Allow more than 2 input channels in AudioProcessing.

webrtc/modules/audio_processing/audio_buffer.h

Index: webrtc/modules/audio_processing/audio_buffer.h
diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h
index 4291fb3eb99832a416d691cecab9c8b8ff9a041e..0ecda828791defa26f586f40fc8c1cf262af0218 100644
--- a/webrtc/modules/audio_processing/audio_buffer.h
+++ b/webrtc/modules/audio_processing/audio_buffer.h
@@ -30,6 +30,72 @@ enum Band {
   kBand16To24kHz = 2
 };
 
+template <typename T, typename Intermediate>
+void DownmixStereoToMono(int num_frames,
+                         T* out,
+                         const T* left,
+                         const T* right) {
+  for (int i = 0; i < num_frames; ++i) {
+    out[i] = (static_cast<Intermediate>(left[i]) + right[i]) / 2;
+  }
+}
+
+template <typename T, typename Intermediate>
+void DownmixToMono(int num_frames,
+                   T* out,
+                   const T* const* input_channels,
+                   int num_channels) {
+  if (num_channels == 2) {
+    DownmixStereoToMono<T, Intermediate>(num_frames, out, input_channels[0],
+                                         input_channels[1]);
+  } else {
+    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,
+                                  T* deinterleaved,
+                                  int num_multichannel_frames,
+                                  int num_channels) {
+  assert(num_channels > 0);
+  assert(num_multichannel_frames > 0);
+
+  const T* const end = interleaved + num_multichannel_frames * num_channels;
+
+  if (num_channels == 1) {
+    std::memmove(
+        deinterleaved, interleaved,
+        num_channels * num_multichannel_frames * sizeof(*deinterleaved));
+  } else if (num_channels == 2) {
+    // Explicitly unroll for the common stereo case.
+    while (interleaved < end) {
+      *deinterleaved++ =
+          (static_cast<Intermediate>(*interleaved) + *(interleaved + 1)) / 2;
+      interleaved += 2;
+    }
+  } else {
+    while (interleaved < end) {
+      const T* const frame_end = interleaved + num_channels;
+
+      Intermediate value = *interleaved++;
+      while (interleaved < frame_end) {
+        value += *interleaved++;
+      }
+
+      *deinterleaved++ = value / num_channels;
+    }
+  }
+}
+
 class AudioBuffer {
  public:
   // TODO(ajm): Switch to take ChannelLayouts.
@@ -112,12 +178,8 @@ class AudioBuffer {
   void InterleaveTo(AudioFrame* frame, bool data_changed) const;
 
   // Use for float deinterleaved data.
-  void CopyFrom(const float* const* data,
-                int num_frames,
-                AudioProcessing::ChannelLayout layout);
-  void CopyTo(int num_frames,
-              AudioProcessing::ChannelLayout layout,
-              float* const* data);
+  void CopyFrom(const float* const* data, const StreamConfig& stream_config);
+  void CopyTo(const StreamConfig& stream_config, float* const* data);
   void CopyLowPassToReference();
 
   // Splits the signal into different bands.