Zijie He 9cad5016a3 Implement IsOccluded() in each DesktopCapturer implementation ...

This change implements IsOccluded() function in each DesktopCapturer
implementation.

This change was reviewed at
https://chromium-review.googlesource.com/c/external/webrtc/+/660817.

Bug: webrtc:7950
Change-Id: Ie4d49b2bfcf5f0da1576f47b752a51ec494442ad
TBR: jamiewalch@chromium.org
Reviewed-on: https://webrtc-review.googlesource.com/1400
Reviewed-by: Zijie He <zijiehe@google.com>
Commit-Queue: Zijie He <zijiehe@google.com>
Cr-Commit-Position: refs/heads/master@{#19840}

2017-09-14 17:42:39 +00:00

..

audio_coding

Add assignment operator to AudioEncoderRuntimeConfig

2017-09-12 16:00:24 +00:00

audio_conference_mixer

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

audio_device

Decoupling audio_device from Obj-C code

2017-09-12 11:45:24 +00:00

audio_mixer

Update thread annotiation macros in modules to use RTC_ prefix

2017-09-07 14:53:45 +00:00

audio_processing

Fixed the overflow in the AGC

2017-09-13 13:28:16 +00:00

bitrate_controller

Update thread annotiation macros in modules to use RTC_ prefix

2017-09-07 14:53:45 +00:00

congestion_controller

Update thread annotiation macros in modules to use RTC_ prefix

2017-09-07 14:53:45 +00:00

desktop_capture

Implement IsOccluded() in each DesktopCapturer implementation

2017-09-14 17:42:39 +00:00

include

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

media_file

Reland "Supporting 48kHz PCM file."

2017-09-01 19:49:24 +00:00

pacing

Update thread annotiation macros in modules to use RTC_ prefix

2017-09-07 14:53:45 +00:00

remote_bitrate_estimator

Only use BBRs pacer if the unit test is running BBR.

2017-09-12 12:47:34 +00:00

rtp_rtcp

Rename rtp::Packet to RtpPacket

2017-09-14 13:37:29 +00:00

utility

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

video_capture

Decoupling video_capture_internal_imlp from Obj-C code

2017-09-04 13:54:54 +00:00

video_coding

Move reencode logic for screenshare bitrate overshoot from generic

2017-09-12 17:24:46 +00:00

video_processing

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

BUILD.gn

Add VideoProcessorIntegrationTest for MediaCodec implementations.

2017-09-07 14:50:06 +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