webrtc/modules/audio_processing/include
Per Åhgren 8ba5861f7e Redesign of the render buffering in AEC3
This CL centralizes the render buffering in AEC3 so that all render
buffers are updated and synchronized/aligned with the render alignment
buffer.

Bug: webrtc:8597, chromium:790905
Change-Id: I8a94e5c1f27316b6100b420eec9652ea31c1a91d
Reviewed-on: https://webrtc-review.googlesource.com/25680
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20989}
2017-12-01 23:14:32 +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 Redesign of the render buffering in AEC3 2017-12-01 23:14:32 +00:00
audio_processing_statistics.cc Use the new AudioProcessing statistics everywhere. 2017-11-24 18:17:39 +00:00
audio_processing_statistics.h Added metric for the delay in AEC3. 2017-11-27 12:52:42 +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 Use the new AudioProcessing statistics everywhere. 2017-11-24 18:17:39 +00:00