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ehmaldonado d64bf6f312 MB: Add Mac Asan (swarming) to FYI waterfall. 
R=kjellander@webrtc.org
BUG=chromium:497757
NOTRY=True

Review-Url: https://codereview.webrtc.org/2491623004
Cr-Commit-Position: refs/heads/master@{#15021}
2016-11-10 14:24:40 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Don't copy loop variables when we don't have to
2016-11-10 13:15:41 +00:00
audio
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
base
Ping the premier connection on each network with higher priority.
2016-11-09 04:04:14 +00:00
build
MB: Add Mac Asan (swarming) to FYI waterfall.
2016-11-10 14:24:40 +00:00
call
Remove audio from BitrateEstimatorTest.
2016-11-08 11:45:01 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Add helper functions for negotiating H264 profile level id
2016-11-10 09:57:42 +00:00
examples
Set videoCapturerStopped when PeerConnectionClient stopped
2016-11-10 14:23:34 +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
Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ )
2016-11-10 11:36:57 +00:00
modules
Replace SequencedTaskChecker in RTPSenderVideo
2016-11-10 14:22:23 +00:00
p2p
First step in providing a UdpTransportChannel.
2016-11-09 21:44:13 +00:00
pc
Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ )
2016-11-10 11:36:57 +00:00
sdk
Bump mac deployment target to 10.9
2016-11-09 15:44:34 +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
Revert of Extract bitrate allocation of spatial/temporal layers out of codec impl. (patchset #17 id:320001 of https://codereview.webrtc.org/2434073003/ )
2016-11-09 14:14:56 +00:00
tools
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
video
Declare VideoCodec.codec_specific_info private
2016-11-10 11:26:45 +00:00
voice_engine
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +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
Declare VideoCodec.codec_specific_info private
2016-11-10 11:26:45 +00:00
common_types.h
Declare VideoCodec.codec_specific_info private
2016-11-10 11:26:45 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Declare VideoCodec.codec_specific_info private
2016-11-10 11:26:45 +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 Extract bitrate allocation of spatial/temporal layers out of codec impl. (patchset #17 id:320001 of https://codereview.webrtc.org/2434073003/ )
2016-11-09 14:14:56 +00:00
video_frame.h
Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ )
2016-11-02 10:40:05 +00:00
video_receive_stream.h
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +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.