terelius 3376c84c90 Add probing to recover faster from large bitrate drops. A single probe at 85% of the original bitrate is sent when transitioning from underusing back to normal state. The actual sending of the probes is disabled by default, and enabled by the field trial string WebRTC-BweRapidRecoveryExperiment/Enabled/. Existing code that did probing after large drops in ALR have been restructured so that it also delays the probe until we are no longer overusing. ...

BUG=webrtc:8015

Review-Url: https://codereview.webrtc.org/2986563002
Cr-Commit-Position: refs/heads/master@{#19187}

2017-07-31 11:23:25 +00:00

..

audio_coding

Removed an obsolete DCHECK in AudioEncoderOpus.

2017-07-28 09:52:59 +00:00

audio_conference_mixer

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio_device

Fixing lint issue

2017-07-21 14:28:41 +00:00

audio_mixer

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio_processing

Integer overflow bug in low_cut_filter.

2017-07-24 11:03:46 +00:00

bitrate_controller

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

congestion_controller

Add probing to recover faster from large bitrate drops. A single probe at 85% of the original bitrate is sent when transitioning from underusing back to normal state. The actual sending of the probes is disabled by default, and enabled by the field trial string WebRTC-BweRapidRecoveryExperiment/Enabled/. Existing code that did probing after large drops in ALR have been restructured so that it also delays the probe until we are no longer overusing.

2017-07-31 11:23:25 +00:00

desktop_capture

Update ResolutionChangeDetector to make it match common practices

2017-07-28 18:08:45 +00:00

include

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

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

media_file

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

pacing

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

remote_bitrate_estimator

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

rtp_rtcp

Take smaller interface for RtpRtcp::Configuration::receive_statistics

2017-07-27 11:44:18 +00:00

utility

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

video_capture

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

video_coding

Revert of Disable SeqNumUnwrapper death tests to avoid breaking downstream builds. (patchset #1 id:1 of https://codereview.chromium.org/2985083002/ )

2017-07-27 11:37:18 +00:00

video_processing

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

BUILD.gn

Remove remains of webrtc/base

2017-07-19 17:40:47 +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