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 47 matching lines @@
   static const float kMinInt16Inverse = 1.f / limits_int16::min();
   return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
 }
 
 void FloatToS16(const float* src, size_t size, int16_t* dest);
 void S16ToFloat(const int16_t* src, size_t size, float* dest);
 void FloatS16ToS16(const float* src, size_t size, int16_t* dest);
 void FloatToFloatS16(const float* src, size_t size, float* dest);
 void FloatS16ToFloat(const float* src, size_t size, float* dest);
 
+// Copy audio from |src| channels to |dest| channels. |src| and |dest| must
+// have the same number of channels, and there must be sufficient space
+// allocated in the |dest| channels.
+template <typename T>
+void CopyAudio(const T* const* src,

[Andrew MacDonald, 2015/08/05 06:18:18]

Perhaps call this CopyAudioIfNeeded() and check if the ptrs point to different
memory.

[ekm, 2015/08/07 06:05:59]

Nice. Done.

+               int samples_per_channel,

[Andrew MacDonald, 2015/08/05 06:18:17]

num_frames is the new way.

[ekm, 2015/08/07 06:05:59]

Done.

+               int num_channels,
+               T* const* dest) {
+  for (int i = 0; i < num_channels; ++i) {
+    const T* src_channel = src[i];
+    T* dest_channel = dest[i];
+    for (int j = 0; j < samples_per_channel; ++j) {
+      dest_channel[j] = src_channel[j];

[Andrew MacDonald, 2015/08/05 06:18:18]

Do a memcpy (or std::copy) instead of individual assignment, so

if (src[i] != dest[i]) {
  std::copy(src[i], src[i] + num_frames, dest[i]);
}

[ekm, 2015/08/07 06:05:59]

Done.

+    }
+  }
+}
+
 // Deinterleave audio from |interleaved| to the channel buffers pointed to
 // by |deinterleaved|. There must be sufficient space allocated in the
 // |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
 // per buffer).
 template <typename T>
 void Deinterleave(const T* interleaved,
                   int samples_per_channel,
                   int num_channels,
                   T* const* deinterleaved) {
   for (int i = 0; i < num_channels; ++i) {
@@ skipping 17 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_