commit d264df587fa2e1d19ab6175f088a530d4e461ea7
author Danil Chapovalov <danilchap@webrtc.org> Thu Jun 14 12:59:38 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Fri Jun 15 09:53:35 2018 +0000
tree a952196041412e01a2b0f7a2b9ea3f223e0defb7
parent 394b4ebe386300f52490dfd26c4c760dd67cc0d4

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

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

This CL generated using script:
#!/bin/bash
dir=modules/rtp_rtcp
find $dir -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $dir -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: Ife720849709959046329c1c9faa3f31aa13274dc
Reviewed-on: https://webrtc-review.googlesource.com/83584
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23624}

modules/rtp_rtcp/source/rtp_receiver_impl.cc

diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 5c5d68f..fb2cc66 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -29,7 +29,7 @@
 namespace webrtc {
 
 namespace {
-bool InOrderPacket(rtc::Optional<uint16_t> latest_sequence_number,
+bool InOrderPacket(absl::optional<uint16_t> latest_sequence_number,
                    uint16_t current_sequence_number) {
   if (!latest_sequence_number)
     return true;
@@ -174,8 +174,8 @@
 
   auto audio_level =
       rtp_header.extension.hasAudioLevel
-          ? rtc::Optional<uint8_t>(rtp_header.extension.audioLevel)
-          : rtc::nullopt;
+          ? absl::optional<uint8_t>(rtp_header.extension.audioLevel)
+          : absl::nullopt;
   UpdateSources(audio_level);
 
   int32_t ret_val = rtp_media_receiver_->ParseRtpPacket(
@@ -323,7 +323,7 @@
 }
 
 void RtpReceiverImpl::UpdateSources(
-    const rtc::Optional<uint8_t>& ssrc_audio_level) {
+    const absl::optional<uint8_t>& ssrc_audio_level) {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   int64_t now_ms = clock_->TimeInMilliseconds();