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webrtc_m130/webrtc
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zijiehe 3045589e5f Remove references of ScreenCapturer and WindowCapturer 
This change removes references of ScreenCapturer and WindowCapturer from WebRTC.
So after this change, ScreenCapturer and WindowCapturer classes can be entirely
removed.

BUG=webrtc:6513

Review-Url: https://codereview.webrtc.org/2489943004
Cr-Commit-Position: refs/heads/master@{#15006}
2016-11-10 00:37:57 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
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
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
Add build flag to enable bitcode on ios
2016-11-09 12:58:19 +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
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
examples
Add screenshare support to AppRTCMobile.
2016-11-09 15:37:32 +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 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
modules
Remove references of ScreenCapturer and WindowCapturer
2016-11-10 00:37:57 +00:00
p2p
First step in providing a UdpTransportChannel.
2016-11-09 21:44:13 +00:00
pc
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +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
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
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
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
common_types.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
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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.