webrtc_m130

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Autoroller b46c552dcd Roll chromium_revision c005394adc..92107f0efb (541159:541279)

Change log: c005394adc..92107f0efb
Full diff: c005394adc..92107f0efb

Changed dependencies:
* src/base: a5a1a585d8..a53e288e7e
* src/build: 02dbfaae98..ca739dc873
* src/ios: 7c93f49550..9e39e1dc9a
* src/testing: 9d8f30c6ce..dc30e2eb78
* src/third_party: e515f9e2f1..e34b3d3468
* src/third_party/ffmpeg: ef99a5d252..4468d4967f
* src/third_party/googletest/src: fe1144246e..703b4a85a2
* src/third_party/libvpx/source/libvpx: edc9a46876..c6fcb9bb94
* src/tools: f267240af1..2ddd4aba2e
DEPS diff: c005394adc..92107f0efb/DEPS

No update to Clang.

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

Change-Id: I8f06baec60003ef294bfe32785e6b82e70c720d3
Reviewed-on: https://webrtc-review.googlesource.com/60443
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22319}

2018-03-07 02:13:12 +00:00

api

Changes name of RtpTransceiverInit's stream_labels to stream_ids.

2018-03-06 23:42:01 +00:00

audio

Production code: Pass codec ID argument to audio codecs

2018-03-05 18:55:19 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Reland "Add hugeFramesSent GetStats metric"

2018-03-06 13:38:11 +00:00

common_audio

Conditionally include real_fourier_openmax.h.

2018-02-23 09:02:36 +00:00

common_video

Drop dependency of common_video on api:libjingle_peerconnection_api.

2018-02-19 13:20:24 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Update the android AppRTC to use PeerConnection Unified Plan API.

2018-03-07 01:56:01 +00:00

infra

Removing android_compile_mips_dbg from CQ.

2018-03-01 09:13:33 +00:00

logging

Add extra RTC_CHECKs in RtcEventLog.

2018-02-27 13:23:49 +00:00

media

Reland "Add hugeFramesSent GetStats metric"

2018-03-06 13:38:11 +00:00

modules

Separating the AEC3 suppressor gain rampup behavior for call startup and in-call resets

2018-03-06 15:48:41 +00:00

ortc

Removed new calls on RtpTransportControllerSend.

2018-02-22 12:54:43 +00:00

p2p

Fix bugs in collecting STUN candidate stats and configuring STUN

2018-03-04 21:37:21 +00:00

Changes name of RtpTransceiverInit's stream_labels to stream_ids.

2018-03-06 23:42:01 +00:00

resources

Adding FourPeople_1280x720_30.yuv.

2018-02-12 15:55:00 +00:00

rtc_base

Fixing data race on vptr of Thread subclasses.

2018-03-05 20:09:20 +00:00

rtc_tools

Update prebuilt AppRTC

2018-03-02 12:16:29 +00:00

sdk

Update the android AppRTC to use PeerConnection Unified Plan API.

2018-03-07 01:56:01 +00:00

stats

Reland "Add hugeFramesSent GetStats metric"

2018-03-06 13:38:11 +00:00

system_wrappers

Fixing issues that affect newest versions of NDK.

2018-02-27 10:30:43 +00:00

test

Delete the VideoCodec::plName string.

2018-03-06 11:17:41 +00:00

tools_webrtc

Removing Android32 MIPS from mb_config.pyl.

2018-03-02 13:11:59 +00:00

video

Reland "Add hugeFramesSent GetStats metric"

2018-03-06 13:38:11 +00:00

.clang-format

Tune ObjC clang-format configuration

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

.git-blame-ignore-revs

Create .git-blame-ignore-revs and add Java format CL to it.

2016-10-20 09:20:39 +00:00

.gitignore

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

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

.gn

Re-enabling 'gn check' on //examples/*.

2018-02-19 15:07:45 +00:00

.vpython

vpython: Specify dependency on pywin32

2018-02-14 13:56:39 +00:00

AUTHORS

Expose a link-local network interfaces enumeration option

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

BUILD.gn

Removing definition of FEATURE_ENABLE_VOICEMAIL.

2018-02-19 15:51:24 +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

Delete the VideoCodec::plName string.

2018-03-06 11:17:41 +00:00

common_types.h

Delete the VideoCodec::plName string.

2018-03-06 11:17:41 +00:00

DEPS

Roll chromium_revision c005394adc..92107f0efb (541159:541279)

2018-03-07 02:13:12 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

Update template to follow chromium copyright style

2013-04-24 01:01:28 +00:00

LICENSE_THIRD_PARTY

Remove custom MD5 / SHA-1 implementations.

2018-02-19 15:03:35 +00:00

native-api.md

Remove legacy VoiceEngine.

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

OWNERS

Add back OWNERS entris that went missing

2018-03-06 10:31:11 +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

Removing obsolete defines.

2018-02-19 14:35:45 +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%