Move optional.h to webrtc/api/

webrtc/test/fuzzers/audio_decoder_fuzzer.cc

 /*
  *  Copyright (c) 2015 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/test/fuzzers/audio_decoder_fuzzer.h"
 
 #include <limits>
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/api/optional.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 namespace {
 template <typename T, unsigned int B = sizeof(T)>
 bool ParseInt(const uint8_t** data, size_t* remaining_size, T* value) {
   static_assert(std::numeric_limits<T>::is_integer, "Type must be an integer.");
   static_assert(sizeof(T) <= sizeof(uint64_t),
                 "Cannot read wider than uint64_t.");
   static_assert(B <= sizeof(T), "T must be at least B bytes wide.");
   if (B > *remaining_size)
@@ skipping 61 matching lines @@
       break;
     if (remaining_size < packet_len)
       break;
     decoder->IncomingPacket(data_ptr, packet_len, rtp_sequence_number,
                             rtp_timestamp, arrival_timestamp);
     data_ptr += packet_len;
     remaining_size -= packet_len;
   }
 }
 }  // namespace webrtc