Move optional.h to webrtc/api/

webrtc/modules/audio_processing/aec3/render_signal_analyzer.h

 /*
  *  Copyright (c) 2017 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_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
 
 #include <array>
 #include <memory>
 
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_processing/aec3/aec3_common.h"
 #include "webrtc/modules/audio_processing/aec3/render_buffer.h"
 #include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
 // Provides functionality for analyzing the properties of the render signal.
 class RenderSignalAnalyzer {
  public:
   RenderSignalAnalyzer();
   ~RenderSignalAnalyzer();
 
   // Updates the render signal analysis with the most recent render signal.
@@ skipping 18 matching lines @@
   std::array<size_t, kFftLengthBy2 - 1> narrow_band_counters_;
   rtc::Optional<int> narrow_peak_band_;
   size_t narrow_peak_counter_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RenderSignalAnalyzer);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_