From 1b3f3ba108a1e50f797f0bd2433e5860f2696f24 Mon Sep 17 00:00:00 2001
From: Miquel Tubau Pires <miquel.tubau@smadex.com>
Date: Sun, 2 Feb 2020 20:52:59 +0100
Subject: [PATCH] data augmentation files uploaded

---
 scripts/augmentate_data.py | 88 ++++++++++++++++++++++++++++++++++++++
 scripts/wav_functions.py   | 80 ++++++++++++++++++++++++++++++++++
 2 files changed, 168 insertions(+)
 create mode 100755 scripts/augmentate_data.py
 create mode 100755 scripts/wav_functions.py

diff --git a/scripts/augmentate_data.py b/scripts/augmentate_data.py
new file mode 100755
index 00000000..8616e95b
--- /dev/null
+++ b/scripts/augmentate_data.py
@@ -0,0 +1,88 @@
+import os
+import argparse
+import string, random
+from shutil import copyfile
+import scipy.io.wavfile as wavfile
+from wav_functions import readwav
+
+printable = set(string.printable)
+
+
+def augment_data(wav_data, factor, audio_seconds):
+    # get different audios until we reach self.data_augmentation
+    audios = []
+    centers = []
+    for id in range(0,factor):
+        # selecting a center randomly but taking into consideration there needs to be data enough "in the left" and "in the right"
+        samples_to_retain = audio_seconds * 16000
+        valid_centers = range(0,len(wav_data))[samples_to_retain/2:-samples_to_retain/2]
+        center = random.choice(valid_centers)
+
+        # if the center is already taken, let's find another one.
+        while center in centers: center = random.choice(valid_centers)
+
+        centers.append(center)
+        audios.append(wav_data[center-samples_to_retain/2:center+samples_to_retain/2])
+
+    return audios
+
+if __name__=="__main__":
+
+    parser = argparse.ArgumentParser()
+
+    parser.add_argument("--folder_path", required = True, help='path of the folder where to take the data and augment')
+    parser.add_argument("--factor", required = True, help = 'data augmentation factor')
+    parser.add_argument('--output_folder', required = True, help = 'folder where to store the augmented data')
+    parser.add_argument('--audio_seconds', required = True, help='audio of the resulting augmented .wav files')
+
+    args = parser.parse_args()
+
+    youtubers = [youtuber for youtuber in os.listdir(args.folder_path) if not youtuber.startswith('.')]
+
+    id_number = 0
+
+    for youtuber in youtubers:
+
+        working_path = os.path.join(args.folder_path,youtuber)
+
+        # creating if necessary a folder with the youtuber name in the output_folder
+        if not os.path.exists(os.path.join(args.output_folder,youtuber)):
+            os.makedirs(os.path.join(args.output_folder,youtuber))
+
+        audios = [audio for audio in os.listdir(working_path) if audio.endswith('.wav')]
+
+        for audio in audios:
+
+            number = audio.split('_')[-1][:-4]
+
+            # getting the paths of the original audio and face
+            audio_path = os.path.join(working_path,audio)
+            corresponding_face = os.path.join(working_path,'cropped_face_frame_'+number+'.png')
+
+            # reading the audio file and converting it into an array
+            fm, _, wav_data = readwav(audio_path)
+
+            if fm != 16000:
+                raise ValueError('Sampling rate is expected to be 16 KHz!')
