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chromium-webrtc-autoroll ae9641b6d8 Roll chromium_revision 70eb5f7c71..4dc8a31053 (747482:747587)

Change log: 70eb5f7c71..4dc8a31053
Full diff: 70eb5f7c71..4dc8a31053

Changed dependencies
* src/base: ab0a88b5de..61e8827cfc
* src/build: 2681c0858d..e393474c8c
* src/ios: aa035f4191..d6be293d4c
* src/testing: f85cfa6a31..bf1933a3b7
* src/third_party: 041588241a..4e9cdeb786
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2e60acad14..b3bfbaa321
* src/third_party/depot_tools: 99df04e8aa..ee8be8a368
* src/third_party/icu: 49ee7b1d18..0b6134378c
* src/tools: b66d37be6a..1ae8daf4c5
Added dependency
* src/third_party/android_deps/libs/com_google_protobuf_protobuf_javalite
Removed dependency
* src/third_party/android_deps/libs/com_google_protobuf_protobuf_lite
DEPS diff: 70eb5f7c71..4dc8a31053/DEPS

No update to Clang.

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

Change-Id: I39c8fe36cd0ad52000db1ad2d760f9d7a3afdce9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169880
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@{#30699}

2020-03-06 06:33:23 +00:00

api

Adds field trial to separate audio and video packets for delay-based overuse detection.

2020-03-05 16:29:55 +00:00

audio

Only update the current time of a played out frame if a new frame is played out.

2020-02-25 12:48:32 +00:00

build_overrides

Remove crbug.com/904400 workaround.

2019-03-15 18:36:23 +00:00

call

Transform encoded frame in RTPSenderVideo.

2020-03-03 08:17:49 +00:00

common_audio

Optimizations and refactoring of the APM 3-band split filter

2020-02-24 13:19:14 +00:00

common_video

Remove wildcard ownership for build files.

2020-02-19 14:05:46 +00:00

data

Remove old data files.

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

docs

Fix links in docs/native-code/index.md.

2020-03-05 08:19:11 +00:00

examples

Remove wildcard ownership for build files.

2020-02-19 14:05:46 +00:00

logging

Replace DataSize and DataRate factories with newer versions

2020-02-18 16:09:50 +00:00

media

Implement new specification for degradation preference

2020-03-05 14:24:25 +00:00

modules

Adds field trial to separate audio and video packets for delay-based overuse detection.

2020-03-05 16:29:55 +00:00

p2p

Make Connection::id() const

2020-03-05 09:30:18 +00:00

Adds field trial to separate audio and video packets for delay-based overuse detection.

2020-03-05 16:29:55 +00:00

resources

iSAC API wrapper unit test fix

2020-02-27 14:27:23 +00:00

rtc_base

Update RTC_CHECK and RTC_LOG macros so they work when called from xxxxx::rtc namespaces

2020-03-04 22:53:34 +00:00

rtc_tools

Add YUV to IVF video converter util

2020-02-19 14:44:21 +00:00

sdk

Implement new specification for degradation preference

2020-03-05 14:24:25 +00:00

stats

Remove wildcard ownership for build files.

2020-02-19 14:05:46 +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 wildcard ownership for build files.

2020-02-19 14:05:46 +00:00

test

Add printout of supported codecs in PC test framework

2020-03-05 18:05:26 +00:00

tools_webrtc

Whitespace change to kick bots.

2020-02-28 06:44:59 +00:00

video

StatsEndToEndTest: Configure bitrate via VideoEncoderConfig.

2020-03-05 08:25:31 +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 new format cleanup.

2019-07-11 16:18:51 +00:00

.gitignore

Add .clangd to .gitignore

2019-10-28 12:27:50 +00:00

.gn

Switch to compiling WebRTC -std=c++14 by default

2019-09-09 19:24:16 +00:00

.vpython

Add source-side perf upload script for WebRTC.

2019-11-18 14:37:01 +00:00

abseil-in-webrtc.md

Fix typo in abseil-in-webrtc.md.

2019-12-18 14:27:34 +00:00

AUTHORS

Reland "Reland "Reland "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."""""

2020-02-26 20:35:54 +00:00

BUILD.gn

Move video_replay under rtc_tools/.

2020-02-07 17:57:30 +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

Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"

2020-02-07 08:23:58 +00:00

DEPS

Roll chromium_revision 70eb5f7c71..4dc8a31053 (747482:747587)

2020-03-06 06:33:23 +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

…

native-api.md

Delete unused I420 "codec"

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

OWNERS

Add #COMPONENT to WebRTC.

2019-10-08 12:20:39 +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

Use source_sets in component builds and static_library in release builds.

2019-10-17 21:17:18 +00:00

PRESUBMIT.py

Fix public_deps presubmit and gn format fighting each other.

2020-01-30 11:22:46 +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 guidance to style guide how to reference a bug in a TODO

2019-12-11 11:55:52 +00:00

WATCHLISTS

Remove benwright@webrtc.org from WATCHLISTS

2020-01-31 18:46:52 +00:00

webrtc_lib_link_test.cc

Rewrite the lib link test to just be a binary.

2019-10-18 07:42:20 +00:00

webrtc.gni

Reformat GN files.

2020-01-21 12:13:11 +00:00

whitespace.txt

Whitespace change

2020-01-28 07:53:15 +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%