srte 3e69e5c2c0 Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock ...

Continues on https://codereview.webrtc.org/2992043002

BUG=webrtc:8033

Review-Url: https://codereview.webrtc.org/2994633002
Cr-Commit-Position: refs/heads/master@{#19286}

2017-08-09 13:13:45 +00:00

..

audio_coding

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +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

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

bitrate_controller

Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock

2017-08-09 13:13:45 +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

desktopCapture: minimized window shouldn't be treated as on-top on Win10

2017-08-08 20:30:01 +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

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

remote_bitrate_estimator

Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock

2017-08-09 13:13:45 +00:00

rtp_rtcp

Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock

2017-08-09 13:13:45 +00:00

utility

Tracking mock_process_thread with a GN target

2017-08-03 12:57:11 +00:00

video_capture

Remove remains of webrtc/base

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

video_coding

Fix the video buffer size should take rtt into consideration

2017-08-09 12:43:08 +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