brandtr 669ea1917e Rename WEBRTC_VIDEOPROCESSOR_H264_TESTS define to WEBRTC_USE_H264. ...

This is the name used in other parts of the code.

BUG=none

Review-Url: https://codereview.webrtc.org/2996463003
Cr-Commit-Position: refs/heads/master@{#19253}

2017-08-07 10:35:13 +00:00

..

audio_coding

Renamed fields in common_types.h/RtcpStatistics.

2017-08-04 12:03:53 +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

TestDataGenerators attempts to create missing input signal files.

2017-08-01 12:44:18 +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

Revert "Track recreation of DxgiTextureStaging"

2017-08-03 10:39:49 +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

Add PacketRouterTest.Sanity_NoModuleRegistered_*

2017-08-02 13:29:00 +00:00

remote_bitrate_estimator

Renamed fields in common_types.h/RtcpStatistics.

2017-08-04 12:03:53 +00:00

rtp_rtcp

Renamed fields in common_types.h/RtcpStatistics.

2017-08-04 12:03:53 +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

Rename WEBRTC_VIDEOPROCESSOR_H264_TESTS define to WEBRTC_USE_H264.

2017-08-07 10:35:13 +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