commit b602123a5aae1a9f0b6b581e611acf44eb525c08
author Danil Chapovalov <danilchap@webrtc.org> Tue Jun 19 13:26:36 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Tue Jun 19 12:46:20 2018 +0000
tree 30d0844e16256401ba98608dfa86f9770e37dc08
parent bbfcc703ad1b5ae141dcd9c6a951241f6fc03967

Replace rtc::Optional with absl::optional in modules/audio_coding

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

This CL generated by running script with parameter 'modules/audio_coding'

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: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}

modules/audio_coding/include/audio_coding_module.h

diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 3c193a4..dfbe459 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -15,9 +15,9 @@
 #include <string>
 #include <vector>
 
+#include "absl/types/optional.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_encoder.h"
-#include "api/optional.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "modules/audio_coding/neteq/include/neteq.h"
@@ -228,7 +228,7 @@
   // Return value:
   //   The send codec, or nothing if we don't have one
   //
-  virtual rtc::Optional<CodecInst> SendCodec() const = 0;
+  virtual absl::optional<CodecInst> SendCodec() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t SendFrequency()
@@ -546,7 +546,7 @@
   virtual int32_t ReceiveCodec(CodecInst* curr_receive_codec) const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // rtc::Optional<SdpAudioFormat> ReceiveFormat()
+  // absl::optional<SdpAudioFormat> ReceiveFormat()
   // Get the format associated with last received payload.
   //
   // Return value:
@@ -554,7 +554,7 @@
   //    received payload.
   //    An empty Optional if no payload has yet been received.
   //
-  virtual rtc::Optional<SdpAudioFormat> ReceiveFormat() const = 0;
+  virtual absl::optional<SdpAudioFormat> ReceiveFormat() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t IncomingPacket()
@@ -631,7 +631,7 @@
   // the latest audio obtained by calling PlayoutData10ms(), or empty if no
   // valid timestamp is available.
   //
-  virtual rtc::Optional<uint32_t> PlayoutTimestamp() = 0;
+  virtual absl::optional<uint32_t> PlayoutTimestamp() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int FilteredCurrentDelayMs()