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/checks.h"
 #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]
 // Float:    float   [-1.0, 1.0]
 // FloatS16: float   [-32768.0, 32767.0]
 static inline int16_t FloatToS16(float v) {
   if (v > 0)
-    return v >= 1 ? limits_int16::max() :
-                    static_cast<int16_t>(v * limits_int16::max() + 0.5f);
-  return v <= -1 ? limits_int16::min() :
-                   static_cast<int16_t>(-v * limits_int16::min() - 0.5f);
+    return v >= 1 ? limits_int16::max()
+                  : static_cast<int16_t>(v * limits_int16::max() + 0.5f);
+  return v <= -1 ? limits_int16::min()
+                 : static_cast<int16_t>(-v * limits_int16::min() - 0.5f);
 }
 
 static inline float S16ToFloat(int16_t v) {
   static const float kMaxInt16Inverse = 1.f / limits_int16::max();
   static const float kMinInt16Inverse = 1.f / limits_int16::min();
   return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
 }
 
 static inline int16_t FloatS16ToS16(float v) {
   static const float kMaxRound = limits_int16::max() - 0.5f;
   static const float kMinRound = limits_int16::min() + 0.5f;
   if (v > 0)
-    return v >= kMaxRound ? limits_int16::max() :
-                            static_cast<int16_t>(v + 0.5f);
-  return v <= kMinRound ? limits_int16::min() :
-                          static_cast<int16_t>(v - 0.5f);
+    return v >= kMaxRound ? limits_int16::max()
+                          : static_cast<int16_t>(v + 0.5f);
+  return v <= kMinRound ? limits_int16::min() : static_cast<int16_t>(v - 0.5f);
 }
 
 static inline float FloatToFloatS16(float v) {
   return v * (v > 0 ? limits_int16::max() : -limits_int16::min());
 }
 
 static inline float FloatS16ToFloat(float v) {
   static const float kMaxInt16Inverse = 1.f / limits_int16::max();
   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);
 
 // 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) {
+void Deinterleave(const T* interleaved,
+                  int samples_per_channel,
+                  int num_channels,
+                  T* const* deinterleaved) {
   for (int i = 0; i < num_channels; ++i) {
     T* channel = deinterleaved[i];
     int interleaved_idx = i;
     for (int j = 0; j < samples_per_channel; ++j) {
       channel[j] = interleaved[interleaved_idx];
       interleaved_idx += num_channels;
     }
   }
 }
 
 // Interleave audio from the channel buffers pointed to by |deinterleaved| to
 // |interleaved|. There must be sufficient space allocated in |interleaved|
 // (|samples_per_channel| * |num_channels|).
 template <typename T>
-void Interleave(const T* const* deinterleaved, int samples_per_channel,
-                int num_channels, T* interleaved) {
+void Interleave(const T* const* deinterleaved,
+                int samples_per_channel,
+                int num_channels,
+                T* interleaved) {
   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_frames,
+                                  int num_channels,
+                                  T* deinterleaved) {
+  DCHECK_GT(num_channels, 0);
+  DCHECK_GT(num_frames, 0);
+
+  const T* const end = interleaved + num_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_frames,
+                              int num_channels,
+                              T* deinterleaved);
+
+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_