Changed folder structure in new mixer and fixed simple lint errors.

webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.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.
  */
 
-#include "webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h"
+#include "webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.h"
 
 #include <algorithm>
 #include <functional>
 
-#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
-#include "webrtc/modules/audio_mixer/include/audio_mixer_defines.h"
+#include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_defines.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/utility.h"
 
 namespace webrtc {
 namespace {
 
 class SourceFrame {
  public:
   SourceFrame(MixerAudioSource* p, AudioFrame* a, bool m, bool was_mixed_before)
       : audio_source_(p),
         audio_frame_(a),
         muted_(m),
         was_mixed_before_(was_mixed_before) {
     if (!muted_) {
-      energy_ = CalculateEnergy(*a);
+      energy_ = NewMixerCalculateEnergy(*a);
     }
   }
 
   // a.shouldMixBefore(b) is used to select mixer participants.
   bool shouldMixBefore(const SourceFrame& other) const {
     if (muted_ != other.muted_) {
       return other.muted_;
     }
 
     auto our_activity = audio_frame_->vad_activity_;
@@ skipping 357 matching lines @@
     if (p.muted_) {
       p.audio_source_->_mixHistory->SetIsMixed(false);
       continue;
     }
 
     // Add frame to result vector for mixing.
     bool is_mixed = false;
     if (maxAudioFrameCounter > 0) {
       --maxAudioFrameCounter;
       if (!p.was_mixed_before_) {
-        RampIn(*p.audio_frame_);
+        NewMixerRampIn(p.audio_frame_);
       }
       result.emplace_back(p.audio_frame_, false);
       is_mixed = true;
     }
 
     // Ramp out unmuted.
     if (p.was_mixed_before_ && !is_mixed) {
-      RampOut(*p.audio_frame_);
+      NewMixerRampOut(p.audio_frame_);
       result.emplace_back(p.audio_frame_, false);
     }
 
     p.audio_source_->_mixHistory->SetIsMixed(is_mixed);
   }
   return result;
 }
 
 void NewAudioConferenceMixerImpl::GetAdditionalAudio(
     AudioFrameList* additionalFramesList) const {
@@ skipping 152 matching lines @@
 
   if (error != limiter_->kNoError) {
     WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
                  "Error from AudioProcessing: %d", error);
     RTC_NOTREACHED();
     return false;
   }
   return true;
 }
 }  // namespace webrtc