Reland "Removing webrtc::AudioFrame::energy_."

webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h

 /*
  *  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 98 matching lines @@
     void GetAdditionalAudio(AudioFrameList* additionalFramesList) const;
 
     // Update the MixHistory of all MixerParticipants. mixedParticipantsList
     // should contain a map of MixerParticipants that have been mixed.
     void UpdateMixedStatus(
         const std::map<int, MixerParticipant*>& mixedParticipantsList) const;
 
     // Clears audioFrameList and reclaims all memory associated with it.
     void ClearAudioFrameList(AudioFrameList* audioFrameList) const;
 
-    // Update the list of MixerParticipants who have a positive VAD. mixList
-    // should be a list of AudioFrames
-    void UpdateVADPositiveParticipants(AudioFrameList* mixList) const;
-
     // This function returns true if it finds the MixerParticipant in the
     // specified list of MixerParticipants.
     bool IsParticipantInList(const MixerParticipant& participant,
                              const MixerParticipantList& participantList) const;
 
     // Add/remove the MixerParticipant to the specified
     // MixerParticipant list.
     bool AddParticipantToList(
         MixerParticipant* participant,
         MixerParticipantList* participantList) const;
@@ skipping 48 matching lines @@
     // Counter keeping track of concurrent calls to process.
     // Note: should never be higher than 1 or lower than 0.
     int16_t _processCalls;
 
     // Used for inhibiting saturation in mixing.
     rtc::scoped_ptr<AudioProcessing> _limiter;
 };
 }  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_