first commit

Author: Fictionarry

.gitignore

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+__pycache__/
+build/
+*.egg-info/
+*.so
+*.mp4
+
+tmp*
+trial*/
+
+data
+data_utils/face_tracking/3DMM/*
+data_utils/face_parsing/79999_iter.pth
+
+pretrained
+*.mp4

LICENSE

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+MIT License
+
+Copyright (c) 2022 hawkey
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.

assets/main.png

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+# Routines for DeepSpeech features processing
+Several routines for [DeepSpeech](https://github.com/mozilla/DeepSpeech) features processing, like speech features generation for [VOCA](https://github.com/TimoBolkart/voca) model.
+
+## Installation
+
+```
+pip3 install -r requirements.txt
+```
+
+## Usage
+
+Generate wav files:
+```
+python3 extract_wav.py --in-video=<you_data_dir>
+```
+
+Generate files with DeepSpeech features:
+```
+python3 extract_ds_features.py --input=<you_data_dir>
+```

data_utils/deepspeech_features/README.md

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+"""
+    DeepSpeech features processing routines.
+    NB: Based on VOCA code. See the corresponding license restrictions.
+"""
+
+__all__ = ['conv_audios_to_deepspeech']
+
+import numpy as np
+import warnings
+import resampy
+from scipy.io import wavfile
+from python_speech_features import mfcc
+import tensorflow.compat.v1 as tf
+tf.disable_v2_behavior()
+
+def conv_audios_to_deepspeech(audios,
+                              out_files,
+                              num_frames_info,
+                              deepspeech_pb_path,
+                              audio_window_size=1,
+                              audio_window_stride=1):
+    """
+    Convert list of audio files into files with DeepSpeech features.
+
+    Parameters
+    ----------
+    audios : list of str or list of None
+        Paths to input audio files.
+    out_files : list of str
+        Paths to output files with DeepSpeech features.
+    num_frames_info : list of int
+        List of numbers of frames.
+    deepspeech_pb_path : str
+        Path to DeepSpeech 0.1.0 frozen model.
+    audio_window_size : int, default 16
+        Audio window size.
+    audio_window_stride : int, default 1
+        Audio window stride.
+    """
+    # deepspeech_pb_path="/disk4/keyu/DeepSpeech/deepspeech-0.9.2-models.pbmm"
+    graph, logits_ph, input_node_ph, input_lengths_ph = prepare_deepspeech_net(
+        deepspeech_pb_path)
+
+    with tf.compat.v1.Session(graph=graph) as sess:
+        for audio_file_path, out_file_path, num_frames in zip(audios, out_files, num_frames_info):
+            print(audio_file_path)
+            print(out_file_path)
+            audio_sample_rate, audio = wavfile.read(audio_file_path)
+            if audio.ndim != 1:
+                warnings.warn(
+                    "Audio has multiple channels, the first channel is used")
+                audio = audio[:, 0]
+            ds_features = pure_conv_audio_to_deepspeech(
+                audio=audio,
+                audio_sample_rate=audio_sample_rate,
+                audio_window_size=audio_window_size,
+                audio_window_stride=audio_window_stride,
+                num_frames=num_frames,
+                net_fn=lambda x: sess.run(
+                    logits_ph,
+                    feed_dict={
+                        input_node_ph: x[np.newaxis, ...],
+                        input_lengths_ph: [x.shape[0]]}))
+
+            net_output = ds_features.reshape(-1, 29)
+            win_size = 16
+            zero_pad = np.zeros((int(win_size / 2), net_output.shape[1]))
+            net_output = np.concatenate(
+                (zero_pad, net_output, zero_pad), axis=0)
+            windows = []
+            for window_index in range(0, net_output.shape[0] - win_size, 2):
+                windows.append(
+                    net_output[window_index:window_index + win_size])
+            print(np.array(windows).shape)
+            np.save(out_file_path, np.array(windows))
+
+
+def prepare_deepspeech_net(deepspeech_pb_path):
+    """
+    Load and prepare DeepSpeech network.
+
+    Parameters
+    ----------
+    deepspeech_pb_path : str
+        Path to DeepSpeech 0.1.0 frozen model.
+
+    Returns
+    -------
+    graph : obj
+        ThensorFlow graph.
+    logits_ph : obj
+        ThensorFlow placeholder for `logits`.
+    input_node_ph : obj
+        ThensorFlow placeholder for `input_node`.
+    input_lengths_ph : obj
+        ThensorFlow placeholder for `input_lengths`.
+    """
+    # Load graph and place_holders:
+    with tf.io.gfile.GFile(deepspeech_pb_path, "rb") as f:
+        graph_def = tf.compat.v1.GraphDef()
+        graph_def.ParseFromString(f.read())
+
+    graph = tf.compat.v1.get_default_graph()
+    tf.import_graph_def(graph_def, name="deepspeech")
+    logits_ph = graph.get_tensor_by_name("deepspeech/logits:0")
+    input_node_ph = graph.get_tensor_by_name("deepspeech/input_node:0")
+    input_lengths_ph = graph.get_tensor_by_name("deepspeech/input_lengths:0")
+
+    return graph, logits_ph, input_node_ph, input_lengths_ph
+
+
