platform-external-webrtc

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chromium-webrtc-autoroll 9444f3a8c8 Roll chromium_revision 6a5b2b19b1..3343618014 (626014:626126)

Change log: 6a5b2b19b1..3343618014
Full diff: 6a5b2b19b1..3343618014

Changed dependencies
* src/base: 9015adf2da..5bbe3caa9f
* src/build: 018911f9a4..838abed988
* src/ios: 528045cd2a..c4af087b33
* src/testing: 5ee5c49371..e69083cab6
* src/third_party: bea3b73746..e2106465bd
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/8e8f250422..6c1b376e1d
* src/third_party/depot_tools: 80b9cf7dfd..60574b5f91
* src/tools: 91febde900..e9cc7fad3f
DEPS diff: 6a5b2b19b1..3343618014/DEPS

Clang version changed 351477:352138
Details: 6a5b2b19b1..3343618014/tools/clang/scripts/update.py

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

Change-Id: Ie9f02f7198f9f87e0f31b23bc976dee021f2bbb5
Reviewed-on: https://webrtc-review.googlesource.com/c/119961
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@{#26413}

2019-01-25 19:38:16 +00:00

api

AEC3: Remove legacy render buffering

2019-01-25 08:31:12 +00:00

audio

Introduces common AudioAllocationSettings class.

2019-01-23 12:13:29 +00:00

build_overrides

Disable hermetic toolchain when building on macOS 10.14.

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

call

Make sure lost packets are removed from FakeNetworkPipe.

2019-01-25 08:57:45 +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

Don't attempt to unwrap RTP timestamps for RTX stream.

2019-01-25 15:59:22 +00:00

media

Negate flag name for prerender smoothing and update comments.

2019-01-24 11:53:26 +00:00

modules

Check "rtc_include_internal_audio_device" before creating unittest for audio_device_ios_objc

2019-01-25 18:51:07 +00:00

p2p

Don't pretend we've received an end-of-candidates indication.

2019-01-23 10:16:43 +00:00

Do not modify media transport config when falling back to RTP

2019-01-25 18:19:17 +00:00

resources

Revert "Opus multistream."

2019-01-17 22:38:57 +00:00

rtc_base

Don't attempt to unwrap RTP timestamps for RTX stream.

2019-01-25 15:59:22 +00:00

rtc_tools

frame_analyzer: exit with status 1 when video files fail to open

2019-01-25 11:31:11 +00:00

sdk

Android: Add option to print native stack traces in PeerConnectionFactory API

2019-01-21 16:07:26 +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

Replace the usage of RTC_HISTOGRAM_COMMON_BLOCK with

2019-01-21 10:44:35 +00:00

test

Improving robustness of stable bandwidth estimate test.

2019-01-25 15:06:17 +00:00

tools_webrtc

Correctly specify Mac version as 10.13 for iOS simulator tests

2019-01-23 10:32:57 +00:00

video

Revert "Always use real VideoStreamsFactory in full stack tests"

2019-01-25 14:27:10 +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

Add Visual Studio Code project folder to gitignore file.

2019-01-21 18:42:33 +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

Adds repeating task class.

2019-01-18 10:55:41 +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

Delete constant RTP_PAYLOAD_NAME_SIZE

2019-01-25 07:59:52 +00:00

DEPS

Roll chromium_revision 6a5b2b19b1..3343618014 (626014:626126)

2019-01-25 19:38:16 +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

Fixing snake_case files that were renamed in PRESUBMIT.py

2019-01-23 18:38:53 +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

Override default manifest from Chromium in WebRTC.

2019-01-24 13:00:49 +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++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%