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/neteq/neteq_impl.h

diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 17589aa..6f69680 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -14,7 +14,7 @@
 #include <memory>
 #include <string>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "api/audio/audio_frame.h"
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
 #include "modules/audio_coding/neteq/defines.h"
@@ -198,13 +198,13 @@
   // Disables post-decode VAD.
   void DisableVad() override;
 
-  rtc::Optional<uint32_t> GetPlayoutTimestamp() const override;
+  absl::optional<uint32_t> GetPlayoutTimestamp() const override;
 
   int last_output_sample_rate_hz() const override;
 
-  rtc::Optional<CodecInst> GetDecoder(int payload_type) const override;
+  absl::optional<CodecInst> GetDecoder(int payload_type) const override;
 
-  rtc::Optional<SdpAudioFormat> GetDecoderFormat(
+  absl::optional<SdpAudioFormat> GetDecoderFormat(
       int payload_type) const override;
 
   int SetTargetNumberOfChannels() override;
@@ -424,8 +424,8 @@
   bool new_codec_ RTC_GUARDED_BY(crit_sect_);
   uint32_t timestamp_ RTC_GUARDED_BY(crit_sect_);
   bool reset_decoder_ RTC_GUARDED_BY(crit_sect_);
-  rtc::Optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
-  rtc::Optional<uint8_t> current_cng_rtp_payload_type_
+  absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
+  absl::optional<uint8_t> current_cng_rtp_payload_type_
       RTC_GUARDED_BY(crit_sect_);
   uint32_t ssrc_ RTC_GUARDED_BY(crit_sect_);
   bool first_packet_ RTC_GUARDED_BY(crit_sect_);