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zijiehe 8d1649d71b MANUAL tests of GDI capturers 
ScreenCapturerWinGdi randomly returns black frames in test environment. The root
cause is still unknown, so change ScreenCapturerWinGdi tests into MANUAL mode to
execute in test environment, but unblock other developers. We can eventually get
a failure ratio and more samples for debugging.

BUG=webrtc:6666, webrtc:6843

Review-Url: https://codereview.webrtc.org/2564173002
Cr-Commit-Position: refs/heads/master@{#15518}
2016-12-10 00:00:10 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
RTCInboundRTPStreamStats.packetsLost set by RTCStatsCollector.
2016-12-09 12:19:50 +00:00
audio
Refactor webrtc/{api,audio} and modules/audio_coding for GN check
2016-12-07 16:23:35 +00:00
base
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
build
Move /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +00:00
call
Disable failing perf test on Android.
2016-12-09 14:15:22 +00:00
common_audio
Split targets mixing .c and .cc sources.
2016-12-06 06:47:52 +00:00
common_video
Sanity check parsed QP values from H264 bitstream
2016-12-02 09:29:53 +00:00
examples
Add FlexFEC settings toggle in Android AppRTCMobile.
2016-12-08 14:51:35 +00:00
libjingle
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
logging
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
media
Reland of Add ability to scale to arbitrary factors (patchset #1 id:1 of https://codereview.webrtc.org/2557323002/ )
2016-12-08 16:04:58 +00:00
modules
MANUAL tests of GDI capturers
2016-12-10 00:00:10 +00:00
p2p
Relanding: Refactoring that removes P2PTransport and DtlsTransport classes.
2016-12-07 00:22:11 +00:00
pc
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
sdk
Modify JavaToStdString to allow ISO-8859-1 encoded strings.
2016-12-09 14:13:48 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
system_wrappers
Add comment to metrics.h
2016-12-06 11:59:08 +00:00
test
Generalize FlexfecReceiveStream::Config.
2016-12-08 12:18:05 +00:00
tools
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
video
Only store sequence numbers for media stream in FlexFEC end-to-end test.
2016-12-09 14:51:43 +00:00
voice_engine
Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h
2016-12-05 10:22:18 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
2016-12-09 10:35:30 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +00:00
voice_engine_configurations.h
Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h
2016-12-05 10:22:18 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.