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webrtc/modules/audio_processing/test
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Mirko Bonadei 317a1f09ed 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}
2019-09-17 15:47:29 +00:00
..
android/apmtest Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
conversational_speech Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
py_quality_assessment Fix wrong-import-order pylint errors in quality_assessment.signal_processing module. 2019-08-28 14:48:28 +00:00
aec_dump_based_simulator.cc Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
aec_dump_based_simulator.h audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
api_call_statistics.cc Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
api_call_statistics.h Added more refined benchmarking code for audioproc_f 2019-04-04 08:37:16 +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 Format almost everything. 2019-07-08 13:45:15 +00:00
audio_processing_simulator.cc Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
audio_processing_simulator.h Adding more refined control over choice of band-splitting 2019-09-14 23:14:17 +00:00
audioproc_float_impl.cc Adding more refined control over choice of band-splitting 2019-09-14 23:14:17 +00:00
audioproc_float_impl.h audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
audioproc_float_main.cc Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
bitexactness_tools.cc Format almost everything. 2019-07-08 13:45:15 +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 Add noise suppression settings to AudioProcessing::Config 2019-01-14 16:17:19 +00:00
debug_dump_replayer.h Store RuntimeSetting in Aec Dumps. 2018-09-10 11:40:28 +00:00
debug_dump_test.cc Fully qualify googletest symbols. 2019-04-09 17:18:20 +00:00
echo_canceller_test_tools.cc AEC3: Make RenderSignalAnalyzer multi-channel 2019-09-13 06:07:09 +00:00
echo_canceller_test_tools.h AEC3: Make RenderSignalAnalyzer multi-channel 2019-09-13 06:07:09 +00:00
echo_canceller_test_tools_unittest.cc AEC3: Make RenderSignalAnalyzer multi-channel 2019-09-13 06:07:09 +00:00
echo_control_mock.h APM unit test: echo path gain change events notified. 2019-01-10 11:06:24 +00:00
fake_recording_device.cc Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +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 std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +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 Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
protobuf_utils.h audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
runtime_setting_util.cc Add PlayoutVolumeChange RuntimeSetting. 2019-05-10 14:12:23 +00:00
runtime_setting_util.h Store RuntimeSetting in Aec Dumps. 2018-09-10 11:40:28 +00:00
simulator_buffers.cc Reland "Simplification and refactoring of the AudioBuffer code" 2019-08-22 10:34:05 +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 audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
test_utils.h audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
unittest.proto Base ApmTest.Process on AudioProcessingStats.output_rms_dbfs 2018-12-18 16:45:03 +00:00
wav_based_simulator.cc Format almost everything. 2019-07-08 13:45:15 +00:00
wav_based_simulator.h Format almost everything. 2019-07-08 13:45:15 +00:00