Removing webrtc::AudioFrame::energy_.

webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc

 /*
  *  Copyright (c) 2012 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 21 matching lines @@
                            0.7089f, 0.7215f, 0.7342f, 0.7468f,
                            0.7595f, 0.7722f, 0.7848f, 0.7975f,
                            0.8101f, 0.8228f, 0.8354f, 0.8481f,
                            0.8608f, 0.8734f, 0.8861f, 0.8987f,
                            0.9114f, 0.9241f, 0.9367f, 0.9494f,
                            0.9620f, 0.9747f, 0.9873f, 1.0000f};
 const size_t rampSize = sizeof(rampArray)/sizeof(rampArray[0]);
 }  // namespace
 
 namespace webrtc {
-void CalculateEnergy(AudioFrame& audioFrame)
+uint32_t CalculateEnergy(const AudioFrame& audioFrame)
 {
-    audioFrame.energy_ = 0;
+    uint32_t energy = 0;
     for(size_t position = 0; position < audioFrame.samples_per_channel_;
         position++)
     {
         // TODO(andrew): this can easily overflow.
-        audioFrame.energy_ += audioFrame.data_[position] *
-                              audioFrame.data_[position];
+        energy += audioFrame.data_[position] * audioFrame.data_[position];
     }
+    return energy;
 }
 
 void RampIn(AudioFrame& audioFrame)
 {
     assert(rampSize <= audioFrame.samples_per_channel_);
     for(size_t i = 0; i < rampSize; i++)
     {
         audioFrame.data_[i] = static_cast<int16_t>(rampArray[i] *
                                                    audioFrame.data_[i]);
     }
 }
 
 void RampOut(AudioFrame& audioFrame)
 {
     assert(rampSize <= audioFrame.samples_per_channel_);
     for(size_t i = 0; i < rampSize; i++)
     {
         const size_t rampPos = rampSize - 1 - i;
         audioFrame.data_[i] = static_cast<int16_t>(rampArray[rampPos] *
                                                    audioFrame.data_[i]);
     }
     memset(&audioFrame.data_[rampSize], 0,
            (audioFrame.samples_per_channel_ - rampSize) *
            sizeof(audioFrame.data_[0]));
 }
 }  // namespace webrtc