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guoweis@webrtc.org b91d0f5130 1. Have IPIsPrivate calling IPIsLinkLocal 
2. Also check the Mac based IPv6
3. move the ip filtering into createnetwork. It shouldn't be done in IsIgnoredNetwork as the IP inside that could change later.

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48509004

Cr-Commit-Position: refs/heads/master@{#8758}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8758 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 14:43:42 +00:00
..
base
1. Have IPIsPrivate calling IPIsLinkLocal
2015-03-17 14:43:42 +00:00
build
Adding build_opus as a switch in GYP.
2015-03-17 14:05:18 +00:00
common_audio
Clean up LappedTransform and Blocker.
2015-03-12 23:24:19 +00:00
common_video
Add I420 buffer pool to avoid unnecessary allocations
2015-03-17 11:41:15 +00:00
examples
Roll chromium_revision 00e438c..8d51d96 (320241:320682)
2015-03-16 09:00:41 +00:00
libjingle
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
2015-02-26 14:43:50 +00:00
modules
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
overrides
Re-landing perf improvement for libjingle logging after reverting the general change.
2015-03-07 12:18:14 +00:00
p2p
Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession.
2015-03-16 21:16:23 +00:00
sound
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
system_wrappers
Move trace_event and associated files to webrtc/base.
2015-03-17 14:26:15 +00:00
test
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
2015-03-16 13:48:18 +00:00
tools
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
2015-03-16 13:48:18 +00:00
video_engine
Change ThreadPosix to use an auto-reset event instead of manual reset now that we know the problem we had with EventWrapper::Wait was simply a bug in the EventWrapper. Also removing |started_| since we can just check the thread_ instead.
2015-03-16 16:06:16 +00:00
voice_engine
Supporting Opus DTX in Voice Engine.
2015-03-13 09:38:55 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Let Chromium declare the mips_dsp_rev build variable.
2015-03-04 09:51:17 +00:00
call.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
config.h
Add CVO support to Vie layer.
2015-03-12 20:51:50 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Remove webrtc/system_wrappers/interface/scoped_ptr.h
2015-03-03 10:17:49 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737)
2015-02-18 10:38:11 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
2015-03-16 13:48:18 +00:00
video_encoder.h
Remove default arguments in EncodedImageCallback.
2015-02-09 09:14:48 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments.
2015-03-16 13:26:41 +00:00
video_receive_stream.h
Add decoder-timing stats to VideoReceiveStream.
2015-02-25 10:42:45 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
2015-03-10 15:13:13 +00:00
webrtc_examples.gyp
Improve cleaning for Android demo applications
2015-02-28 11:18:20 +00:00
webrtc_perf_tests.isolate
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
rtc_unittests on Android
2015-03-04 08:48:17 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.