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PaddleSpeech
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README.md
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# Deep Speech 2 on PaddlePaddle
## Installation
Please replace `$PADDLE_INSTALL_DIR` with your own paddle installation directory.
```
sh setup.sh
export LD_LIBRARY_PATH=$PADDLE_INSTALL_DIR/Paddle/third_party/install/warpctc/lib:$LD_LIBRARY_PATH
```
For some machines, we also need to install libsndfile1. Details to be added.
## Usage
### Preparing Data
```
cd datasets
sh run_all.sh
cd ..
```
`sh run_all.sh` prepares all ASR datasets (currently, only LibriSpeech available). After running, we have several summarization manifest files in json-format.
A manifest file summarizes a speech data set, with each line containing the meta data (i.e. audio filepath, transcript text, audio duration) of each audio file within the data set, in json format. Manifest file serves as an interface informing our system of where and what to read the speech samples.
More help for arguments:
```
python datasets/librispeech/librispeech.py --help
```
### Preparing for Training
```
python compute_mean_std.py
```
`python compute_mean_std.py` computes mean and stdandard deviation for audio features, and save them to a file with a default name `./mean_std.npz`. This file will be used in both training and inferencing.
More help for arguments:
```
python compute_mean_std.py --help
```
### Training
For GPU Training:
```
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 python train.py
```
For CPU Training:
```
python train.py --use_gpu False
```
More help for arguments:
```
python train.py --help
```
### Inferencing
```
CUDA_VISIBLE_DEVICES=0 python infer.py
```
More help for arguments:
```
python infer.py --help
```

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compute_mean_std.py
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"""Compute mean and std for feature normalizer, and save to file."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import argparse
from data_utils.normalizer import FeatureNormalizer
from data_utils.augmentor.augmentation import AugmentationPipeline
from data_utils.featurizer.audio_featurizer import AudioFeaturizer
parser = argparse.ArgumentParser(
description='Computing mean and stddev for feature normalizer.')
parser.add_argument(
"--manifest_path",
default='datasets/manifest.train',
type=str,
help="Manifest path for computing normalizer's mean and stddev."
"(default: %(default)s)")
parser.add_argument(
"--num_samples",
default=2000,
type=int,
help="Number of samples for computing mean and stddev. "
"(default: %(default)s)")
parser.add_argument(
"--augmentation_config",
default='{}',
type=str,
help="Augmentation configuration in json-format. "
"(default: %(default)s)")
parser.add_argument(
"--output_file",
default='mean_std.npz',
type=str,
help="Filepath to write mean and std to (.npz)."
"(default: %(default)s)")
args = parser.parse_args()
def main():
augmentation_pipeline = AugmentationPipeline(args.augmentation_config)
audio_featurizer = AudioFeaturizer()
def augment_and_featurize(audio_segment):
augmentation_pipeline.transform_audio(audio_segment)
return audio_featurizer.featurize(audio_segment)
normalizer = FeatureNormalizer(
mean_std_filepath=None,
manifest_path=args.manifest_path,
featurize_func=augment_and_featurize,
num_samples=args.num_samples)
normalizer.write_to_file(args.output_file)
if __name__ == '__main__':
main()

