webrtc/modules/audio_processing/agc
Artem Titov 333a50562c Move fft4g to proper third_party directory
Bug: webrtc:8366
Change-Id: I98d3ae56a1d14b3ecacd85a4b3d234e215c8bc58
Reviewed-on: https://webrtc-review.googlesource.com/85642
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24103}
2018-07-25 15:44:53 +00:00
..
legacy Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
agc.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
agc.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
agc_manager_direct.cc Reset level estimator when analog gain changes. 2018-07-20 14:18:38 +00:00
agc_manager_direct.h Allow AGC2 level estimation in AgcManagerDirect. 2018-07-06 14:18:18 +00:00
agc_manager_direct_unittest.cc Reset level estimator when analog gain changes. 2018-07-20 14:18:38 +00:00
BUILD.gn Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
gain_map_internal.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
loudness_histogram.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
loudness_histogram.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
loudness_histogram_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mock_agc.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
utility.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
utility.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00