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webrtc/modules/audio_processing/aec_dump
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Karl Wiberg 6a4d411023 Move file_wrapper.h to rtc_base/system/ 
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.

BUG=webrtc:8445

Change-Id: I440974da4d347b09ff042478720d7983056b62b9
Reviewed-on: https://webrtc-review.googlesource.com/21226
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22579}
2018-03-23 11:17:15 +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 Add FixedGainController and move GainController2 in APM. 2018-02-16 10:56:38 +00:00
aec_dump_impl.h Move file_wrapper.h to rtc_base/system/ 2018-03-23 11:17:15 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
BUILD.gn Move file_wrapper.h to rtc_base/system/ 2018-03-23 11:17:15 +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 Add FixedGainController and move GainController2 in APM. 2018-02-16 10:56:38 +00:00
mock_aec_dump.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mock_aec_dump.h Add FixedGainController and move GainController2 in APM. 2018-02-16 10:56:38 +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