platform-external-webrtc

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chromium-webrtc-autoroll 262cf691b1 Roll chromium_revision fa85f826d0..dd5a54c29b (736081:736224)

Change log: fa85f826d0..dd5a54c29b
Full diff: fa85f826d0..dd5a54c29b

Changed dependencies
* src/base: 72eb032b0e..ce806f00e6
* src/build: 5dc5e02620..2f17606c25
* src/ios: 0de24610a1..31829ea7dd
* src/testing: b83b82bf65..403d2930e5
* src/third_party: aa87e20519..18f4ad54fc
* src/third_party/depot_tools: 9d635962bc..ea8b58b970
* src/tools: 03d0d67bf3..39d70a2950
DEPS diff: fa85f826d0..dd5a54c29b/DEPS

No update to Clang.

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

Change-Id: I5ab5844dcfc879598a31aeb24f6b8687ec497fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167780
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@{#30398}

2020-01-29 04:51:45 +00:00

api

Remove iceRegatherIntervalRange

2020-01-28 19:16:18 +00:00

audio

Reland "Only include overhead if using send side bandwidth estimation."

2020-01-28 10:36:39 +00:00

build_overrides

Remove crbug.com/904400 workaround.

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

call

Pass picture_id from generic packetizer through codec-specific field

2020-01-28 19:26:28 +00:00

common_audio

Reformat GN files.

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

common_video

Reformat GN files.

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

data

Remove old data files.

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

docs

remove mention of prebuilt libraries from docs/

2020-01-28 16:08:44 +00:00

examples

Reformat GN files.

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

logging

Reformat GN files.

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

media

Revert "Reland "Reland "Distinguish between send and receive codecs"""

2020-01-24 16:44:17 +00:00

modules

Pass picture_id from generic packetizer through codec-specific field

2020-01-28 19:26:28 +00:00

p2p

Remove iceRegatherIntervalRange

2020-01-28 19:16:18 +00:00

Remove iceRegatherIntervalRange

2020-01-28 19:16:18 +00:00

resources

Reformat GN files.

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

rtc_base

Remove iceRegatherIntervalRange

2020-01-28 19:16:18 +00:00

rtc_tools

Reformat GN files.

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

sdk

Remove iceRegatherIntervalRange

2020-01-28 19:16:18 +00:00

stats

[Stats] Include fecPackets[Reeceived/Discarded] in Members()

2020-01-28 11:22:09 +00:00

style-guide

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

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

system_wrappers

Reformat GN files.

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

test

Adds fake video codec mode to PeerScenarioClient

2020-01-27 18:07:45 +00:00

tools_webrtc

Remove strip_absolute_paths_from_debug_symbols from mb.

2020-01-27 09:17:46 +00:00

video

[Overuse] Move GetCpuOveruseOptions() to adaption module.

2020-01-27 13:51:40 +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

Fix typo in abseil-in-webrtc.md.

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

AUTHORS

Update Android camera switch API to allow specifying a name

2020-01-09 16:04:09 +00:00

BUILD.gn

Fix video_replay to build and actually work

2020-01-22 13:16:28 +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 fa85f826d0..dd5a54c29b (736081:736224)

2020-01-29 04:51:45 +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

Extract an interface from the perf results logger.

2020-01-14 06:05:02 +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

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

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++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%