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data_utils/__init__.py
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data_utils/audio.py
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"""Contains the audio segment class."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import numpy as np
import io
import soundfile
import resampy
from scipy import signal
import random
import copy
class AudioSegment(object):
"""Monaural audio segment abstraction.
:param samples: Audio samples [num_samples x num_channels].
:type samples: ndarray.float32
:param sample_rate: Audio sample rate.
:type sample_rate: int
:raises TypeError: If the sample data type is not float or int.
"""
def __init__(self, samples, sample_rate):
"""Create audio segment from samples.
Samples are convert float32 internally, with int scaled to [-1, 1].
"""
self._samples = self._convert_samples_to_float32(samples)
self._sample_rate = sample_rate
if self._samples.ndim >= 2:
self._samples = np.mean(self._samples, 1)
def __eq__(self, other):
"""Return whether two objects are equal."""
if type(other) is not type(self):
return False
if self._sample_rate != other._sample_rate:
return False
if self._samples.shape != other._samples.shape:
return False
if np.any(self.samples != other._samples):
return False
return True
def __ne__(self, other):
"""Return whether two objects are unequal."""
return not self.__eq__(other)
def __str__(self):
"""Return human-readable representation of segment."""
return ("%s: num_samples=%d, sample_rate=%d, duration=%.2fsec, "
"rms=%.2fdB" % (type(self), self.num_samples, self.sample_rate,
self.duration, self.rms_db))
@classmethod
def from_file(cls, file):
"""Create audio segment from audio file.
:param filepath: Filepath or file object to audio file.
:type filepath: basestring|file
:return: Audio segment instance.
:rtype: AudioSegment
"""
samples, sample_rate = soundfile.read(file, dtype='float32')
return cls(samples, sample_rate)
@classmethod
def slice_from_file(cls, file, start=None, end=None):
"""Loads a small section of an audio without having to load
the entire file into the memory which can be incredibly wasteful.
:param file: Input audio filepath or file object.
:type file: basestring|file
:param start: Start time in seconds. If start is negative, it wraps
around from the end. If not provided, this function
reads from the very beginning.
:type start: float
:param end: End time in seconds. If end is negative, it wraps around
from the end. If not provided, the default behvaior is
to read to the end of the file.
:type end: float
:return: AudioSegment instance of the specified slice of the input
audio file.
:rtype: AudioSegment
:raise ValueError: If start or end is incorrectly set, e.g. out of
bounds in time.
"""
sndfile = soundfile.SoundFile(file)
sample_rate = sndfile.samplerate
duration = float(len(sndfile)) / sample_rate
start = 0. if start is None else start
end = 0. if end is None else end
if start < 0.0:
start += duration
if end < 0.0:
end += duration
if start < 0.0:
raise ValueError("The slice start position (%f s) is out of "
"bounds." % start)
if end < 0.0:
raise ValueError("The slice end position (%f s) is out of bounds." %
end)
if start > end:
raise ValueError("The slice start position (%f s) is later than "
"the slice end position (%f s)." % (start, end))
if end > duration:
raise ValueError("The slice end position (%f s) is out of bounds "
"(> %f s)" % (end, duration))
start_frame = int(start * sample_rate)
end_frame = int(end * sample_rate)
sndfile.seek(start_frame)
data = sndfile.read(frames=end_frame - start_frame, dtype='float32')
return cls(data, sample_rate)
@classmethod
def from_bytes(cls, bytes):
"""Create audio segment from a byte string containing audio samples.
:param bytes: Byte string containing audio samples.
:type bytes: str
:return: Audio segment instance.
:rtype: AudioSegment
"""
samples, sample_rate = soundfile.read(
io.BytesIO(bytes), dtype='float32')
return cls(samples, sample_rate)
@classmethod
def concatenate(cls, *segments):
"""Concatenate an arbitrary number of audio segments together.
:param *segments: Input audio segments to be concatenated.
:type *segments: tuple of AudioSegment
:return: Audio segment instance as concatenating results.
:rtype: AudioSegment
:raises ValueError: If the number of segments is zero, or if the
sample_rate of any segments does not match.
:raises TypeError: If any segment is not AudioSegment instance.
"""
# Perform basic sanity-checks.
if len(segments) == 0:
raise ValueError("No audio segments are given to concatenate.")
sample_rate = segments[0]._sample_rate
for seg in segments:
if sample_rate != seg._sample_rate:
raise ValueError("Can't concatenate segments with "
"different sample rates")
if type(seg) is not cls:
raise TypeError("Only audio segments of the same type "
"can be concatenated.")
samples = np.concatenate([seg.samples for seg in segments])
return cls(samples, sample_rate)
@classmethod
def make_silence(cls, duration, sample_rate):
"""Creates a silent audio segment of the given duration and sample rate.
:param duration: Length of silence in seconds.
:type duration: float
:param sample_rate: Sample rate.
:type sample_rate: float
:return: Silent AudioSegment instance of the given duration.
:rtype: AudioSegment
"""
samples = np.zeros(int(duration * sample_rate))
return cls(samples, sample_rate)
def to_wav_file(self, filepath, dtype='float32'):
"""Save audio segment to disk as wav file.
:param filepath: WAV filepath or file object to save the
audio segment.
:type filepath: basestring|file
:param dtype: Subtype for audio file. Options: 'int16', 'int32',
'float32', 'float64'. Default is 'float32'.
:type dtype: str
:raises TypeError: If dtype is not supported.
"""
samples = self._convert_samples_from_float32(self._samples, dtype)
subtype_map = {
'int16': 'PCM_16',
'int32': 'PCM_32',
'float32': 'FLOAT',
'float64': 'DOUBLE'
}
soundfile.write(
filepath,
samples,
self._sample_rate,
format='WAV',
subtype=subtype_map[dtype])
def superimpose(self, other):
"""Add samples from another segment to those of this segment
(sample-wise addition, not segment concatenation).
Note that this is an in-place transformation.
:param other: Segment containing samples to be added in.
:type other: AudioSegments
:raise TypeError: If type of two segments don't match.
:raise ValueError: If the sample rates of the two segments are not
equal, or if the lengths of segments don't match.
"""
if type(self) != type(other):
raise TypeError("Cannot add segments of different types: %s "
"and %s." % (type(self), type(other)))
if self._sample_rate != other._sample_rate:
raise ValueError("Sample rates must match to add segments.")
if len(self._samples) != len(other._samples):
raise ValueError("Segment lengths must match to add segments.")
self._samples += other._samples
def to_bytes(self, dtype='float32'):
"""Create a byte string containing the audio content.
:param dtype: Data type for export samples. Options: 'int16', 'int32',
'float32', 'float64'. Default is 'float32'.
:type dtype: str
:return: Byte string containing audio content.
:rtype: str
"""
samples = self._convert_samples_from_float32(self._samples, dtype)
return samples.tostring()
def gain_db(self, gain):
"""Apply gain in decibels to samples.
Note that this is an in-place transformation.
:param gain: Gain in decibels to apply to samples.
:type gain: float
"""
self._samples *= 10.**(gain / 20.)
def change_speed(self, speed_rate):
"""Change the audio speed by linear interpolation.
Note that this is an in-place transformation.
:param speed_rate: Rate of speed change:
speed_rate > 1.0, speed up the audio;
speed_rate = 1.0, unchanged;
speed_rate < 1.0, slow down the audio;
speed_rate <= 0.0, not allowed, raise ValueError.
:type speed_rate: float
:raises ValueError: If speed_rate <= 0.0.
"""
if speed_rate <= 0:
raise ValueError("speed_rate should be greater than zero.")
old_length = self._samples.shape[0]
new_length = int(old_length / speed_rate)
old_indices = np.arange(old_length)
new_indices = np.linspace(start=0, stop=old_length, num=new_length)
self._samples = np.interp(new_indices, old_indices, self._samples)
def normalize(self, target_db=-20, max_gain_db=300.0):
"""Normalize audio to be of the desired RMS value in decibels.
Note that this is an in-place transformation.
:param target_db: Target RMS value in decibels. This value should be
less than 0.0 as 0.0 is full-scale audio.
:type target_db: float
:param max_gain_db: Max amount of gain in dB that can be applied for
normalization. This is to prevent nans when
attempting to normalize a signal consisting of
all zeros.
:type max_gain_db: float
:raises ValueError: If the required gain to normalize the segment to
the target_db value exceeds max_gain_db.
"""
gain = target_db - self.rms_db
if gain > max_gain_db:
raise ValueError(
"Unable to normalize segment to %f dB because the "
"the probable gain have exceeds max_gain_db (%f dB)" %
(target_db, max_gain_db))
self.gain_db(min(max_gain_db, target_db - self.rms_db))
def normalize_online_bayesian(self,
target_db,
prior_db,
prior_samples,
startup_delay=0.0):
"""Normalize audio using a production-compatible online/causal
algorithm. This uses an exponential likelihood and gamma prior to
make online estimates of the RMS even when there are very few samples.
Note that this is an in-place transformation.
:param target_db: Target RMS value in decibels.
:type target_bd: float
:param prior_db: Prior RMS estimate in decibels.
:type prior_db: float
:param prior_samples: Prior strength in number of samples.
:type prior_samples: float
:param startup_delay: Default 0.0s. If provided, this function will
accrue statistics for the first startup_delay
seconds before applying online normalization.
:type startup_delay: float
"""
# Estimate total RMS online.
startup_sample_idx = min(self.num_samples - 1,
int(self.sample_rate * startup_delay))
prior_mean_squared = 10.**(prior_db / 10.)
prior_sum_of_squares = prior_mean_squared * prior_samples
cumsum_of_squares = np.cumsum(self.samples**2)
sample_count = np.arange(self.num_samples) + 1
if startup_sample_idx > 0:
cumsum_of_squares[:startup_sample_idx] = \
cumsum_of_squares[startup_sample_idx]
sample_count[:startup_sample_idx] = \
sample_count[startup_sample_idx]
mean_squared_estimate = ((cumsum_of_squares + prior_sum_of_squares) /
(sample_count + prior_samples))
rms_estimate_db = 10 * np.log10(mean_squared_estimate)
# Compute required time-varying gain.
gain_db = target_db - rms_estimate_db
self.gain_db(gain_db)
def resample(self, target_sample_rate, filter='kaiser_best'):
"""Resample the audio to a target sample rate.
Note that this is an in-place transformation.
:param target_sample_rate: Target sample rate.
:type target_sample_rate: int
:param filter: The resampling filter to use one of {'kaiser_best',
'kaiser_fast'}.
:type filter: str
"""
self._samples = resampy.resample(
self.samples, self.sample_rate, target_sample_rate, filter=filter)
self._sample_rate = target_sample_rate
def pad_silence(self, duration, sides='both'):
"""Pad this audio sample with a period of silence.
Note that this is an in-place transformation.
:param duration: Length of silence in seconds to pad.
:type duration: float
:param sides: Position for padding:
'beginning' - adds silence in the beginning;
'end' - adds silence in the end;
'both' - adds silence in both the beginning and the end.
:type sides: str
:raises ValueError: If sides is not supported.
"""
if duration == 0.0:
return self
cls = type(self)
silence = self.make_silence(duration, self._sample_rate)
if sides == "beginning":
padded = cls.concatenate(silence, self)
elif sides == "end":
padded = cls.concatenate(self, silence)
elif sides == "both":
padded = cls.concatenate(silence, self, silence)
else:
raise ValueError("Unknown value for the sides %s" % sides)
self._samples = padded._samples
def shift(self, shift_ms):
"""Shift the audio in time. If `shift_ms` is positive, shift with time
advance; if negative, shift with time delay. Silence are padded to
keep the duration unchanged.
Note that this is an in-place transformation.
:param shift_ms: Shift time in millseconds. If positive, shift with
time advance; if negative; shift with time delay.
:type shift_ms: float
:raises ValueError: If shift_ms is longer than audio duration.
"""
if abs(shift_ms) / 1000.0 > self.duration:
raise ValueError("Absolute value of shift_ms should be smaller "
"than audio duration.")
shift_samples = int(shift_ms * self._sample_rate / 1000)
if shift_samples > 0:
# time advance
self._samples[:-shift_samples] = self._samples[shift_samples:]
self._samples[-shift_samples:] = 0
elif shift_samples < 0:
# time delay
self._samples[-shift_samples:] = self._samples[:shift_samples]
self._samples[:-shift_samples] = 0
def subsegment(self, start_sec=None, end_sec=None):
"""Cut the AudioSegment between given boundaries.
Note that this is an in-place transformation.
:param start_sec: Beginning of subsegment in seconds.
:type start_sec: float
:param end_sec: End of subsegment in seconds.
:type end_sec: float
:raise ValueError: If start_sec or end_sec is incorrectly set, e.g. out
of bounds in time.
"""
start_sec = 0.0 if start_sec is None else start_sec
end_sec = self.duration if end_sec is None else end_sec
if start_sec < 0.0:
start_sec = self.duration + start_sec
if end_sec < 0.0:
end_sec = self.duration + end_sec
if start_sec < 0.0:
raise ValueError("The slice start position (%f s) is out of "
"bounds." % start_sec)
if end_sec < 0.0:
raise ValueError("The slice end position (%f s) is out of bounds." %
end_sec)
if start_sec > end_sec:
raise ValueError("The slice start position (%f s) is later than "
"the end position (%f s)." % (start_sec, end_sec))
if end_sec > self.duration:
raise ValueError("The slice end position (%f s) is out of bounds "
"(> %f s)" % (end_sec, self.duration))
start_sample = int(round(start_sec * self._sample_rate))
end_sample = int(round(end_sec * self._sample_rate))
self._samples = self._samples[start_sample:end_sample]
def random_subsegment(self, subsegment_length, rng=None):
"""Cut the specified length of the audiosegment randomly.
Note that this is an in-place transformation.
:param subsegment_length: Subsegment length in seconds.
:type subsegment_length: float
:param rng: Random number generator state.
:type rng: random.Random
:raises ValueError: If the length of subsegment is greater than
the origineal segemnt.
"""
rng = random.Random() if rng is None else rng
if subsegment_length > self.duration:
raise ValueError("Length of subsegment must not be greater "
"than original segment.")
start_time = rng.uniform(0.0, self.duration - subsegment_length)
self.subsegment(start_time, start_time + subsegment_length)
def convolve(self, impulse_segment, allow_resample=False):
"""Convolve this audio segment with the given impulse segment.
Note that this is an in-place transformation.
:param impulse_segment: Impulse response segments.
:type impulse_segment: AudioSegment
:param allow_resample: Indicates whether resampling is allowed when
the impulse_segment has a different sample
rate from this signal.
:type allow_resample: bool
:raises ValueError: If the sample rate is not match between two
audio segments when resample is not allowed.
"""
if allow_resample and self.sample_rate != impulse_segment.sample_rate:
impulse_segment = impulse_segment.resample(self.sample_rate)
if self.sample_rate != impulse_segment.sample_rate:
raise ValueError("Impulse segment's sample rate (%d Hz) is not"
"equal to base signal sample rate (%d Hz)." %
(impulse_segment.sample_rate, self.sample_rate))
samples = signal.fftconvolve(self.samples, impulse_segment.samples,
"full")
self._samples = samples
def convolve_and_normalize(self, impulse_segment, allow_resample=False):
"""Convolve and normalize the resulting audio segment so that it
has the same average power as the input signal.
Note that this is an in-place transformation.
:param impulse_segment: Impulse response segments.
:type impulse_segment: AudioSegment
:param allow_resample: Indicates whether resampling is allowed when
the impulse_segment has a different sample
rate from this signal.
:type allow_resample: bool
"""
target_db = self.rms_db
self.convolve(impulse_segment, allow_resample=allow_resample)
self.normalize(target_db)
def add_noise(self,
noise,
snr_dB,
allow_downsampling=False,
max_gain_db=300.0,
rng=None):
"""Add the given noise segment at a specific signal-to-noise ratio.
If the noise segment is longer than this segment, a random subsegment
of matching length is sampled from it and used instead.
Note that this is an in-place transformation.
:param noise: Noise signal to add.
:type noise: AudioSegment
:param snr_dB: Signal-to-Noise Ratio, in decibels.
:type snr_dB: float
:param allow_downsampling: Whether to allow the noise signal to be
downsampled to match the base signal sample
rate.
:type allow_downsampling: bool
:param max_gain_db: Maximum amount of gain to apply to noise signal
before adding it in. This is to prevent attempting
to apply infinite gain to a zero signal.
:type max_gain_db: float
:param rng: Random number generator state.
:type rng: None|random.Random
:raises ValueError: If the sample rate does not match between the two
audio segments when downsampling is not allowed, or
if the duration of noise segments is shorter than
original audio segments.
"""
rng = random.Random() if rng is None else rng
if allow_downsampling and noise.sample_rate > self.sample_rate:
noise = noise.resample(self.sample_rate)
if noise.sample_rate != self.sample_rate:
raise ValueError("Noise sample rate (%d Hz) is not equal to base "
"signal sample rate (%d Hz)." % (noise.sample_rate,
self.sample_rate))
if noise.duration < self.duration:
raise ValueError("Noise signal (%f sec) must be at least as long as"
" base signal (%f sec)." %
(noise.duration, self.duration))
noise_gain_db = min(self.rms_db - noise.rms_db - snr_dB, max_gain_db)
noise_new = copy.deepcopy(noise)
noise_new.random_subsegment(self.duration, rng=rng)
noise_new.gain_db(noise_gain_db)
self.superimpose(noise_new)
@property
def samples(self):
"""Return audio samples.
:return: Audio samples.
:rtype: ndarray
"""
return self._samples.copy()
@property
def sample_rate(self):
"""Return audio sample rate.
:return: Audio sample rate.
:rtype: int
"""
return self._sample_rate
@property
def num_samples(self):
"""Return number of samples.
:return: Number of samples.
:rtype: int
"""
return self._samples.shape[0]
@property
def duration(self):
"""Return audio duration.
:return: Audio duration in seconds.
:rtype: float
"""
return self._samples.shape[0] / float(self._sample_rate)
@property
def rms_db(self):
"""Return root mean square energy of the audio in decibels.
:return: Root mean square energy in decibels.
:rtype: float
"""
# square root => multiply by 10 instead of 20 for dBs
mean_square = np.mean(self._samples**2)
return 10 * np.log10(mean_square)
def _convert_samples_to_float32(self, samples):
"""Convert sample type to float32.
Audio sample type is usually integer or float-point.
Integers will be scaled to [-1, 1] in float32.
"""
float32_samples = samples.astype('float32')
if samples.dtype in np.sctypes['int']:
bits = np.iinfo(samples.dtype).bits
float32_samples *= (1. / 2**(bits - 1))
elif samples.dtype in np.sctypes['float']:
pass
else:
raise TypeError("Unsupported sample type: %s." % samples.dtype)
return float32_samples
def _convert_samples_from_float32(self, samples, dtype):
"""Convert sample type from float32 to dtype.
Audio sample type is usually integer or float-point. For integer
type, float32 will be rescaled from [-1, 1] to the maximum range
supported by the integer type.
This is for writing a audio file.
"""
dtype = np.dtype(dtype)
output_samples = samples.copy()
if dtype in np.sctypes['int']:
bits = np.iinfo(dtype).bits
output_samples *= (2**(bits - 1) / 1.)
min_val = np.iinfo(dtype).min
max_val = np.iinfo(dtype).max
output_samples[output_samples > max_val] = max_val
output_samples[output_samples < min_val] = min_val
elif samples.dtype in np.sctypes['float']:
min_val = np.finfo(dtype).min
max_val = np.finfo(dtype).max
output_samples[output_samples > max_val] = max_val
output_samples[output_samples < min_val] = min_val
else:
raise TypeError("Unsupported sample type: %s." % samples.dtype)
return output_samples.astype(dtype)

