commit 0bc58cf8769643d7ad6f24d106754336c7d29385
author Danil Chapovalov <danilchap@webrtc.org> Thu Jun 21 13:32:56 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Thu Jun 21 12:50:03 2018 +0000
tree c735ccc1082636510b643f38cad5bc1ab4d838b2
parent 1ff41eb784ffe919982124fae0c3f294467aafc8

Replace rtc::Optional with absl::optional in api

This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}

api/audio_codecs/opus/audio_encoder_opus.h

diff --git a/api/audio_codecs/opus/audio_encoder_opus.h b/api/audio_codecs/opus/audio_encoder_opus.h
index 6325269..20aaaf7 100644
--- a/api/audio_codecs/opus/audio_encoder_opus.h
+++ b/api/audio_codecs/opus/audio_encoder_opus.h
@@ -14,11 +14,11 @@
 #include <memory>
 #include <vector>
 
+#include "absl/types/optional.h"
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_encoder_opus_config.h"
-#include "api/optional.h"
 
 namespace webrtc {
 
@@ -28,14 +28,14 @@
 // NOTE: This struct is still under development and may change without notice.
 struct AudioEncoderOpus {
   using Config = AudioEncoderOpusConfig;
-  static rtc::Optional<AudioEncoderOpusConfig> SdpToConfig(
+  static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
       const SdpAudioFormat& audio_format);
   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
   static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       const AudioEncoderOpusConfig& config,
       int payload_type,
-      rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
 };
 
 }  // namespace webrtc