webrtc_m130

Go to file

Autoroller afe9eb33dd Roll chromium_revision b36a6c8d2c..bb5e87b52c (540587:540725)

Change log: b36a6c8d2c..bb5e87b52c
Full diff: b36a6c8d2c..bb5e87b52c

Changed dependencies:
* src/base: a97a91025e..e0b732554d
* src/build: 2b99ad9823..a81ca97020
* src/ios: abb3531752..8111fe6961
* src/testing: c117dfdc8a..0dcf0bcd14
* src/third_party: f4cfcfca97..0aeb75e705
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/8df8e64205..f8058d4114
* src/third_party/ffmpeg: 9ca66eafaf..ef99a5d252
* src/third_party/libyuv: 6630558875..98a0a157dc
* src/tools: 59edaf442d..571b1312e0
DEPS diff: b36a6c8d2c..bb5e87b52c/DEPS

No update to Clang.

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

Change-Id: Ifc77cea363948e5206ff398291423cef5cf942c7
Reviewed-on: https://webrtc-review.googlesource.com/59502
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22277}

2018-03-03 03:33:29 +00:00

api

Name change from stream label to stream id for spec compliance.

2018-03-02 20:44:48 +00:00

audio

Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 1.

2018-02-28 15:38:15 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Enables configuration of transmission max bitrate multiplier and fec protection level.

2018-03-02 18:05:29 +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 iOS AppRTC to use PeerConnection Unified Plan

2018-03-01 17:25:18 +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

Name change from stream label to stream id for spec compliance.

2018-03-02 20:44:48 +00:00

modules

Merge SimulcastStream and SpatialLayer structures.

2018-03-02 16:30:19 +00:00

ortc

Removed new calls on RtpTransportControllerSend.

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

p2p

Fixing typo in log messages.

2018-03-02 19:02:09 +00:00

Name change from stream label to stream id for spec compliance.

2018-03-02 20:44:48 +00:00

resources

Adding FourPeople_1280x720_30.yuv.

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

rtc_base

Renamed rtc::RateLimiter to rtrc::DataRateLimiter.

2018-02-28 19:58:24 +00:00

rtc_tools

Update prebuilt AppRTC

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

sdk

Updating android/README.

2018-03-01 20:22:48 +00:00

stats

Removing useless dependencies on //testing/gmock.

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

system_wrappers

Fixing issues that affect newest versions of NDK.

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

test

Template argument and corpora for Audio Processing Fuzzer.

2018-03-02 14:00:39 +00:00

tools_webrtc

Removing Android32 MIPS from mb_config.pyl.

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

video

Merge SimulcastStream and SpatialLayer structures.

2018-03-02 16:30:19 +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

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

Add flexfec payload name to string-type conversions

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

common_types.h

Merge SimulcastStream and SpatialLayer structures.

2018-03-02 16:30:19 +00:00

DEPS

Roll chromium_revision b36a6c8d2c..bb5e87b52c (540587:540725)

2018-03-03 03:33:29 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

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 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

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%