Allow more than 2 input channels in AudioProcessing.

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.
  */
 
 #ifndef WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
 #define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
 
 #include <limits>
+#include <cstring>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 typedef std::numeric_limits<int16_t> limits_int16;
 
 // The conversion functions use the following naming convention:
 // S16:      int16_t [-32768, 32767]
@@ skipping 65 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;
     }
   }
 }
 
+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];
+    }
+    out[i] = value / num_channels;
+  }
+}
+
+// Downmixes an interleaved multichannel signal to a single channel by averaging
+// all channels.
+template <typename T, typename Intermediate>
+void DownmixInterleavedToMonoImpl(const T* interleaved,
+                                  int num_multichannel_frames,

[Andrew MacDonald, 2015/07/22 22:47:21]

Why "multichannel"? To differentiate from the "frame == block of audio"
interpretation? That could still be misconstrued.

I think we should go use num_frames or stick to the old samples_per_channel.

[mgraczyk, 2015/07/23 00:16:54]

Done.

+                                  int num_channels,
+                                  T* deinterleaved) {
+  assert(num_channels > 0);

[Andrew MacDonald, 2015/07/22 22:47:21]

For new code we should use DCHECK.

[mgraczyk, 2015/07/23 00:16:54]

Done.

+  assert(num_multichannel_frames > 0);
+
+  const T* const end = interleaved + num_multichannel_frames * num_channels;
+
+  while (interleaved < end) {
+    const T* const frame_end = interleaved + num_channels;
+
+    Intermediate value = *interleaved++;
+    while (interleaved < frame_end) {
+      value += *interleaved++;
+    }
+
+    *deinterleaved++ = value / num_channels;
+  }
+}
+
+template <typename T>
+void DownmixInterleavedToMono(const T* interleaved,
+                              int num_multichannel_frames,
+                              int num_channels,
+                              T* deinterleaved);
+
+template <>
+void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
+                                       int num_multichannel_frames,
+                                       int num_channels,
+                                       int16_t* deinterleaved);
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_