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data_utils/augmentor/__init__.py
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data_utils/augmentor/augmentation.py
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"""Contains the data augmentation pipeline."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import json
import random
from data_utils.augmentor.volume_perturb import VolumePerturbAugmentor
from data_utils.augmentor.shift_perturb import ShiftPerturbAugmentor
from data_utils.augmentor.speed_perturb import SpeedPerturbAugmentor
from data_utils.augmentor.resample import ResampleAugmentor
from data_utils.augmentor.online_bayesian_normalization import \
OnlineBayesianNormalizationAugmentor
class AugmentationPipeline(object):
"""Build a pre-processing pipeline with various augmentation models.Such a
data augmentation pipeline is oftern leveraged to augment the training
samples to make the model invariant to certain types of perturbations in the
real world, improving model's generalization ability.
The pipeline is built according the the augmentation configuration in json
string, e.g.
.. code-block::
'[{"type": "volume",
"params": {"min_gain_dBFS": -15,
"max_gain_dBFS": 15},
"prob": 0.5},
{"type": "speed",
"params": {"min_speed_rate": 0.8,
"max_speed_rate": 1.2},
"prob": 0.5}
]'
This augmentation configuration inserts two augmentation models
into the pipeline, with one is VolumePerturbAugmentor and the other
SpeedPerturbAugmentor. "prob" indicates the probability of the current
augmentor to take effect.
:param augmentation_config: Augmentation configuration in json string.
:type augmentation_config: str
:param random_seed: Random seed.
:type random_seed: int
:raises ValueError: If the augmentation json config is in incorrect format".
"""
def __init__(self, augmentation_config, random_seed=0):
self._rng = random.Random(random_seed)
self._augmentors, self._rates = self._parse_pipeline_from(
augmentation_config)
def transform_audio(self, audio_segment):
"""Run the pre-processing pipeline for data augmentation.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to process.
:type audio_segment: AudioSegmenet|SpeechSegment
"""
for augmentor, rate in zip(self._augmentors, self._rates):
if self._rng.uniform(0., 1.) <= rate:
augmentor.transform_audio(audio_segment)
def _parse_pipeline_from(self, config_json):
"""Parse the config json to build a augmentation pipelien."""
try:
configs = json.loads(config_json)
augmentors = [
self._get_augmentor(config["type"], config["params"])
for config in configs
]
rates = [config["prob"] for config in configs]
except Exception as e:
raise ValueError("Failed to parse the augmentation config json: "
"%s" % str(e))
return augmentors, rates
def _get_augmentor(self, augmentor_type, params):
"""Return an augmentation model by the type name, and pass in params."""
if augmentor_type == "volume":
return VolumePerturbAugmentor(self._rng, **params)
elif augmentor_type == "shift":
return ShiftPerturbAugmentor(self._rng, **params)
elif augmentor_type == "speed":
return SpeedPerturbAugmentor(self._rng, **params)
elif augmentor_type == "resample":
return ResampleAugmentor(self._rng, **params)
elif augmentor_type == "bayesian_normal":
return OnlineBayesianNormalizationAugmentor(self._rng, **params)
else:
raise ValueError("Unknown augmentor type [%s]." % augmentor_type)

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data_utils/augmentor/base.py
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"""Contains the abstract base class for augmentation models."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from abc import ABCMeta, abstractmethod
class AugmentorBase(object):
"""Abstract base class for augmentation model (augmentor) class.
All augmentor classes should inherit from this class, and implement the
following abstract methods.
"""
__metaclass__ = ABCMeta
@abstractmethod
def __init__(self):
pass
@abstractmethod
def transform_audio(self, audio_segment):
"""Adds various effects to the input audio segment. Such effects
will augment the training data to make the model invariant to certain
types of perturbations in the real world, improving model's
generalization ability.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegmenet|SpeechSegment
"""
pass

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data_utils/augmentor/online_bayesian_normalization.py
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"""Contain the online bayesian normalization augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
class OnlineBayesianNormalizationAugmentor(AugmentorBase):
"""Augmentation model for adding online bayesian normalization.
:param rng: Random generator object.
:type rng: random.Random
:param target_db: Target RMS value in decibels.
:type target_db: float
:param prior_db: Prior RMS estimate in decibels.
:type prior_db: float
:param prior_samples: Prior strength in number of samples.
:type prior_samples: int
:param startup_delay: Default 0.0s. If provided, this function will
accrue statistics for the first startup_delay
seconds before applying online normalization.
:type starup_delay: float.
"""
def __init__(self,
rng,
target_db,
prior_db,
prior_samples,
startup_delay=0.0):
self._target_db = target_db
self._prior_db = prior_db
self._prior_samples = prior_samples
self._rng = rng
self._startup_delay = startup_delay
def transform_audio(self, audio_segment):
"""Normalizes the input audio using the online Bayesian approach.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegment|SpeechSegment
"""
audio_segment.normalize_online_bayesian(self._target_db, self._prior_db,
self._prior_samples,
self._startup_delay)

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data_utils/augmentor/resample.py
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"""Contain the resample augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
class ResampleAugmentor(AugmentorBase):
"""Augmentation model for resampling.
See more info here:
https://ccrma.stanford.edu/~jos/resample/index.html
:param rng: Random generator object.
:type rng: random.Random
:param new_sample_rate: New sample rate in Hz.
:type new_sample_rate: int
"""
def __init__(self, rng, new_sample_rate):
self._new_sample_rate = new_sample_rate
self._rng = rng
def transform_audio(self, audio_segment):
"""Resamples the input audio to a target sample rate.
Note that this is an in-place transformation.
:param audio: Audio segment to add effects to.
:type audio: AudioSegment|SpeechSegment
"""
audio_segment.resample(self._new_sample_rate)

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data_utils/augmentor/shift_perturb.py
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"""Contains the volume perturb augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
class ShiftPerturbAugmentor(AugmentorBase):
"""Augmentation model for adding random shift perturbation.
:param rng: Random generator object.
:type rng: random.Random
:param min_shift_ms: Minimal shift in milliseconds.
:type min_shift_ms: float
:param max_shift_ms: Maximal shift in milliseconds.
:type max_shift_ms: float
"""
def __init__(self, rng, min_shift_ms, max_shift_ms):
self._min_shift_ms = min_shift_ms
self._max_shift_ms = max_shift_ms
self._rng = rng
def transform_audio(self, audio_segment):
"""Shift audio.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegmenet|SpeechSegment
"""
shift_ms = self._rng.uniform(self._min_shift_ms, self._max_shift_ms)
audio_segment.shift(shift_ms)

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data_utils/augmentor/speed_perturb.py
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"""Contain the speech perturbation augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
class SpeedPerturbAugmentor(AugmentorBase):
"""Augmentation model for adding speed perturbation.
See reference paper here:
http://www.danielpovey.com/files/2015_interspeech_augmentation.pdf
:param rng: Random generator object.
:type rng: random.Random
:param min_speed_rate: Lower bound of new speed rate to sample and should
not be smaller than 0.9.
:type min_speed_rate: float
:param max_speed_rate: Upper bound of new speed rate to sample and should
not be larger than 1.1.
:type max_speed_rate: float
"""
def __init__(self, rng, min_speed_rate, max_speed_rate):
if min_speed_rate < 0.9:
raise ValueError(
"Sampling speed below 0.9 can cause unnatural effects")
if max_speed_rate > 1.1:
raise ValueError(
"Sampling speed above 1.1 can cause unnatural effects")
self._min_speed_rate = min_speed_rate
self._max_speed_rate = max_speed_rate
self._rng = rng
def transform_audio(self, audio_segment):
"""Sample a new speed rate from the given range and
changes the speed of the given audio clip.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegment|SpeechSegment
"""
sampled_speed = self._rng.uniform(self._min_speed_rate,
self._max_speed_rate)
audio_segment.change_speed(sampled_speed)

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data_utils/augmentor/volume_perturb.py
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"""Contains the volume perturb augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
class VolumePerturbAugmentor(AugmentorBase):
"""Augmentation model for adding random volume perturbation.
This is used for multi-loudness training of PCEN. See
https://arxiv.org/pdf/1607.05666v1.pdf
for more details.
:param rng: Random generator object.
:type rng: random.Random
:param min_gain_dBFS: Minimal gain in dBFS.
:type min_gain_dBFS: float
:param max_gain_dBFS: Maximal gain in dBFS.
:type max_gain_dBFS: float
"""
def __init__(self, rng, min_gain_dBFS, max_gain_dBFS):
self._min_gain_dBFS = min_gain_dBFS
self._max_gain_dBFS = max_gain_dBFS
self._rng = rng
def transform_audio(self, audio_segment):
"""Change audio loadness.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegmenet|SpeechSegment
"""
gain = self._rng.uniform(self._min_gain_dBFS, self._max_gain_dBFS)
audio_segment.gain_db(gain)

