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webrtc_m130/webrtc
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ehmaldonado 1bf0ff36ea Roll chromium_revision f156b499f7..f439921f66 (493756:494089) 
Change log: f156b499f7..f439921f66
Full diff: f156b499f7..f439921f66

Changed dependencies:
* src/base: c9ab1936b1..feac46e933
* src/build: 5fecec2d69..221820676e
* src/ios: bf72566bf8..e437e37fec
* src/testing: 1330967db2..3f5325f618
* src/third_party: 0a591a99a3..058ff821a8
* src/third_party/catapult: 0eeb5baed7..122dd5e91b
* src/tools: 3136678749..e78bdaf8db
DEPS diff: f156b499f7..f439921f66/DEPS

Clang version changed 309984:310694
Details: f156b499f7..f439921f66/tools/clang/scripts/update.py

TBR=
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Review-Url: https://codereview.webrtc.org/3001673002
Cr-Commit-Position: refs/heads/master@{#19350}
2017-08-15 10:16:50 +00:00
..
api
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
audio
Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
2017-07-26 14:28:51 +00:00
base
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
call
Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ )
2017-08-14 15:03:17 +00:00
common_audio
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
common_video
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
examples
Add --canary flag to gradle project test and update gradle.
2017-08-11 08:03:13 +00:00
logging
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
media
Default enable content type rtp header extension
2017-08-15 08:32:51 +00:00
modules
Roll chromium_revision f156b499f7..f439921f66 (493756:494089)
2017-08-15 10:16:50 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
pc
Revert of Trace the stats report as JSON instead of each stat separately. (patchset #3 id:100001 of https://codereview.webrtc.org/2986453002/ )
2017-08-15 06:48:03 +00:00
rtc_base
Implementation of SSL caching; tests in separate CL.
2017-08-15 00:37:05 +00:00
rtc_tools
Improved UI for event_log_analyzer tool
2017-08-15 09:04:02 +00:00
sdk
Add VideoSink interface to SurfaceViewRenderer.
2017-08-15 08:56:02 +00:00
stats
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
system_wrappers
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
test
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
video
Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ )
2017-08-14 15:03:17 +00:00
voice_engine
Add thread annotations to AudioLevel
2017-08-10 12:01:21 +00:00
.gitignore
…
BUILD.gn
Removing NO_SOUND_SYSTEM
2017-08-14 07:40:58 +00:00
codereview.settings
…
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_receive_stream.h
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
Removing unused declared arg
2017-08-03 17:45:31 +00:00
whitespace.txt
…

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.