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PaddleSpeech/deepspeech/frontend/audio.py

722 lines
29 KiB

4 years ago
# Copyright (c) 2021 PaddlePaddle Authors. All Rights Reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
"""Contains the audio segment class."""
E2E/Streaming Transformer/Conformer ASR (#578) * add cmvn and label smoothing loss layer * add layer for transformer * add glu and conformer conv * add torch compatiable hack, mask funcs * not hack size since it exists * add test; attention * add attention, common utils, hack paddle * add audio utils * conformer batch padding mask bug fix #223 * fix typo, python infer fix rnn mem opt name error and batchnorm1d, will be available at 2.0.2 * fix ci * fix ci * add encoder * refactor egs * add decoder * refactor ctc, add ctc align, refactor ckpt, add warmup lr scheduler, cmvn utils * refactor docs * add fix * fix readme * fix bugs, refactor collator, add pad_sequence, fix ckpt bugs * fix docstring * refactor data feed order * add u2 model * refactor cmvn, test * add utils * add u2 config * fix bugs * fix bugs * fix autograd maybe has problem when using inplace operation * refactor data, build vocab; add format data * fix text featurizer * refactor build vocab * add fbank, refactor feature of speech * refactor audio feat * refactor data preprare * refactor data * model init from config * add u2 bins * flake8 * can train * fix bugs, add coverage, add scripts * test can run * fix data * speed perturb with sox * add spec aug * fix for train * fix train logitc * fix logger * log valid loss, time dataset process * using np for speed perturb, remove some debug log of grad clip * fix logger * fix build vocab * fix logger name * using module logger as default * fix * fix install * reorder imports * fix board logger * fix logger * kaldi fbank and mfcc * fix cmvn and print prarams * fix add_eos_sos and cmvn * fix cmvn compute * fix logger and cmvn * fix subsampling, label smoothing loss, remove useless * add notebook test * fix log * fix tb logger * multi gpu valid * fix log * fix log * fix config * fix compute cmvn, need paddle 2.1 * add cmvn notebook * fix layer tools * fix compute cmvn * add rtf * fix decoding * fix layer tools * fix log, add avg script * more avg and test info * fix dataset pickle problem; using 2.1 paddle; num_workers can > 0; ckpt save in exp dir;fix setup.sh; * add vimrc * refactor tiny script, add transformer and stream conf * spm demo; librisppech scripts and confs * fix log * add librispeech scripts * refactor data pipe; fix conf; fix u2 default params * fix bugs * refactor aishell scripts * fix test * fix cmvn * fix s0 scripts * fix ds2 scripts and bugs * fix dev & test dataset filter * fix dataset filter * filter dev * fix ckpt path * filter test, since librispeech will cause OOM, but all test wer will be worse, since mismatch train with test * add comment * add syllable doc * fix ds2 configs * add doc * add pypinyin tools * fix decoder using blank_id=0 * mmseg with pybind11 * format code
4 years ago
import copy
import io
E2E/Streaming Transformer/Conformer ASR (#578) * add cmvn and label smoothing loss layer * add layer for transformer * add glu and conformer conv * add torch compatiable hack, mask funcs * not hack size since it exists * add test; attention * add attention, common utils, hack paddle * add audio utils * conformer batch padding mask bug fix #223 * fix typo, python infer fix rnn mem opt name error and batchnorm1d, will be available at 2.0.2 * fix ci * fix ci * add encoder * refactor egs * add decoder * refactor ctc, add ctc align, refactor ckpt, add warmup lr scheduler, cmvn utils * refactor docs * add fix * fix readme * fix bugs, refactor collator, add pad_sequence, fix ckpt bugs * fix docstring * refactor data feed order * add u2 model * refactor cmvn, test * add utils * add u2 config * fix bugs * fix bugs * fix autograd maybe has problem when using inplace operation * refactor data, build vocab; add format data * fix text featurizer * refactor build vocab * add fbank, refactor feature of speech * refactor audio feat * refactor data preprare * refactor data * model init from config * add u2 bins * flake8 * can train * fix bugs, add coverage, add scripts * test can run * fix data * speed perturb with sox * add spec aug * fix for train * fix train logitc * fix logger * log valid loss, time dataset process * using np for speed perturb, remove some debug log of grad clip * fix logger * fix build vocab * fix logger name * using module logger as default * fix * fix install * reorder imports * fix board logger * fix logger * kaldi fbank and mfcc * fix cmvn and print prarams * fix add_eos_sos and cmvn * fix cmvn compute * fix logger and cmvn * fix subsampling, label smoothing loss, remove useless * add notebook test * fix log * fix tb logger * multi gpu valid * fix log * fix log * fix config * fix compute cmvn, need paddle 2.1 * add cmvn notebook * fix layer tools * fix compute cmvn * add rtf * fix decoding * fix layer tools * fix log, add avg script * more avg and test info * fix dataset pickle problem; using 2.1 paddle; num_workers can > 0; ckpt save in exp dir;fix setup.sh; * add vimrc * refactor tiny script, add transformer and stream conf * spm demo; librisppech scripts and confs * fix log * add librispeech scripts * refactor data pipe; fix conf; fix u2 default params * fix bugs * refactor aishell scripts * fix test * fix cmvn * fix s0 scripts * fix ds2 scripts and bugs * fix dev & test dataset filter * fix dataset filter * filter dev * fix ckpt path * filter test, since librispeech will cause OOM, but all test wer will be worse, since mismatch train with test * add comment * add syllable doc * fix ds2 configs * add doc * add pypinyin tools * fix decoder using blank_id=0 * mmseg with pybind11 * format code
