webrtc_m130

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Autoroller ecfe2e86a6 Roll chromium_revision feadf7258e..6fcad09d4a (535428:535596)

Change log: feadf7258e..6fcad09d4a
Full diff: feadf7258e..6fcad09d4a

Changed dependencies:
* src/base: 69eb3646b6..83ec53d6ec
* src/build: 19190cb080..03220ef360
* src/ios: 075f6c8518..0959011404
* src/testing: 453c6a4ddb..7d3f4563c0
* src/third_party: 10c57e38e2..efea12dde9
* src/third_party/android_tools: https://chromium.googlesource.com/android_tools.git/+log/c9f9bbf0a6..9a70d48fcd
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/7e5dd25d47..61dedd6815
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/aa41a69e72..e653c4b823
* src/third_party/depot_tools: 6a60d7586a..71236c3af6
* src/third_party/libyuv: ffec313dbe..b792e0dbc1
* src/tools: 500944dbb2..ffaf09e5ff
DEPS diff: feadf7258e..6fcad09d4a/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I7b6d4ac17cbaed4c0736a9d0cb876ee4f5c3019c
Reviewed-on: https://webrtc-review.googlesource.com/50040
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21964}

2018-02-09 01:12:44 +00:00

api

Added ptime to RtpEncodingParameters.

2018-02-07 17:33:41 +00:00

audio

Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.

2018-02-07 10:07:28 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +00:00

call

Reland "Moved congestion controller to task queue."

2018-02-06 08:38:49 +00:00

common_audio

Stop using public_deps in common_audio.

2018-02-06 09:44:20 +00:00

common_video

Fix clang style errors in rtp_rtcp and dependant targets

2018-02-07 09:48:28 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

2016-11-22 18:43:05 +00:00

examples

Update AppRTCMobile AAR-project to SDK version 27.

2018-02-07 16:22:31 +00:00

infra

Remove win_chromium_webrtc_compile_rel_ng from CQ

2018-02-01 19:26:24 +00:00

logging

Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.

2018-02-07 10:07:28 +00:00

media

Update several tests: FakeVideoCapturer -> FakeVideoCapturerWithTaskQueue.

2018-02-08 09:03:58 +00:00

modules

Remove unused field |output_filename| from TestConfig.

2018-02-08 15:23:19 +00:00

ortc

Change FakePeriodicVideoCapturer to use a TaskQueue instead of Thread.

2018-02-07 20:51:51 +00:00

p2p

Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32."

2018-02-08 16:25:31 +00:00

Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32."

2018-02-08 16:25:31 +00:00

resources

Add a new NetEq decoding unit test for Opus with DTX

2017-11-28 10:45:38 +00:00

rtc_base

Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32."

2018-02-08 16:25:31 +00:00

rtc_tools

Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.

2018-02-07 10:07:28 +00:00

sdk

Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32."

2018-02-08 16:25:31 +00:00

stats

Removing useless dependencies on //testing/gmock.

2018-01-26 13:34:12 +00:00

system_wrappers

Reland Use runtime enabled features API to enable dual stream mode

2018-01-18 12:22:49 +00:00

test

Simplify CodecSettings helper function.

2018-02-08 14:38:59 +00:00

tools_webrtc

MB: Add a way to run tests on swarming without using gtest-parallel.

2018-02-08 19:37:19 +00:00

video

Testing that media is resumed when congestion ends.

2018-02-07 15:31:26 +00:00

.clang-format

Tune ObjC clang-format configuration

2017-05-11 09:14:18 +00:00

.git-blame-ignore-revs

…

.gitignore

Reland "Make it possible to run video_quality_loopback_test in swarming."

2018-01-23 13:03:17 +00:00

.gn

Temporarily skipping examples/* from gn check.

2018-02-06 12:42:23 +00:00

.vpython

Add psutil to vpython dependencies (used on builder bots)

2017-09-04 08:04:16 +00:00

AUTHORS

Expose a link-local network interfaces enumeration option

2018-02-06 19:12:04 +00:00

BUILD.gn

Enable building WebRTC without built-in software codecs

2018-01-31 08:33:59 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

2017-05-16 12:09:13 +00:00

codereview.settings

Make Gerrit the default for WebRTC changes

2017-09-29 01:38:07 +00:00

common_types.cc

Add flexfec payload name to string-type conversions

2018-01-31 08:58:39 +00:00

common_types.h

Reland "Rename stereo video codec to multiplex"

2018-01-29 20:37:59 +00:00

DEPS

Roll chromium_revision feadf7258e..6fcad09d4a (535428:535596)

2018-02-09 01:12:44 +00:00

LICENSE

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

license_template.txt

…

LICENSE_THIRD_PARTY

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

native-api.md

Remove legacy VoiceEngine.

2018-01-12 11:31:52 +00:00

OWNERS

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +00:00

PATENTS

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

presubmit_test_mocks.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

presubmit_test.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

PRESUBMIT.py

Delete unused MediaFile module.

2018-01-29 11:18:18 +00:00

pylintrc

Removing invalid-name from disabled pylint checks.

2017-10-11 08:06:49 +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

Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"""

2018-01-10 15:55:04 +00:00

typedefs.h

Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix

2018-02-05 11:24:59 +00:00

WATCHLISTS

Move remaining traces of VoiceEngine

2018-01-17 13:27:47 +00:00

webrtc.gni

Enable building WebRTC without built-in software codecs

2018-01-31 08:33:59 +00:00

whitespace.txt

Moving src/webrtc into src/.

2017-09-15 04:25:06 +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

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Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%