Revert of Added new mixer to |check_targets| in .gn and fixed include/depend 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/new_audio_conference_mixer_impl.h"
 
 #include <algorithm>
 #include <functional>
 
 #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),
@@ skipping 102 matching lines @@
       output_frequency_(kDefaultFrequency),
       sample_size_(0),
       audio_source_list_(),
       additional_audio_source_list_(),
       num_mixed_audio_sources_(0),
       use_limiter_(true),
       time_stamp_(0) {
   thread_checker_.DetachFromThread();
 }
 
-NewAudioConferenceMixerImpl::~NewAudioConferenceMixerImpl() {}
-
 bool NewAudioConferenceMixerImpl::Init() {
   crit_.reset(CriticalSectionWrapper::CreateCriticalSection());
   if (crit_.get() == NULL)
     return false;
 
   cb_crit_.reset(CriticalSectionWrapper::CreateCriticalSection());
   if (cb_crit_.get() == NULL)
     return false;
 
   Config config;
@@ skipping 429 matching lines @@
 
   if (error != limiter_->kNoError) {
     WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
                  "Error from AudioProcessing: %d", error);
     RTC_NOTREACHED();
     return false;
   }
   return true;
 }
 }  // namespace webrtc