webrtc_m130

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Autoroller 12325c5b83 Roll chromium_revision 3d1e7a12b7..8b6898fe4e (517813:518117)

Change log: 3d1e7a12b7..8b6898fe4e
Full diff: 3d1e7a12b7..8b6898fe4e

Changed dependencies:
* src/base: 4c3d6272f8..b2b46604dd
* src/build: 571871636f..6167734e03
* src/ios: 865de15eaf..d6ba2ebe6e
* src/testing: 14acaa118c..c5ed5cc935
* src/third_party: d43ee09615..de6ce59aae
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/6cc352e216..8793942c5c
* src/third_party/depot_tools: ddfead3483..7227d2175a
* src/third_party/ffmpeg: 252244150a..168dfaa19b
* src/tools: bce4039f1c..2c0f60c1ad
DEPS diff: 3d1e7a12b7..8b6898fe4e/DEPS

No update to Clang.

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

Change-Id: I7115e7c9e4bc14b1fbe46554413d9f45954a590c
Reviewed-on: https://webrtc-review.googlesource.com/24760
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20807}

2017-11-21 06:13:34 +00:00

api

Add SDP semantics option to RTCConfiguration

2017-11-21 02:09:03 +00:00

audio

Optional: Use nullopt and implicit construction in /audio

2017-11-17 15:56:17 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Add stats googHasEnteredLowResolution.

2017-11-17 13:02:07 +00:00

common_audio

Fix circular deps in common_audio.

2017-11-17 11:20:17 +00:00

common_video

In generic encoder ensure capture timestamp is preserved.

2017-11-17 16:06:17 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Android: Fix warnings

2017-11-16 18:25:33 +00:00

infra

Remove linux_memcheck from commit queue

2017-11-17 17:52:47 +00:00

logging

Revert "Encode log events periodically instead of for every event."

2017-11-17 21:02:02 +00:00

media

Created the DtlsSrtpTransport.

2017-11-20 23:18:22 +00:00

modules

Implement Opus bandwidth adjustment behind a FieldTrial

2017-11-20 20:04:19 +00:00

ortc

Fix ortc_api circular deps.

2017-11-15 13:31:51 +00:00

p2p

Created the DtlsSrtpTransport.

2017-11-20 23:18:22 +00:00

Add SDP semantics option to RTCConfiguration

2017-11-21 02:09:03 +00:00

resources

Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process

2017-10-23 14:25:37 +00:00

rtc_base

Add performance tests for rtc::Event.

2017-11-20 16:09:59 +00:00

rtc_tools

Fix some missed stdlib includes

2017-11-16 08:50:44 +00:00

sdk

Android: Generate JNI code for VideoDecoder

2017-11-20 22:48:02 +00:00

stats

Added RTCMediaStreamTrackStats.jitterBufferDelay for audio

2017-10-02 10:47:00 +00:00

system_wrappers

Reland of "Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator""

2017-11-16 12:10:04 +00:00

test

On-fly calculation of quality metrics.

2017-11-20 16:13:59 +00:00

tools_webrtc

[webrtc] Add mb configs for new bots in 2 internal masters

2017-11-17 09:08:11 +00:00

video

Reland "Update internal encoder factory to new interface"

2017-11-18 10:48:28 +00:00

voice_engine

Revert "Encode log events periodically instead of for every event."

2017-11-17 21:02:02 +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

Roll chromium_revision f93b8b19f2..adf969a7cb (513366:514871) and more.

2017-11-08 21:42:48 +00:00

.gn

Remove remaining mentions of gflags

2017-09-25 15:34:41 +00:00

.vpython

Add psutil to vpython dependencies (used on builder bots)

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

AUTHORS

Delete wrapper API ConvertToI420 for YUV conversion to I420

2017-11-15 11:10:20 +00:00

BUILD.gn

Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)."

2017-09-29 13:48:29 +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

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

common_types.h

Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled.

2017-10-06 13:41:14 +00:00

DEPS

Roll chromium_revision 3d1e7a12b7..8b6898fe4e (517813:518117)

2017-11-21 06:13:34 +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

Tell users where they can find the native API headers

2017-11-14 10:36:46 +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

Tell users where they can find the native API headers

2017-11-14 10:36:46 +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

Style guide: Attempt to make the L2 and L3 headings more visually distinct

2017-09-09 03:52:23 +00:00

typedefs.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

WATCHLISTS

Add myself to the watchlist for webrtc/api/ and webrtc/base/

2017-05-04 13:22:46 +00:00

webrtc.gni

Reland: "Make javac warnings errors for WebRTC targets."

2017-11-07 15:36:46 +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

Languages

C++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%