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denicija 77bfd7c1b8 Add ARDSettingsModelTests to apprtcmobile_test target. 
Also extract all iOS sources into a static library configuration
so it's easier to include them in the test target as well.
Also, fix a wrong test that was undiscovered because the
tests were not running

BUG=webrtc:6707

Review-Url: https://codereview.webrtc.org/2502623002
Cr-Commit-Position: refs/heads/master@{#15076}
2016-11-15 08:41:31 +00:00
..
androidjunit
…
api
Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed].
2016-11-14 09:41:56 +00:00
audio
Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
2016-11-14 19:30:16 +00:00
base
Ping the premier connection on each network with higher priority.
2016-11-09 04:04:14 +00:00
build
MB: Add new Win8 and Win10 bots.
2016-11-11 07:07:22 +00:00
call
Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
2016-11-14 19:30:16 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Negotiate H264 profiles in SDP
2016-11-12 17:53:08 +00:00
examples
Add ARDSettingsModelTests to apprtcmobile_test target.
2016-11-15 08:41:31 +00:00
libjingle
GN: Move xmllite_xmpp_unittests into webrtc/libjingle
2016-11-08 11:18:43 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
2016-11-14 19:30:16 +00:00
modules
DirectX capturer flickers on the second monitor
2016-11-15 02:20:41 +00:00
p2p
Re-enable the P2PTransportChannelMultihomedTest.TestBasic
2016-11-10 21:59:50 +00:00
pc
Negotiate H264 profiles in SDP
2016-11-12 17:53:08 +00:00
sdk
Negotiate H264 profiles in SDP
2016-11-12 17:53:08 +00:00
stats
RTCMediaStream[Track]Stats added.
2016-11-08 14:29:26 +00:00
system_wrappers
Add stats for frequency offset when converting RTP timestamp to NTP time.
2016-11-02 14:14:10 +00:00
test
Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
2016-11-14 19:30:16 +00:00
tools
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
video
Use correct define in H264 end-to-end tests.
2016-11-14 12:11:30 +00:00
voice_engine
Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
2016-11-14 19:30:16 +00:00
.gitignore
…
BUILD.gn
First step in providing a UdpTransportChannel.
2016-11-09 21:44:13 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
Revert of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2494683006/ )
2016-11-11 14:03:26 +00:00
common_types.h
Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ )
2016-11-12 16:54:50 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Revert of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2494683006/ )
2016-11-11 14:03:26 +00:00
config.h
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Revert of Issue 2434073003: Extract bitrate allocation ... (patchset #4 id:60001 of https://codereview.webrtc.org/2488833004/ )
2016-11-10 16:30:39 +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
Remove RED/RTX workaround from sender/receiver and VideoEngine2.
2016-11-11 11:28:38 +00:00
video_send_stream.h
Let ViEEncoder express resolution requests as Sinkwants.
2016-11-01 18:45:54 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Moving stun_prober target from webrtc/p2p to webrtc/examples
2016-11-01 10:09:19 +00:00
webrtc_tests.gypi
…
webrtc.gyp
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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.