Revert of Added new mixer to |check_targets| in .gn and fixed include/depend errors.

webrtc/modules/audio_mixer/new_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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_MIXER_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_MIXER_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
 
 #include <list>
 #include <map>
 #include <memory>
 #include <vector>
 
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_mixer/new_audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
 #include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 class AudioProcessing;
 class CriticalSectionWrapper;
 
 struct FrameAndMuteInfo {
   FrameAndMuteInfo(AudioFrame* f, bool m) : frame(f), muted(m) {}
   AudioFrame* frame;
   bool muted;
@@ skipping 24 matching lines @@
   bool is_mixed_;
 };
 
 class NewAudioConferenceMixerImpl : public NewAudioConferenceMixer {
  public:
   // AudioProcessing only accepts 10 ms frames.
   enum { kProcessPeriodicityInMs = 10 };
 
   explicit NewAudioConferenceMixerImpl(int id);
 
-  ~NewAudioConferenceMixerImpl() override;
-
   // Must be called after ctor.
   bool Init();
 
   // NewAudioConferenceMixer functions
   int32_t SetMixabilityStatus(MixerAudioSource* audio_source,
                               bool mixable) override;
   bool MixabilityStatus(const MixerAudioSource& audio_source) const override;
   int32_t SetAnonymousMixabilityStatus(MixerAudioSource* audio_source,
                                        bool mixable) override;
   void Mix(int sample_rate,
@@ skipping 71 matching lines @@
 
   // Ensures that Mix is called from the same thread.
   rtc::ThreadChecker thread_checker_;
 
   // Used for inhibiting saturation in mixing.
   std::unique_ptr<AudioProcessing> limiter_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_