Move optional.h to webrtc/api/

webrtc/api/audio_codecs/opus/audio_encoder_opus_config.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_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
 #define WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
 
 #include <stddef.h>
 
 #include <vector>
 
-#include "webrtc/rtc_base/optional.h"
+#include "webrtc/api/optional.h"
 
 namespace webrtc {
 
 // NOTE: This struct is still under development and may change without notice.
 struct AudioEncoderOpusConfig {
   static constexpr int kDefaultFrameSizeMs = 20;
 
   // Opus API allows a min bitrate of 500bps, but Opus documentation suggests
   // bitrate should be in the range of 6000 to 510000, inclusive.
   static constexpr int kMinBitrateBps = 6000;
@@ skipping 35 matching lines @@
   int uplink_bandwidth_update_interval_ms;
 
   // NOTE: This member isn't necessary, and will soon go away. See
   // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
   int payload_type;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_