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webrtc_m130/webrtc
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zhihuang 45efce01c7 Add the url attribute to the IceCandidate (Java Wrapper) 
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.

BUG=webrtc:7128

Review-Url: https://codereview.webrtc.org/2690593002
Cr-Original-Commit-Position: refs/heads/master@{#16593}
Committed: 8586c8ee88
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16615}
2017-02-14 20:23:34 +00:00
..
api
Changing some PeerConnection-related comments.
2017-02-13 17:49:58 +00:00
audio
Clean out platform specific things from voice_engine_defines.h.
2017-02-13 12:42:52 +00:00
base
Delete httpclient.c and related files.
2017-02-14 08:23:10 +00:00
call
Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.
2017-02-13 13:59:46 +00:00
common_audio
Fix for left shift of potentially negative value.
2017-02-09 11:05:59 +00:00
common_video
Remove henrik.lundin from webrtc/common_video/OWNERS
2017-02-10 16:44:31 +00:00
examples
Add the url attribute to the IceCandidate (Java Wrapper)
2017-02-14 20:23:34 +00:00
logging
Move AudioDecoder and related stuff to the api/ directory
2017-02-10 16:15:44 +00:00
media
Handle InitDecode and reset in fallback decoder.
2017-02-13 22:11:08 +00:00
modules
Added handling of 'agc_compression_gain' flag in audioproc_f.
2017-02-14 10:07:49 +00:00
p2p
Fix the build break by keeping the old Port::AddAddress method since the
2017-02-14 01:33:27 +00:00
pc
Changing some PeerConnection-related comments.
2017-02-13 17:49:58 +00:00
sdk
Add the url attribute to the IceCandidate (Java Wrapper)
2017-02-14 20:23:34 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
replace NtpTime->Clock with Clock->NtpTime dependency
2017-02-09 19:15:25 +00:00
test
Remove VoEExternalMedia interface.
2017-02-14 12:55:00 +00:00
tools
Move AudioDecoder and related stuff to the api/ directory
2017-02-10 16:15:44 +00:00
video
Fix flaky ViEEncoder unit test.
2017-02-14 19:57:25 +00:00
voice_engine
Remove VoEExternalMedia interface.
2017-02-14 12:55:00 +00:00
.gitignore
…
BUILD.gn
Added gn target for rtc_event_log2rtp_dump.
2017-02-08 08:28:09 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Remove VoEExternalMedia interface.
2017-02-14 12:55:00 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +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
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +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
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_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Move iOS tests to XCTest from gtest.
2017-02-14 12:58:56 +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.