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

webrtc/modules/audio_mixer/test/audio_mixer_unittest.cc

 /*
  *  Copyright (c) 2016 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 <memory>
 #include <utility>
 
 #include "testing/gmock/include/gmock/gmock.h"
 
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
 #include "webrtc/modules/audio_mixer/audio_mixer.h"
-#include "webrtc/modules/audio_mixer/include/audio_mixer_defines.h"
-#include "webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h"
-#include "webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_defines.h"
+#include "webrtc/modules/audio_mixer/new_audio_conference_mixer.h"
+#include "webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.h"
 
 using testing::_;
 using testing::Exactly;
 using testing::Invoke;
 using testing::Return;
 
 using webrtc::voe::AudioMixer;
 
 namespace webrtc {
 class MockAudioMixerParticipant : public MixerParticipant {
@@ skipping 415 matching lines @@
   AddParticipant(&second_frame, MixerParticipant::AudioFrameInfo::kMuted);
   AddParticipant(&third_frame, MixerParticipant::AudioFrameInfo::kMuted);
 
   for (int i = 0; i < 3; ++i) {
     MixAndCompare();
   }
 }
 
 TEST_F(CompareWithOldMixerTest, ManyParticipantsDifferentFrames) {
   Reset();
-  constexpr int num_participants = 20;
-  AudioFrame audio_frames[num_participants];
+  constexpr int kNumParticipants = 20;
+  AudioFrame audio_frames[kNumParticipants];
 
-  for (int i = 0; i < num_participants; ++i) {
+  for (int i = 0; i < kNumParticipants; ++i) {
     ResetFrame(&audio_frames[i]);
     audio_frames[i].id_ = 1;
     audio_frames[i].data_[10] = 100 * (i % 5);
     audio_frames[i].data_[100] = 100 * (i % 5);
     if (i % 2 == 0) {
       audio_frames[i].vad_activity_ = AudioFrame::kVadPassive;
     }
   }
 
-  for (int i = 0; i < num_participants; ++i) {
+  for (int i = 0; i < kNumParticipants; ++i) {
     if (i % 2 == 0) {
       AddParticipant(&audio_frames[i],
                      MixerParticipant::AudioFrameInfo::kMuted);
     } else {
       AddParticipant(&audio_frames[i],
                      MixerParticipant::AudioFrameInfo::kNormal);
     }
     MixAndCompare();
   }
 }
 
 }  // namespace webrtc