webrtc/modules/audio_processing/test
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
..
android/apmtest Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
conversational_speech Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
py_quality_assessment Fix pylint presubmit errors and warnings from untouched modules. 2018-06-27 09:31:29 +00:00
aec_dump_based_simulator.cc Added an audioproc option to not report the stream delay 2018-05-28 13:22:29 +00:00
aec_dump_based_simulator.h Restructure the audioproc_f tool into a library with a thin executable wrapper. 2018-03-09 18:06:04 +00:00
apmtest.m Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
audio_buffer_tools.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_buffer_tools.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_processing_simulator.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
audio_processing_simulator.h Flags and settings for AGC2 in AgcManagerDirect. 2018-07-02 13:20:39 +00:00
audioproc_float_impl.cc Flags and settings for AGC2 in AgcManagerDirect. 2018-07-02 13:20:39 +00:00
audioproc_float_impl.h Moved audioproc_f interface into api directory. 2018-03-15 12:31:37 +00:00
audioproc_float_main.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
bitexactness_tools.cc Add DCHECK and documentation to disallow trying to read more than two audio channels in helper function. 2018-01-12 15:27:50 +00:00
bitexactness_tools.h Add DCHECK and documentation to disallow trying to read more than two audio channels in helper function. 2018-01-12 15:27:50 +00:00
debug_dump_replayer.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
debug_dump_replayer.h Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
debug_dump_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
echo_canceller_test_tools.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_canceller_test_tools.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_canceller_test_tools_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
fake_recording_device.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
fake_recording_device.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fake_recording_device_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
performance_timer.cc Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
performance_timer.h Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
protobuf_utils.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
protobuf_utils.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
simulator_buffers.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
simulator_buffers.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
test_utils.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
test_utils.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
unittest.proto Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
wav_based_simulator.cc Added an audioproc option to not report the stream delay 2018-05-28 13:22:29 +00:00
wav_based_simulator.h Restructure the audioproc_f tool into a library with a thin executable wrapper. 2018-03-09 18:06:04 +00:00