4 years ago
import random
import re
E2E/Streaming Transformer/Conformer ASR (#578) * add cmvn and label smoothing loss layer * add layer for transformer * add glu and conformer conv * add torch compatiable hack, mask funcs * not hack size since it exists * add test; attention * add attention, common utils, hack paddle * add audio utils * conformer batch padding mask bug fix #223 * fix typo, python infer fix rnn mem opt name error and batchnorm1d, will be available at 2.0.2 * fix ci * fix ci * add encoder * refactor egs * add decoder * refactor ctc, add ctc align, refactor ckpt, add warmup lr scheduler, cmvn utils * refactor docs * add fix * fix readme * fix bugs, refactor collator, add pad_sequence, fix ckpt bugs * fix docstring * refactor data feed order * add u2 model * refactor cmvn, test * add utils * add u2 config * fix bugs * fix bugs * fix autograd maybe has problem when using inplace operation * refactor data, build vocab; add format data * fix text featurizer * refactor build vocab * add fbank, refactor feature of speech * refactor audio feat * refactor data preprare * refactor data * model init from config * add u2 bins * flake8 * can train * fix bugs, add coverage, add scripts * test can run * fix data * speed perturb with sox * add spec aug * fix for train * fix train logitc * fix logger * log valid loss, time dataset process * using np for speed perturb, remove some debug log of grad clip * fix logger * fix build vocab * fix logger name * using module logger as default * fix * fix install * reorder imports * fix board logger * fix logger * kaldi fbank and mfcc * fix cmvn and print prarams * fix add_eos_sos and cmvn * fix cmvn compute * fix logger and cmvn * fix subsampling, label smoothing loss, remove useless * add notebook test * fix log * fix tb logger * multi gpu valid * fix log * fix log * fix config * fix compute cmvn, need paddle 2.1 * add cmvn notebook * fix layer tools * fix compute cmvn * add rtf * fix decoding * fix layer tools * fix log, add avg script * more avg and test info * fix dataset pickle problem; using 2.1 paddle; num_workers can > 0; ckpt save in exp dir;fix setup.sh; * add vimrc * refactor tiny script, add transformer and stream conf * spm demo; librisppech scripts and confs * fix log * add librispeech scripts * refactor data pipe; fix conf; fix u2 default params * fix bugs * refactor aishell scripts * fix test * fix cmvn * fix s0 scripts * fix ds2 scripts and bugs * fix dev & test dataset filter * fix dataset filter * filter dev * fix ckpt path * filter test, since librispeech will cause OOM, but all test wer will be worse, since mismatch train with test * add comment * add syllable doc * fix ds2 configs * add doc * add pypinyin tools * fix decoder using blank_id=0 * mmseg with pybind11 * format code
4 years ago
import struct
import numpy as np
import resampy
E2E/Streaming Transformer/Conformer ASR (#578) * add cmvn and label smoothing loss layer * add layer for transformer * add glu and conformer conv * add torch compatiable hack, mask funcs * not hack size since it exists * add test; attention * add attention, common utils, hack paddle * add audio utils * conformer batch padding mask bug fix #223 * fix typo, python infer fix rnn mem opt name error and batchnorm1d, will be available at 2.0.2 * fix ci * fix ci * add encoder * refactor egs * add decoder * refactor ctc, add ctc align, refactor ckpt, add warmup lr scheduler, cmvn utils * refactor docs * add fix * fix readme * fix bugs, refactor collator, add pad_sequence, fix ckpt bugs * fix docstring * refactor data feed order * add u2 model * refactor cmvn, test * add utils * add u2 config * fix bugs * fix bugs * fix autograd maybe has problem when using inplace operation * refactor data, build vocab; add format data * fix text featurizer * refactor build vocab * add fbank, refactor feature of speech * refactor audio feat * refactor data preprare * refactor data * model init from config * add u2 bins * flake8 * can train * fix bugs, add coverage, add scripts * test can run * fix data * speed perturb with sox * add spec aug * fix for train * fix train logitc * fix logger * log valid loss, time dataset process * using np for speed perturb, remove some debug log of grad clip * fix logger * fix build vocab * fix logger name * using module logger as default * fix * fix install * reorder imports * fix