platform-external-webrtc

Go to file

chromium-webrtc-autoroll 0179a3d224 Roll chromium_revision bfd7bcf815..9a34b2cc2d (632040:632146)

Change log: bfd7bcf815..9a34b2cc2d
Full diff: bfd7bcf815..9a34b2cc2d

Changed dependencies
* src/base: 21b2987f40..8f4487a1e9
* src/build: d7b4a0981f..7f6757275f
* src/ios: 21b7d265e8..c34eafde76
* src/testing: 15d911e392..847bd5d9c1
* src/third_party: 97e8569c41..1f019ff756
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/63f9b85be5..8ba5518f87
* src/tools: 56007f95ed..69818a7ff0
DEPS diff: bfd7bcf815..9a34b2cc2d/DEPS

No update to Clang.

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

Change-Id: I12a5dc02459e823a67aa9660e6434ebc6862b075
Reviewed-on: https://webrtc-review.googlesource.com/c/123024
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@{#26680}

2019-02-14 10:36:27 +00:00

api

Add FullStackTest with simulated encoder overshooting

2019-02-13 22:55:50 +00:00

audio

Adds CallEncoder to ChannelSend.

2019-02-13 15:01:53 +00:00

build_overrides

Disable hermetic toolchain when building on macOS 10.14.

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

call

Adds audio priority bitrate field trial parameter.

2019-02-12 16:03:22 +00:00

common_audio

Stop using Googletest legacy APIs.

2019-01-31 13:23:33 +00:00

common_video

Always add/rewrite VUI and set max_num_reorder_frames to 0.

2019-02-04 09:18:46 +00:00

data

Remove old data files.

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

examples

Remove unused file.

2019-02-06 10:24:07 +00:00

logging

Replace NOTREACHED with a break.

2019-02-14 09:59:49 +00:00

media

Revert "Delete video source proxying in WebRtcVideoSendStream"

2019-02-13 22:42:43 +00:00

modules

Update the resolution check for VP8 simulcast.

2019-02-14 09:56:40 +00:00

p2p

Remove TCPPort incoming_only_ member.

2019-02-04 19:46:30 +00:00

Delete obsolete usage of FakeConstraints

2019-02-12 12:27:04 +00:00

resources

Reland of Opus multistream.

2019-01-29 12:16:19 +00:00

rtc_base

Use a provider in rtc::Network to access the mDNS responder.

2019-02-14 00:59:06 +00:00

rtc_tools

Fix -Wunused-result warnings

2019-02-13 09:24:08 +00:00

sdk

Delete unused media constraints

2019-02-08 14:45:00 +00:00

stats

Expose video freeze metrics in GetStats.

2019-02-04 09:58:08 +00:00

style-guide

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

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

system_wrappers

Makes Clock interface fully mutable.

2019-01-31 08:12:05 +00:00

test

Introduce default video quality analyzer

2019-02-14 09:08:31 +00:00

tools_webrtc

Delete obsolete usage of FakeConstraints

2019-02-12 12:27:04 +00:00

video

Add FullStackTest with simulated encoder overshooting

2019-02-13 22:55:50 +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

Allow use of functions in absl/algorithms

2019-01-28 18:42:16 +00:00

AUTHORS

Add ',' between elements in RTCStatsReport::ToJson

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

BUILD.gn

Add gn configs to remove the dependency to audio and video codecs.

2019-02-12 20:41:25 +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 bfd7bcf815..9a34b2cc2d (632040:632146)

2019-02-14 10:36:27 +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

Add PRESUBMIT to enforce usage of new Googletest APIs.

2019-02-06 11:06:02 +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

Remove rule that discourages passing optional by const reference

2019-02-05 11:58:05 +00:00

WATCHLISTS

Modify pc/ WATCHLISTS definition

2019-01-31 22:09:16 +00:00

webrtc.gni

Add gn configs to remove the dependency to audio and video codecs.

2019-02-12 20:41:25 +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%