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data_utils/data.py
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"""Contains data generator for orgnaizing various audio data preprocessing
pipeline and offering data reader interface of PaddlePaddle requirements.
"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import random
import numpy as np
import multiprocessing
import paddle.v2 as paddle
from data_utils import utils
from data_utils.augmentor.augmentation import AugmentationPipeline
from data_utils.featurizer.speech_featurizer import SpeechFeaturizer
from data_utils.speech import SpeechSegment
from data_utils.normalizer import FeatureNormalizer
class DataGenerator(object):
"""
DataGenerator provides basic audio data preprocessing pipeline, and offers
data reader interfaces of PaddlePaddle requirements.
:param vocab_filepath: Vocabulary filepath for indexing tokenized
transcripts.
:type vocab_filepath: basestring
:param mean_std_filepath: File containing the pre-computed mean and stddev.
:type mean_std_filepath: None|basestring
:param augmentation_config: Augmentation configuration in json string.
Details see AugmentationPipeline.__doc__.
:type augmentation_config: str
:param max_duration: Audio with duration (in seconds) greater than
this will be discarded.
:type max_duration: float
:param min_duration: Audio with duration (in seconds) smaller than
this will be discarded.
:type min_duration: float
:param stride_ms: Striding size (in milliseconds) for generating frames.
:type stride_ms: float
:param window_ms: Window size (in milliseconds) for generating frames.
:type window_ms: float
:param max_freq: Used when specgram_type is 'linear', only FFT bins
corresponding to frequencies between [0, max_freq] are
returned.
:types max_freq: None|float
:param specgram_type: Specgram feature type. Options: 'linear'.
:type specgram_type: str
:param use_dB_normalization: Whether to normalize the audio to -20 dB
before extracting the features.
:type use_dB_normalization: bool
:param num_threads: Number of CPU threads for processing data.
:type num_threads: int
:param random_seed: Random seed.
:type random_seed: int
"""
def __init__(self,
vocab_filepath,
mean_std_filepath,
augmentation_config='{}',
max_duration=float('inf'),
min_duration=0.0,
stride_ms=10.0,
window_ms=20.0,
max_freq=None,
specgram_type='linear',
use_dB_normalization=True,
num_threads=multiprocessing.cpu_count(),
random_seed=0):
self._max_duration = max_duration
self._min_duration = min_duration
self._normalizer = FeatureNormalizer(mean_std_filepath)
self._augmentation_pipeline = AugmentationPipeline(
augmentation_config=augmentation_config, random_seed=random_seed)
self._speech_featurizer = SpeechFeaturizer(
vocab_filepath=vocab_filepath,
specgram_type=specgram_type,
stride_ms=stride_ms,
window_ms=window_ms,
max_freq=max_freq,
use_dB_normalization=use_dB_normalization)
self._num_threads = num_threads
self._rng = random.Random(random_seed)
self._epoch = 0
def batch_reader_creator(self,
manifest_path,
batch_size,
min_batch_size=1,
padding_to=-1,
flatten=False,
sortagrad=False,
shuffle_method="batch_shuffle"):
"""
Batch data reader creator for audio data. Return a callable generator
function to produce batches of data.
Audio features within one batch will be padded with zeros to have the
same shape, or a user-defined shape.
:param manifest_path: Filepath of manifest for audio files.
:type manifest_path: basestring
:param batch_size: Number of instances in a batch.
:type batch_size: int
:param min_batch_size: Any batch with batch size smaller than this will
be discarded. (To be deprecated in the future.)
:type min_batch_size: int
:param padding_to: If set -1, the maximun shape in the batch
will be used as the target shape for padding.
Otherwise, `padding_to` will be the target shape.
:type padding_to: int
:param flatten: If set True, audio features will be flatten to 1darray.
:type flatten: bool
:param sortagrad: If set True, sort the instances by audio duration
in the first epoch for speed up training.
:type sortagrad: bool
:param shuffle_method: Shuffle method. Options:
'' or None: no shuffle.
'instance_shuffle': instance-wise shuffle.
'batch_shuffle': similarly-sized instances are
put into batches, and then
batch-wise shuffle the batches.
For more details, please see
``_batch_shuffle.__doc__``.
'batch_shuffle_clipped': 'batch_shuffle' with
head shift and tail
clipping. For more
details, please see
``_batch_shuffle``.
If sortagrad is True, shuffle is disabled
for the first epoch.
:type shuffle_method: None|str
:return: Batch reader function, producing batches of data when called.
:rtype: callable
"""
def batch_reader():
# read manifest
manifest = utils.read_manifest(
manifest_path=manifest_path,
max_duration=self._max_duration,
min_duration=self._min_duration)
# sort (by duration) or batch-wise shuffle the manifest
if self._epoch == 0 and sortagrad:
manifest.sort(key=lambda x: x["duration"])
else:
if shuffle_method == "batch_shuffle":
manifest = self._batch_shuffle(
manifest, batch_size, clipped=False)
elif shuffle_method == "batch_shuffle_clipped":
manifest = self._batch_shuffle(
manifest, batch_size, clipped=True)
elif shuffle_method == "instance_shuffle":
self._rng.shuffle(manifest)
elif not shuffle_method:
pass
else:
raise ValueError("Unknown shuffle method %s." %
shuffle_method)
# prepare batches
instance_reader = self._instance_reader_creator(manifest)
batch = []
for instance in instance_reader():
batch.append(instance)
if len(batch) == batch_size:
yield self._padding_batch(batch, padding_to, flatten)
batch = []
if len(batch) >= min_batch_size:
yield self._padding_batch(batch, padding_to, flatten)
self._epoch += 1
return batch_reader
@property
def feeding(self):
"""Returns data reader's feeding dict.
:return: Data feeding dict.
:rtype: dict
"""
return {"audio_spectrogram": 0, "transcript_text": 1}
@property
def vocab_size(self):
"""Return the vocabulary size.
:return: Vocabulary size.
:rtype: int
"""
return self._speech_featurizer.vocab_size
@property
def vocab_list(self):
"""Return the vocabulary in list.
:return: Vocabulary in list.
:rtype: list
"""
return self._speech_featurizer.vocab_list
def _process_utterance(self, filename, transcript):
"""Load, augment, featurize and normalize for speech data."""
speech_segment = SpeechSegment.from_file(filename, transcript)
self._augmentation_pipeline.transform_audio(speech_segment)
specgram, text_ids = self._speech_featurizer.featurize(speech_segment)
specgram = self._normalizer.apply(specgram)
return specgram, text_ids
def _instance_reader_creator(self, manifest):
"""
Instance reader creator. Create a callable function to produce
instances of data.
Instance: a tuple of ndarray of audio spectrogram and a list of
token indices for transcript.
"""
def reader():
for instance in manifest:
yield instance
def mapper(instance):
return self._process_utterance(instance["audio_filepath"],
instance["text"])
return paddle.reader.xmap_readers(
mapper, reader, self._num_threads, 1024, order=True)
def _padding_batch(self, batch, padding_to=-1, flatten=False):
"""
Padding audio features with zeros to make them have the same shape (or
a user-defined shape) within one bach.
If ``padding_to`` is -1, the maximun shape in the batch will be used
as the target shape for padding. Otherwise, `padding_to` will be the
target shape (only refers to the second axis).
If `flatten` is True, features will be flatten to 1darray.
"""
new_batch = []
# get target shape
max_length = max([audio.shape[1] for audio, text in batch])
if padding_to != -1:
if padding_to < max_length:
raise ValueError("If padding_to is not -1, it should be larger "
"than any instance's shape in the batch")
max_length = padding_to
# padding
for audio, text in batch:
padded_audio = np.zeros([audio.shape[0], max_length])
padded_audio[:, :audio.shape[1]] = audio
if flatten:
padded_audio = padded_audio.flatten()
new_batch.append((padded_audio, text))
return new_batch
def _batch_shuffle(self, manifest, batch_size, clipped=False):
"""Put similarly-sized instances into minibatches for better efficiency
and make a batch-wise shuffle.
1. Sort the audio clips by duration.
2. Generate a random number `k`, k in [0, batch_size).
3. Randomly shift `k` instances in order to create different batches
for different epochs. Create minibatches.
4. Shuffle the minibatches.
:param manifest: Manifest contents. List of dict.
:type manifest: list
:param batch_size: Batch size. This size is also used for generate
a random number for batch shuffle.
:type batch_size: int
:param clipped: Whether to clip the heading (small shift) and trailing
(incomplete batch) instances.
:type clipped: bool
:return: Batch shuffled mainifest.
:rtype: list
"""
manifest.sort(key=lambda x: x["duration"])
shift_len = self._rng.randint(0, batch_size - 1)
batch_manifest = zip(*[iter(manifest[shift_len:])] * batch_size)
self._rng.shuffle(batch_manifest)
batch_manifest = list(sum(batch_manifest, ()))
if not clipped:
res_len = len(manifest) - shift_len - len(batch_manifest)
batch_manifest.extend(manifest[-res_len:])
batch_manifest.extend(manifest[0:shift_len])
return batch_manifest

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data_utils/featurizer/__init__.py
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data_utils/featurizer/audio_featurizer.py
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"""Contains the audio featurizer class."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import numpy as np
from data_utils import utils
from data_utils.audio import AudioSegment
class AudioFeaturizer(object):
"""Audio featurizer, for extracting features from audio contents of
AudioSegment or SpeechSegment.
Currently, it only supports feature type of linear spectrogram.
:param specgram_type: Specgram feature type. Options: 'linear'.
:type specgram_type: str
:param stride_ms: Striding size (in milliseconds) for generating frames.
:type stride_ms: float
:param window_ms: Window size (in milliseconds) for generating frames.
:type window_ms: float
:param max_freq: Used when specgram_type is 'linear', only FFT bins
corresponding to frequencies between [0, max_freq] are
returned.
:types max_freq: None|float
:param target_sample_rate: Audio are resampled (if upsampling or
downsampling is allowed) to this before
extracting spectrogram features.
:type target_sample_rate: float
:param use_dB_normalization: Whether to normalize the audio to a certain
decibels before extracting the features.
:type use_dB_normalization: bool
:param target_dB: Target audio decibels for normalization.
:type target_dB: float
"""
def __init__(self,
specgram_type='linear',
stride_ms=10.0,
window_ms=20.0,
max_freq=None,
target_sample_rate=16000,
use_dB_normalization=True,
target_dB=-20):
self._specgram_type = specgram_type
self._stride_ms = stride_ms
self._window_ms = window_ms
self._max_freq = max_freq
self._target_sample_rate = target_sample_rate
self._use_dB_normalization = use_dB_normalization
self._target_dB = target_dB
def featurize(self,
audio_segment,
allow_downsampling=True,
allow_upsamplling=True):
"""Extract audio features from AudioSegment or SpeechSegment.
:param audio_segment: Audio/speech segment to extract features from.
:type audio_segment: AudioSegment|SpeechSegment
:param allow_downsampling: Whether to allow audio downsampling before
featurizing.
:type allow_downsampling: bool
:param allow_upsampling: Whether to allow audio upsampling before
featurizing.
:type allow_upsampling: bool
:return: Spectrogram audio feature in 2darray.
:rtype: ndarray
:raises ValueError: If audio sample rate is not supported.
"""
# upsampling or downsampling
if ((audio_segment.sample_rate > self._target_sample_rate and
allow_downsampling) or
(audio_segment.sample_rate < self._target_sample_rate and
allow_upsampling)):
audio_segment.resample(self._target_sample_rate)
if audio_segment.sample_rate != self._target_sample_rate:
raise ValueError("Audio sample rate is not supported. "
"Turn allow_downsampling or allow up_sampling on.")
# decibel normalization
if self._use_dB_normalization:
audio_segment.normalize(target_db=self._target_dB)
# extract spectrogram
return self._compute_specgram(audio_segment.samples,
audio_segment.sample_rate)
def _compute_specgram(self, samples, sample_rate):
"""Extract various audio features."""
if self._specgram_type == 'linear':
return self._compute_linear_specgram(
samples, sample_rate, self._stride_ms, self._window_ms,
self._max_freq)
else:
raise ValueError("Unknown specgram_type %s. "
"Supported values: linear." % self._specgram_type)
def _compute_linear_specgram(self,
samples,
sample_rate,
stride_ms=10.0,
window_ms=20.0,
max_freq=None,
eps=1e-14):
"""Compute the linear spectrogram from FFT energy."""
if max_freq is None:
max_freq = sample_rate / 2
if max_freq > sample_rate / 2:
raise ValueError("max_freq must be greater than half of "
"sample rate.")
if stride_ms > window_ms:
raise ValueError("Stride size must not be greater than "
"window size.")
stride_size = int(0.001 * sample_rate * stride_ms)
window_size = int(0.001 * sample_rate * window_ms)
specgram, freqs = self._specgram_real(
samples,
window_size=window_size,
stride_size=stride_size,
sample_rate=sample_rate)
ind = np.where(freqs <= max_freq)[0][-1] + 1
return np.log(specgram[:ind, :] + eps)
def _specgram_real(self, samples, window_size, stride_size, sample_rate):
"""Compute the spectrogram for samples from a real signal."""
# extract strided windows
truncate_size = (len(samples) - window_size) % stride_size
samples = samples[:len(samples) - truncate_size]
nshape = (window_size, (len(samples) - window_size) // stride_size + 1)
nstrides = (samples.strides[0], samples.strides[0] * stride_size)
windows = np.lib.stride_tricks.as_strided(
samples, shape=nshape, strides=nstrides)
assert np.all(
windows[:, 1] == samples[stride_size:(stride_size + window_size)])
# window weighting, squared Fast Fourier Transform (fft), scaling
weighting = np.hanning(window_size)[:, None]
fft = np.fft.rfft(windows * weighting, axis=0)
fft = np.absolute(fft)**2
scale = np.sum(weighting**2) * sample_rate
fft[1:-1, :] *= (2.0 / scale)
fft[(0, -1), :] /= scale
# prepare fft frequency list
freqs = float(sample_rate) / window_size * np.arange(fft.shape[0])
return fft, freqs

