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chromium-webrtc-autoroll ea9845f16a Roll chromium_revision 3a9216d2da..e291b99915 (623639:623778)

Change log: 3a9216d2da..e291b99915
Full diff: 3a9216d2da..e291b99915

Changed dependencies
* src/base: 7a599db692..99b4f188ee
* src/build: 5da693fe57..fed0626a8d
* src/ios: d36794522f..646c6b87c2
* src/testing: 735158b947..93b5ba2dc8
* src/third_party: ef8e4d1296..9e3a2a27e8
* src/third_party/android_build_tools/bundletool: iBp9dwYQET2-P96y1HPJikezjXIprC6C4i6vUyviPcUC..GpDcZ4f-UcI4u6Kk7Ud-vtiLMltFr8vW63-NThsB5tEC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a9729f3347..a0e5e3328c
* src/third_party/libvpx/source/libvpx: 858fe955ae..9ecc0e779a
* src/tools: c1667501ec..f9f06f0f06
DEPS diff: 3a9216d2da..e291b99915/DEPS

No update to Clang.

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

Change-Id: I3232b8d7adc26e7c0b5a7f8885b793e2a843160c
Reviewed-on: https://webrtc-review.googlesource.com/c/118161
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@{#26303}

2019-01-17 19:55:56 +00:00

api

Add ability for VideoEncoder to signal frame rate allocation.

2019-01-17 15:40:53 +00:00

audio

Fix chromium roll into WebRTC.

2019-01-16 15:19:57 +00:00

build_overrides

Disable hermetic toolchain when building on macOS 10.14.

2018-12-07 13:57:28 +00:00

call

Expose jitterBufferEmittedCount in addition to the existing jitterBufferDelay for getStats().

2019-01-16 11:44:10 +00:00

common_audio

(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries

2019-01-11 17:11:39 +00:00

common_video

(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries

2019-01-11 17:11:39 +00:00

data

Remove old data files.

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

examples

Remove legacy video codec factories.

2019-01-14 14:56:40 +00:00

infra

Re-add MSVC debug bots to CQ

2018-12-05 12:20:56 +00:00

logging

Add DTLSTransport info into sender/receiver state.

2019-01-17 10:21:32 +00:00

media

Add ability for VideoEncoder to signal frame rate allocation.

2019-01-17 15:40:53 +00:00

modules

Add ability for VideoEncoder to signal frame rate allocation.

2019-01-17 15:40:53 +00:00

p2p

DTLS transport needs to forward network route change.

2019-01-17 19:25:25 +00:00

Add DTLSTransport info into sender/receiver state.

2019-01-17 10:21:32 +00:00

resources

Opus multistream.

2019-01-17 12:23:23 +00:00

rtc_base

Introduce webrtc::TaskQueue and TaskQueueFactory interfaces

2019-01-17 13:10:14 +00:00

rtc_tools

(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries

2019-01-11 17:11:39 +00:00

sdk

Move some video codec constants to separate file.

2019-01-17 15:29:53 +00:00

stats

Expose jitterBufferEmittedCount in addition to the existing jitterBufferDelay for getStats().

2019-01-16 11:44:10 +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 definition of macro WEBRTC_THREAD_RR (it's unused).

2019-01-14 15:01:38 +00:00

test

Introduce ExampleVideoQualityAnalyzer.

2019-01-17 16:50:38 +00:00

tools_webrtc

Increase timeout for internal iOS perf, like for public android and desktop

2019-01-17 13:46:18 +00:00

video

Move some video codec constants to separate file.

2019-01-17 15:29:53 +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

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 ',' between elements in RTCStatsReport::ToJson

2019-01-08 16:05:23 +00:00

BUILD.gn

Remove legacy video codec factories.

2019-01-14 14:56:40 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

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

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 some video codec constants to separate file.

2019-01-17 15:29:53 +00:00

DEPS

Roll chromium_revision 3a9216d2da..e291b99915 (623639:623778)

2019-01-17 19:55:56 +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

Moving src/webrtc into src/.

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

license_template.txt

Update template to follow chromium copyright style

2013-04-24 01:01:28 +00:00

native-api.md

Delete unused I420 "codec"

2018-12-18 12:30:58 +00:00

OWNERS

Clean up root OWNERS.

2018-11-09 14:23:59 +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

Delete WebRtcVideoCapturer and related classes.

2019-01-03 15:15:48 +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 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 //third_party/ffmpeg and disable H264 when MSVC is used.

2019-01-15 10:25:01 +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%