commit 918f50c5d14040385e912a676e739e9248b03aef
author Karl Wiberg <kwiberg@webrtc.org> Thu Jul 05 11:40:33 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Thu Jul 05 10:59:49 2018 +0000
tree 3e306a31ea350d50c6255a7be29ecfa929e5f4ec
parent 431f14ef69f54b960bbce09f2382e4a95039f767

Use absl::make_unique and absl::WrapUnique directly

Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}

audio/BUILD.gn

diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 709fd7f..22ceeb6 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -86,6 +86,7 @@
     "../system_wrappers:field_trial_api",
     "../system_wrappers:metrics_api",
     "utility:audio_frame_operations",
+    "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -155,6 +156,7 @@
       "../test:test_support",
       "utility:utility_tests",
       "//testing/gtest",
+      "//third_party/abseil-cpp/absl/memory",
     ]
 
     if (!rtc_use_memcheck) {
@@ -258,6 +260,7 @@
       "../test:test_common",
       "../test:test_main",
       "//testing/gtest",
+      "//third_party/abseil-cpp/absl/memory",
     ]
 
     data = [

audio/audio_send_stream_unittest.cc

diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index f84aaaf..2341d4e 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -12,6 +12,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/memory/memory.h"
 #include "api/units/time_delta.h"
 #include "audio/audio_send_stream.h"
 #include "audio/audio_state.h"
@@ -27,7 +28,6 @@
 #include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
 #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "rtc_base/fakeclock.h"
-#include "rtc_base/ptr_util.h"
 #include "rtc_base/task_queue.h"
 #include "test/gtest.h"
 #include "test/mock_audio_encoder.h"

audio/audio_state.cc

diff --git a/audio/audio_state.cc b/audio/audio_state.cc
index bff818e..35ea03d 100644
--- a/audio/audio_state.cc
+++ b/audio/audio_state.cc
@@ -14,12 +14,12 @@
 #include <utility>
 #include <vector>
 
+#include "absl/memory/memory.h"
 #include "audio/audio_receive_stream.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "rtc_base/atomicops.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/ptr_util.h"
 #include "rtc_base/thread.h"
 
 namespace webrtc {
@@ -121,7 +121,8 @@
       }
     } else {
       config_.audio_device_module->StopPlayout();
-      null_audio_poller_ = rtc::MakeUnique<NullAudioPoller>(&audio_transport_);
+      null_audio_poller_ =
+          absl::make_unique<NullAudioPoller>(&audio_transport_);
     }
   }
 }

audio/channel.cc

diff --git a/audio/channel.cc b/audio/channel.cc
index cb32b92..297c111 100644
--- a/audio/channel.cc
+++ b/audio/channel.cc
@@ -17,6 +17,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/memory/memory.h"
 #include "api/array_view.h"
 #include "audio/utility/audio_frame_operations.h"
 #include "call/rtp_transport_controller_send_interface.h"
@@ -38,7 +39,6 @@
 #include "rtc_base/format_macros.h"
 #include "rtc_base/location.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/ptr_util.h"
 #include "rtc_base/rate_limiter.h"
 #include "rtc_base/task_queue.h"
 #include "rtc_base/thread_checker.h"
@@ -393,7 +393,7 @@
 
   unsigned int ssrc;
   RTC_CHECK_EQ(GetRemoteSSRC(ssrc), 0);
-  event_log_proxy_->Log(rtc::MakeUnique<RtcEventAudioPlayout>(ssrc));
+  event_log_proxy_->Log(absl::make_unique<RtcEventAudioPlayout>(ssrc));
   // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
   bool muted;
   if (audio_coding_->PlayoutData10Ms(audio_frame->sample_rate_hz_, audio_frame,

audio/test/audio_bwe_integration_test.cc

diff --git a/audio/test/audio_bwe_integration_test.cc b/audio/test/audio_bwe_integration_test.cc
index ecdd162..4d8ac80 100644
--- a/audio/test/audio_bwe_integration_test.cc
+++ b/audio/test/audio_bwe_integration_test.cc
@@ -10,8 +10,8 @@
 
 #include "audio/test/audio_bwe_integration_test.h"
 
+#include "absl/memory/memory.h"
 #include "common_audio/wav_file.h"
-#include "rtc_base/ptr_util.h"
 #include "system_wrappers/include/sleep.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"