board logger * fix logger * kaldi fbank and mfcc * fix cmvn and print prarams * fix add_eos_sos and cmvn * fix cmvn compute * fix logger and cmvn * fix subsampling, label smoothing loss, remove useless * add notebook test * fix log * fix tb logger * multi gpu valid * fix log * fix log * fix config * fix compute cmvn, need paddle 2.1 * add cmvn notebook * fix layer tools * fix compute cmvn * add rtf * fix decoding * fix layer tools * fix log, add avg script * more avg and test info * fix dataset pickle problem; using 2.1 paddle; num_workers can > 0; ckpt save in exp dir;fix setup.sh; * add vimrc * refactor tiny script, add transformer and stream conf * spm demo; librisppech scripts and confs * fix log * add librispeech scripts * refactor data pipe; fix conf; fix u2 default params * fix bugs * refactor aishell scripts * fix test * fix cmvn * fix s0 scripts * fix ds2 scripts and bugs * fix dev & test dataset filter * fix dataset filter * filter dev * fix ckpt path * filter test, since librispeech will cause OOM, but all test wer will be worse, since mismatch train with test * add comment * add syllable doc * fix ds2 configs * add doc * add pypinyin tools * fix decoder using blank_id=0 * mmseg with pybind11 * format code
4 years ago
import soundfile
import soxbindings as sox
8 years ago
from scipy import signal
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.
4 years ago
:type filepath: str|file
:return: Audio segment instance.
:rtype: AudioSegment
"""
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if isinstance(file, str) and re.findall(r".seqbin_\d+$", file):
return cls.from_sequence_file(file)
else:
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.
4 years ago
:type file: str|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
4 years ago
end = duration 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_sequence_file(cls, filepath):
"""Create audio segment from sequence file. Sequence file is a binary
file containing a collection of multiple audio files, with several
header bytes in the head indicating the offsets of each audio byte data
chunk.
The format is:
4 bytes (int, version),
4 bytes (int, num of utterance),
4 bytes (int, bytes per header),
[bytes_per_header*(num_utterance+1)] bytes (offsets for each audio),
audio_bytes_data_of_1st_utterance,
audio_bytes_data_of_2nd_utterance,
......
Sequence file name must end with ".seqbin". And the filename of the 5th
utterance's audio file in sequence file "xxx.seqbin" must be
"xxx.seqbin_5", with "5" indicating the utterance index within this
sequence file (starting from 1).
:param filepath: Filepath of sequence file.
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:type filepath: str
:return: Audio segment instance.
:rtype: AudioSegment
"""
# parse filepath
matches = re.match(r"(.+\.seqbin)_(\d+)", filepath)
if matches is None:
raise IOError("File type of %s is not supported" % filepath)
filename = matches.group(1)
fileno = int(matches.group(2))
# read headers
f = io.open(filename, mode='rb', encoding='utf8')
version = f.read(4)
num_utterances = struct.unpack("i", f.read(4))[0]
bytes_per_header = struct.unpack("i", f.read(4))[0]
header_bytes = f.read(bytes_per_header * (num_utterances + 1))
header = [
struct.unpack("i", header_bytes[bytes_per_header * i:
bytes_per_header * (i + 1)])[0]
for i in range(num_utterances + 1)
]
# read audio bytes
f.seek(header[fileno - 1])
audio_bytes = f.read(header[fileno] - header[fileno - 1])
f.close()
# create audio segment
try:
return cls.from_bytes(audio_bytes)
except Exception as e:
samples = np.frombuffer(audio_bytes, dtype='int16')
return cls(samples=samples, sample_rate=8000)
@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):
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"""Concatenate an arbitrary number of audio segments together.
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:param *segments: Input audio segments to be concatenated.
:type *segments: tuple of AudioSegment
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:return: Audio segment instance as concatenating results.
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:rtype: AudioSegment
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:raises ValueError: If the number of segments is zero, or if the
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sample_rate of any segments does not match.
:raises TypeError: If any segment is not AudioSegment instance.
8 years ago
"""
# Perform basic sanity-checks.
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if len(segments) == 0:
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raise ValueError("No audio segments are given to concatenate.")
sample_rate = segments[0]._sample_rate
8 years ago
for seg in segments:
if sample_rate != seg._sample_rate:
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raise ValueError("Can't concatenate segments with "
"different sample rates")
if type(seg) is not cls:
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raise TypeError("Only audio segments of the same type "
8 years ago
"can be concatenated.")
