_filename_placeholder = tf.placeholder(tf.string, [])
wav_loader = io_ops.read_file(self.wav_filename_placeholder)
# already pads/crops
wav_decoder = contrib_audio.decode_wav(wav_loader,
desired_channels=1,
desired_samples=desired_samples)
spectrogram = contrib_audio.audio_spectrogram(
wav_decoder.audio,
window_size=window_size_samples,
stride=window_stride_samples,
magnitude_squared=True)
self.mfcc = contrib_audio.mfcc(spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=40)
def build_data_generator():
# Build data generator pipeline
desired_samples = model_settings['desired_samples']
wav_filename_placeholder_ = tf.placeholder(
tf.string, [], name="wav_filename_placeholder_")
wav_loader = io_ops.read_file(wav_filename_placeholder_)
wav_decoder = contrib_audio.decode_wav(wav_loader,
desired_channels=1,
desired_samples=desired_samples)
# Allow the audio sample's volume to be adjusted.
foreground_volume_placeholder_ = tf.placeholder(
tf.float32, [], name="foreground_volume_placeholder_")
scaled_foreground = tf.multiply(wav_decoder.audio,
foreground_volume_placeholder_)
# Shift the sample's start position, and pad any gaps with zeros.
time_shift_padding_placeholder_ = tf.placeholder(
tf.int32, [2, 2], name="time_shift_padding_placeholder_")
time_shift_offset_placeholder_ = tf.placeholder(tf.int32, [2])
padded_foreground = tf.pad(scaled_foreground,
time_shift_padding_placeholder_,
mode='CONSTANT')
sliced_foreground = tf.slice(padded_foreground,
time_shift_offset_placeholder_,
[desired_samples, -1])
# Mix in background noise.
background_data_placeholder_ = tf.placeholder(tf.float32,
[desired_samples, 1])
background_volume_placeholder_ = tf.placeholder(tf.float32, [])
background_mul = tf.multiply(background_data_placeholder_,
background_volume_placeholder_)
background_add = tf.add(background_mul, sliced_foreground)
background_clamp = tf.clip_by_value(background_add, -1.0, 1.0)
# Run the spectrogram and MFCC ops to get a 2D 'fingerprint' of the audio.
spectrogram = contrib_audio.audio_spectrogram(
background_clamp,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
mfcc_ = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=model_settings['dct_coefficient_count'])
return wav_filename_placeholder_, foreground_volume_placeholder_, time_shift_padding_placeholder_, time_shift_offset_placeholder_, background_data_placeholder_, background_volume_placeholder_, mfcc_
def decode_audio(audio_str):
wav_decoder = contrib_audio.decode_wav(
audio_str,
desired_channels=1,
desired_samples=self.desired_samples)
spectrogram = contrib_audio.audio_spectrogram(
wav_decoder.audio,
window_size=self.window_size_samples,
stride=self.window_stride_samples,
magnitude_squared=True)
mfcc_fingerprint = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=self.dct_coefficient_count)
return mfcc_fingerprint
def prepare_processing_graph(self):
self.wav_filename_placeholder_ = tf.placeholder(tf.string, [])
wav_loader = io_ops.read_file(self.wav_filename_placeholder_)
wav_decoder = contrib_audio.decode_wav(wav_loader,
desired_channels=1,
desired_samples=desired_samples)
spectrogram = contrib_audio.audio_spectrogram(
wav_decoder.audio,
window_size=window_size_samples,
stride=window_stride_samples,
magnitude_squared=True)
print 'spectrogram', spectrogram
print 'dct_coefficient_count', dct_coefficient_count
print 'wav_decoder.sample_rate', wav_decoder.sample_rate
self.mfcc_ = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=dct_coefficient_count)
print 'self.mfcc_', self.mfcc_
def create_inference_graph(wanted_words, sample_rate, clip_duration_ms,
clip_stride_ms, window_size_ms, window_stride_ms,
dct_coefficient_count, model_architecture):
graph = tf.Graph()
with graph.as_default():
words_list = input_data.prepare_words_list(wanted_words.split(','))
model_settings = models.prepare_model_settings(
len(words_list), sample_rate, clip_duration_ms, window_size_ms,
window_stride_ms, dct_coefficient_count)
runtime_settings = {'clip_stride_ms': clip_stride_ms}
wav_data_placeholder = tf.placeholder(tf.string, [], name='wav_data')
decoded_sample_data = contrib_audio.decode_wav(
wav_data_placeholder,
desired_channels=1,
desired_samples=model_settings['desired_samples'],
name='decoded_sample_data')
spectrogram = contrib_audio.audio_spectrogram(
decoded_sample_data.audio,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
fingerprint_input = contrib_audio.mfcc(
spectrogram,
decoded_sample_data.sample_rate,
dct_coefficient_count=dct_coefficient_count)
fingerprint_frequency_size = model_settings['dct_coefficient_count']
fingerprint_time_size = model_settings['spectrogram_length']
reshaped_input = tf.reshape(
fingerprint_input,
[-1, fingerprint_time_size * fingerprint_frequency_size])
logits = models.create_model(reshaped_input,
model_settings,
model_architecture,
is_training=False,
runtime_settings=runtime_settings)
