# This code is adpated from: https://github.com/wiseman/py-webrtcvad/blob/master/example.py import os import tqdm import glob import argparse import pathlib import collections import contextlib import sys import wave import numpy as np import webrtcvad from tqdm.contrib.concurrent import process_map import multiprocessing from itertools import chain 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 # sys.stdout.write('+(%s)' % (ring_buffer[0][0].timestamp,)) # 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, s 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: #sys.stdout.write('-(%s)' % (frame.timestamp + frame.duration)) triggered = False yield b''.join([f.bytes for f in voiced_frames]) 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]) def remove_silence(filepath): filename = os.path.basename(filepath) output_path = filepath.replace(os.path.join(args.input_dir, ''),os.path.join(args.output_dir, '')) # ignore if the file exists if os.path.exists(output_path) and not args.force: return False # create all directory structure pathlib.Path(output_path).parent.mkdir(parents=True, exist_ok=True) padding_duration_ms = 300 # default 300 audio, sample_rate = read_wave(filepath) vad = webrtcvad.Vad(int(args.aggressiveness)) frames = frame_generator(30, audio, sample_rate) frames = list(frames) segments = vad_collector(sample_rate, 30, padding_duration_ms, vad, frames) flag = False segments = list(segments) num_segments = len(segments) if num_segments != 0: for i, segment in reversed(list(enumerate(segments))): if i >= 1: if flag == False: concat_segment = segment flag = True else: concat_segment = segment + concat_segment else: if flag: segment = segment + concat_segment write_wave(output_path, segment, sample_rate) print(output_path) return True else: print("> Just Copying the file to:", output_path) # if fail to remove silence just write the file write_wave(output_path, audio, sample_rate) def preprocess_audios(): files = sorted(glob.glob(os.path.join(args.input_dir, args.glob), recursive=True)) print("> Number of files: ", len(files)) if not args.force: print("> Ignoring files that already exist in the output directory.") if files: # create threads num_threads = multiprocessing.cpu_count() process_map(remove_silence, files, max_workers=num_threads, chunksize=15) else: print("> No files Found !") if __name__ == "__main__": """ usage python remove_silence.py -i=VCTK-Corpus-bk/ -o=../VCTK-Corpus-removed-silence -g=wav48/*/*.wav -a=2 """ parser = argparse.ArgumentParser() parser.add_argument('-i', '--input_dir', type=str, default='../VCTK-Corpus', help='Dataset root dir') parser.add_argument('-o', '--output_dir', type=str, default='../VCTK-Corpus-removed-silence', help='Output Dataset dir') parser.add_argument('-f', '--force', type=bool, default=True, help='Force the replace of exists files') parser.add_argument('-g', '--glob', type=str, default='**/*.wav', help='path in glob format for acess wavs from input_dir. ex: wav48/*/*.wav') parser.add_argument('-a', '--aggressiveness', type=int, default=2, help='set its aggressiveness mode, which is an integer between 0 and 3. 0 is the least aggressive about filtering out non-speech, 3 is the most aggressive.') args = parser.parse_args() preprocess_audios()