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chromium-webrtc-autoroll 7e6abf0053 Roll chromium_revision 5ac2340a23..977e732442 (703358:703537)

Change log: 5ac2340a23..977e732442
Full diff: 5ac2340a23..977e732442

Changed dependencies
* src/base: e562576635..05b43c3ab0
* src/build: 30e445c75c..ae142b53b6
* src/ios: d22d06eed7..ecf8848b0a
* src/testing: aaaa705d50..65fc5a314d
* src/third_party: c7e10c69d3..68f42f8961
* src/third_party/depot_tools: 4102985e14..3306bbe476
* src/tools: 6c2e4f90a1..7ad0ae5537
DEPS diff: 5ac2340a23..977e732442/DEPS

No update to Clang.

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

Change-Id: Ie9d0b3f534dfc7fd2ceef6e327bafc1b7a6416a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156040
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@{#29399}

2019-10-08 00:45:05 +00:00

api

Added support for skipping get_audio events, adding dummy packets and setting a field trial string.

2019-10-07 12:26:44 +00:00

audio

Removes rtp_transport checks in AudioSendStream

2019-10-07 12:58:55 +00:00

build_overrides

Remove crbug.com/904400 workaround.

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

call

Adds RTPSenderVideo::Config struct with red/ulpfec config

2019-10-04 14:19:49 +00:00

common_audio

Use std::make_unique instead of absl::make_unique.

2019-09-17 15:47:29 +00:00

common_video

Always pass arguments to INSTANTIATE_TEST_SUITE_P.

2019-09-30 12:52:07 +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

New build target api:media_interface

2019-09-19 09:32:27 +00:00

logging

Cleanup includes in modules/include/module_common_types.h

2019-10-07 16:06:26 +00:00

media

Add/remove receive streams with SSRC 0 from media channels

2019-10-07 23:01:28 +00:00

modules

Cleanup includes in modules/include/module_common_types.h

2019-10-07 16:06:26 +00:00

p2p

Update RTC_LOGs in DtlsTransport to be able to distinguish errors.

2019-10-04 12:13:52 +00:00

Add/remove receive streams with SSRC 0 from media channels

2019-10-07 23:01:28 +00:00

resources

Delete voice_detection() pointer to submodule

2019-10-07 13:06:05 +00:00

rtc_base

Removing AudioAllocationSettings moving functionality to AudioSendStream.

2019-10-03 10:52:16 +00:00

rtc_tools

Use std::make_unique instead of absl::make_unique.

2019-09-17 15:47:29 +00:00

sdk

Avoid memcpy in JavaToNativeEncodedImage

2019-10-01 12:55:44 +00:00

stats

Add qualityLimitationResolutionChanges stat

2019-09-09 15:22:57 +00:00

style-guide

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

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

system_wrappers

2019-09-10 12:34:38 +00:00

test

Delete voice_detection() pointer to submodule

2019-10-07 13:06:05 +00:00

tools_webrtc

Replacing /target:target with /target in BUILD autofix.

2019-10-04 10:07:42 +00:00

video

Cleanup includes in modules/include/module_common_types.h

2019-10-07 16:06:26 +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

Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH.

2019-09-04 18:49:28 +00:00

.gn

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

2019-09-09 19:24:16 +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

Update style guide for absl::make_unique.

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

AUTHORS

pipewire: handle deleting the capturer while a D-Bus call is in progress

2019-09-26 18:58:56 +00:00

BUILD.gn

Cleanup of simple TODO(srte) comments.

2019-10-04 07:57:16 +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 5ac2340a23..977e732442 (703358:703537)

2019-10-08 00:45:05 +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 juberti@ to webrtc root owners

2019-05-17 18:11:58 +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

absl::make_unique presubmit check.

2019-09-17 17:47:31 +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.gni

Remove rtc_use_lto GN arg.

2019-08-20 14:00:49 +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%