# Create an output to use for inference.
tf.nn.softmax(logits, name='labels_softmax')
return graph
def create_decoder_graph(): #may need to pass in session
"""Creates the input of the CNN model based off of this paper https://www.isca-speech.org/archive/interspeech_2015/papers/i15_1478.pdf
Returns:
input node and output node
"""
words_list = prepare_words_list(FLAGS.wanted_words.split(','))
model_settings = prepare_model_settings(len(words_list), FLAGS.sample_rate,
FLAGS.clip_duration_ms,
FLAGS.window_size_ms,
FLAGS.window_stride_ms,
FLAGS.dct_coefficient_count)
runtime_settings = {'clip_stride_ms': FLAGS.clip_stride_ms}
wav_data_placeholder = tf.placeholder(tf.string, [], name='wav_data')
decoded_sample_data = contrib_audio.decode_wav(
wav_data_placeholder,
desired_channels=1,
desired_samples=model_settings['desired_samples'],
name='decoded_sample_data')
spectrogram = contrib_audio.audio_spectrogram(
decoded_sample_data.audio,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
fingerprint_input = contrib_audio.mfcc(
spectrogram,
decoded_sample_data.sample_rate,
dct_coefficient_count=FLAGS.dct_coefficient_count)
fingerprint_frequency_size = model_settings['dct_coefficient_count']
fingerprint_time_size = model_settings['spectrogram_length']
reshaped_input = tf.reshape(
fingerprint_input,
[-1, fingerprint_time_size * fingerprint_frequency_size])
input_frequency_size = model_settings['dct_coefficient_count']
input_time_size = model_settings['spectrogram_length']
fingerprint_4d = tf.reshape(reshaped_input,
[-1, input_time_size, input_frequency_size, 1])
return wav_data_placeholder, fingerprint_4d
def process_audio(audio_data):
with tf.Session() as sess:
data_input = tf.compat.v1.placeholder(dtype=tf.float32,
shape=(16000, 1),
name=None)
spectrum = audio_ops.audio_spectrogram(input=data_input,
window_size=640,
stride=320,
magnitude_squared=True,
name='AudioSpectrogram')
final = audio_ops.mfcc(spectrogram=spectrum,
sample_rate=RATE,
upper_frequency_limit=4000.0,
lower_frequency_limit=20.0,
filterbank_channel_count=40,
dct_coefficient_count=10,
name='Mfcc')
data_out = sess.run(final, feed_dict={data_input: audio_data})
return data_out
def Features(conf):
sound = tf.placeholder(tf.float32, [None, None], name='wav')
