webrtc/modules
Per Åhgren fc63c9e273 AEC3: Allow filter adaptation even though the estimated echo is saturated
This CL removes the constraint that freezes the filter adaptation
whenever the estimated echo or the prediction error is saturated. This
allows for much more rapid filter recovery in cases where the echo path
gain for some reason changes, such as when the analog AGC gain is
adjusted or the loudspeaker volume is changed.

TBR: devicentepena@webrtc.org
Bug: webrtc:9466,chromium:857426
Change-Id: Ic0b3b03f41f12e9a607aaadd2ee91cbaa16cac52
Reviewed-on: https://webrtc-review.googlesource.com/86124
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23775}
2018-06-28 22:45:18 +00:00
..
audio_coding Extract fft to separate target to be able to move it to third_party 2018-06-27 09:08:19 +00:00
audio_device Replace rtc::Optional with absl::optional 2018-06-21 09:32:56 +00:00
audio_mixer Remove APM limiter in Audio Mixer. 2018-06-25 14:06:11 +00:00
audio_processing AEC3: Allow filter adaptation even though the estimated echo is saturated 2018-06-28 22:45:18 +00:00
bitrate_controller Removing usage of //build/config/compiler:no_size_t_to_int_warning. 2018-06-20 13:44:26 +00:00
congestion_controller Use field trial parser for BBR Experiment. 2018-06-28 07:52:58 +00:00
desktop_capture Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
include Add RTPVideoHeader const accessor. 2018-06-21 09:49:40 +00:00
pacing Adds field trial for disabling pacer queue draining. 2018-06-28 13:46:22 +00:00
remote_bitrate_estimator Removing some MSVC warning suppression flags. 2018-06-20 12:41:46 +00:00
rtp_rtcp Remove StreamStatistician::IsPacketInOrder 2018-06-28 08:44:40 +00:00
utility Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
video_capture Delete unused file. 2018-06-28 12:53:17 +00:00
video_coding Revert two instances of num_active_spatial_layers. 2018-06-27 10:49:00 +00:00
video_processing Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
BUILD.gn Replace rtc::Optional with absl::optional 2018-06-21 09:32:56 +00:00
module_common_types_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS