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Honghai Zhang f4ae6dc763 Fix an issue in IPv6 support. 
When creating connections on turn port, check whether the local and remote candidates have the same IP address family, instead of checking the address family of the local socket against the remote candidate.

BUG=5871
R=deadbeef@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2083803002 .

Cr-Commit-Position: refs/heads/master@{#13269}
2016-06-23 05:35:06 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
Adding IceConfig option to assume TURN/TURN candidate pairs will work.
2016-06-23 00:14:20 +00:00
audio
Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ )
2016-06-20 18:53:09 +00:00
base
Reland of Do not delete a connection in the turn port with permission error or refresh error. (patchset #1 id:1 of https://codereview.webrtc.org/2090833002/ )
2016-06-22 23:27:08 +00:00
build
Remove dead GYP target audio_device_module_java
2016-06-20 08:43:42 +00:00
call
Remove ViEEncoder::Pause / Start
2016-06-17 14:27:23 +00:00
common_audio
GN: Add common_audio_unittests and common_video_unittests
2016-06-08 08:28:15 +00:00
common_video
Delete deprecated VideoFrameBuffer methods.
2016-06-21 10:59:32 +00:00
examples
Add RTCEventLog API to ObjC.
2016-06-14 22:03:19 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
modules
Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #5 id:80001 of https://codereview.chromium.org/2037623002/ )
2016-06-22 23:29:58 +00:00
p2p
Fix an issue in IPv6 support.
2016-06-23 05:35:06 +00:00
pc
Delete GetExecutablePath and related unused code.
2016-06-16 10:44:44 +00:00
sdk
Revert of Delete method cricket::VideoFrame::Copy. (patchset #7 id:120001 of https://codereview.webrtc.org/2080253002/ )
2016-06-21 21:03:01 +00:00
system_wrappers
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
test
Add fuzzer corpora.
2016-06-22 13:43:30 +00:00
tools
New rtc dump analyzing tool in Python
2016-06-20 14:39:21 +00:00
video
Explicitly logging video suspend state.
2016-06-22 15:42:41 +00:00
voice_engine
Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #5 id:80001 of https://codereview.chromium.org/2037623002/ )
2016-06-22 23:29:58 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
audio_send_stream.h
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 17:53:28 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Implement timestamp translation/filter in VideoCapturer.
2016-06-22 15:36:58 +00:00
call.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add kwiberg@webrtc.org as root owner.
2016-06-14 09:39:40 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
video_receive_stream.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Workaround java.gypi inclusion error in Chromium builds.
2016-06-20 08:34:11 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Fix missing resource file in webrtc_perf_tests.isolate
2016-06-17 07:39:49 +00:00
webrtc_tests.gypi
Implement timestamp translation/filter in VideoCapturer.
2016-06-22 15:36:58 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.