# Run the spectrogram and MFCC ops to get a 2D 'fingerprint' of the audio.
spectrogram = contrib_audio.audio_spectrogram(
sound,
window_size=conf['window_size_samples'],
stride=conf['window_stride_samples'],
magnitude_squared=True,
name='spectogram')
mfcc = contrib_audio.mfcc(
spectrogram,
conf['sample_rate'],
dct_coefficient_count=conf['dct_coefficient_count'],
name='mfcc')
spect_norm = spectrogram / tf.reduce_sum(spectrogram, [1, 2])
mfcc_norm = mfcc / tf.reduce_sum(mfcc, [1, 2])
return sound, spectrogram, mfcc, spect_norm, mfcc_norm
def wav2Input(filename, model_settings):
#print('wav2Input ', filename)
desired_samples = model_settings['desired_samples']
foreground_volume = 1
time_shift_padding = [[0, 0], [0, 0]]
time_shift_offset = [0, 0]
#background_data=0
#background_volume=0
wav_loader = io_ops.read_file(filename)
wav_decoder = contrib_audio.decode_wav(wav_loader,
desired_channels=1,
desired_samples=desired_samples)
#scaled_foreground = tf.multiply(wav_decoder.audio, foreground_volume)
#padded_foreground = tf.pad(scaled_foreground, time_shift_padding, mode='CONSTANT')
#sliced_foreground = tf.slice(padded_foreground, time_shift_offset, [desired_samples, -1])
#background_mul = tf.multiply(background_data, background_volume)
#background_add = tf.add(background_mul, sliced_foreground)
background_clamp = tf.clip_by_value(wav_decoder.audio, -1.0, 1.0)
spectrogram = contrib_audio.audio_spectrogram(
background_clamp,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
mfcc = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=model_settings['dct_coefficient_count'])
mfcc = tf.clip_by_value(mfcc, -10.0, 127.0)
with tf.Session() as sess:
mfcc = sess.run(mfcc)
return mfcc
def create_inference_graph(wanted_words, sample_rate, clip_duration_ms,
clip_stride_ms, window_size_ms, window_stride_ms,
feature_bin_count, model_architecture, preprocess):
"""Creates an audio model with the nodes needed for inference.
Uses the supplied arguments to create a model, and inserts the input and
output nodes that are needed to use the graph for inference.
Args:
wanted_words: Comma-separated list of the words we're trying to recognize.
sample_rate: How many samples per second are in the input audio files.
clip_duration_ms: How many samples to analyze for the audio pattern.
clip_stride_ms: How often to run recognition. Useful for models with cache.
window_size_ms: Time slice duration to estimate frequencies from.
window_stride_ms: How far apart time slices should be.
feature_bin_count: Number of frequency bands to analyze.
model_architecture: Name of the kind of model to generate.
preprocess: How the spectrogram is processed to produce features, for
example 'mfcc' or 'average'.
Raises:
Exception: If the preprocessing mode isn't recognized.
"""
words_list = input_data.prepare_words_list(wanted_words.split(','))
model_settings = models.prepare_model_settings(
len(words_list), sample_rate, clip_duration_ms, window_size_ms,
window_stride_ms, feature_bin_count, preprocess)
runtime_settings = {'clip_stride_ms': clip_stride_ms}
wav_data_placeholder = tf.placeholder(tf.string, [], name='wav_data')
decoded_sample_data = contrib_audio.decode_wav(
wav_data_placeholder,
desired_channels=1,
desired_samples=model_settings['desired_samples'],
name='decoded_sample_data')
spectrogram = contrib_audio.audio_spectrogram(
decoded_sample_data.audio,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
if preprocess == 'average':
fingerprint_input = tf.nn.pool(
tf.expand_dims(spectrogram, -1),
window_shape=[1, model_settings['average_window_width']],
strides=[1, model_settings['average_window_width']],
pooling_type='AVG',
padding='SAME')
elif preprocess == 'mfcc':
fingerprint_input = contrib_audio.mfcc(
spectrogram,
sample_rate,
dct_coefficient_count=model_settings['fingerprint_width'])
else:
raise Exception('Unknown preprocess mode "%s" (should be "mfcc" or'
' "average")' % (preprocess))
fingerprint_size = model_settings['fingerprint_size']
reshaped_input = tf.reshape(fingerprint_input, [-1, fingerprint_size])
logits = models.create_model(
reshaped_input, model_settings, model_architecture, is_training=False,
runtime_settings=runtime_settings)
# Create an output to use for inference.
tf.nn.softmax(logits, name='labels_softmax')
def create_inference_graph(wanted_words, sample_rate, clip_duration_ms,
clip_stride_ms, window_size_ms, window_stride_ms,
feature_bin_count, model_architecture, preprocess):
"""Creates an audio model with the nodes needed for inference.