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data_utils/featurizer/speech_featurizer.py
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"""Contains the speech featurizer class."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.featurizer.audio_featurizer import AudioFeaturizer
from data_utils.featurizer.text_featurizer import TextFeaturizer
class SpeechFeaturizer(object):
"""Speech featurizer, for extracting features from both audio and transcript
contents of SpeechSegment.
Currently, for audio parts, it only supports feature type of linear
spectrogram; for transcript parts, it only supports char-level tokenizing
and conversion into a list of token indices. Note that the token indexing
order follows the given vocabulary file.
:param vocab_filepath: Filepath to load vocabulary for token indices
conversion.
:type specgram_type: basestring
:param specgram_type: Specgram feature type. Options: 'linear'.
:type specgram_type: str
:param stride_ms: Striding size (in milliseconds) for generating frames.
:type stride_ms: float
:param window_ms: Window size (in milliseconds) for generating frames.
:type window_ms: float
:param max_freq: Used when specgram_type is 'linear', only FFT bins
corresponding to frequencies between [0, max_freq] are
returned.
:types max_freq: None|float
:param target_sample_rate: Speech are resampled (if upsampling or
downsampling is allowed) to this before
extracting spectrogram features.
:type target_sample_rate: float
:param use_dB_normalization: Whether to normalize the audio to a certain
decibels before extracting the features.
:type use_dB_normalization: bool
:param target_dB: Target audio decibels for normalization.
:type target_dB: float
"""
def __init__(self,
vocab_filepath,
specgram_type='linear',
stride_ms=10.0,
window_ms=20.0,
max_freq=None,
target_sample_rate=16000,
use_dB_normalization=True,
target_dB=-20):
self._audio_featurizer = AudioFeaturizer(
specgram_type=specgram_type,
stride_ms=stride_ms,
window_ms=window_ms,
max_freq=max_freq,
target_sample_rate=target_sample_rate,
use_dB_normalization=use_dB_normalization,
target_dB=target_dB)
self._text_featurizer = TextFeaturizer(vocab_filepath)
def featurize(self, speech_segment):
"""Extract features for speech segment.
1. For audio parts, extract the audio features.
2. For transcript parts, convert text string to a list of token indices
in char-level.
:param audio_segment: Speech segment to extract features from.
:type audio_segment: SpeechSegment
:return: A tuple of 1) spectrogram audio feature in 2darray, 2) list of
char-level token indices.
:rtype: tuple
"""
audio_feature = self._audio_featurizer.featurize(speech_segment)
text_ids = self._text_featurizer.featurize(speech_segment.transcript)
return audio_feature, text_ids
@property
def vocab_size(self):
"""Return the vocabulary size.
:return: Vocabulary size.
:rtype: int
"""
return self._text_featurizer.vocab_size
@property
def vocab_list(self):
"""Return the vocabulary in list.
:return: Vocabulary in list.
:rtype: list
"""
return self._text_featurizer.vocab_list

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data_utils/featurizer/text_featurizer.py
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"""Contains the text featurizer class."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import os
class TextFeaturizer(object):
"""Text featurizer, for processing or extracting features from text.
Currently, it only supports char-level tokenizing and conversion into
a list of token indices. Note that the token indexing order follows the
given vocabulary file.
:param vocab_filepath: Filepath to load vocabulary for token indices
conversion.
:type specgram_type: basestring
"""
def __init__(self, vocab_filepath):
self._vocab_dict, self._vocab_list = self._load_vocabulary_from_file(
vocab_filepath)
def featurize(self, text):
"""Convert text string to a list of token indices in char-level.Note
that the token indexing order follows the given vocabulary file.
:param text: Text to process.
:type text: basestring
:return: List of char-level token indices.
:rtype: list
"""
tokens = self._char_tokenize(text)
return [self._vocab_dict[token] for token in tokens]
@property
def vocab_size(self):
"""Return the vocabulary size.
:return: Vocabulary size.
:rtype: int
"""
return len(self._vocab_list)
@property
def vocab_list(self):
"""Return the vocabulary in list.
:return: Vocabulary in list.
:rtype: list
"""
return self._vocab_list
def _char_tokenize(self, text):
"""Character tokenizer."""
return list(text.strip())
def _load_vocabulary_from_file(self, vocab_filepath):
"""Load vocabulary from file."""
vocab_lines = []
with open(vocab_filepath, 'r') as file:
vocab_lines.extend(file.readlines())
vocab_list = [line[:-1] for line in vocab_lines]
vocab_dict = dict(
[(token, id) for (id, token) in enumerate(vocab_list)])
return vocab_dict, vocab_list

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data_utils/normalizer.py
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"""Contains feature normalizers."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import numpy as np
import random
import data_utils.utils as utils
from data_utils.audio import AudioSegment
class FeatureNormalizer(object):
"""Feature normalizer. Normalize features to be of zero mean and unit
stddev.
if mean_std_filepath is provided (not None), the normalizer will directly
initilize from the file. Otherwise, both manifest_path and featurize_func
should be given for on-the-fly mean and stddev computing.
:param mean_std_filepath: File containing the pre-computed mean and stddev.
:type mean_std_filepath: None|basestring
:param manifest_path: Manifest of instances for computing mean and stddev.
:type meanifest_path: None|basestring
:param featurize_func: Function to extract features. It should be callable
with ``featurize_func(audio_segment)``.
:type featurize_func: None|callable
:param num_samples: Number of random samples for computing mean and stddev.
:type num_samples: int
:param random_seed: Random seed for sampling instances.
:type random_seed: int
:raises ValueError: If both mean_std_filepath and manifest_path
(or both mean_std_filepath and featurize_func) are None.
"""
def __init__(self,
mean_std_filepath,
manifest_path=None,
featurize_func=None,
num_samples=500,
random_seed=0):
if not mean_std_filepath:
if not (manifest_path and featurize_func):
raise ValueError("If mean_std_filepath is None, meanifest_path "
"and featurize_func should not be None.")
self._rng = random.Random(random_seed)
self._compute_mean_std(manifest_path, featurize_func, num_samples)
else:
self._read_mean_std_from_file(mean_std_filepath)
def apply(self, features, eps=1e-14):
"""Normalize features to be of zero mean and unit stddev.
:param features: Input features to be normalized.
:type features: ndarray
:param eps: added to stddev to provide numerical stablibity.
:type eps: float
:return: Normalized features.
:rtype: ndarray
"""
return (features - self._mean) / (self._std + eps)
def write_to_file(self, filepath):
"""Write the mean and stddev to the file.
:param filepath: File to write mean and stddev.
:type filepath: basestring
"""
np.savez(filepath, mean=self._mean, std=self._std)
def _read_mean_std_from_file(self, filepath):
"""Load mean and std from file."""
npzfile = np.load(filepath)
self._mean = npzfile["mean"]
self._std = npzfile["std"]
def _compute_mean_std(self, manifest_path, featurize_func, num_samples):
"""Compute mean and std from randomly sampled instances."""
manifest = utils.read_manifest(manifest_path)
sampled_manifest = self._rng.sample(manifest, num_samples)
features = []
for instance in sampled_manifest:
features.append(
featurize_func(
AudioSegment.from_file(instance["audio_filepath"])))
features = np.hstack(features)
self._mean = np.mean(features, axis=1).reshape([-1, 1])
self._std = np.std(features, axis=1).reshape([-1, 1])

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data_utils/speech.py
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"""Contains the speech segment class."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.audio import AudioSegment
class SpeechSegment(AudioSegment):
"""Speech segment abstraction, a subclass of AudioSegment,
with an additional transcript.
:param samples: Audio samples [num_samples x num_channels].
:type samples: ndarray.float32
:param sample_rate: Audio sample rate.
:type sample_rate: int
:param transcript: Transcript text for the speech.
:type transript: basestring
:raises TypeError: If the sample data type is not float or int.
"""
def __init__(self, samples, sample_rate, transcript):
AudioSegment.__init__(self, samples, sample_rate)
self._transcript = transcript
def __eq__(self, other):
"""Return whether two objects are equal.
"""
if not AudioSegment.__eq__(self, other):
return False
if self._transcript != other._transcript:
return False
return True
def __ne__(self, other):
"""Return whether two objects are unequal."""
return not self.__eq__(other)
@classmethod
def from_file(cls, filepath, transcript):
"""Create speech segment from audio file and corresponding transcript.
:param filepath: Filepath or file object to audio file.
:type filepath: basestring|file
:param transcript: Transcript text for the speech.
:type transript: basestring
:return: Audio segment instance.
:rtype: AudioSegment
"""
audio = AudioSegment.from_file(filepath)
return cls(audio.samples, audio.sample_rate, transcript)
@classmethod
def from_bytes(cls, bytes, transcript):
"""Create speech segment from a byte string and corresponding
transcript.
:param bytes: Byte string containing audio samples.
:type bytes: str
:param transcript: Transcript text for the speech.
:type transript: basestring
:return: Audio segment instance.
:rtype: AudioSegment
"""
audio = AudioSegment.from_bytes(bytes)
return cls(audio.samples, audio.sample_rate, transcript)
@classmethod
def concatenate(cls, *segments):
"""Concatenate an arbitrary number of speech segments together, both
audio and transcript will be concatenated.
:param *segments: Input speech segments to be concatenated.
:type *segments: tuple of SpeechSegment
:return: Speech segment instance.
:rtype: SpeechSegment
:raises ValueError: If the number of segments is zero, or if the
sample_rate of any two segments does not match.
:raises TypeError: If any segment is not SpeechSegment instance.
"""
if len(segments) == 0:
raise ValueError("No speech segments are given to concatenate.")
sample_rate = segments[0]._sample_rate
transcripts = ""
for seg in segments:
if sample_rate != seg._sample_rate:
raise ValueError("Can't concatenate segments with "
"different sample rates")
if type(seg) is not cls:
raise TypeError("Only speech segments of the same type "
"instance can be concatenated.")
transcripts += seg._transcript
samples = np.concatenate([seg.samples for seg in segments])
return cls(samples, sample_rate, transcripts)
@classmethod
def slice_from_file(cls, filepath, transcript, start=None, end=None):
"""Loads a small section of an speech without having to load
the entire file into the memory which can be incredibly wasteful.
:param filepath: Filepath or file object to audio file.
:type filepath: basestring|file
:param start: Start time in seconds. If start is negative, it wraps
around from the end. If not provided, this function
reads from the very beginning.
:type start: float
:param end: End time in seconds. If end is negative, it wraps around
from the end. If not provided, the default behvaior is
to read to the end of the file.
:type end: float
:param transcript: Transcript text for the speech. if not provided,
the defaults is an empty string.
:type transript: basestring
:return: SpeechSegment instance of the specified slice of the input
speech file.
:rtype: SpeechSegment
"""
audio = Audiosegment.slice_from_file(filepath, start, end)
return cls(audio.samples, audio.sample_rate, transcript)
@classmethod
def make_silence(cls, duration, sample_rate):
"""Creates a silent speech segment of the given duration and
sample rate, transcript will be an empty string.
:param duration: Length of silence in seconds.
:type duration: float
:param sample_rate: Sample rate.
:type sample_rate: float
:return: Silence of the given duration.
:rtype: SpeechSegment
"""
audio = AudioSegment.make_silence(duration, sample_rate)
return cls(audio.samples, audio.sample_rate, "")
@property
def transcript(self):
"""Return the transcript text.
:return: Transcript text for the speech.
:rtype: basestring
"""
return self._transcript

