webrtc/modules/audio_processing/aec_dump
Minyue Li 656d609a95 Add UTC time to init event in AEC debug dump.
Bug: webrtc:9616
Change-Id: I1350212f0b8835fb64427483269da96d51670c01
Reviewed-on: https://webrtc-review.googlesource.com/92620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24267}
2018-08-11 20:29:07 +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 UTC time to init event in AEC debug dump. 2018-08-11 20:29:07 +00:00
aec_dump_impl.h Add UTC time to init event in AEC debug dump. 2018-08-11 20:29:07 +00:00
aec_dump_integration_test.cc Add UTC time to init event in AEC debug dump. 2018-08-11 20:29:07 +00:00
aec_dump_unittest.cc Add UTC time to init event in AEC debug dump. 2018-08-11 20:29:07 +00:00
BUILD.gn Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +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 Add UTC time to init event in AEC debug dump. 2018-08-11 20:29:07 +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