+
+            if len(wav_data) < 16000 * int(args.audio_seconds):
+                raise ValueError('The original audio is shorter than the desired output')
+
+            # obtaining the audios as a result of the data augmentation
+            wav_vectors = augment_data(wav_data, int(args.factor), int(args.audio_seconds))
+
+            for wav_vec in wav_vectors:
+
+                new_path = os.path.join(args.output_folder,youtuber)
+
+                # converting wav_vec into a .wav file and storing it
+                wav_file = wavfile.write(filename=os.path.join(new_path,'preprocessed_frame_'+str(id_number)+'.wav'), rate=16000, data=wav_vec)
+                new_face_path = os.path.join(new_path,'cropped_face_frame_'+str(id_number)+'.png')
+
+
+                # copying the faces from the older path to the newer
+
+                copyfile(corresponding_face, new_face_path)
+
+                # updating id_number to avoid overwritting files
+                id_number += 1
diff --git a/scripts/wav_functions.py b/scripts/wav_functions.py
new file mode 100755
index 00000000..99540a92
--- /dev/null
+++ b/scripts/wav_functions.py
@@ -0,0 +1,80 @@
+# wavio.py
+# Author: Warren Weckesser
+# License: BSD 3-Clause (http://opensource.org/licenses/BSD-3-Clause)
+
+import wave
+import numpy as np
+
+
+def _wav2array(nchannels, sampwidth, data):
+    """data must be the string containing the bytes from the wav file."""
+    num_samples, remainder = divmod(len(data), sampwidth * nchannels)
+    if remainder > 0:
+        raise ValueError('The length of data is not a multiple of '
+                         'sampwidth * num_channels.')
+    if sampwidth > 4:
+        raise ValueError("sampwidth must not be greater than 4.")
+
+    if sampwidth == 3:
+        a = np.empty((num_samples, nchannels, 4), dtype=np.uint8)
+        raw_bytes = np.fromstring(data, dtype=np.uint8)
+        a[:, :, :sampwidth] = raw_bytes.reshape(-1, nchannels, sampwidth)
+        a[:, :, sampwidth:] = (a[:, :, sampwidth - 1:sampwidth] >> 7) * 255
+        result = a.view('<i4').reshape(a.shape[:-1])
+    else:
+        # 8 bit samples are stored as unsigned ints; others as signed ints.
+        dt_char = 'u' if sampwidth == 1 else 'i'
+        a = np.fromstring(data, dtype='<%s%d' % (dt_char, sampwidth))
+        result = a.reshape(-1, nchannels)
+    return result
+
+
+def readwav(file):
+    """
+    Read a wav file.
+    Returns the frame rate, sample width (in bytes) and a numpy array
+    containing the data.
+    This function does not read compressed wav files.
+    """
+    wav = wave.open(file)
+    rate = wav.getframerate()
+    nchannels = wav.getnchannels()
+    sampwidth = wav.getsampwidth()
+    nframes = wav.getnframes()
+    data = wav.readframes(nframes)
+    wav.close()
+    array = _wav2array(nchannels, sampwidth, data)
+    return rate, sampwidth, array
+
+
+def writewav24(filename, rate, data):
+    """Create a 24 bit wav file.
+    data must be "array-like", either 1- or 2-dimensional.  If it is 2-d,
+    the rows are the frames (i.e. samples) and the columns are the channels.
+    The data is assumed to be signed, and the values are assumed to be
+    within the range of a 24 bit integer.  Floating point values are
+    converted to integers.  The data is not rescaled or normalized before
+    writing it to the file.
+    Example: Create a 3 second 440 Hz sine wave.
+    >>> rate = 22050  # samples per second
+    >>> T = 3         # sample duration (seconds)
+    >>> f = 440.0     # sound frequency (Hz)
+    >>> t = np.linspace(0, T, T*rate, endpoint=False)
+    >>> x = (2**23 - 1) * np.sin(2 * np.pi * f * t)
+    >>> writewav24("sine24.wav", rate, x)
+    """
+    a32 = np.asarray(data, dtype=np.int32)
+    if a32.ndim == 1:
+        # Convert to a 2D array with a single column.
+        a32.shape = a32.shape + (1,)
+    # By shifting first 0 bits, then 8, then 16, the resulting output
+    # is 24 bit little-endian.
+    a8 = (a32.reshape(a32.shape + (1,)) >> np.array([0, 8, 16])) & 255
+    wavdata = a8.astype(np.uint8).tostring()
+
+    w = wave.open(filename, 'wb')
+    w.setnchannels(a32.shape[1])
+    w.setsampwidth(3)
+    w.setframerate(rate)
+    w.writeframes(wavdata)
+    w.close()
\ No newline at end of file