Uses the supplied arguments to create a model, and inserts the input and
output nodes that are needed to use the graph for inference.
Args:
wanted_words: Comma-separated list of the words we're trying to recognize.
sample_rate: How many samples per second are in the input audio files.
clip_duration_ms: How many samples to analyze for the audio pattern.
clip_stride_ms: How often to run recognition. Useful for models with cache.
window_size_ms: Time slice duration to estimate frequencies from.
window_stride_ms: How far apart time slices should be.
feature_bin_count: Number of frequency bands to analyze.
model_architecture: Name of the kind of model to generate.
preprocess: How the spectrogram is processed to produce features, for
example 'mfcc', 'average', or 'micro'.
Raises:
Exception: If the preprocessing mode isn't recognized.
"""
words_list = input_data.prepare_words_list(wanted_words.split(','))
model_settings = models.prepare_model_settings(
len(words_list), sample_rate, clip_duration_ms, window_size_ms,
window_stride_ms, feature_bin_count, preprocess)
runtime_settings = {'clip_stride_ms': clip_stride_ms}
wav_data_placeholder = tf.placeholder(tf.string, [], name='wav_data')
decoded_sample_data = contrib_audio.decode_wav(
wav_data_placeholder,
desired_channels=1,
desired_samples=model_settings['desired_samples'],
name='decoded_sample_data')
spectrogram = contrib_audio.audio_spectrogram(
decoded_sample_data.audio,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
if preprocess == 'average':
fingerprint_input = tf.nn.pool(
tf.expand_dims(spectrogram, -1),
window_shape=[1, model_settings['average_window_width']],
strides=[1, model_settings['average_window_width']],
pooling_type='AVG',
padding='SAME')
elif preprocess == 'mfcc':
fingerprint_input = contrib_audio.mfcc(
spectrogram,
sample_rate,
dct_coefficient_count=model_settings['fingerprint_width'])
elif preprocess == 'micro':
if not frontend_op:
raise Exception(
'Micro frontend op is currently not available when running TensorFlow'
' directly from Python, you need to build and run through Bazel, for'
' example'
' `bazel run tensorflow/examples/speech_commands:freeze_graph`'
)
sample_rate = model_settings['sample_rate']
window_size_ms = (model_settings['window_size_samples'] *
1000) / sample_rate
window_step_ms = (model_settings['window_stride_samples'] *
1000) / sample_rate
int16_input = tf.cast(
tf.multiply(decoded_sample_data.audio, 32767), tf.int16)
micro_frontend = frontend_op.audio_microfrontend(
int16_input,
sample_rate=sample_rate,
window_size=window_size_ms,
window_step=window_step_ms,
num_channels=model_settings['fingerprint_width'],
out_scale=1,
out_type=tf.float32)
fingerprint_input = tf.multiply(micro_frontend, (10.0 / 256.0))
else:
raise Exception('Unknown preprocess mode "%s" (should be "mfcc",'
' "average", or "micro")' % (preprocess))
fingerprint_size = model_settings['fingerprint_size']
reshaped_input = tf.reshape(fingerprint_input, [-1, fingerprint_size])
logits = models.create_model(
reshaped_input, model_settings, model_architecture, is_training=False,
runtime_settings=runtime_settings)
# Create an output to use for inference.
tf.nn.softmax(logits, name='labels_softmax')
def prepare_processing_graph(self, model_settings, summaries_dir):
"""Builds a TensorFlow graph to apply the input distortions.
Creates a graph that loads a WAVE file, decodes it, scales the volume,
shifts it in time, adds in background noise, calculates a spectrogram, and
then builds an MFCC fingerprint from that.
This must be called with an active TensorFlow session running, and it
creates multiple placeholder inputs, and one output:
- wav_filename_placeholder_: Filename of the WAV to load.