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data_utils/utils.py
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"""Contains data helper functions."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import json
def read_manifest(manifest_path, max_duration=float('inf'), min_duration=0.0):
"""Load and parse manifest file.
Instances with durations outside [min_duration, max_duration] will be
filtered out.
:param manifest_path: Manifest file to load and parse.
:type manifest_path: basestring
:param max_duration: Maximal duration in seconds for instance filter.
:type max_duration: float
:param min_duration: Minimal duration in seconds for instance filter.
:type min_duration: float
:return: Manifest parsing results. List of dict.
:rtype: list
:raises IOError: If failed to parse the manifest.
"""
manifest = []
for json_line in open(manifest_path):
try:
json_data = json.loads(json_line)
except Exception as e:
raise IOError("Error reading manifest: %s" % str(e))
if (json_data["duration"] <= max_duration and
json_data["duration"] >= min_duration):
manifest.append(json_data)
return manifest

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datasets/librispeech/librispeech.py
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"""Prepare Librispeech ASR datasets.
Download, unpack and create manifest files.
Manifest file is a json-format file with each line containing the
meta data (i.e. audio filepath, transcript and audio duration)
of each audio file in the data set.
"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import distutils.util
import os
import wget
import tarfile
import argparse
import soundfile
import json
from paddle.v2.dataset.common import md5file
DATA_HOME = os.path.expanduser('~/.cache/paddle/dataset/speech')
URL_ROOT = "http://www.openslr.org/resources/12"
URL_TEST_CLEAN = URL_ROOT + "/test-clean.tar.gz"
URL_TEST_OTHER = URL_ROOT + "/test-other.tar.gz"
URL_DEV_CLEAN = URL_ROOT + "/dev-clean.tar.gz"
URL_DEV_OTHER = URL_ROOT + "/dev-other.tar.gz"
URL_TRAIN_CLEAN_100 = URL_ROOT + "/train-clean-100.tar.gz"
URL_TRAIN_CLEAN_360 = URL_ROOT + "/train-clean-360.tar.gz"
URL_TRAIN_OTHER_500 = URL_ROOT + "/train-other-500.tar.gz"
MD5_TEST_CLEAN = "32fa31d27d2e1cad72775fee3f4849a9"
MD5_TEST_OTHER = "fb5a50374b501bb3bac4815ee91d3135"
MD5_DEV_CLEAN = "42e2234ba48799c1f50f24a7926300a1"
MD5_DEV_OTHER = "c8d0bcc9cca99d4f8b62fcc847357931"
MD5_TRAIN_CLEAN_100 = "2a93770f6d5c6c964bc36631d331a522"
MD5_TRAIN_CLEAN_360 = "c0e676e450a7ff2f54aeade5171606fa"
MD5_TRAIN_OTHER_500 = "d1a0fd59409feb2c614ce4d30c387708"
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument(
"--target_dir",
default=DATA_HOME + "/Libri",
type=str,
help="Directory to save the dataset. (default: %(default)s)")
parser.add_argument(
"--manifest_prefix",
default="manifest",
type=str,
help="Filepath prefix for output manifests. (default: %(default)s)")
parser.add_argument(
"--full_download",
default="True",
type=distutils.util.strtobool,
help="Download all datasets for Librispeech."
" If False, only download a minimal requirement (test-clean, dev-clean"
" train-clean-100). (default: %(default)s)")
args = parser.parse_args()
def download(url, md5sum, target_dir):
"""
Download file from url to target_dir, and check md5sum.
"""
if not os.path.exists(target_dir): os.makedirs(target_dir)
filepath = os.path.join(target_dir, url.split("/")[-1])
if not (os.path.exists(filepath) and md5file(filepath) == md5sum):
print("Downloading %s ..." % url)
wget.download(url, target_dir)
print("\nMD5 Chesksum %s ..." % filepath)
if not md5file(filepath) == md5sum:
raise RuntimeError("MD5 checksum failed.")
else:
print("File exists, skip downloading. (%s)" % filepath)
return filepath
def unpack(filepath, target_dir):
"""
Unpack the file to the target_dir.
"""
print("Unpacking %s ..." % filepath)
tar = tarfile.open(filepath)
tar.extractall(target_dir)
tar.close()
def create_manifest(data_dir, manifest_path):
"""
Create a manifest json file summarizing the data set, with each line
containing the meta data (i.e. audio filepath, transcription text, audio
duration) of each audio file within the data set.
"""
print("Creating manifest %s ..." % manifest_path)
json_lines = []
for subfolder, _, filelist in sorted(os.walk(data_dir)):
text_filelist = [
filename for filename in filelist if filename.endswith('trans.txt')
]
if len(text_filelist) > 0:
text_filepath = os.path.join(data_dir, subfolder, text_filelist[0])
for line in open(text_filepath):
segments = line.strip().split()
text = ' '.join(segments[1:]).lower()
audio_filepath = os.path.join(data_dir, subfolder,
segments[0] + '.flac')
audio_data, samplerate = soundfile.read(audio_filepath)
duration = float(len(audio_data)) / samplerate
json_lines.append(
json.dumps({
'audio_filepath': audio_filepath,
'duration': duration,
'text': text
}))
with open(manifest_path, 'w') as out_file:
for line in json_lines:
out_file.write(line + '\n')
def prepare_dataset(url, md5sum, target_dir, manifest_path):
"""
Download, unpack and create summmary manifest file.
"""
if not os.path.exists(os.path.join(target_dir, "LibriSpeech")):
# download
filepath = download(url, md5sum, target_dir)
# unpack
unpack(filepath, target_dir)
else:
print("Skip downloading and unpacking. Data already exists in %s." %
target_dir)
# create manifest json file
create_manifest(target_dir, manifest_path)
def main():
prepare_dataset(
url=URL_TEST_CLEAN,
md5sum=MD5_TEST_CLEAN,
target_dir=os.path.join(args.target_dir, "test-clean"),
manifest_path=args.manifest_prefix + ".test-clean")
prepare_dataset(
url=URL_DEV_CLEAN,
md5sum=MD5_DEV_CLEAN,
target_dir=os.path.join(args.target_dir, "dev-clean"),
manifest_path=args.manifest_prefix + ".dev-clean")
prepare_dataset(
url=URL_TRAIN_CLEAN_100,
md5sum=MD5_TRAIN_CLEAN_100,
target_dir=os.path.join(args.target_dir, "train-clean-100"),
manifest_path=args.manifest_prefix + ".train-clean-100")
if args.full_download:
prepare_dataset(
url=URL_TEST_OTHER,
md5sum=MD5_TEST_OTHER,
target_dir=os.path.join(args.target_dir, "test-other"),
manifest_path=args.manifest_prefix + ".test-other")
prepare_dataset(
url=URL_DEV_OTHER,
md5sum=MD5_DEV_OTHER,
target_dir=os.path.join(args.target_dir, "dev-other"),
manifest_path=args.manifest_prefix + ".dev-other")
prepare_dataset(
url=URL_TRAIN_CLEAN_360,
md5sum=MD5_TRAIN_CLEAN_360,
target_dir=os.path.join(args.target_dir, "train-clean-360"),
manifest_path=args.manifest_prefix + ".train-clean-360")
prepare_dataset(
url=URL_TRAIN_OTHER_500,
md5sum=MD5_TRAIN_OTHER_500,
target_dir=os.path.join(args.target_dir, "train-other-500"),
manifest_path=args.manifest_prefix + ".train-other-500")
if __name__ == '__main__':
main()

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datasets/run_all.sh
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cd librispeech
python librispeech.py
if [ $? -ne 0 ]; then
echo "Prepare LibriSpeech failed. Terminated."
exit 1
fi
cd -
cat librispeech/manifest.train* | shuf > manifest.train
cat librispeech/manifest.dev-clean > manifest.dev
cat librispeech/manifest.test-clean > manifest.test
echo "All done."

28
datasets/vocab/eng_vocab.txt
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'
a
b
c
d
e
f
g
h
i
j
k
l
m
n
o
p
q
r
s
t
u
v
w
x
y
z

59
decoder.py
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"""Contains various CTC decoder."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import numpy as np
from itertools import groupby
def ctc_best_path_decode(probs_seq, vocabulary):
"""Best path decoding, also called argmax decoding or greedy decoding.
Path consisting of the most probable tokens are further post-processed to
remove consecutive repetitions and all blanks.
:param probs_seq: 2-D list of probabilities over the vocabulary for each
character. Each element is a list of float probabilities
for one character.
:type probs_seq: list
:param vocabulary: Vocabulary list.
:type vocabulary: list
:return: Decoding result string.
:rtype: baseline
"""
# dimension verification
for probs in probs_seq:
if not len(probs) == len(vocabulary) + 1:
raise ValueError("probs_seq dimension mismatchedd with vocabulary")
# argmax to get the best index for each time step
max_index_list = list(np.array(probs_seq).argmax(axis=1))
# remove consecutive duplicate indexes
index_list = [index_group[0] for index_group in groupby(max_index_list)]
# remove blank indexes
blank_index = len(vocabulary)
index_list = [index for index in index_list if index != blank_index]
# convert index list to string
return ''.join([vocabulary[index] for index in index_list])
def ctc_decode(probs_seq, vocabulary, method):
"""CTC-like sequence decoding from a sequence of likelihood probablilites.
:param probs_seq: 2-D list of probabilities over the vocabulary for each
character. Each element is a list of float probabilities
for one character.
:type probs_seq: list
:param vocabulary: Vocabulary list.
:type vocabulary: list
:param method: Decoding method name, with options: "best_path".
:type method: basestring
:return: Decoding result string.
:rtype: baseline
"""
for prob_list in probs_seq:
if not len(prob_list) == len(vocabulary) + 1:
raise ValueError("probs dimension mismatchedd with vocabulary")
if method == "best_path":
return ctc_best_path_decode(probs_seq, vocabulary)
else:
raise ValueError("Decoding method [%s] is not supported.")

