commit b9b146c9feaf6fd610c3d96b6ed532c447f4bb89
author Danil Chapovalov <danilchap@webrtc.org> Fri Jun 15 10:28:07 2018
committer Commit Bot <commit-bot@chromium.org> Fri Jun 15 12:09:49 2018
tree 27ba8995c071edf50c8b8cbc1e2d61c12b31a096
parent e61d72b37c64d970851976a490dc9d65061e1568

Replace rtc::Optional with absl::optional in audio, call and video

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

This CL generated by running script with parameters 'audio call video':
#!/bin/bash
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: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}

call/call.cc

diff --git a/call/call.cc b/call/call.cc
index e0105ab..be94f6a 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -16,7 +16,7 @@
 #include <utility>
 #include <vector>
 
-#include "api/optional.h"
+#include "absl/types/optional.h"
 #include "api/transport/network_control.h"
 #include "audio/audio_receive_stream.h"
 #include "audio/audio_send_stream.h"
@@ -343,10 +343,10 @@
   RateCounter received_audio_bytes_per_second_counter_;
   RateCounter received_video_bytes_per_second_counter_;
   RateCounter received_rtcp_bytes_per_second_counter_;
-  rtc::Optional<int64_t> first_received_rtp_audio_ms_;
-  rtc::Optional<int64_t> last_received_rtp_audio_ms_;
-  rtc::Optional<int64_t> first_received_rtp_video_ms_;
-  rtc::Optional<int64_t> last_received_rtp_video_ms_;
+  absl::optional<int64_t> first_received_rtp_audio_ms_;
+  absl::optional<int64_t> last_received_rtp_audio_ms_;
+  absl::optional<int64_t> first_received_rtp_video_ms_;
+  absl::optional<int64_t> last_received_rtp_video_ms_;
   TimeInterval sent_rtp_audio_timer_ms_;
 
   rtc::CriticalSection last_bandwidth_bps_crit_;
@@ -584,7 +584,7 @@
   event_log_->Log(rtc::MakeUnique<RtcEventAudioSendStreamConfig>(
       CreateRtcLogStreamConfig(config)));
 
-  rtc::Optional<RtpState> suspended_rtp_state;
+  absl::optional<RtpState> suspended_rtp_state;
   {
     const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);
     if (iter != suspended_audio_send_ssrcs_.end()) {