Integrate Intelligibility with APM

webrtc/common_audio/include/audio_util.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 84 matching lines @@
   for (int i = 0; i < num_channels; ++i) {
     const T* channel = deinterleaved[i];
     int interleaved_idx = i;
     for (int j = 0; j < samples_per_channel; ++j) {
       interleaved[interleaved_idx] = channel[j];
       interleaved_idx += num_channels;
     }
   }
 }
 
+// Copies audio from a single channel buffer pointed to by |mono| to each
+// channel of |interleaved|. There must be sufficient space allocated in
+// |interleaved| (|samples_per_channel| * |num_channels|).
+template <typename T>
+void UpmixMonoToInterleaved(const T* mono,
+                            int num_frames,
+                            int num_channels,
+                            T* interleaved) {
+  int interleaved_idx = 0;
+  for (int i = 0; i < num_frames; ++i) {
+    for (int j = 0; j < num_channels; ++j) {
+      interleaved[interleaved_idx++] = mono[i];
+    }
+  }
+}
+
 template <typename T, typename Intermediate>
 void DownmixToMono(const T* const* input_channels,
                    int num_frames,
                    int num_channels,
                    T* out) {
   for (int i = 0; i < num_frames; ++i) {
     Intermediate value = input_channels[0][i];
     for (int j = 1; j < num_channels; ++j) {
       value += input_channels[j][i];
     }
@@ skipping 33 matching lines @@
 
 template <>
 void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
                                        int num_frames,
                                        int num_channels,
                                        int16_t* deinterleaved);
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_