webrtc/modules/audio_processing/aec_dump
Yves Gerey 665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
..
aec_dump_factory.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
aec_dump_impl.cc Merge :audio_processing and :aec_dump_interface. 2018-05-15 14:22:52 +00:00
aec_dump_impl.h Merge :audio_processing and :aec_dump_interface. 2018-05-15 14:22:52 +00:00
aec_dump_integration_test.cc Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
aec_dump_unittest.cc Merge :audio_processing and :aec_dump_interface. 2018-05-15 14:22:52 +00:00
BUILD.gn Merge :audio_processing and :aec_dump_interface. 2018-05-15 14:22:52 +00:00
capture_stream_info.cc Add FixedGainController and move GainController2 in APM. 2018-02-16 10:56:38 +00:00
capture_stream_info.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
mock_aec_dump.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_aec_dump.h Merge :audio_processing and :aec_dump_interface. 2018-05-15 14:22:52 +00:00
null_aec_dump_factory.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
write_to_file_task.cc Stop using ByteSize (deprecated) to get the size of a proto message. 2017-11-30 14:27:50 +00:00
write_to_file_task.h Move file_wrapper.h to rtc_base/system/ 2018-03-23 11:17:15 +00:00