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# Copyright (c) Facebook, Inc. and its affiliates.
#
# This source code is licensed under the MIT license found in the
# LICENSE file in the root directory of this source tree.
import collections
import contextlib
import wave
try:
import webrtcvad
except ImportError:
raise ImportError("Please install py-webrtcvad: pip install webrtcvad")
import argparse
import os
import logging
from tqdm import tqdm
AUDIO_SUFFIX = '.wav'
FS_MS = 30
SCALE = 6e-5
THRESHOLD = 0.3
def read_wave(path):
"""Reads a .wav file.
Takes the path, and returns (PCM audio data, sample rate).
"""
with contextlib.closing(wave.open(path, 'rb')) as wf:
num_channels = wf.getnchannels()
assert num_channels == 1
sample_width = wf.getsampwidth()
assert sample_width == 2
sample_rate = wf.getframerate()
assert sample_rate in (8000, 16000, 32000, 48000)
pcm_data = wf.readframes(wf.getnframes())
return pcm_data, sample_rate
def write_wave(path, audio, sample_rate):
"""Writes a .wav file.
Takes path, PCM audio data, and sample rate.
"""
with contextlib.closing(wave.open(path, 'wb')) as wf:
wf.setnchannels(1)
wf.setsampwidth(2)
wf.setframerate(sample_rate)
wf.writeframes(audio)
class Frame(object):
"""Represents a "frame" of audio data."""
def __init__(self, bytes, timestamp, duration):
self.bytes = bytes
self.timestamp = timestamp
self.duration = duration
def frame_generator(frame_duration_ms, audio, sample_rate):
"""Generates audio frames from PCM audio data.
Takes the desired frame duration in milliseconds, the PCM data, and
the sample rate.
Yields Frames of the requested duration.
"""
n = int(sample_rate * (frame_duration_ms / 1000.0) * 2)
offset = 0
timestamp = 0.0
duration = (float(n) / sample_rate) / 2.0
while offset + n < len(audio):
yield Frame(audio[offset:offset + n], timestamp, duration)
timestamp += duration
offset += n
def vad_collector(sample_rate, frame_duration_ms,
padding_duration_ms, vad, frames):
"""Filters out non-voiced audio frames.
Given a webrtcvad.Vad and a source of audio frames, yields only
the voiced audio.
Uses a padded, sliding window algorithm over the audio frames.
When more than 90% of the frames in the window are voiced (as
reported by the VAD), the collector triggers and begins yielding
audio frames. Then the collector waits until 90% of the frames in
the window are unvoiced to detrigger.
The window is padded at the front and back to provide a small
amount of silence or the beginnings/endings of speech around the
voiced frames.
Arguments:
sample_rate - The audio sample rate, in Hz.
frame_duration_ms - The frame duration in milliseconds.
padding_duration_ms - The amount to pad the window, in milliseconds.
vad - An instance of webrtcvad.Vad.
frames - a source of audio frames (sequence or generator).
Returns: A generator that yields PCM audio data.
"""
num_padding_frames = int(padding_duration_ms / frame_duration_ms)
# We use a deque for our sliding window/ring buffer.
ring_buffer = collections.deque(maxlen=num_padding_frames)
# We have two states: TRIGGERED and NOTTRIGGERED. We start in the
# NOTTRIGGERED state.
triggered = False
voiced_frames = []
for frame in frames:
is_speech = vad.is_speech(frame.bytes, sample_rate)
# sys.stdout.write('1' if is_speech else '0')
if not triggered:
ring_buffer.append((frame, is_speech))
num_voiced = len([f for f, speech in ring_buffer if speech])
# If we're NOTTRIGGERED and more than 90% of the frames in
# the ring buffer are voiced frames, then enter the
# TRIGGERED state.
if num_voiced > 0.9 * ring_buffer.maxlen:
triggered = True
# We want to yield all the audio we see from now until
# we are NOTTRIGGERED, but we have to start with the
# audio that's already in the ring buffer.
for f, _ in ring_buffer:
voiced_frames.append(f)
ring_buffer.clear()
else:
# We're in the TRIGGERED state, so collect the audio data
# and add it to the ring buffer.
voiced_frames.append(frame)
ring_buffer.append((frame, is_speech))
num_unvoiced = len([f for f, speech in ring_buffer if not speech])
# If more than 90% of the frames in the ring buffer are
# unvoiced, then enter NOTTRIGGERED and yield whatever
# audio we've collected.
if num_unvoiced > 0.9 * ring_buffer.maxlen:
triggered = False
yield [b''.join([f.bytes for f in voiced_frames]),
voiced_frames[0].timestamp, voiced_frames[-1].timestamp]
ring_buffer.clear()
voiced_frames = []
# If we have any leftover voiced audio when we run out of input,
# yield it.
if voiced_frames:
yield [b''.join([f.bytes for f in voiced_frames]),
voiced_frames[0].timestamp, voiced_frames[-1].timestamp]
def main(args):
# create output folder
try:
cmd = f"mkdir -p {args.out_path}"
os.system(cmd)
except Exception:
logging.error("Can not create output folder")
exit(-1)
# build vad object
vad = webrtcvad.Vad(int(args.agg))
# iterating over wavs in dir
for file in tqdm(os.listdir(args.in_path)):
if file.endswith(AUDIO_SUFFIX):
audio_inpath = os.path.join(args.in_path, file)
audio_outpath = os.path.join(args.out_path, file)
audio, sample_rate = read_wave(audio_inpath)
frames = frame_generator(FS_MS, audio, sample_rate)
frames = list(frames)
segments = vad_collector(sample_rate, FS_MS, 300, vad, frames)
merge_segments = list()
timestamp_start = 0.0
timestamp_end = 0.0
# removing start, end, and long sequences of sils
for i, segment in enumerate(segments):
merge_segments.append(segment[0])
if i and timestamp_start:
sil_duration = segment[1] - timestamp_end
if sil_duration > THRESHOLD:
merge_segments.append(int(THRESHOLD / SCALE)*(b'\x00'))
else:
merge_segments.append(int((sil_duration / SCALE))*(b'\x00'))
timestamp_start = segment[1]
timestamp_end = segment[2]
segment = b''.join(merge_segments)
write_wave(audio_outpath, segment, sample_rate)
if __name__ == '__main__':
parser = argparse.ArgumentParser(description='Apply vad to a file of fils.')
parser.add_argument('in_path', type=str, help='Path to the input files')
parser.add_argument('out_path', type=str,
help='Path to save the processed files')
parser.add_argument('--agg', type=int, default=3,
help='The level of aggressiveness of the VAD: [0-3]')
args = parser.parse_args()
main(args)
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