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samples = np.concatenate([seg.samples for seg in segments])
return cls(samples, sample_rate)
8 years ago
@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.
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:type filepath: str|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])
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def superimpose(self, other):
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"""Add samples from another segment to those of this segment
(sample-wise addition, not segment concatenation).
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Note that this is an in-place transformation.
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:param other: Segment containing samples to be added in.
:type other: AudioSegments
:raise TypeError: If type of two segments don't match.
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:raise ValueError: If the sample rates of the two segments are not
equal, or if the lengths of segments don't match.
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"""
if isinstance(other, type(self)):
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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
8 years ago
def to_bytes(self, dtype='float32'):
"""Create a byte string containing the audio content.
8 years ago
: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()
E2E/Streaming Transformer/Conformer ASR (#578) * add cmvn and label smoothing loss layer * add layer for transformer * add glu and conformer conv * add torch compatiable hack, mask funcs * not hack size since it exists * add test; attention * add attention, common utils, hack paddle * add audio utils * conformer batch padding mask bug fix #223 * fix typo, python infer fix rnn mem opt name error and batchnorm1d, will be available at 2.0.2 * fix ci * fix ci * add encoder * refactor egs * add decoder * refactor ctc, add ctc align, refactor ckpt, add warmup lr scheduler, cmvn utils * refactor docs * add fix * fix readme * fix bugs, refactor collator, add pad_sequence, fix ckpt bugs * fix docstring * refactor data feed order * add u2 model * refactor cmvn, test * add utils * add u2 config * fix bugs * fix bugs * fix autograd maybe has problem when using inplace operation * refactor data, build vocab; add format data * fix text featurizer * refactor build vocab * add fbank, refactor feature of speech * refactor audio feat * refactor data preprare * refactor data * model init from config * add u2 bins * flake8 * can train * fix bugs, add coverage, add scripts * test can run * fix data * speed perturb with sox * add spec aug * fix for train * fix train logitc * fix logger * log valid loss, time dataset process * using np for speed perturb, remove some debug log of grad clip * fix logger * fix build vocab * fix logger name * using module logger as default * fix * fix install * reorder imports * fix board logger * fix logger * kaldi fbank and mfcc * fix cmvn and print prarams * fix add_eos_sos and cmvn * fix cmvn compute * fix logger and cmvn * fix subsampling, label smoothing loss, remove useless * add notebook test * fix log * fix tb logger * multi gpu valid * fix log * fix log * fix config * fix compute cmvn, need paddle 2.1 * add cmvn notebook * fix layer tools * fix compute cmvn * add rtf * fix decoding * fix layer tools * fix log, add avg script * more avg and test info * fix dataset pickle problem; using 2.1 paddle; num_workers can > 0; ckpt save in exp dir;fix setup.sh; * add vimrc * refactor tiny script, add transformer and stream conf * spm demo; librisppech scripts and confs * fix log * add librispeech scripts * refactor data pipe; fix conf; fix u2 default params * fix bugs * refactor aishell scripts * fix test * fix cmvn * fix s0 scripts * fix ds2 scripts and bugs * fix dev & test dataset filter * fix dataset filter * filter dev * fix ckpt path * filter test, since librispeech will cause OOM, but all test wer will be worse, since mismatch train with test * add comment * add syllable doc * fix ds2 configs * add doc * add pypinyin tools * fix decoder using blank_id=0 * mmseg with pybind11 * format code
4 years ago
def to(self, dtype='int16'):
"""Create a `dtype` audio content.
:param dtype: Data type for export samples. Options: 'int16', 'int32',
'float32', 'float64'. Default is 'float32'.
:type dtype: str
:return: np.ndarray containing `dtype` audio content.
:rtype: str
"""
samples = self._convert_samples_from_float32(self._samples, dtype)
return samples
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|1darray
"""
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.
