platform-external-webrtc

Go to file

Oleh Prypin cdab13d9a3 Roll chromium_revision c27b32b2fd..1a597bc4e4 (629510:629788)

Change log: c27b32b2fd..1a597bc4e4
Full diff: c27b32b2fd..1a597bc4e4

Changed dependencies
* src/base: 5b5c6b774e..1161fb86d1
* src/build: ae9b018ef6..e148b4cffc
* src/ios: 472ec5538f..3f7730ca8d
* src/testing: 73be5bfc61..bbb6ca1af1
* src/third_party: 778cfca8c4..a8a90adc37
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/1a51a5b4a6..70fe610556
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/78b1c61fff..11e283fea2
* src/third_party/depot_tools: 484d74fa21..61d0c29253
* src/third_party/freetype/src: 1e7a8f30c2..a6feefdfef
* src/tools: 662d7014c7..a02273b65b
DEPS diff: c27b32b2fd..1a597bc4e4/DEPS

Clang version changed 352138:353250
Details: c27b32b2fd..1a597bc4e4/tools/clang/scripts/update.py

Bug: None
Change-Id: I471a19927cefe25b3b2aa893cb9385a38fc9d5bf
Reviewed-on: https://webrtc-review.googlesource.com/c/121702
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26581}

2019-02-07 08:39:57 +00:00

api

Delete MediaTransport method SetNetworkChangeCallback

2019-02-06 17:46:19 +00:00

audio

Propagate base minimum delay to audio_receiver_stream

2019-02-06 11:07:42 +00:00

build_overrides

Disable hermetic toolchain when building on macOS 10.14.

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

call

Propagate base minimum delay to audio_receiver_stream

2019-02-06 11:07:42 +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

Add events processing to GetIceEvents.

2019-02-06 14:14:09 +00:00

media

Reland: Remove dead code from stream_params.h

2019-02-06 19:18:15 +00:00

modules

Move RtpSenderVideo tests to separate file.

2019-02-06 18:00:39 +00:00

p2p

Remove TCPPort incoming_only_ member.

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

Allow more than one registered network change callback in MediaTransport

2019-02-06 16:14:45 +00:00

resources

Reland of Opus multistream.

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

rtc_base

Implement encoder overshoot detector and rate adjuster.

2019-02-06 15:54:11 +00:00

rtc_tools

Do not build rtp_generator in Chromium builds.

2019-02-04 14:48:58 +00:00

sdk

Separate AndroidVideoTrackSource::OnFrameCaptured from adaptation

2019-02-06 14:33:59 +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

Fix test FrameGenerator to work with a single file source

2019-02-06 11:59:03 +00:00

tools_webrtc

Pause rolling buildtools

2019-02-07 08:38:52 +00:00

video

Implement encoder overshoot detector and rate adjuster.

2019-02-06 15:54:11 +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

Remove HAVE_WEBRTC_VOICE.

2019-02-06 18:39:45 +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 c27b32b2fd..1a597bc4e4 (629510:629788)

2019-02-07 08:39:57 +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

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%