commit 66cadcc6b924f027afb60a9b5932d4af3dc6a1d5
author Danil Chapovalov <danilchap@webrtc.org> Tue Jun 19 14:47:43 2018
committer Commit Bot <commit-bot@chromium.org> Tue Jun 19 20:55:07 2018
tree 3d0629af5f9456e4546c818165f6a93c55312c42
parent 751a8170440312f24f91fe2dde3bd58ae5c9c376

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

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

This CL generated by running script with parameter 'pc'

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: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12
Reviewed-on: https://webrtc-review.googlesource.com/84181
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23669}

pc/channel_unittest.cc

diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index f854826..29d2bd7 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -863,7 +863,7 @@
       // The transport channel becomes disconnected.
       fake_rtp_dtls_transport1_->ice_transport()->SignalNetworkRouteChanged(
 
-          rtc::Optional<rtc::NetworkRoute>(network_route));
+          absl::optional<rtc::NetworkRoute>(network_route));
     });
     WaitForThreads();
     EXPECT_EQ(1, media_channel1->num_network_route_changes());
@@ -880,7 +880,7 @@
       // The transport channel becomes connected.
       fake_rtp_dtls_transport1_->ice_transport()->SignalNetworkRouteChanged(
 
-          rtc::Optional<rtc::NetworkRoute>(network_route));
+          absl::optional<rtc::NetworkRoute>(network_route));
     });
     WaitForThreads();
     EXPECT_EQ(1, media_channel1->num_network_route_changes());
@@ -1348,7 +1348,7 @@
     return channel1_->SetRemoteContent(&content, SdpType::kOffer, NULL);
   }
 
-  webrtc::RtpParameters BitrateLimitedParameters(rtc::Optional<int> limit) {
+  webrtc::RtpParameters BitrateLimitedParameters(absl::optional<int> limit) {
     webrtc::RtpParameters parameters;
     webrtc::RtpEncodingParameters encoding;
     encoding.max_bitrate_bps = std::move(limit);
@@ -1357,7 +1357,7 @@
   }
 
   void VerifyMaxBitrate(const webrtc::RtpParameters& parameters,
-                        rtc::Optional<int> expected_bitrate) {
+                        absl::optional<int> expected_bitrate) {
     EXPECT_EQ(1UL, parameters.encodings.size());
     EXPECT_EQ(expected_bitrate, parameters.encodings[0].max_bitrate_bps);
   }
@@ -1368,7 +1368,7 @@
                                            SdpType::kOffer, NULL));
     EXPECT_EQ(media_channel1_->max_bps(), -1);
     VerifyMaxBitrate(media_channel1_->GetRtpSendParameters(kSsrc1),
-                     rtc::nullopt);
+                     absl::nullopt);
   }
 
   // Test that when a channel gets new RtpTransport with a call to