webrtc_m130

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Autoroller 28d7ce80f5 Roll chromium_revision 840e0f7269..7f86d891ed (520123:520726)

Change log: 840e0f7269..7f86d891ed
Full diff: 840e0f7269..7f86d891ed

Changed dependencies:
* src/base: 5dfdb70192..f24ce599a9
* src/build: b1a63aeccd..0d99c1a9d9
* src/ios: 597d6a0451..fdc84c6ac7
* src/testing: 119295dad5..83301a6384
* src/third_party: 38215cc4ef..c6fa5edc84
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/8793942c5c..fc9c67599d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b0b1ce2c6e..c3fc7aa9c6
* src/third_party/depot_tools: 9e51906ffb..8076c2854d
* src/third_party/ffmpeg: 18c815f814..23e0b491fa
* src/tools: d5795c8019..2552ec6e90
DEPS diff: 840e0f7269..7f86d891ed/DEPS

No update to Clang.

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

Change-Id: Ia110bc25905714237fd929b4d40b655a9016bf1a
Reviewed-on: https://webrtc-review.googlesource.com/27760
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20959}

2017-11-30 23:12:11 +00:00

api

Require audio codec API types to have a Config type member

2017-11-29 14:14:19 +00:00

audio

Use the new AudioProcessing statistics everywhere.

2017-11-24 18:17:39 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Made functions on BitrateAllocator::ObserverConfig member functions

2017-11-29 12:51:49 +00:00

common_audio

Optional: Use nullopt and implicit construction in /common_audio

2017-11-28 15:46:48 +00:00

common_video

Reland "Add stereo codec header and pass it through RTP"

2017-11-30 01:44:19 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Roll Chromium + Fix Android lint suppressions

2017-11-30 16:59:50 +00:00

infra

Adding win_asan to CQ.

2017-11-22 09:12:18 +00:00

logging

Stop using ByteSize (deprecated) to get the size of a proto message.

2017-11-30 14:27:50 +00:00

media

Resolve circular dependency in rtc_media_base.

2017-11-28 15:28:58 +00:00

modules

Roll Chromium + Fix Android lint suppressions

2017-11-30 16:59:50 +00:00

ortc

Replace MediaContentDirection with RtpTransceiverDirection

2017-11-28 23:44:28 +00:00

p2p

Fix cpplint errors in port/port_unittest

2017-11-29 19:57:09 +00:00

Make the DtlsSrtpTransport cache the RtpAbsSendTimeHeaderExtension.

2017-11-30 02:17:09 +00:00

resources

Add a new NetEq decoding unit test for Opus with DTX

2017-11-28 10:45:38 +00:00

rtc_base

Optional: Use nullopt and implicit construction in /rtc_base/rate_statistics.cc

2017-11-29 11:37:29 +00:00

rtc_tools

Reland "Replaced magic numbers with constants for PacketFeedback."

2017-11-30 16:04:20 +00:00

sdk

Optional: Use nullopt and implicit construction in /sdk/objc

2017-11-30 15:52:50 +00:00

stats

Add network_type to local RTCIceCandidateStats

2017-11-21 19:58:37 +00:00

system_wrappers

Reland of "Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator""

2017-11-16 12:10:04 +00:00

test

Add a function to report perf results in JSON format.

2017-11-30 11:20:00 +00:00

tools_webrtc

Roll Chromium + Fix Android lint suppressions

2017-11-30 16:59:50 +00:00

video

Reland "Add stereo codec header and pass it through RTP"

2017-11-30 01:44:19 +00:00

voice_engine

Remove voice_engine_defines.h

2017-11-30 10:22:10 +00:00

.clang-format

Tune ObjC clang-format configuration

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

.git-blame-ignore-revs

Create .git-blame-ignore-revs and add Java format CL to it.

2016-10-20 09:20:39 +00:00

.gitignore

Roll chromium_revision f93b8b19f2..adf969a7cb (513366:514871) and more.

2017-11-08 21:42:48 +00:00

.gn

Roll chromium_revision 5df4e1bfe7..5bd5874cbf (518692:519731) + iOS fix

2017-11-28 18:42:38 +00:00

.vpython

Add psutil to vpython dependencies (used on builder bots)

2017-09-04 08:04:16 +00:00

AUTHORS

Delete wrapper API ConvertToI420 for YUV conversion to I420

2017-11-15 11:10:20 +00:00

BUILD.gn

Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)."

2017-09-29 13:48:29 +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

Add a new function to BitrateAllocation: HasBitrate.

2017-11-23 15:00:08 +00:00

common_types.h

Reland "Add stereo codec header and pass it through RTP"

2017-11-30 01:44:19 +00:00

DEPS

Roll chromium_revision 840e0f7269..7f86d891ed (520123:520726)

2017-11-30 23:12:11 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

LICENSE_THIRD_PARTY

Moving src/webrtc into src/.

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

native-api.md

Tell users where they can find the native API headers

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

OWNERS

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +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

Tell users where they can find the native API headers

2017-11-14 10:36:46 +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

Style guide: Attempt to make the L2 and L3 headings more visually distinct

2017-09-09 03:52:23 +00:00

typedefs.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

WATCHLISTS

Add myself to the watchlist for webrtc/api/ and webrtc/base/

2017-05-04 13:22:46 +00:00

webrtc.gni

Removing assert in rtc_test template.

2017-11-22 08:26:37 +00:00

whitespace.txt

Moving src/webrtc into src/.

2017-09-15 04:25:06 +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

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Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%