Move optional.h to webrtc/api/

webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
 
 #include <memory>
 
 #include "webrtc/api/array_view.h"
 #include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 namespace test {
 
 // Provides an AudioDecoder implementation that delivers audio data from a file.
 // The "encoded" input should contain information about what RTP timestamp the
 // encoding represents, and how many samples the decoder should produce for that
 // encoding. A helper method PrepareEncoded is provided to prepare such
 // encodings. If packets are missing, as determined from the timestamps, the
 // file reading will skip forward to match the loss.
@@ skipping 34 matching lines @@
   rtc::Optional<uint32_t> next_timestamp_from_input_;
   const int sample_rate_hz_;
   const bool stereo_;
   size_t last_decoded_length_ = 0;
   bool cng_mode_ = false;
 };
 
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_