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zhihuang 38989e593c Parse the connection data in SDP (c= line). 
Extract the remote addresses from SDP c= line on both session level and
media level. The media level address will overwrite the session level one if
exists.

WebRTC is not using c= and this is used for new SDP parsing API.

BUG=webrtc:7311

Review-Url: https://codereview.webrtc.org/2742903002
Cr-Commit-Position: refs/heads/master@{#17326}
2017-03-21 18:04:53 +00:00
..
api
Make lint errors fatal in presubmit and fix files in whitelisted paths
2017-03-21 14:52:41 +00:00
audio
Delete class MockCongestionController.
2017-03-21 16:41:07 +00:00
base
Add a missing DCHECK to PlatformThread::SetPriority.
2017-03-20 10:47:17 +00:00
call
Split CongestionController into send- and receive-side classes.
2017-03-21 13:41:12 +00:00
common_audio
iSAC fix entropy coder: Recently added DCHECK could in fact trigger
2017-03-13 12:28:47 +00:00
common_video
Add FullStack test for simulcast screenshare mode.
2017-03-09 17:23:30 +00:00
examples
Add video codec setting to AppRTCMobile on iOS.
2017-03-17 16:01:59 +00:00
logging
Enable GN check for webrtc/logging
2017-03-08 10:12:11 +00:00
media
(Re)move VoE specific enums from common_types.h.
2017-03-16 08:20:23 +00:00
modules
Delete class MockCongestionController.
2017-03-21 16:41:07 +00:00
ortc
Add the function ToRtpParameters.
2017-03-09 01:15:06 +00:00
p2p
Parse the connection data in SDP (c= line).
2017-03-21 18:04:53 +00:00
pc
Parse the connection data in SDP (c= line).
2017-03-21 18:04:53 +00:00
sdk
Better handling of error condition in MediaCodecVideoEncoder.
2017-03-15 12:53:14 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Delete utf_util_win.h.
2017-03-14 08:32:50 +00:00
test
TransportFeedbackPacketLossTracker to receive std::vector<PacketFeedback> in place of the entire feedback
2017-03-21 14:31:35 +00:00
tools
Improve error handling for ffmpeg operations
2017-03-15 15:27:31 +00:00
video
Split CongestionController into send- and receive-side classes.
2017-03-21 13:41:12 +00:00
voice_engine
TransportFeedbackPacketLossTracker to receive std::vector<PacketFeedback> in place of the entire feedback
2017-03-21 14:31:35 +00:00
.gitignore
…
BUILD.gn
Remove voe_cmd_test.
2017-03-07 09:43:21 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Remove dead code in vp8_impl files.
2017-03-21 16:35:51 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Use RTC_UNUSED instead of conditional compilation in BWE simulator tool.
2017-03-17 13:34:47 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.