platform-external-webrtc

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chromium-webrtc-autoroll 395d29944a Roll chromium_revision 2e2ded718a..8c0cc38022 (621632:621736)

Change log: 2e2ded718a..8c0cc38022
Full diff: 2e2ded718a..8c0cc38022

Changed dependencies
* src/base: c4e5b7ca9d..155eaadd00
* src/build: 7b20546cf8..67630827e1
* src/ios: cd569bf30b..a2ac2bd4c2
* src/testing: e091f08842..d1c310b6d6
* src/third_party: 695a8d6bb4..2da31fafff
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/daced929e5..775ce3b01f
* src/third_party/depot_tools: b1be3782a4..80a1cf66b8
* src/tools: 043d1c8fe4..7ebeeeb997
DEPS diff: 2e2ded718a..8c0cc38022/DEPS

No update to Clang.

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

Change-Id: I1e01ebad4cd94de0b04c73a97d09dec2b2bab89b
Reviewed-on: https://webrtc-review.googlesource.com/c/116860
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@{#26206}

2019-01-10 21:23:53 +00:00

api

Add PeerConnection option to enable RTX handling in the audio jitter buffer.

2019-01-10 16:28:43 +00:00

audio

Add PeerConnection option to enable RTX handling in the audio jitter buffer.

2019-01-10 16:28:43 +00:00

build_overrides

Disable hermetic toolchain when building on macOS 10.14.

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

call

Add PeerConnection option to enable RTX handling in the audio jitter buffer.

2019-01-10 16:28:43 +00:00

common_audio

[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.

2018-11-28 18:25:07 +00:00

common_video

Introduce VideoFrame::id to keep track of frames inside application.

2019-01-04 08:59:26 +00:00

data

Remove old data files.

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

examples

Fix macOS demo privacy crash.

2019-01-10 10:50:49 +00:00

infra

Re-add MSVC debug bots to CQ

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

logging

Parse logs without RTX SSRC even if there is an RTX payload type.

2019-01-10 14:43:39 +00:00

media

Add PeerConnection option to enable RTX handling in the audio jitter buffer.

2019-01-10 16:28:43 +00:00

modules

Refactor FrameBuffer to store decoded frames history separately

2019-01-10 15:11:15 +00:00

p2p

Include absl/memory/memory.h if absl::make_unique is used

2019-01-08 20:08:32 +00:00

Add PeerConnection option to enable RTX handling in the audio jitter buffer.

2019-01-10 16:28:43 +00:00

resources

Base ApmTest.Process on AudioProcessingStats.output_rms_dbfs

2018-12-18 16:45:03 +00:00

rtc_base

Remove TaskQueue::PostAndReply as unused

2019-01-10 16:06:57 +00:00

rtc_tools

Delete NetEqTest::ExtDecoderMap

2019-01-08 16:25:05 +00:00

sdk

Include absl/memory/memory.h if absl::make_unique is used

2019-01-08 20:08:32 +00:00

stats

Add ',' between elements in RTCStatsReport::ToJson

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

style-guide

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

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

system_wrappers

[Fuchsia] Implement detection of available cores.

2018-12-17 19:37:46 +00:00

test

Revert "Add a high bitrate full stack test with fake codec"

2019-01-10 11:49:05 +00:00

tools_webrtc

mb: Turn LUCI desktop perf bots into builder + tester pairs

2018-12-21 13:43:04 +00:00

video

Minor change to the Json Config format for the replay file.

2019-01-10 18:38:08 +00:00

.clang-format

…

.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

Reland "Default to dlopening the PipeWire."

2019-01-08 08:09:42 +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

Remove CodecInst pt.3

2018-12-18 07:42:21 +00:00

DEPS

Roll chromium_revision 2e2ded718a..8c0cc38022 (621632:621736)

2019-01-10 21:23:53 +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

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

Reland "Default to dlopening the PipeWire."

2019-01-08 08:09:42 +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%