webrtc_m130

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Autoroller cf06a53152 Roll chromium_revision 381f71a417..b13129e4a5 (547202:547655)

Change log: 381f71a417..b13129e4a5
Full diff: 381f71a417..b13129e4a5

Changed dependencies:
* src/base: ce3710c94d..187e6fe890
* src/build: fd402752c1..e8dd3a198e
* src/ios: adfc442c5c..4ebeebf55f
* src/testing: bf9442f946..1e1ec9d9b4
* src/third_party: 6b5c78334f..1aaec09102
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/a6bfc45b62..eb7c3008cc
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d95849b996..29a751cece
* src/third_party/depot_tools: a16b4ccd55..c7d0b34084
* src/third_party/libvpx/source/libvpx: f4b1eca53e..d636fe53af
* src/third_party/winsdk_samples: 2d31a1cbec..601401003b
* src/tools: aaaaac29fb..d9299c5672
DEPS diff: 381f71a417..b13129e4a5/DEPS

No update to Clang.

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

Change-Id: I89359a021e0999ab0f5ed76a1fa92b04e741c5b9
Reviewed-on: https://webrtc-review.googlesource.com/66420
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22704}

2018-04-03 10:27:20 +00:00

api

Reland "Reland "Adds support for multiple or no media stream ids.""

2018-04-03 01:10:07 +00:00

audio

Fix MID not always getting set with audio

2018-03-29 20:22:28 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Don't check MIDs when demuxing RTP packets in Call

2018-03-29 20:36:08 +00:00

common_audio

Move aligned memory utilities to rtc_base/memory/

2018-03-22 14:13:24 +00:00

common_video

Remove the public_deps to fileutils from test_support.

2018-03-16 09:06:27 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

2016-11-22 18:43:05 +00:00

examples

Android: Rename AudioDeviceModule to JavaAudioDeviceModule

2018-03-29 10:55:37 +00:00

infra

Reland "Add linux_internal_compile_lite trybot to CQ"

2018-03-26 10:12:59 +00:00

logging

Provide the option of injecting rtc::TaskQueue when creating RtcEventLogImpl via factory methods.

2018-03-20 18:06:18 +00:00

media

Add nullptr check in SctpTransport.

2018-04-03 03:04:07 +00:00

modules

Remove a couple of unnecessary winsock2.h includes

2018-04-03 08:49:58 +00:00

ortc

Use new TransportController implementation in PeerConnection.

2018-03-30 18:41:19 +00:00

p2p

Remove LOG_J and LOG_JV, tweak p2p logs.

2018-03-29 08:21:27 +00:00

Add nullptr check in SctpTransport.

2018-04-03 03:04:07 +00:00

resources

Adding FourPeople_1280x720_30.yuv.

2018-02-12 15:55:00 +00:00

rtc_base

Annotate rest of WebRTC with @Nullable.

2018-03-28 08:30:06 +00:00

rtc_tools

Fix path to proto in py_event_log_analyzer/pb_parse.py

2018-03-19 07:42:35 +00:00

sdk

Revert "Add unit tests for RTCCVPixelBuffer and ObjCVideoTrackSource."

2018-03-30 00:49:48 +00:00

stats

PeerConnectionInterface::GetStats() with selector argument added.

2018-03-26 12:08:20 +00:00

style-guide

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

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

system_wrappers

Remove a couple of unnecessary winsock2.h includes

2018-04-03 08:49:58 +00:00

test

Revert "Reland "Move rtp-specific config out of EncoderSettings.""

2018-03-29 11:45:18 +00:00

tools_webrtc

Start autorolling winsdk_samples

2018-04-03 09:18:07 +00:00

video

Call RegisterStatsObserver after initializing video_stream_decoder_

2018-03-30 08:03:38 +00:00

.clang-format

Tune ObjC clang-format configuration

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

.git-blame-ignore-revs

…

.gitignore

Reland "Make it possible to run video_quality_loopback_test in swarming."

2018-01-23 13:03:17 +00:00

.gn

Re-enabling 'gn check' on //examples/*.

2018-02-19 15:07:45 +00:00

.vpython

vpython: Specify dependency on pywin32

2018-02-14 13:56:39 +00:00

AUTHORS

Expose a link-local network interfaces enumeration option

2018-02-06 19:12:04 +00:00

BUILD.gn

Removing -Wno-strict-overflow from main BUILD.gn.

2018-03-28 11:25:07 +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.cc

Adding layering configurator and rate allocator for VP9 SVC.

2018-03-29 10:16:47 +00:00

common_types.h

Adding layering configurator and rate allocator for VP9 SVC.

2018-03-29 10:16:47 +00:00

DEPS

Roll chromium_revision 381f71a417..b13129e4a5 (547202:547655)

2018-04-03 10:27:20 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

LICENSE_THIRD_PARTY

Remove custom MD5 / SHA-1 implementations.

2018-02-19 15:03:35 +00:00

native-api.md

Remove legacy VoiceEngine.

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

OWNERS

Add back OWNERS entris that went missing

2018-03-06 10:31:11 +00:00

PATENTS

Moving src/webrtc into src/.

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

presubmit_test_mocks.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

presubmit_test.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

PRESUBMIT.py

Revert "Relaxing no-streams presubmit check (streams are allowed in tests)."

2018-03-19 10:32:02 +00:00

pylintrc

Removing invalid-name from disabled pylint checks.

2017-10-11 08:06:49 +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

typedefs.h

Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix

2018-02-05 11:24:59 +00:00

WATCHLISTS

Move remaining traces of VoiceEngine

2018-01-17 13:27:47 +00:00

webrtc.gni

Annotate rest of WebRTC with @Nullable.

2018-03-28 08:30:06 +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%