webrtc_m130

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Autoroller 29ef9f066c Roll chromium_revision 1cf758f803..29f6e2e8b6 (538618:538933)

Change log: 1cf758f803..29f6e2e8b6
Full diff: 1cf758f803..29f6e2e8b6

Changed dependencies:
* src/base: 61c367532a..32ea59df65
* src/build: 3834707f87..b1d6c28b4a
* src/ios: 836bde6a10..377cd538f5
* src/testing: 2a421a40d7..438a3381cc
* src/third_party: 8ccff6e774..78ac06f599
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/085955c567..672f6fc248
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/aa397a9110..8a42ad3cb1
* src/third_party/openmax_dl: b611996df3..63d8cf4708
* src/tools: bb161327b3..63a486a5e6
DEPS diff: 1cf758f803..29f6e2e8b6/DEPS

No update to Clang.

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

Change-Id: I54e88477c0577dca3ab22f2440ed142f0ce01935
Reviewed-on: https://webrtc-review.googlesource.com/57280
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22176}

2018-02-23 23:23:26 +00:00

api

Add application_data field(s) to RtpPacketToSend and PacketOptions.

2018-02-23 17:20:46 +00:00

audio

Separate build targets for aec3 and aec3_unittests

2018-02-23 13:16:16 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Added some margin to ramp down target in perf test.

2018-02-23 12:15:15 +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

Add support for saving local audio input to file in AppRTCMobile

2018-02-21 14:09:56 +00:00

infra

Shorten Chromium compile trybot names

2018-02-22 10:06:20 +00:00

logging

Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.

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

media

Move default thresholds from QualityScaler to encoders.

2018-02-23 13:12:36 +00:00

modules

Multiplex Codec Bug Fix: Padding Needed For H264

2018-02-23 18:15:46 +00:00

ortc

Removed new calls on RtpTransportControllerSend.

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

p2p

Create the JsepTransportController and JsepTransport2.

2018-02-23 00:13:45 +00:00

Create the JsepTransportController and JsepTransport2.

2018-02-23 00:13:45 +00:00

resources

Adding FourPeople_1280x720_30.yuv.

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

rtc_base

ClosePlatformFile() on non-Windows: Return true on success, false on failure

2018-02-22 14:18:49 +00:00

rtc_tools

Preparing for task queue in congenstion controller

2018-02-20 12:35:15 +00:00

sdk

Move default thresholds from QualityScaler to encoders.

2018-02-23 13:12:36 +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

Separate build targets for aec3 and aec3_unittests

2018-02-23 13:16:16 +00:00

tools_webrtc

Delete unused sample project code.

2018-02-20 09:51:52 +00:00

video

Move default thresholds from QualityScaler to encoders.

2018-02-23 13:12:36 +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

Add flexfec payload name to string-type conversions

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

common_types.h

Remove unused fields from VideoCodecVP8.

2018-02-09 15:55:59 +00:00

DEPS

Roll chromium_revision 1cf758f803..29f6e2e8b6 (538618:538933)

2018-02-23 23:23:26 +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 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%