sprang 89c4a7e57d Wire up experiment for improved screenshare bwe. ...

Also adds some full stack test variants with the experiment enabled.

BUG=webrtc:7694

Review-Url: https://codereview.webrtc.org/2949553002
Cr-Commit-Position: refs/heads/master@{#18869}

2017-06-30 20:27:40 +00:00

..

audio_coding

Reland "Adding ANA config event to debug dump."

2017-06-30 14:32:23 +00:00

audio_conference_mixer

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

audio_device

Disable AudioDeviceTest.StartStopRecording on iOS

2017-06-30 12:08:40 +00:00

audio_mixer

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

audio_processing

Added the ability to adjust the AEC3 performance for large rooms

2017-06-30 14:06:10 +00:00

bitrate_controller

Tune loss-based BWE to be more compatible with the low frequency loss reports of audio streams.

2017-06-16 14:47:00 +00:00

congestion_controller

Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )

2017-06-30 15:11:44 +00:00

desktop_capture

Ensure Dxgi duplicator works correctly in session 0

2017-06-28 05:04:21 +00:00

include

Implement timing frames.

2017-06-19 14:18:55 +00:00

media_file

Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ )

2017-04-26 10:38:35 +00:00

pacing

Wire up experiment for improved screenshare bwe.

2017-06-30 20:27:40 +00:00

remote_bitrate_estimator

Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )

2017-06-30 15:11:44 +00:00

rtp_rtcp

Fix FecTest.FlexfecTest flakiness caused by seq. num. wraparound.

2017-06-30 14:22:15 +00:00

utility

Remove deprecation warning from JVM::Initialize with the context parameter.

2017-06-01 11:59:52 +00:00

video_capture

Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface

2017-06-14 13:46:38 +00:00

video_coding

Move webrtc/{tools => rtc_tools}

2017-06-30 10:04:59 +00:00

video_processing

Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface

2017-06-14 13:46:38 +00:00

BUILD.gn

This is an initial cl, which contains small amount of implemented functions, and large amount of unimplemented ones.

2017-06-05 13:01:26 +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