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error_rate.py
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# -*- coding: utf-8 -*-
"""This module provides functions to calculate error rate in different level.
e.g. wer for word-level, cer for char-level.
"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import numpy as np
def _levenshtein_distance(ref, hyp):
"""Levenshtein distance is a string metric for measuring the difference between
two sequences. Informally, the levenshtein disctance is defined as the minimum
number of single-character edits (substitutions, insertions or deletions)
required to change one word into the other. We can naturally extend the edits to
word level when calculate levenshtein disctance for two sentences.
"""
ref_len = len(ref)
hyp_len = len(hyp)
# special case
if ref == hyp:
return 0
if ref_len == 0:
return hyp_len
if hyp_len == 0:
return ref_len
distance = np.zeros((ref_len + 1, hyp_len + 1), dtype=np.int32)
# initialize distance matrix
for j in xrange(hyp_len + 1):
distance[0][j] = j
for i in xrange(ref_len + 1):
distance[i][0] = i
# calculate levenshtein distance
for i in xrange(1, ref_len + 1):
for j in xrange(1, hyp_len + 1):
if ref[i - 1] == hyp[j - 1]:
distance[i][j] = distance[i - 1][j - 1]
else:
s_num = distance[i - 1][j - 1] + 1
i_num = distance[i][j - 1] + 1
d_num = distance[i - 1][j] + 1
distance[i][j] = min(s_num, i_num, d_num)
return distance[ref_len][hyp_len]
def wer(reference, hypothesis, ignore_case=False, delimiter=' '):
"""Calculate word error rate (WER). WER compares reference text and
hypothesis text in word-level. WER is defined as:
.. math::
WER = (Sw + Dw + Iw) / Nw
where
.. code-block:: text
Sw is the number of words subsituted,
Dw is the number of words deleted,
Iw is the number of words inserted,
Nw is the number of words in the reference
We can use levenshtein distance to calculate WER. Please draw an attention that
empty items will be removed when splitting sentences by delimiter.
:param reference: The reference sentence.
:type reference: basestring
:param hypothesis: The hypothesis sentence.
:type hypothesis: basestring
:param ignore_case: Whether case-sensitive or not.
:type ignore_case: bool
:param delimiter: Delimiter of input sentences.
:type delimiter: char
:return: Word error rate.
:rtype: float
:raises ValueError: If the reference length is zero.
"""
if ignore_case == True:
reference = reference.lower()
hypothesis = hypothesis.lower()
ref_words = filter(None, reference.split(delimiter))
hyp_words = filter(None, hypothesis.split(delimiter))
if len(ref_words) == 0:
raise ValueError("Reference's word number should be greater than 0.")
edit_distance = _levenshtein_distance(ref_words, hyp_words)
wer = float(edit_distance) / len(ref_words)
return wer
def cer(reference, hypothesis, ignore_case=False):
"""Calculate charactor error rate (CER). CER compares reference text and
hypothesis text in char-level. CER is defined as:
.. math::
CER = (Sc + Dc + Ic) / Nc
where
.. code-block:: text
Sc is the number of characters substituted,
Dc is the number of characters deleted,
Ic is the number of characters inserted
Nc is the number of characters in the reference
We can use levenshtein distance to calculate CER. Chinese input should be
encoded to unicode. Please draw an attention that the leading and tailing
white space characters will be truncated and multiple consecutive white
space characters in a sentence will be replaced by one white space character.
:param reference: The reference sentence.
:type reference: basestring
:param hypothesis: The hypothesis sentence.
:type hypothesis: basestring
:param ignore_case: Whether case-sensitive or not.
:type ignore_case: bool
:return: Character error rate.
:rtype: float
:raises ValueError: If the reference length is zero.
"""
if ignore_case == True:
reference = reference.lower()
hypothesis = hypothesis.lower()
reference = ' '.join(filter(None, reference.split(' ')))
hypothesis = ' '.join(filter(None, hypothesis.split(' ')))
if len(reference) == 0:
raise ValueError("Length of reference should be greater than 0.")
edit_distance = _levenshtein_distance(reference, hypothesis)
cer = float(edit_distance) / len(reference)
return cer

137
infer.py
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"""Inferer for DeepSpeech2 model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import argparse
import gzip
import distutils.util
import multiprocessing
import paddle.v2 as paddle
from data_utils.data import DataGenerator
from model import deep_speech2
from decoder import ctc_decode
import utils
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument(
"--num_samples",
default=10,
type=int,
help="Number of samples for inference. (default: %(default)s)")
parser.add_argument(
"--num_conv_layers",
default=2,
type=int,
help="Convolution layer number. (default: %(default)s)")
parser.add_argument(
"--num_rnn_layers",
default=3,
type=int,
help="RNN layer number. (default: %(default)s)")
parser.add_argument(
"--rnn_layer_size",
default=512,
type=int,
help="RNN layer cell number. (default: %(default)s)")
parser.add_argument(
"--use_gpu",
default=True,
type=distutils.util.strtobool,
help="Use gpu or not. (default: %(default)s)")
parser.add_argument(
"--num_threads_data",
default=multiprocessing.cpu_count(),
type=int,
help="Number of cpu threads for preprocessing data. (default: %(default)s)")
parser.add_argument(
"--mean_std_filepath",
default='mean_std.npz',
type=str,
help="Manifest path for normalizer. (default: %(default)s)")
parser.add_argument(
"--decode_manifest_path",
default='datasets/manifest.test',
type=str,
help="Manifest path for decoding. (default: %(default)s)")
parser.add_argument(
"--model_filepath",
default='checkpoints/params.latest.tar.gz',
type=str,
help="Model filepath. (default: %(default)s)")
parser.add_argument(
"--vocab_filepath",
default='datasets/vocab/eng_vocab.txt',
type=str,
help="Vocabulary filepath. (default: %(default)s)")
args = parser.parse_args()
def infer():
"""Max-ctc-decoding for DeepSpeech2."""
# initialize data generator
data_generator = DataGenerator(
vocab_filepath=args.vocab_filepath,
mean_std_filepath=args.mean_std_filepath,
augmentation_config='{}',
num_threads=args.num_threads_data)
# create network config
# paddle.data_type.dense_array is used for variable batch input.
# The size 161 * 161 is only an placeholder value and the real shape
# of input batch data will be induced during training.
audio_data = paddle.layer.data(
name="audio_spectrogram", type=paddle.data_type.dense_array(161 * 161))
text_data = paddle.layer.data(
name="transcript_text",
type=paddle.data_type.integer_value_sequence(data_generator.vocab_size))
output_probs = deep_speech2(
audio_data=audio_data,
text_data=text_data,
dict_size=data_generator.vocab_size,
num_conv_layers=args.num_conv_layers,
num_rnn_layers=args.num_rnn_layers,
rnn_size=args.rnn_layer_size,
is_inference=True)
# load parameters
parameters = paddle.parameters.Parameters.from_tar(
gzip.open(args.model_filepath))
# prepare infer data
batch_reader = data_generator.batch_reader_creator(
manifest_path=args.decode_manifest_path,
batch_size=args.num_samples,
sortagrad=False,
shuffle_method=None)
infer_data = batch_reader().next()
# run inference
infer_results = paddle.infer(
output_layer=output_probs, parameters=parameters, input=infer_data)
num_steps = len(infer_results) // len(infer_data)
probs_split = [
infer_results[i * num_steps:(i + 1) * num_steps]
for i in xrange(len(infer_data))
]
# decode and print
for i, probs in enumerate(probs_split):
output_transcription = ctc_decode(
probs_seq=probs,
vocabulary=data_generator.vocab_list,
method="best_path")
target_transcription = ''.join(
[data_generator.vocab_list[index] for index in infer_data[i][1]])
print("Target Transcription: %s \nOutput Transcription: %s \n" %
(target_transcription, output_transcription))
def main():
utils.print_arguments(args)
paddle.init(use_gpu=args.use_gpu, trainer_count=1)
infer()
if __name__ == '__main__':
main()

143
model.py
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"""Contains DeepSpeech2 model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import paddle.v2 as paddle
def conv_bn_layer(input, filter_size, num_channels_in, num_channels_out, stride,
padding, act):
"""
Convolution layer with batch normalization.
"""
conv_layer = paddle.layer.img_conv(
input=input,
filter_size=filter_size,
num_channels=num_channels_in,
num_filters=num_channels_out,
stride=stride,
padding=padding,
act=paddle.activation.Linear(),
bias_attr=False)
return paddle.layer.batch_norm(input=conv_layer, act=act)
def bidirectional_simple_rnn_bn_layer(name, input, size, act):
"""
Bidirectonal simple rnn layer with sequence-wise batch normalization.
The batch normalization is only performed on input-state weights.
"""
# input-hidden weights shared across bi-direcitonal rnn.
input_proj = paddle.layer.fc(
input=input, size=size, act=paddle.activation.Linear(), bias_attr=False)
# batch norm is only performed on input-state projection
input_proj_bn = paddle.layer.batch_norm(
input=input_proj, act=paddle.activation.Linear())
# forward and backward in time
forward_simple_rnn = paddle.layer.recurrent(
input=input_proj_bn, act=act, reverse=False)
backward_simple_rnn = paddle.layer.recurrent(
input=input_proj_bn, act=act, reverse=True)
return paddle.layer.concat(input=[forward_simple_rnn, backward_simple_rnn])
def conv_group(input, num_stacks):
"""
Convolution group with several stacking convolution layers.
"""
conv = conv_bn_layer(
input=input,
filter_size=(11, 41),
num_channels_in=1,
num_channels_out=32,
stride=(3, 2),
padding=(5, 20),
act=paddle.activation.BRelu())
for i in xrange(num_stacks - 1):
conv = conv_bn_layer(
input=conv,
filter_size=(11, 21),
num_channels_in=32,
num_channels_out=32,
stride=(1, 2),
padding=(5, 10),
act=paddle.activation.BRelu())
output_num_channels = 32
output_height = 160 // pow(2, num_stacks) + 1
return conv, output_num_channels, output_height
def rnn_group(input, size, num_stacks):
"""
RNN group with several stacking RNN layers.
"""
output = input
for i in xrange(num_stacks):
output = bidirectional_simple_rnn_bn_layer(
name=str(i), input=output, size=size, act=paddle.activation.BRelu())
return output
def deep_speech2(audio_data,
text_data,
dict_size,
num_conv_layers=2,
num_rnn_layers=3,
rnn_size=256,
is_inference=False):
"""
The whole DeepSpeech2 model structure (a simplified version).
:param audio_data: Audio spectrogram data layer.
:type audio_data: LayerOutput
:param text_data: Transcription text data layer.
:type text_data: LayerOutput
:param dict_size: Dictionary size for tokenized transcription.
:type dict_size: int
:param num_conv_layers: Number of stacking convolution layers.
:type num_conv_layers: int
:param num_rnn_layers: Number of stacking RNN layers.
:type num_rnn_layers: int
:param rnn_size: RNN layer size (number of RNN cells).
:type rnn_size: int
:param is_inference: False in the training mode, and True in the
inferene mode.
:type is_inference: bool
:return: If is_inference set False, return a ctc cost layer;
if is_inference set True, return a sequence layer of output
probability distribution.
:rtype: tuple of LayerOutput
"""
# convolution group
conv_group_output, conv_group_num_channels, conv_group_height = conv_group(
input=audio_data, num_stacks=num_conv_layers)
# convert data form convolution feature map to sequence of vectors
conv2seq = paddle.layer.block_expand(
input=conv_group_output,
num_channels=conv_group_num_channels,
stride_x=1,
stride_y=1,
block_x=1,
block_y=conv_group_height)
# rnn group
rnn_group_output = rnn_group(
input=conv2seq, size=rnn_size, num_stacks=num_rnn_layers)
fc = paddle.layer.fc(
input=rnn_group_output,
size=dict_size + 1,
act=paddle.activation.Linear(),
bias_attr=True)
if is_inference:
# probability distribution with softmax
return paddle.layer.mixed(
input=paddle.layer.identity_projection(input=fc),
act=paddle.activation.Softmax())
else:
# ctc cost
return paddle.layer.warp_ctc(
input=fc,
label=text_data,
size=dict_size + 1,
blank=dict_size,
norm_by_times=True)

