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webrtc/modules/audio_processing/include
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Per Åhgren d980c57c80 Adding more conservative AEC3 suppressor behavior initially in calls 
Bug: webrtc:8746
Change-Id: I47def88f8d6092fcb6b1a4bd14478e8d5ccd5320
Reviewed-on: https://webrtc-review.googlesource.com/39840
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21631}
2018-01-16 09:32:52 +00:00
..
aec_dump.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
aec_dump.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_processing.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_processing.h Adding more conservative AEC3 suppressor behavior initially in calls 2018-01-16 09:32:52 +00:00
audio_processing_statistics.cc Use the new AudioProcessing statistics everywhere. 2017-11-24 18:17:39 +00:00
audio_processing_statistics.h Make delay stat optional. 2017-12-15 14:23:06 +00:00
config.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
config.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mock_audio_processing.h Render-side pre-processing in APM. 2017-12-18 16:11:03 +00:00