"""
E2E/Streaming Transformer/Conformer ASR (#578) * add cmvn and label smoothing loss layer * add layer for transformer * add glu and conformer conv * add torch compatiable hack, mask funcs * not hack size since it exists * add test; attention * add attention, common utils, hack paddle * add audio utils * conformer batch padding mask bug fix #223 * fix typo, python infer fix rnn mem opt name error and batchnorm1d, will be available at 2.0.2 * fix ci * fix ci * add encoder * refactor egs * add decoder * refactor ctc, add ctc align, refactor ckpt, add warmup lr scheduler, cmvn utils * refactor docs * add fix * fix readme * fix bugs, refactor collator, add pad_sequence, fix ckpt bugs * fix docstring * refactor data feed order * add u2 model * refactor cmvn, test * add utils * add u2 config * fix bugs * fix bugs * fix autograd maybe has problem when using inplace operation * refactor data, build vocab; add format data * fix text featurizer * refactor build vocab * add fbank, refactor feature of speech * refactor audio feat * refactor data preprare * refactor data * model init from config * add u2 bins * flake8 * can train * fix bugs, add coverage, add scripts * test can run * fix data * speed perturb with sox * add spec aug * fix for train * fix train logitc * fix logger * log valid loss, time dataset process * using np for speed perturb, remove some debug log of grad clip * fix logger * fix build vocab * fix logger name * using module logger as default * fix * fix install * reorder imports * fix board logger * fix logger * kaldi fbank and mfcc * fix cmvn and print prarams * fix add_eos_sos and cmvn * fix cmvn compute * fix logger and cmvn * fix subsampling, label smoothing loss, remove useless * add notebook test * fix log * fix tb logger * multi gpu valid * fix log * fix log * fix config * fix compute cmvn, need paddle 2.1 * add cmvn notebook * fix layer tools * fix compute cmvn * add rtf * fix decoding * fix layer tools * fix log, add avg script * more avg and test info * fix dataset pickle problem; using 2.1 paddle; num_workers can > 0; ckpt save in exp dir;fix setup.sh; * add vimrc * refactor tiny script, add transformer and stream conf * spm demo; librisppech scripts and confs * fix log * add librispeech scripts * refactor data pipe; fix conf; fix u2 default params * fix bugs * refactor aishell scripts * fix test * fix cmvn * fix s0 scripts * fix ds2 scripts and bugs * fix dev & test dataset filter * fix dataset filter * filter dev * fix ckpt path * filter test, since librispeech will cause OOM, but all test wer will be worse, since mismatch train with test * add comment * add syllable doc * fix ds2 configs * add doc * add pypinyin tools * fix decoder using blank_id=0 * mmseg with pybind11 * format code
4 years ago
if speed_rate == 1.0:
return
if speed_rate <= 0:
raise ValueError("speed_rate should be greater than zero.")
E2E/Streaming Transformer/Conformer ASR (#578) * add cmvn and label smoothing loss layer * add layer for transformer * add glu and conformer conv * add torch compatiable hack, mask funcs * not hack size since it exists * add test; attention * add attention, common utils, hack paddle * add audio utils * conformer batch padding mask bug fix #223 * fix typo, python infer fix rnn mem opt name error and batchnorm1d, will be available at 2.0.2 * fix ci * fix ci * add encoder * refactor egs * add decoder * refactor ctc, add ctc align, refactor ckpt, add warmup lr scheduler, cmvn utils * refactor docs * add fix * fix readme * fix bugs, refactor collator, add pad_sequence, fix ckpt bugs * fix docstring * refactor data feed order * add u2 model * refactor cmvn, test * add utils * add u2 config * fix bugs * fix bugs * fix autograd maybe has problem when using inplace operation * refactor data, build vocab; add format data * fix text featurizer * refactor build vocab * add fbank, refactor feature of speech * refactor audio feat * refactor data preprare * refactor data * model init from config * add u2 bins * flake8 * can train * fix bugs, add coverage, add scripts * test can run * fix data * speed perturb with sox * add spec aug * fix for train * fix train logitc * fix logger * log valid loss, time dataset process * using np for speed perturb, remove some debug log of grad clip * fix logger * fix build vocab * fix logger name * using module logger as default * fix * fix install * reorder imports * fix board logger * fix logger * kaldi fbank and mfcc * fix cmvn and print prarams * fix add_eos_sos and cmvn * fix cmvn compute * fix logger and cmvn * fix subsampling, label smoothing loss, remove useless * add notebook test * fix log * fix tb logger * multi gpu valid * fix log * fix log * fix config * fix compute cmvn, need paddle 2.1 * add cmvn notebook * fix layer tools * fix compute cmvn * add rtf * fix decoding * fix layer tools * fix log, add avg script * more avg and test info * fix dataset pickle problem; using 2.1 paddle; num_workers can > 0; ckpt save in exp dir;fix setup.sh; * add vimrc * refactor tiny script, add transformer and stream conf * spm demo; librisppech scripts and confs * fix log * add librispeech scripts * refactor data pipe; fix conf; fix u2 default params * fix bugs * refactor aishell scripts * fix test * fix cmvn * fix s0 scripts * fix ds2 scripts and bugs * fix dev & test dataset filter * fix dataset filter * filter dev * fix ckpt path * filter test, since librispeech will cause OOM, but all test wer will be worse, since mismatch train with test * add comment * add syllable doc * fix ds2 configs * add doc * add pypinyin tools * fix decoder using blank_id=0 * mmseg with pybind11 * format code
4 years ago
