Rename Maybe to Optional

webrtc/modules/audio_coding/main/include/audio_coding_module.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_
 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_
 
 #include <vector>
 
-#include "webrtc/base/maybe.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // forward declarations
@@ skipping 179 matching lines @@
   virtual void RegisterExternalSendCodec(
       AudioEncoder* external_speech_encoder) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t SendCodec()
   // Get parameters for the codec currently registered as send codec.
   //
   // Return value:
   //   The send codec, or nothing if we don't have one
   //
-  virtual rtc::Maybe<CodecInst> SendCodec() const = 0;
+  virtual rtc::Optional<CodecInst> SendCodec() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t SendFrequency()
   // Get the sampling frequency of the current encoder in Hertz.
   //
   // Return value:
   //   positive; sampling frequency [Hz] of the current encoder.
   //   -1 if an error has happened.
   //
   virtual int32_t SendFrequency() const = 0;
@@ skipping 505 matching lines @@
   virtual std::vector<uint16_t> GetNackList(
       int64_t round_trip_time_ms) const = 0;
 
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_