Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.
This CL has been created with the following steps:
git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt
diff --new-line-format="" --unchanged-line-format="" \
/tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
/tmp/only_make_unique.txt /tmp/memory.txt | \
xargs grep -l "absl/memory" > /tmp/add-memory.txt
git grep -l "\babsl::make_unique\b" | \
xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"
git checkout PRESUBMIT.py abseil-in-webrtc.md
cat /tmp/add-memory.txt | \
xargs sed -i \
's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>
cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
xargs sed -i '/#include "absl\/memory\/memory.h"/d'
git ls-files | grep BUILD.gn | \
xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'
python tools_webrtc/gn_check_autofix.py \
-m tryserver.webrtc -b linux_rel
# Repead the gn_check_autofix step for other platforms
git ls-files | grep BUILD.gn | \
xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format
Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index 02d7452..9026cfc 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -109,7 +109,7 @@
// test RtpSenders/RtpReceivers.
media_engine_(new cricket::FakeMediaEngine()),
channel_manager_(absl::WrapUnique(media_engine_),
- absl::make_unique<cricket::RtpDataEngine>(),
+ std::make_unique<cricket::RtpDataEngine>(),
worker_thread_,
network_thread_),
fake_call_(),
@@ -117,7 +117,7 @@
// Create channels to be used by the RtpSenders and RtpReceivers.
channel_manager_.Init();
bool srtp_required = true;
- rtp_dtls_transport_ = absl::make_unique<cricket::FakeDtlsTransport>(
+ rtp_dtls_transport_ = std::make_unique<cricket::FakeDtlsTransport>(
"fake_dtls_transport", cricket::ICE_CANDIDATE_COMPONENT_RTP);
rtp_transport_ = CreateDtlsSrtpTransport();
@@ -163,7 +163,7 @@
}
std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
- auto dtls_srtp_transport = absl::make_unique<webrtc::DtlsSrtpTransport>(
+ auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
/*rtcp_mux_required=*/true);
dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(),
/*rtcp_dtls_transport=*/nullptr);
@@ -196,7 +196,7 @@
audio_track_ = AudioTrack::Create(kAudioTrackId, source);
EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
- absl::make_unique<MockSetStreamsObserver>();
+ std::make_unique<MockSetStreamsObserver>();
audio_rtp_sender_ =
AudioRtpSender::Create(worker_thread_, audio_track_->id(), nullptr,
set_streams_observer.get());
@@ -261,7 +261,7 @@
void CreateVideoRtpSender(bool is_screencast, uint32_t ssrc = kVideoSsrc) {
AddVideoTrack(is_screencast);
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
- absl::make_unique<MockSetStreamsObserver>();
+ std::make_unique<MockSetStreamsObserver>();
video_rtp_sender_ = VideoRtpSender::Create(
worker_thread_, video_track_->id(), set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
@@ -855,7 +855,7 @@
EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
- absl::make_unique<MockSetStreamsObserver>();
+ std::make_unique<MockSetStreamsObserver>();
audio_rtp_sender_ = AudioRtpSender::Create(
worker_thread_, audio_track_->id(), nullptr, set_streams_observer.get());
ASSERT_TRUE(audio_rtp_sender_->SetTrack(audio_track_));
@@ -1086,7 +1086,7 @@
AddVideoTrack(false);
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
- absl::make_unique<MockSetStreamsObserver>();
+ std::make_unique<MockSetStreamsObserver>();
video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
@@ -1127,7 +1127,7 @@
AddVideoTrack(false);
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
- absl::make_unique<MockSetStreamsObserver>();
+ std::make_unique<MockSetStreamsObserver>();
video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
@@ -1555,7 +1555,7 @@
PropagatesVideoTrackContentHintSetBeforeEnabling) {
AddVideoTrack();
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
- absl::make_unique<MockSetStreamsObserver>();
+ std::make_unique<MockSetStreamsObserver>();
// Setting detailed overrides the default non-screencast mode. This should be
// applied even if the track is set on construction.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