# numpy
# 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)
E2E/Streaming Transformer/Conformer ASR (#578) * add cmvn and label smoothing loss layer * add layer for transformer * add glu and conformer conv * add torch compatiable hack, mask funcs * not hack size since it exists * add test; attention * add attention, common utils, hack paddle * add audio utils * conformer batch padding mask bug fix #223 * fix typo, python infer fix rnn mem opt name error and batchnorm1d, will be available at 2.0.2 * fix ci * fix ci * add encoder * refactor egs * add decoder * refactor ctc, add ctc align, refactor ckpt, add warmup lr scheduler, cmvn utils * refactor docs * add fix * fix readme * fix bugs, refactor collator, add pad_sequence, fix ckpt bugs * fix docstring * refactor data feed order * add u2 model * refactor cmvn, test * add utils * add u2 config * fix bugs * fix bugs * fix autograd maybe has problem when using inplace operation * refactor data, build vocab; add format data * fix text featurizer * refactor build vocab * add fbank, refactor feature of speech * refactor audio feat * refactor data preprare * refactor data * model init from config * add u2 bins * flake8 * can train * fix bugs, add coverage, add scripts * test can run * fix data * speed perturb with sox * add spec aug * fix for train * fix train logitc * fix logger * log valid loss, time dataset process * using np for speed perturb, remove some debug log of grad clip * fix logger * fix build vocab * fix logger name * using module logger as default * fix * fix install * reorder imports * fix board logger * fix logger * kaldi fbank and mfcc * fix cmvn and print prarams * fix add_eos_sos and cmvn * fix cmvn compute * fix logger and cmvn * fix subsampling, label smoothing loss, remove useless * add notebook test * fix log * fix tb logger * multi gpu valid * fix log * fix log * fix config * fix compute cmvn, need paddle 2.1 * add cmvn notebook * fix layer tools * fix compute cmvn * add rtf * fix decoding * fix layer tools * fix log, add avg script * more avg and test info * fix dataset pickle problem; using 2.1 paddle; num_workers can > 0; ckpt save in exp dir;fix setup.sh; * add vimrc * refactor tiny script, add transformer and stream conf * spm demo; librisppech scripts and confs * fix log * add librispeech scripts * refactor data pipe; fix conf; fix u2 default params * fix bugs * refactor aishell scripts * fix test * fix cmvn * fix s0 scripts * fix ds2 scripts and bugs * fix dev & test dataset filter * fix dataset filter * filter dev * fix ckpt path * filter test, since librispeech will cause OOM, but all test wer will be worse, since mismatch train with test * add comment * add syllable doc * fix ds2 configs * add doc * add pypinyin tools * fix decoder using blank_id=0 * mmseg with pybind11 * format code
4 years ago
# sox, slow
tfm = sox.Transformer()
tfm.set_globals(multithread=False)
tfm.speed(speed_rate)
self._samples = tfm.build_array(
input_array=self._samples,
sample_rate_in=self._sample_rate).squeeze(-1).astype(
np.float32).copy()
E2E/Streaming Transformer/Conformer ASR (#578) * add cmvn and label smoothing loss layer * add layer for transformer * add glu and conformer conv * add torch compatiable hack, mask funcs * not hack size since it exists * add test; attention * add attention, common utils, hack paddle * add audio utils * conformer batch padding mask bug fix #223 * fix typo, python infer fix rnn mem opt name error and batchnorm1d, will be available at 2.0.2 * fix ci * fix ci * add encoder * refactor egs * add decoder * refactor ctc, add ctc align, refactor ckpt, add warmup lr scheduler, cmvn utils * refactor docs * add fix * fix readme * fix bugs, refactor collator, add pad_sequence, fix ckpt bugs * fix docstring * refactor data feed order * add u2 model * refactor cmvn, test * add utils * add u2 config * fix bugs * fix bugs * fix autograd maybe has problem when using inplace operation * refactor data, build vocab; add format data * fix text featurizer * refactor build vocab * add fbank, refactor feature of speech * refactor audio feat * refactor data preprare * refactor data * model init from config * add u2 bins * flake8 * can train * fix bugs, add coverage, add scripts * test can run * fix data * speed perturb with sox * add spec aug * fix for train * fix train logitc * fix logger * log valid loss, time dataset process * using np for speed perturb, remove some debug log of grad clip * fix logger * fix build vocab * fix logger name * using module logger as default * fix * fix install * reorder imports * fix board logger * fix logger * kaldi fbank and mfcc * fix cmvn and print prarams * fix add_eos_sos and cmvn * fix cmvn compute * fix logger and cmvn * fix subsampling, label smoothing loss, remove useless * add notebook test * fix log * fix tb logger * multi gpu valid * fix log * fix log * fix config * fix compute cmvn, need paddle 2.1 * add cmvn notebook * fix layer tools * fix compute cmvn * add rtf * fix decoding * fix layer tools * fix log, add avg script * more avg and test info * fix dataset pickle problem; using 2.1 paddle; num_workers can > 0; ckpt save in exp dir;fix setup.sh; * add vimrc * refactor tiny script, add transformer and stream conf * spm demo; librisppech scripts and confs * fix log * add librispeech scripts * refactor data pipe; fix conf; fix u2 default params * fix bugs * refactor aishell scripts * fix test * fix cmvn * fix s0 scripts * fix ds2 scripts and bugs * fix dev & test dataset filter * fix dataset filter * filter dev * fix ckpt path * filter test, since librispeech will cause OOM, but all test wer will be worse, since mismatch train with test * add comment * add syllable doc * fix ds2 configs * add doc * add pypinyin tools * fix decoder using blank_id=0 * mmseg with pybind11 * format code
4 years ago
8 years ago
def normalize(self, target_db=-20, max_gain_db=300.0):
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"""Normalize audio to be of the desired RMS value in decibels.
