webrtc/modules/audio_device/android
Karl Wiberg 918f50c5d1 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}
2018-07-05 10:59:49 +00:00
..
java/src/org/webrtc/voiceengine Temporary suppress bytebuffer warnings. 2018-04-20 11:45:28 +00:00
aaudio_player.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
aaudio_player.h Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +00:00
aaudio_recorder.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
aaudio_recorder.h Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +00:00
aaudio_wrapper.cc Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +00:00
aaudio_wrapper.h Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +00:00
audio_common.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_device_template.h Removes usage of AGC APIs in the ADM. 2017-12-13 16:32:21 +00:00
audio_device_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_manager.cc Force alignment of JVM called functions. 2018-03-23 10:20:55 +00:00
audio_manager.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_manager_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_record_jni.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_record_jni.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_track_jni.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_track_jni.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
build_info.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
build_info.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ensure_initialized.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ensure_initialized.h Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
opensles_common.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
opensles_common.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opensles_player.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
opensles_player.h FineAudioBuffer now uses 16-bit audio samples to match the AudioDeviceBuffer. 2018-04-19 12:20:28 +00:00
opensles_recorder.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
opensles_recorder.h FineAudioBuffer now uses 16-bit audio samples to match the AudioDeviceBuffer. 2018-04-19 12:20:28 +00:00