+def pure_conv_audio_to_deepspeech(audio,
+                                  audio_sample_rate,
+                                  audio_window_size,
+                                  audio_window_stride,
+                                  num_frames,
+                                  net_fn):
+    """
+    Core routine for converting audion into DeepSpeech features.
+
+    Parameters
+    ----------
+    audio : np.array
+        Audio data.
+    audio_sample_rate : int
+        Audio sample rate.
+    audio_window_size : int
+        Audio window size.
+    audio_window_stride : int
+        Audio window stride.
+    num_frames : int or None
+        Numbers of frames.
+    net_fn : func
+        Function for DeepSpeech model call.
+
+    Returns
+    -------
+    np.array
+        DeepSpeech features.
+    """
+    target_sample_rate = 16000
+    if audio_sample_rate != target_sample_rate:
+        resampled_audio = resampy.resample(
+            x=audio.astype(np.float),
+            sr_orig=audio_sample_rate,
+            sr_new=target_sample_rate)
+    else:
+        resampled_audio = audio.astype(np.float)
+    input_vector = conv_audio_to_deepspeech_input_vector(
+        audio=resampled_audio.astype(np.int16),
+        sample_rate=target_sample_rate,
+        num_cepstrum=26,
+        num_context=9)
+
+    network_output = net_fn(input_vector)
+    # print(network_output.shape)
+
+    deepspeech_fps = 50
+    video_fps = 50  # Change this option if video fps is different
+    audio_len_s = float(audio.shape[0]) / audio_sample_rate
+    if num_frames is None:
+        num_frames = int(round(audio_len_s * video_fps))
+    else:
+        video_fps = num_frames / audio_len_s
+    network_output = interpolate_features(
+        features=network_output[:, 0],
+        input_rate=deepspeech_fps,
+        output_rate=video_fps,
+        output_len=num_frames)
+
+    # Make windows:
+    zero_pad = np.zeros((int(audio_window_size / 2), network_output.shape[1]))
+    network_output = np.concatenate(
+        (zero_pad, network_output, zero_pad), axis=0)
+    windows = []
+    for window_index in range(0, network_output.shape[0] - audio_window_size, audio_window_stride):
+        windows.append(
+            network_output[window_index:window_index + audio_window_size])
+
+    return np.array(windows)
+
+
+def conv_audio_to_deepspeech_input_vector(audio,
+                                          sample_rate,
+                                          num_cepstrum,
+                                          num_context):
+    """
+    Convert audio raw data into DeepSpeech input vector.
+
+    Parameters
+    ----------
+    audio : np.array
+        Audio data.
+    audio_sample_rate : int
+        Audio sample rate.
+    num_cepstrum : int
+        Number of cepstrum.
+    num_context : int
+        Number of context.
+
+    Returns
+    -------
+    np.array
+        DeepSpeech input vector.
+    """
+    # Get mfcc coefficients:
+    features = mfcc(
+        signal=audio,
+        samplerate=sample_rate,
+        numcep=num_cepstrum)
+
+    # We only keep every second feature (BiRNN stride = 2):
+    features = features[::2]
+
+    # One stride per time step in the input:
+    num_strides = len(features)
+
+    # Add empty initial and final contexts:
+    empty_context = np.zeros((num_context, num_cepstrum), dtype=features.dtype)
+    features = np.concatenate((empty_context, features, empty_context))
+
+    # Create a view into the array with overlapping strides of size
+    # numcontext (past) + 1 (present) + numcontext (future):
+    window_size = 2 * num_context + 1
+    train_inputs = np.lib.stride_tricks.as_strided(
+        features,
+        shape=(num_strides, window_size, num_cepstrum),
+        strides=(features.strides[0],
+                 features.strides[0], features.strides[1]),
+        writeable=False)
+
+    # Flatten the second and third dimensions:
+    train_inputs = np.reshape(train_inputs, [num_strides, -1])
+
+    train_inputs = np.copy(train_inputs)
+    train_inputs = (train_inputs - np.mean(train_inputs)) / \
+        np.std(train_inputs)
+
+    return train_inputs
+
+
+def interpolate_features(features,
+                         input_rate,
+                         output_rate,
+                         output_len):
+    """
+    Interpolate DeepSpeech features.
+
+    Parameters
+    ----------
+    features : np.array
+        DeepSpeech features.
+    input_rate : int
+        input rate (FPS).
+    output_rate : int
+        Output rate (FPS).
+    output_len : int
+        Output data length.
+
+    Returns
+    -------
+    np.array
+        Interpolated data.
+    """
+    input_len = features.shape[0]
+    num_features = features.shape[1]
+    input_timestamps = np.arange(input_len) / float(input_rate)
+    output_timestamps = np.arange(output_len) / float(output_rate)
+    output_features = np.zeros((output_len, num_features))
+    for feature_idx in range(num_features):
+        output_features[:, feature_idx] = np.interp(
+            x=output_timestamps,
+            xp=input_timestamps,
+            fp=features[:, feature_idx])
+    return output_features