8 years ago
Note that this is an in-place transformation.
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:param target_db: Target RMS value in decibels. This value should be
less than 0.0 as 0.0 is full-scale audio.
8 years ago
:type target_db: float
:param max_gain_db: Max amount of gain in dB that can be applied for
8 years ago
normalization. This is to prevent nans when
attempting to normalize a signal consisting of
all zeros.
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:type max_gain_db: float
:raises ValueError: If the required gain to normalize the segment to
the target_db value exceeds max_gain_db.
8 years ago
"""
gain = target_db - self.rms_db
if gain > max_gain_db:
raise ValueError(
8 years ago
"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))
8 years ago
def normalize_online_bayesian(self,
target_db,
prior_db,
prior_samples,
startup_delay=0.0):
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"""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.
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Note that this is an in-place transformation.
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:param target_db: Target RMS value in decibels.
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:type target_bd: float
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:param prior_db: Prior RMS estimate in decibels.
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:type prior_db: float
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:param prior_samples: Prior strength in number of samples.
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:type prior_samples: float
:param startup_delay: Default 0.0s. If provided, this function will
8 years ago
accrue statistics for the first startup_delay
seconds before applying online normalization.
8 years ago
:type startup_delay: float
8 years ago
"""
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# Estimate total RMS online.
8 years ago
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
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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)
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# Compute required time-varying gain.
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gain_db = target_db - rms_estimate_db
self.gain_db(gain_db)
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def resample(self, target_sample_rate, filter='kaiser_best'):
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"""Resample the audio to a target sample rate.
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Note that this is an in-place transformation.
8 years ago
:param target_sample_rate: Target sample rate.
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:type target_sample_rate: int
:param filter: The resampling filter to use one of {'kaiser_best',
'kaiser_fast'}.
:type filter: str
8 years ago
"""
self._samples = resampy.resample(
self.samples, self.sample_rate, target_sample_rate, filter=filter)
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self._sample_rate = target_sample_rate
def pad_silence(self, duration, sides='both'):
8 years ago
"""Pad this audio sample with a period of silence.
8 years ago
Note that this is an in-place transformation.
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:param duration: Length of silence in seconds to pad.
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:type duration: float
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:param sides: Position for padding:
'beginning' - adds silence in the beginning;
'end' - adds silence in the end;
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'both' - adds silence in both the beginning and the end.
:type sides: str
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:raises ValueError: If sides is not supported.
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"""
if duration == 0.0:
return self
cls = type(self)
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silence = self.make_silence(duration, self._sample_rate)
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if sides == "beginning":
padded = cls.concatenate(silence, self)
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elif sides == "end":
padded = cls.concatenate(self, silence)
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elif sides == "both":
padded = cls.concatenate(silence, self, silence)
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else:
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raise ValueError("Unknown value for the sides %s" % sides)
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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):
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"""Cut the AudioSegment between given boundaries.
Note that this is an in-place transformation.
8 years ago
8 years ago
:param start_sec: Beginning of subsegment in seconds.
8 years ago
:type start_sec: float
8 years ago
:param end_sec: End of subsegment in seconds.
8 years ago
:type end_sec: float
8 years ago
:raise ValueError: If start_sec or end_sec is incorrectly set, e.g. out
of bounds in time.
