platform-external-webrtc

Go to file

Yves Gerey 2115d2d268 Roll chromium_revision 34a43a356e..5939567173 (722057:722888)

Manual tweak: Do not roll src/ios, since it breaks ios_sim_x64_dbg_ios10.

Change log: 34a43a356e..5939567173
Full diff: 34a43a356e..5939567173

Changed dependencies
* src/base: ad02e24051..4a67f656da
* src/build: fae06de3dd..b1050d1e6a
* src/testing: 0775600850..2363b239d0
* src/third_party: ca4f6358dd..244bb7a24b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c6bece5e5b..8953fbe6c5
* src/third_party/depot_tools: 9212599f6a..6b52dc21e1
* src/third_party/lss: https://chromium.googlesource.com/linux-syscall-support.git/+log/726d71ec08..7bde79cc27
* src/tools: b7dec18459..3f49cabf04
* src/tools/luci-go: git_revision:7d11fd9e66407c49cb6c8546a2ae45ea993a240c..git_revision:37a855b64d59b7f079c9a0e5368f2757099d14d3
DEPS diff: 34a43a356e..5939567173/DEPS

No update to Clang.

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

Change-Id: I0d3509efa554a5f8090678b22448f8ee960ac912
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161554
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30043}

2019-12-09 15:19:23 +00:00

api

Revert "Distinguish between send and receive video codecs"

2019-12-09 14:48:55 +00:00

audio

Minor fixes to ChannelSend.

2019-12-02 09:30:51 +00:00

build_overrides

Remove crbug.com/904400 workaround.

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

call

Add directive to make webrtc metrics optional.

2019-12-09 13:55:50 +00:00

common_audio

Add ability to disable detailed error message in RTC_CHECKs

2019-11-28 17:51:00 +00:00

common_video

Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class

2019-12-07 00:54:26 +00:00

data

Remove old data files.

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

docs

Fixing some typos.

2019-09-10 10:03:50 +00:00

examples

Fix errorprone issues preventing Chromium Roll.

2019-11-27 12:52:48 +00:00

logging

Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_.

2019-11-29 09:45:50 +00:00

media

Revert "Distinguish between send and receive video codecs"

2019-12-09 14:48:55 +00:00

modules

Add directive to make webrtc metrics optional.

2019-12-09 13:55:50 +00:00

p2p

Add directive to make webrtc metrics optional.

2019-12-09 13:55:50 +00:00

Revert "Distinguish between send and receive video codecs"

2019-12-09 14:48:55 +00:00

resources

Changed the digital AGC1 gain to properly support multichannel

2019-11-23 08:42:59 +00:00

rtc_base

Allow setting the initial congestion window size by config.

2019-12-09 11:00:10 +00:00

rtc_tools

Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class

2019-12-07 00:54:26 +00:00

sdk

Revert "Distinguish between send and receive video codecs"

2019-12-09 14:48:55 +00:00

stats

Add totalInterFrameDelay to RTCInboundRTPStreamStats

2019-11-25 10:50:37 +00:00

style-guide

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

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

system_wrappers

Add directive to make webrtc metrics optional.

2019-12-09 13:55:50 +00:00

test

Remove deprecated legacy AEC code

2019-12-09 10:37:49 +00:00

tools_webrtc

Revert "Remove temporary workaround for generate_licenses."

2019-11-27 17:56:45 +00:00

video

Add directive to make webrtc metrics optional.

2019-12-09 13:55:50 +00:00

.clang-format

…

.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

Update style guide for absl::make_unique.

2019-09-18 06:10:58 +00:00

AUTHORS

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

2019-12-09 11:26:20 +00:00

BUILD.gn

Add directive to make webrtc metrics optional.

2019-12-09 13:55:50 +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

Format almost everything.

2019-07-08 13:45:15 +00:00

DEPS

Roll chromium_revision 34a43a356e..5939567173 (722057:722888)

2019-12-09 15:19:23 +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

Add possibility to skip check_includes presubmit check.

2019-10-22 19:35:31 +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

Update WebRTC's C++ style guide to reflect the switch to C++14.

2019-09-16 11:45:35 +00:00

WATCHLISTS

Add saza to audio watchlists

2019-09-03 14:55:43 +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

Add directive to make webrtc metrics optional.

2019-12-09 13:55:50 +00:00

whitespace.txt

Revert "Whitespace change"

2019-11-11 14:58:20 +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++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%