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marpan 5f4aaeb0a4 Re-enable FullStackTest.ScreenshareSlidesVP9_2SL test. 
TBR=sprang@webrtc.org
BUG=webrtc:7401

Review-Url: https://codereview.webrtc.org/2799063003
Cr-Commit-Position: refs/heads/master@{#17571}
2017-04-06 17:52:23 +00:00
..
api
Injectable audio encoders: BuiltinAudioEncoderFactory
2017-04-06 17:03:21 +00:00
audio
Let PacketRouter separate send and receive modules.
2017-03-31 12:44:52 +00:00
base
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
call
Make RtpTransportControllerSend::send_side_cc_ a direct member.
2017-04-06 13:34:25 +00:00
common_audio
iSAC fix entropy coder: Recently added DCHECK could in fact trigger
2017-03-13 12:28:47 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Use new RTCCameraVideoCapturer in AppRTCMobile.
2017-04-05 19:17:48 +00:00
logging
Resolve cyclic dependency between audio network adaptor and event log api
2017-04-06 12:59:10 +00:00
media
Injectable audio encoders: BuiltinAudioEncoderFactory
2017-04-06 17:03:21 +00:00
modules
Adding AudioDeviceDataObserver interface
2017-04-06 17:17:02 +00:00
ortc
Disable the ORTC integration tests on TSan.
2017-03-28 20:45:18 +00:00
p2p
Disable possibly flaky AudioCodingModuleTest.TestPacketLossStereo for Linux.
2017-03-28 10:00:07 +00:00
pc
Injectable audio encoders: BuiltinAudioEncoderFactory
2017-04-06 17:03:21 +00:00
sdk
Move AndroidVideoTrackSourceObserver from API to src
2017-04-06 11:31:26 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Delete system_wrappers logging facility.
2017-04-06 10:53:06 +00:00
test
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
tools
Resolve cyclic dependency between audio network adaptor and event log api
2017-04-06 12:59:10 +00:00
video
Re-enable FullStackTest.ScreenshareSlidesVP9_2SL test.
2017-04-06 17:52:23 +00:00
voice_engine
Injectable audio encoders: BuiltinAudioEncoderFactory
2017-04-06 17:03:21 +00:00
.gitignore
…
BUILD.gn
Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
2017-03-31 10:08:07 +00:00
call.h
…
codereview.settings
…
common_types.cc
…
common_types.h
Revert of Supporting 48kHz PCM file. (patchset #1 id:1 of https://codereview.webrtc.org/2790493004/ )
2017-03-31 23:32:28 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
DEPS
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_decoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_encoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream.h
Add number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.