3
requirements.txt
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wget==3.2
scipy==0.13.1
resampy==0.1.5

33
setup.sh
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#!/bin/bash
# install python dependencies
if [ -f "requirements.txt" ]; then
pip install -r requirements.txt
fi
if [ $? != 0 ]; then
echo "Install python dependencies failed !!!"
exit 1
fi
# install package Soundfile
curl -O "http://www.mega-nerd.com/libsndfile/files/libsndfile-1.0.28.tar.gz"
if [ $? != 0 ]; then
echo "Download libsndfile-1.0.28.tar.gz failed !!!"
exit 1
fi
tar -zxvf libsndfile-1.0.28.tar.gz
cd libsndfile-1.0.28
./configure && make && make install
cd -
rm -rf libsndfile-1.0.28
rm libsndfile-1.0.28.tar.gz
pip install SoundFile==0.9.0.post1
if [ $? != 0 ]; then
echo "Install SoundFile failed !!!"
exit 1
fi
# prepare ./checkpoints
mkdir checkpoints
echo "Install all dependencies successfully."

59
tests/test_error_rate.py
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# -*- coding: utf-8 -*-
"""Test error rate."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import unittest
import error_rate
class TestParse(unittest.TestCase):
def test_wer_1(self):
ref = 'i UM the PHONE IS i LEFT THE portable PHONE UPSTAIRS last night'
hyp = 'i GOT IT TO the FULLEST i LOVE TO portable FROM OF STORES last night'
word_error_rate = error_rate.wer(ref, hyp)
self.assertTrue(abs(word_error_rate - 0.769230769231) < 1e-6)
def test_wer_2(self):
ref = 'i UM the PHONE IS i LEFT THE portable PHONE UPSTAIRS last night'
word_error_rate = error_rate.wer(ref, ref)
self.assertEqual(word_error_rate, 0.0)
def test_wer_3(self):
ref = ' '
hyp = 'Hypothesis sentence'
with self.assertRaises(ValueError):
word_error_rate = error_rate.wer(ref, hyp)
def test_cer_1(self):
ref = 'werewolf'
hyp = 'weae wolf'
char_error_rate = error_rate.cer(ref, hyp)
self.assertTrue(abs(char_error_rate - 0.25) < 1e-6)
def test_cer_2(self):
ref = 'werewolf'
char_error_rate = error_rate.cer(ref, ref)
self.assertEqual(char_error_rate, 0.0)
def test_cer_3(self):
ref = u'我是中国人'
hyp = u'我是 美洲人'
char_error_rate = error_rate.cer(ref, hyp)
self.assertTrue(abs(char_error_rate - 0.6) < 1e-6)
def test_cer_4(self):
ref = u'我是中国人'
char_error_rate = error_rate.cer(ref, ref)
self.assertFalse(char_error_rate, 0.0)
def test_cer_5(self):
ref = ''
hyp = 'Hypothesis'
with self.assertRaises(ValueError):
char_error_rate = error_rate.cer(ref, hyp)
if __name__ == '__main__':
unittest.main()

226
train.py
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"""Trainer for DeepSpeech2 model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import sys
import os
import argparse
import gzip
import time
import distutils.util
import multiprocessing
import paddle.v2 as paddle
from model import deep_speech2
from data_utils.data import DataGenerator
import utils
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument(
"--batch_size", default=256, type=int, help="Minibatch size.")
parser.add_argument(
"--num_passes",
default=200,
type=int,
help="Training pass number. (default: %(default)s)")
parser.add_argument(
"--num_conv_layers",
default=2,
type=int,
help="Convolution layer number. (default: %(default)s)")
parser.add_argument(
"--num_rnn_layers",
default=3,
type=int,
help="RNN layer number. (default: %(default)s)")
parser.add_argument(
"--rnn_layer_size",
default=512,
type=int,
help="RNN layer cell number. (default: %(default)s)")
parser.add_argument(
"--adam_learning_rate",
default=5e-4,
type=float,
help="Learning rate for ADAM Optimizer. (default: %(default)s)")
parser.add_argument(
"--use_gpu",
default=True,
type=distutils.util.strtobool,
help="Use gpu or not. (default: %(default)s)")
parser.add_argument(
"--use_sortagrad",
default=True,
type=distutils.util.strtobool,
help="Use sortagrad or not. (default: %(default)s)")
parser.add_argument(
"--max_duration",
default=27.0,
type=float,
help="Audios with duration larger than this will be discarded. "
"(default: %(default)s)")
parser.add_argument(
"--min_duration",
default=0.0,
type=float,
help="Audios with duration smaller than this will be discarded. "
"(default: %(default)s)")
parser.add_argument(
"--shuffle_method",
default='batch_shuffle_clipped',
type=str,
help="Shuffle method: 'instance_shuffle', 'batch_shuffle', "
"'batch_shuffle_batch'. (default: %(default)s)")
parser.add_argument(
"--trainer_count",
default=8,
type=int,
help="Trainer number. (default: %(default)s)")
parser.add_argument(
"--num_threads_data",
default=multiprocessing.cpu_count(),
type=int,
help="Number of cpu threads for preprocessing data. (default: %(default)s)")
parser.add_argument(
"--mean_std_filepath",
default='mean_std.npz',
type=str,
help="Manifest path for normalizer. (default: %(default)s)")
parser.add_argument(
"--train_manifest_path",
default='datasets/manifest.train',
type=str,
help="Manifest path for training. (default: %(default)s)")
parser.add_argument(
"--dev_manifest_path",
default='datasets/manifest.dev',
type=str,
help="Manifest path for validation. (default: %(default)s)")
parser.add_argument(
"--vocab_filepath",
default='datasets/vocab/eng_vocab.txt',
type=str,
help="Vocabulary filepath. (default: %(default)s)")
parser.add_argument(
"--init_model_path",
default=None,
type=str,
help="If set None, the training will start from scratch. "
"Otherwise, the training will resume from "
"the existing model of this path. (default: %(default)s)")
parser.add_argument(
"--augmentation_config",
default='[{"type": "shift", '
'"params": {"min_shift_ms": -5, "max_shift_ms": 5},'
'"prob": 1.0}]',
type=str,
help="Augmentation configuration in json-format. "
"(default: %(default)s)")
args = parser.parse_args()
def train():
"""DeepSpeech2 training."""
# initialize data generator
def data_generator():
return DataGenerator(
vocab_filepath=args.vocab_filepath,
mean_std_filepath=args.mean_std_filepath,
augmentation_config=args.augmentation_config,
max_duration=args.max_duration,
min_duration=args.min_duration,
num_threads=args.num_threads_data)
train_generator = data_generator()
test_generator = data_generator()
# create network config
# paddle.data_type.dense_array is used for variable batch input.
# The size 161 * 161 is only an placeholder value and the real shape
# of input batch data will be induced during training.
audio_data = paddle.layer.data(
name="audio_spectrogram", type=paddle.data_type.dense_array(161 * 161))
text_data = paddle.layer.data(
name="transcript_text",
type=paddle.data_type.integer_value_sequence(
train_generator.vocab_size))
cost = deep_speech2(
audio_data=audio_data,
text_data=text_data,
dict_size=train_generator.vocab_size,
num_conv_layers=args.num_conv_layers,
num_rnn_layers=args.num_rnn_layers,
rnn_size=args.rnn_layer_size,
is_inference=False)
# create/load parameters and optimizer
if args.init_model_path is None:
parameters = paddle.parameters.create(cost)
else:
if not os.path.isfile(args.init_model_path):
raise IOError("Invalid model!")
parameters = paddle.parameters.Parameters.from_tar(
gzip.open(args.init_model_path))
optimizer = paddle.optimizer.Adam(
learning_rate=args.adam_learning_rate, gradient_clipping_threshold=400)
trainer = paddle.trainer.SGD(
cost=cost, parameters=parameters, update_equation=optimizer)
# prepare data reader
train_batch_reader = train_generator.batch_reader_creator(
manifest_path=args.train_manifest_path,
batch_size=args.batch_size,
min_batch_size=args.trainer_count,
sortagrad=args.use_sortagrad if args.init_model_path is None else False,
shuffle_method=args.shuffle_method)
test_batch_reader = test_generator.batch_reader_creator(
manifest_path=args.dev_manifest_path,
batch_size=args.batch_size,
min_batch_size=1, # must be 1, but will have errors.
sortagrad=False,
shuffle_method=None)
# create event handler
def event_handler(event):
global start_time, cost_sum, cost_counter
if isinstance(event, paddle.event.EndIteration):
cost_sum += event.cost
cost_counter += 1
if (event.batch_id + 1) % 100 == 0:
print("\nPass: %d, Batch: %d, TrainCost: %f" % (
event.pass_id, event.batch_id + 1, cost_sum / cost_counter))
cost_sum, cost_counter = 0.0, 0
with gzip.open("checkpoints/params.latest.tar.gz", 'w') as f:
parameters.to_tar(f)
else:
sys.stdout.write('.')
sys.stdout.flush()
if isinstance(event, paddle.event.BeginPass):
start_time = time.time()
cost_sum, cost_counter = 0.0, 0
if isinstance(event, paddle.event.EndPass):
result = trainer.test(
reader=test_batch_reader, feeding=test_generator.feeding)
print("\n------- Time: %d sec, Pass: %d, ValidationCost: %s" %
(time.time() - start_time, event.pass_id, result.cost))
with gzip.open("checkpoints/params.pass-%d.tar.gz" % event.pass_id,
'w') as f:
parameters.to_tar(f)
# run train
trainer.train(
reader=train_batch_reader,
event_handler=event_handler,
num_passes=args.num_passes,
feeding=train_generator.feeding)
def main():
utils.print_arguments(args)
paddle.init(use_gpu=args.use_gpu, trainer_count=args.trainer_count)
train()
if __name__ == '__main__':
main()

25
utils.py
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"""Contains common utility functions."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
def print_arguments(args):
"""Print argparse's arguments.
Usage:
.. code-block:: python
parser = argparse.ArgumentParser()
parser.add_argument("name", default="Jonh", type=str, help="User name.")
args = parser.parse_args()
print_arguments(args)
:param args: Input argparse.Namespace for printing.
:type args: argparse.Namespace
"""
print("----- Configuration Arguments -----")
for arg, value in vars(args).iteritems():
print("%s: %s" % (arg, value))
print("------------------------------------")
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