Alessio Bazzica 8da7b350cf AGC2 adaptive digital false by default ...

Avoid that the client code relies on the adaptive digital mode being
enabled by default (error prone).

Bug: webrtc:7494
Change-Id: I765fecf535cf31a2163e10595a42520473c233b6
Reviewed-on: https://webrtc-review.googlesource.com/c/111586
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25728}

2018-11-21 14:20:15 +00:00

..

aec_dump.cc

Removing the intelligibility enhancer.

2018-08-30 21:29:57 +00:00

aec_dump.h

[Cleanup] Add missing #include. Remove useless ones.

2018-10-23 11:32:56 +00:00

audio_frame_view.h

Add namespace 'webrtc' to AudioFrameView.

2018-05-14 12:33:49 +00:00

audio_generator_factory.cc

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

audio_generator_factory.h

Add stub draft of audio generator to APM

2018-03-05 09:28:52 +00:00

audio_generator.h

Add stub draft of audio generator to APM

2018-03-05 09:28:52 +00:00

audio_processing_statistics.cc

Use the new AudioProcessing statistics everywhere.

2017-11-24 18:17:39 +00:00

audio_processing_statistics.h

[Cleanup] Add missing #include. Remove useless ones.

2018-10-23 11:32:56 +00:00

audio_processing.cc

[Cleanup] Add missing #include. Remove useless ones.

2018-10-23 11:32:56 +00:00

audio_processing.h

AGC2 adaptive digital false by default

2018-11-21 14:20:15 +00:00

config.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

config.h

Reland "Export symbols needed by the Chromium component build (part 1)."

2018-10-11 09:50:21 +00:00

gain_control.h

Break out Agc code from audio_processing.

2018-07-06 13:29:43 +00:00

mock_audio_processing.h

Remove deprecated AudioProcessing::GetStatistics function

2018-11-01 11:21:15 +00:00