webrtc_m130

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chromium-webrtc-autoroll ad44b75a7c Roll chromium_revision e1ec78e27e..60cc82f9b7 (663034:663509)

Change log: e1ec78e27e..60cc82f9b7
Full diff: e1ec78e27e..60cc82f9b7

Changed dependencies
* src/base: 8e5cc6374c..e34acbc60c
* src/build: 912c7b060f..323d12f978
* src/ios: 3af8884c08..823f58ee81
* src/testing: 98c1282560..9f14178c61
* src/third_party: 6bd5381759..0a5d09d1d0
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/441284b3fb..a7b3312467
* src/third_party/depot_tools: 54434e7e1d..26af0d34d2
* src/tools: 7cf65e88e0..5264b01658
DEPS diff: e1ec78e27e..60cc82f9b7/DEPS

Clang version changed 67510fac36d27b2e22c7cd955fc167136b737b93:342571e8d6eb1afb151ae1103431798e3d24054f
Details: e1ec78e27e..60cc82f9b7/tools/clang/scripts/update.py

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

Change-Id: I01e8f27414daf9e225d414443a6dba582455f2a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138800
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@{#28068}

2019-05-27 10:31:27 +00:00

api

Delete the remaining ORTC interfaces.

2019-05-24 18:17:37 +00:00

audio

Add a config flag to disable the audio ALR probing request.

2019-05-23 11:23:43 +00:00

build_overrides

Remove crbug.com/904400 workaround.

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

call

Negotiation of LNTF controls instantiation of RTPSenderVideo::rtp_sequence_number_map_

2019-05-24 13:02:06 +00:00

common_audio

Add ability to play audio in circle for TestAudioDevice wav file capturer

2019-04-16 15:33:03 +00:00

common_video

Remove deprecated interface in I420BufferInterface::GetI420

2019-05-03 12:16:34 +00:00

crypto

Adding new top-level directory crypto/

2019-03-08 00:35:05 +00:00

data

Remove old data files.

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

examples

Use single argument PeerConnectionFactory factory in objc code

2019-05-21 08:20:04 +00:00

logging

Fix for crash in event log when using scenario tests.

2019-05-22 15:22:49 +00:00

media

Negotiate use of RTCP loss notification feedback (LNTF)

2019-05-24 12:44:14 +00:00

modules

VP9 screenshare: drop base layer separately

2019-05-24 15:28:02 +00:00

p2p

Delete the remaining ORTC interfaces.

2019-05-24 18:17:37 +00:00

Remove rtp_ and rtcp_packet_transport() from the RtpTransport interface.

2019-05-24 23:58:46 +00:00

resources

RNN VAD: clean-up unit tests

2019-04-29 12:55:02 +00:00

rtc_base

Cleanup of AudioAllocationSettings flags.

2019-05-24 14:14:08 +00:00

rtc_tools

Make unpack_aecdump unpack RuntimeSettings

2019-05-21 12:38:15 +00:00

sdk

Negotiate use of RTCP loss notification feedback (LNTF)

2019-05-24 12:44:14 +00:00

stats

Implement RTCMediaSourceStats and friends in standard getStats().

2019-05-22 16:03:41 +00:00

style-guide

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

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

system_wrappers

Ensure CpuInfo::DetectNumberOfCores is > 0 and thread safe.

2019-05-24 12:59:14 +00:00

test

LogToSderr by default in WebRTC tests

2019-05-24 19:39:18 +00:00

tools_webrtc

Fix chromium autoroller to parse new clang revision format

2019-05-23 17:10:04 +00:00

video

Negotiate use of RTCP loss notification feedback (LNTF)

2019-05-24 12:44:14 +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

Remove last mention of ortc from the codebase.

2019-05-25 07:28:05 +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

Ban absl::StrSplit and absl::StrJoin

2019-02-26 00:45:11 +00:00

AUTHORS

Import proto_library.gni when rtc_enable_protobuf is true

2019-02-27 09:56:42 +00:00

BUILD.gn

Delete rtc_base/unittest_main.cc

2019-05-21 14:44:11 +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

Move kRtpCsrcSize from common_types.h to rtp_headers.h

2019-05-10 09:43:39 +00:00

DEPS

Roll chromium_revision e1ec78e27e..60cc82f9b7 (663034:663509)

2019-05-27 10:31:27 +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

Add juberti@ to webrtc root owners

2019-05-17 18:11:58 +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

Do not run CheckNoStreamUsageIsAdded on tests.

2019-05-07 15:52:35 +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

Add peah to more audio watchlists

2019-04-26 12:45:06 +00:00

webrtc.gni

Testing no /DUNICODE assumptions with Win more configs bots.

2019-03-28 08:46:37 +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++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%