- foreground_volume_placeholder_: How loud the main clip should be.
- time_shift_padding_placeholder_: Where to pad the clip.
- time_shift_offset_placeholder_: How much to move the clip in time.
- background_data_placeholder_: PCM sample data for background noise.
- background_volume_placeholder_: Loudness of mixed-in background.
- output_: Output 2D fingerprint of processed audio.
Args:
model_settings: Information about the current model being trained.
summaries_dir: Path to save training summary information to.
Raises:
ValueError: If the preprocessing mode isn't recognized.
"""
with tf.get_default_graph().name_scope('data'):
desired_samples = model_settings['desired_samples']
self.wav_filename_placeholder_ = tf.placeholder(
tf.string, [], name='wav_filename')
wav_loader = io_ops.read_file(self.wav_filename_placeholder_)
wav_decoder = contrib_audio.decode_wav(
wav_loader, desired_channels=1, desired_samples=desired_samples)
# Allow the audio sample's volume to be adjusted.
self.foreground_volume_placeholder_ = tf.placeholder(
tf.float32, [], name='foreground_volume')
scaled_foreground = tf.multiply(wav_decoder.audio,
self.foreground_volume_placeholder_)
# Shift the sample's start position, and pad any gaps with zeros.
self.time_shift_padding_placeholder_ = tf.placeholder(
tf.int32, [2, 2], name='time_shift_padding')
self.time_shift_offset_placeholder_ = tf.placeholder(
tf.int32, [2], name='time_shift_offset')
padded_foreground = tf.pad(
scaled_foreground,
self.time_shift_padding_placeholder_,
mode='CONSTANT')
sliced_foreground = tf.slice(padded_foreground,
self.time_shift_offset_placeholder_,
[desired_samples, -1])
# Mix in background noise.
self.background_data_placeholder_ = tf.placeholder(
tf.float32, [desired_samples, 1], name='background_data')
self.background_volume_placeholder_ = tf.placeholder(
tf.float32, [], name='background_volume')
background_mul = tf.multiply(self.background_data_placeholder_,
self.background_volume_placeholder_)
background_add = tf.add(background_mul, sliced_foreground)
background_clamp = tf.clip_by_value(background_add, -1.0, 1.0)
# Run the spectrogram and MFCC ops to get a 2D 'fingerprint' of the audio.
spectrogram = contrib_audio.audio_spectrogram(
background_clamp,
window_size=model_settings['window_size_samples'],
stride=model_settings['window_stride_samples'],
magnitude_squared=True)
tf.summary.image(
'spectrogram', tf.expand_dims(spectrogram, -1), max_outputs=1)
# The number of buckets in each FFT row in the spectrogram will depend on
# how many input samples there are in each window. This can be quite
# large, with a 160 sample window producing 127 buckets for example. We
# don't need this level of detail for classification, so we often want to
# shrink them down to produce a smaller result. That's what this section
# implements. One method is to use average pooling to merge adjacent
# buckets, but a more sophisticated approach is to apply the MFCC
# algorithm to shrink the representation.
if model_settings['preprocess'] == 'average':
self.output_ = tf.nn.pool(
tf.expand_dims(spectrogram, -1),
window_shape=[1, model_settings['average_window_width']],
strides=[1, model_settings['average_window_width']],
pooling_type='AVG',
padding='SAME')
tf.summary.image('shrunk_spectrogram', self.output_, max_outputs=1)
elif model_settings['preprocess'] == 'mfcc':
self.output_ = contrib_audio.mfcc(
spectrogram,
wav_decoder.sample_rate,
dct_coefficient_count=model_settings['fingerprint_width'])
tf.summary.image(
'mfcc', tf.expand_dims(self.output_, -1), max_outputs=1)
else:
raise ValueError('Unknown preprocess mode "%s" (should be "mfcc" or'
' "average")' % (model_settings['preprocess']))
# Merge all the summaries and write them out to /tmp/retrain_logs (by
# default)
self.merged_summaries_ = tf.summary.merge_all(scope='data')
self.summary_writer_ = tf.summary.FileWriter(summaries_dir + '/data',
tf.get_default_graph())
