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webrtc_m130/webrtc
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sakal e19649b8ce Fix Android lint error. 
BUG=webrtc:6711
NOTRY=True

Review-Url: https://codereview.webrtc.org/2504643002
Cr-Commit-Position: refs/heads/master@{#15095}
2016-11-15 20:32:26 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
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: Run test with gtest-parallel on swarming.
2016-11-15 20:20:35 +00:00
call
Start probes only after network is connected.
2016-11-15 20:25:37 +00:00
common_audio
Avoid left-shifting negative values in a number of places
2016-11-15 09:45:59 +00:00
common_video
Negotiate H264 profiles in SDP
2016-11-12 17:53:08 +00:00
examples
Fix Android lint error.
2016-11-15 20:32:26 +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
Start probes only after network is connected.
2016-11-15 20:25:37 +00:00
modules
Start probes only after network is connected.
2016-11-15 20:25:37 +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
Prepare iOS H264 HW encoder for High Profile
2016-11-15 17:57:02 +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
Start probes only after network is connected.
2016-11-15 20:25:37 +00:00
tools
Simplify creating RtpHeaderExtensionMap in EventLogAnalyzer
2016-11-15 17:21:03 +00:00
video
Start probes only after network is connected.
2016-11-15 20:25:37 +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
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
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
Move VideoEncoderSoftwareFallbackWrapper from webrtc/video_encoder.h to webrtc/media/engine/
2016-11-15 14:31:01 +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
Wire up FlexfecSender in RTP module and VideoSendStream.
2016-11-15 13:25:44 +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
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
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.