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chromium-webrtc-autoroll ab20b3f364 Roll chromium_revision 81c26a093b..aed902039c (611047:611312)

Change log: 81c26a093b..aed902039c
Full diff: 81c26a093b..aed902039c

Changed dependencies
* src/base: 3c9ac70552..577be0657e
* src/build: 076d347a56..e1cd03d1a2
* src/ios: aa06ff9dc9..cd3f4cdae4
* src/testing: eeb483a834..1070bcfcd9
* src/third_party: 3a20b86d38..23cd046553
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/f241a59dcc..6965d25602
* src/third_party/depot_tools: 6c18a1afa1..44d4b29082
* src/tools: 627b8cb5af..fb15cfbb94
* src/tools/swarming_client: b6e9e23e4e..157bec8a25
DEPS diff: 81c26a093b..aed902039c/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I2ae5dbcc633fbe1cccbd0bc28011ccbbdcb2828f
Reviewed-on: https://webrtc-review.googlesource.com/c/112215
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25808}

2018-11-27 22:58:35 +00:00

api

Add PeerConnection option to configure minimum audio jitter buffer delay.

2018-11-27 19:49:48 +00:00

audio

Add PeerConnection option to configure minimum audio jitter buffer delay.

2018-11-27 19:49:48 +00:00

build_overrides

nit: Missing space in build_overrides/build.gni

2018-11-15 14:17:12 +00:00

call

Add PeerConnection option to configure minimum audio jitter buffer delay.

2018-11-27 19:49:48 +00:00

common_audio

Delete unneeded includes of common_types.h and gn deps on webrtc_common.

2018-11-20 16:28:39 +00:00

common_video

Delete deprecated class WrappedI420Buffer

2018-11-13 10:59:10 +00:00

data

Remove old data files.

2018-10-05 14:40:21 +00:00

examples

Increase test timeouts in TCPChannelClientTest

2018-11-22 13:16:18 +00:00

infra

Remove ios32_sim_ios9_dbg from CQ.

2018-10-15 06:59:19 +00:00

logging

Remove RSID from stream configs in new event log format.

2018-11-22 17:54:06 +00:00

media

Add PeerConnection option to configure minimum audio jitter buffer delay.

2018-11-27 19:49:48 +00:00

modules

AEC3: Add metrics for API call jitter

2018-11-27 19:52:08 +00:00

p2p

Delay call to Destroy until after SignalDone has finished firing.

2018-11-27 19:55:48 +00:00

Add PeerConnection option to configure minimum audio jitter buffer delay.

2018-11-27 19:49:48 +00:00

resources

Removing ancient and unused test scripts and data files

2018-11-05 16:08:46 +00:00

rtc_base

Rebase std::is_trivially_* with absl::is_trivially_*

2018-11-26 19:20:27 +00:00

rtc_tools

Add magjed as owner of rtc_tools.

2018-11-26 11:30:34 +00:00

sdk

Remove unused BWE field trial strings.

2018-11-27 13:05:43 +00:00

stats

Expose delayed packet outage as a cumulative metric of samples in the new getStats API.

2018-11-27 15:10:09 +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 unneeded includes of common_types.h and gn deps on webrtc_common.

2018-11-20 16:28:39 +00:00

test

Change HdrMetadataExtension to ColorSpaceExtension

2018-11-27 14:05:31 +00:00

tools_webrtc

Fix webrtc-internal configs to run perf tests on separate bots

2018-11-27 16:36:30 +00:00

video

Move RtcpStatistics from common_types.h to a new header file

2018-11-27 13:46:42 +00:00

.clang-format

…

.git-blame-ignore-revs

Let git-hyper-blame ignore format commit.

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

.gitignore

Reland "Compile frame analyzer for the host machine on perf tests."

2018-09-18 09:51:19 +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

abseil-in-webrtc.md

Rebase std::is_trivially_* with absl::is_trivially_*

2018-11-26 19:20:27 +00:00

AUTHORS

add cstring include need for strncmp

2018-11-26 20:49:36 +00:00

BUILD.gn

Decouple //rtc_base:rtc_base_tests_utils from gunit.

2018-11-23 12:52:46 +00:00

CODE_OF_CONDUCT.md

…

codereview.settings

Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit

2018-10-25 08:19:53 +00:00

common_types.h

Move RtcpStatistics from common_types.h to a new header file

2018-11-27 13:46:42 +00:00

DEPS

Roll chromium_revision 81c26a093b..aed902039c (611047:611312)

2018-11-27 22:58:35 +00:00

ENG_REVIEW_OWNERS

Enforce LGTM from owners of depends-on paths in DEPS via presubmit.

2018-09-28 12:49:54 +00:00

LICENSE

…

license_template.txt

…

native-api.md

Add documentation about field_trial/metrics custom impl.

2018-09-18 11:27:59 +00:00

OWNERS

Clean up root OWNERS.

2018-11-09 14:23:59 +00:00

PATENTS

…

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

Add a presubmit check for absl/memory/memory.h inclusion

2018-11-15 12:15:40 +00:00

pylintrc

Fixing py lint errors

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

README.chromium

…

README.md

Tell users where they can find the native API headers

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

style-guide.md

Add a style rule about not using const optional<T>& arguments

2018-11-08 11:57:35 +00:00

WATCHLISTS

Remove likely obsolete entries from WATCHLISTS

2018-10-30 07:46:29 +00:00

webrtc.gni

Roll chromium_revision f9be7d3d66..d298cced6c (610432:610627)

2018-11-23 17:47:35 +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%