webrtc_m130

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Autoroller ed125fb2f3 Roll chromium_revision 60eb76e65c..858c0de5ad (583181:583305)

Change log: 60eb76e65c..858c0de5ad
Full diff: 60eb76e65c..858c0de5ad

Changed dependencies:
* src/build: 3ae3c03905..e322d9cf51
* src/ios: 2197360460..8402344fe9
* src/testing: b797c58e93..ffe8c848fd
* src/third_party: f506abeeed..f94ca0cb97
* src/third_party/depot_tools: 52c7211807..af03ae194f
* src/tools: a4b36d2aaf..49e9e90d54
DEPS diff: 60eb76e65c..858c0de5ad/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal_compile_lite

Change-Id: I0a38e1866d866d3cb877b8bc528ba20d46cc22ad
Reviewed-on: https://webrtc-review.googlesource.com/94337
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24296}

2018-08-15 18:03:57 +00:00

api

Use NetworkSimulationInterface instead of SimulatedNetwork.

2018-08-15 10:21:10 +00:00

audio

Makes treatment of received reports of packets lost signed.

2018-08-15 14:27:23 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +00:00

call

Use NetworkSimulationInterface instead of SimulatedNetwork.

2018-08-15 10:21:10 +00:00

common_audio

Move spl_sqrt_floor dep to proper third_party directory

2018-08-07 10:33:13 +00:00

common_video

Delete root header file typedef.h.

2018-07-25 14:59:26 +00:00

data

…

examples

Add certificate generate/set functionality to bring iOS closer to JS API

2018-08-13 22:25:15 +00:00

infra

cq_name is no longer used and can and should be removed

2018-08-09 12:12:35 +00:00

logging

Print timestamp-to-UTC map when event_log starts.

2018-08-10 10:04:12 +00:00

media

Revert "Add initial support for RtpEncodingParameters max_framerate."

2018-08-14 07:25:23 +00:00

modules

Optimize execution time of RTPSender::UpdateDelayStatistics

2018-08-15 17:53:04 +00:00

ortc

Delete root header file typedef.h.

2018-07-25 14:59:26 +00:00

p2p

Remove definition and usage of macro GTEST_RELATIVE_PATH.

2018-08-08 11:00:11 +00:00

Add a missing #include from jsepicecandidate.cc to absl/memory/memory.h

2018-08-15 15:19:07 +00:00

resources

Fix gitignore pattern to never exclude sha1 files

2018-08-03 12:45:45 +00:00

rtc_base

Remove public_deps from rtc_base:protobuf_utils.

2018-08-13 14:06:31 +00:00

rtc_tools

Add UTC time to init event in AEC debug dump.

2018-08-11 20:29:07 +00:00

sdk

Add certificate generate/set functionality to bring iOS closer to JS API

2018-08-13 22:25:15 +00:00

stats

Removing clang:find_bad_constructs from stats/.

2018-07-16 14:27:07 +00:00

style-guide

Add style guide rule about paired .h and .cc files

2018-03-14 13:02:35 +00:00

system_wrappers

Delete root header file typedef.h.

2018-07-25 14:59:26 +00:00

test

Remove RTPVideoHeader::vp9() accessors.

2018-08-09 10:53:28 +00:00

tools_webrtc

Temporarily skip linux_internal on autoroller tryjobs.

2018-08-10 12:55:08 +00:00

video

In (new) estimator of encoder cpu load, count max per input frame.

2018-08-15 15:16:47 +00:00

.clang-format

Tune ObjC clang-format configuration

2017-05-11 09:14:18 +00:00

.git-blame-ignore-revs

Let git-hyper-blame ignore format commit.

2018-06-20 09:26:44 +00:00

.gitignore

Roll back checking in the third_party directory

2018-06-27 13:04:08 +00:00

.gn

Re-enable gtest absl pretty printers.

2018-08-13 13:54:05 +00:00

.vpython

Add vpython dependencies needed to run presubmit tests on LUCI

2018-05-18 08:10:25 +00:00

AUTHORS

Add certificate generate/set functionality to bring iOS closer to JS API

2018-08-13 22:25:15 +00:00

BUILD.gn

Remove definition and usage of macro GTEST_RELATIVE_PATH.

2018-08-08 11:00:11 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

2017-05-16 12:09:13 +00:00

codereview.settings

Make Gerrit the default for WebRTC changes

2017-09-29 01:38:07 +00:00

common_types.h

Delete struct webrtc::PacketTime.

2018-08-07 10:07:15 +00:00

DEPS

Roll chromium_revision 60eb76e65c..858c0de5ad (583181:583305)

2018-08-15 18:03:57 +00:00

LICENSE

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

license_template.txt

…

native-api.md

Remove legacy VoiceEngine.

2018-01-12 11:31:52 +00:00

OWNERS

Add mbonadei@ to build configs OWNERS.

2018-06-20 12:39:11 +00:00

PATENTS

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

presubmit_test_mocks.py

Reland: Add presubmit check for changes in 3pp

2018-05-22 13:11:18 +00:00

presubmit_test.py

Fixing py lint errors

2018-07-23 15:28:48 +00:00

PRESUBMIT.py

Remove memcheck.

2018-08-07 07:40:08 +00:00

pylintrc

Fixing py lint errors

2018-07-23 15:28:48 +00:00

README.chromium

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

README.md

Tell users where they can find the native API headers

2017-11-14 10:36:46 +00:00

style-guide.md

Add style guidance about forward declarations.

2018-03-28 20:58:27 +00:00

WATCHLISTS

Adding mbonadei@ to build_files WATCHLIST.

2018-06-20 12:38:06 +00:00

webrtc.gni

Remove memcheck.

2018-08-07 07:40:08 +00:00

whitespace.txt

Whitespace change

2018-02-23 10:34:16 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info

Official web site: http://www.webrtc.org
Master source code repo: https://webrtc.googlesource.com/src
Samples and reference apps: https://github.com/webrtc
Mailing list: http://groups.google.com/group/discuss-webrtc
Continuous build: http://build.chromium.org/p/client.webrtc
Coding style guide
Code of conduct

Languages

C++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%