peah 5e6685ff35 Robustification of the AEC3 echo removal in the first part of the call ...

This CL robustifies the echo removal in AEC3 during the initial parts
of a call in two ways:
-By extending the period until which a headset is deemed to be used.
-By increasing the assumed echo path gain for unknown echo paths at
higher frequencies.

BUG=webrtc:7971

Review-Url: https://codereview.webrtc.org/2974883002
Cr-Commit-Position: refs/heads/master@{#18967}

2017-07-11 11:19:58 +00:00

..

audio_coding

Remove verbose logs from audio_coding_module.cc.

2017-07-10 17:36:28 +00:00

audio_conference_mixer

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

audio_device

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

audio_mixer

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

audio_processing

Robustification of the AEC3 echo removal in the first part of the call

2017-07-11 11:19:58 +00:00

bitrate_controller

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

congestion_controller

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

desktop_capture

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

include

Add AudioFrame::ResetWithoutMuting() to address performance regression.

2017-07-08 23:36:26 +00:00

media_file

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

pacing

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

remote_bitrate_estimator

Remove unused headers from remote_bitrate_estimator_single_stream.cc

2017-07-10 19:46:25 +00:00

rtp_rtcp

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

utility

Now uses CallStaticObjectMethodV to an variable argument list argument

2017-07-11 10:52:09 +00:00

video_capture

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

video_coding

Revert of Refactor timing frame logic to work with encoders with internal sources (patchset #2 id:20001 of https://codereview.webrtc.org/2968153002/ )

2017-07-10 19:25:29 +00:00

video_processing

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

BUILD.gn

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

module_common_types_unittest.cc

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00