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"""
8 years ago
start_sec = 0.0 if start_sec is None else start_sec
end_sec = self.duration if end_sec is None else end_sec
8 years ago
if start_sec < 0.0:
start_sec = self.duration + start_sec
if end_sec < 0.0:
end_sec = self.duration + end_sec
8 years ago
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))
8 years ago
start_sample = int(round(start_sec * self._sample_rate))
end_sample = int(round(end_sec * self._sample_rate))
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self._samples = self._samples[start_sample:end_sample]
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def random_subsegment(self, subsegment_length, rng=None):
8 years ago
"""Cut the specified length of the audiosegment randomly.
Note that this is an in-place transformation.
8 years ago
:param subsegment_length: Subsegment length in seconds.
8 years ago
:type subsegment_length: float
8 years ago
:param rng: Random number generator state.
8 years ago
:type rng: random.Random
8 years ago
:raises ValueError: If the length of subsegment is greater than
the origineal segemnt.
8 years ago
"""
8 years ago
rng = random.Random() if rng is None else rng
8 years ago
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)
8 years ago
self.subsegment(start_time, start_time + subsegment_length)
8 years ago
8 years ago
def convolve(self, impulse_segment, allow_resample=False):
8 years ago
"""Convolve this audio segment with the given impulse segment.
8 years ago
8 years ago
Note that this is an in-place transformation.
8 years ago
8 years ago
:param impulse_segment: Impulse response segments.
:type impulse_segment: AudioSegment
8 years ago
:param allow_resample: Indicates whether resampling is allowed when
the impulse_segment has a different sample
rate from this signal.
:type allow_resample: bool
8 years ago
:raises ValueError: If the sample rate is not match between two
8 years ago
audio segments when resample is not allowed.
8 years ago
"""
if allow_resample and self.sample_rate != impulse_segment.sample_rate:
impulse_segment.resample(self.sample_rate)
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if self.sample_rate != impulse_segment.sample_rate:
raise ValueError("Impulse segment's sample rate (%d Hz) is not "
8 years ago
"equal to base signal sample rate (%d Hz)." %
(impulse_segment.sample_rate, self.sample_rate))
samples = signal.fftconvolve(self.samples, impulse_segment.samples,
"full")
8 years ago
self._samples = samples
8 years ago
def convolve_and_normalize(self, impulse_segment, allow_resample=False):
8 years ago
"""Convolve and normalize the resulting audio segment so that it
has the same average power as the input signal.
8 years ago
Note that this is an in-place transformation.
8 years ago
:param impulse_segment: Impulse response segments.
:type impulse_segment: AudioSegment
8 years ago
:param allow_resample: Indicates whether resampling is allowed when
the impulse_segment has a different sample
rate from this signal.
:type allow_resample: bool
8 years ago
"""
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target_db = self.rms_db
self.convolve(impulse_segment, allow_resample=allow_resample)
self.normalize(target_db)
8 years ago
def add_noise(self,
noise,
snr_dB,
allow_downsampling=False,
max_gain_db=300.0,
rng=None):
8 years ago
"""Add the given noise segment at a specific signal-to-noise ratio.
8 years ago
If the noise segment is longer than this segment, a random subsegment
of matching length is sampled from it and used instead.
8 years ago
Note that this is an in-place transformation.
8 years ago
:param noise: Noise signal to add.
8 years ago
:type noise: AudioSegment
8 years ago
:param snr_dB: Signal-to-Noise Ratio, in decibels.
8 years ago
:type snr_dB: float
8 years ago
: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.
8 years ago
:type max_gain_db: float
8 years ago
:param rng: Random number generator state.
8 years ago
:type rng: None|random.Random
:raises ValueError: If the sample rate does not match between the two
8 years ago
audio segments when downsampling is not allowed, or
if the duration of noise segments is shorter than
8 years ago
original audio segments.
8 years ago
"""
8 years ago
rng = random.Random() if rng is None else rng
8 years ago
if allow_downsampling and noise.sample_rate > self.sample_rate:
noise = noise.resample(self.sample_rate)
if noise.sample_rate != self.sample_rate:
8 years ago
raise ValueError("Noise sample rate (%d Hz) is not equal to base "
"signal sample rate (%d Hz)." % (noise.sample_rate,
self.sample_rate))
8 years ago
if noise.duration < self.duration:
8 years ago
raise ValueError("Noise signal (%f sec) must be at least as long as"
" base signal (%f sec)." %
8 years ago
(noise.duration, self.duration))
8 years ago
noise_gain_db = min(self.rms_db - noise.rms_db - snr_dB, max_gain_db)
8 years ago
noise_new = copy.deepcopy(noise)
noise_new.random_subsegment(self.duration, rng=rng)
noise_new.gain_db(noise_gain_db)
8 years ago
self.superimpose(noise_new)
8 years ago
@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
"""
8 years ago
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.
8 years ago
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)