platform-external-webrtc

Go to file

Mirko Bonadei 64cf731ce4 Roll chromium_revision 2c98648a24..37c4da4be1 (538114:538199)

This CL also includes a fix in: webrtc/test/android/AndroidManifest.xml.

Change log: 2c98648a24..37c4da4be1
Full diff: 2c98648a24..37c4da4be1

Changed dependencies:
* src/base: ed313e8c6c..6afa983e37
* src/build: b734510a01..1e64514e9a
* src/ios: d48cc0d3d6..1f2dde49c3
* src/testing: 7e6cab0619..b4bd3e1fde
* src/third_party: bcdd2c72a7..d5ab621035
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6a7c1ed24c..3994526859
* src/tools: 22c4d769bf..0f9e34ac82
DEPS diff: 2c98648a24..37c4da4be1/DEPS

No update to Clang.

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

Forward fixing the Chromium Roll.

Change-Id: If36b97067fa43dc13f43e85fca706d0b5526c3d6
Reviewed-on: https://webrtc-review.googlesource.com/56640
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22148}

2018-02-22 13:58:58 +00:00

api

Removed Die mock from MockAudioEncoder

2018-02-22 12:53:38 +00:00

audio

Moved bitrate configuration to rtp controller

2018-02-21 13:55:16 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Moved bitrate config interface from Call class.

2018-02-21 15:03:45 +00:00

common_audio

Add decibel conversion functions to //common_audio:common_audio

2018-02-16 10:46:48 +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

Moved bitrate config interface from Call class.

2018-02-21 15:03:45 +00:00

modules

Removed Die mock from MockAudioEncoder

2018-02-22 12:53:38 +00:00

ortc

Removed new calls on RtpTransportControllerSend.

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

p2p

Collect packet loss and RTT stats of STUN binding requests.

2018-02-21 00:49:26 +00:00

Forward the SignalNetworkRouteChanged from DtlsSrtpTransport to BaseChannel.

2018-02-21 19:18:19 +00:00

resources

Adding FourPeople_1280x720_30.yuv.

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

rtc_base

Collect packet loss and RTT stats of STUN binding requests.

2018-02-21 00:49:26 +00:00

rtc_tools

Preparing for task queue in congenstion controller

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

sdk

Update JavaI420Buffer.allocate to use native allocations.

2018-02-21 08:40:26 +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

Roll chromium_revision 2c98648a24..37c4da4be1 (538114:538199)

2018-02-22 13:58:58 +00:00

tools_webrtc

Delete unused sample project code.

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

video

Revert "Reduce locking in VideoReceiver and check the threading model."

2018-02-21 19:37:21 +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 2c98648a24..37c4da4be1 (538114:538199)

2018-02-22 13:58:58 +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

Revert "Whitespace change."

2018-02-17 22:47:22 +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++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%