Rename Maybe to Optional

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

 /*
  *  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.
  */
 
@@ skipping 190 matching lines @@
   return codec_manager_.RegisterEncoder(send_codec);
 }
 
 void AudioCodingModuleImpl::RegisterExternalSendCodec(
     AudioEncoder* external_speech_encoder) {
   CriticalSectionScoped lock(acm_crit_sect_.get());
   codec_manager_.RegisterEncoder(external_speech_encoder);
 }
 
 // Get current send codec.
-rtc::Maybe<CodecInst> AudioCodingModuleImpl::SendCodec() const {
+rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
   CriticalSectionScoped lock(acm_crit_sect_.get());
   return codec_manager_.GetCodecInst();
 }
 
 // Get current send frequency.
 int AudioCodingModuleImpl::SendFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "SendFrequency()");
   CriticalSectionScoped lock(acm_crit_sect_.get());
 
@@ skipping 561 matching lines @@
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
 }  // namespace webrtc