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chromium-webrtc-autoroll 54c03266f7 Roll chromium_revision a42eacf137..78591f12ff (692288:692389)

Change log: a42eacf137..78591f12ff
Full diff: a42eacf137..78591f12ff

Changed dependencies
* src/build: 5f1456d718..5dd17829f4
* src/third_party: d2680ce0c3..57d158d40f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9e1c92c073..9f64c5cb49
* src/third_party/depot_tools: 17016be940..355e97e300
* src/tools: 9f3ef015d3..ea54c5157c
DEPS diff: a42eacf137..78591f12ff/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I9ddd9da7fd658294d8841a401f3e4deb61901c29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151145
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29032}

2019-09-02 02:37:14 +00:00

api

Revert "Add core multi-channel pipeline in AEC3"

2019-08-30 10:19:29 +00:00

audio

Delete unused logic for audio RtcpMode::kOff

2019-08-30 10:35:06 +00:00

build_overrides

Remove crbug.com/904400 workaround.

2019-03-15 18:36:23 +00:00

call

Delete audio methods SignalNetworkState

2019-08-30 09:27:30 +00:00

common_audio

Remove rtc_use_lto GN arg.

2019-08-20 14:00:49 +00:00

common_video

Add new FrameRateEstimator utility class for more precis FPS estimation.

2019-08-14 12:15:06 +00:00

data

Remove old data files.

2018-10-05 14:40:21 +00:00

examples

Make the RtpHeaderParserImpl available to tests and tools only.

2019-08-29 15:56:40 +00:00

logging

Delete unneeded dependencies on libjingle_peerconnection_api

2019-08-29 10:52:42 +00:00

media

Make the RtpHeaderParserImpl available to tests and tools only.

2019-08-29 15:56:40 +00:00

modules

Filter out small packets from delay-based overuse detection.

2019-09-01 17:57:01 +00:00

p2p

Adding CreateTcpClientSocket without user_agent and proxy_info.

2019-08-30 14:21:52 +00:00

Revert "Refactor SCTP data channels to use DataChannelTransportInterface."

2019-08-30 12:31:21 +00:00

resources

Use the AEC3 high-pass filter for the whole APM

2019-08-23 20:04:10 +00:00

rtc_base

Add support for unsigned parameters in FieldTrialParser

2019-08-30 08:23:37 +00:00

rtc_tools

Increased event log visualizer RTP clock estimation tolerance.

2019-08-29 10:08:43 +00:00

sdk

Interface for monitoring ref counts of texture buffers created by SurfaceTextureHelper.

2019-08-30 10:36:11 +00:00

stats

Add support for RTCTransportStats.selectedCandidatePairChanges

2019-08-28 13:22:08 +00:00

style-guide

Add style guide rule about paired .h and .cc files

2018-03-14 13:02:35 +00:00

system_wrappers

Add helper functions to convert between integer milliseconds and fixed-point seconds.

2019-08-16 14:49:46 +00:00

test

Revert "Refactor SCTP data channels to use DataChannelTransportInterface."

2019-08-30 12:31:21 +00:00

tools_webrtc

Add license for android_ndk

2019-08-28 14:57:28 +00:00

video

Make the RtpHeaderParserImpl available to tests and tools only.

2019-08-29 15:56:40 +00:00

.clang-format

Tune ObjC clang-format configuration

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

.git-blame-ignore-revs

Let git-hyper-blame ignore new format cleanup.

2019-07-11 16:18:51 +00:00

.gitignore

Add Visual Studio Code project folder to gitignore file.

2019-01-21 18:42:33 +00:00

.gn

Remove last mention of ortc from the codebase.

2019-05-25 07:28:05 +00:00

.vpython

Add vpython dependencies needed to run presubmit tests on LUCI

2018-05-18 08:10:25 +00:00

abseil-in-webrtc.md

Allowing buffering a LNTF (loss notification) feedback message in RTCPSender

2019-06-03 16:28:34 +00:00

AUTHORS

Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly

2019-06-28 19:12:14 +00:00

BUILD.gn

Make the RtpHeaderParserImpl available to tests and tools only.

2019-08-29 15:56:40 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

2017-05-16 12:09:13 +00:00

codereview.settings

Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit

2018-10-25 08:19:53 +00:00

common_types.h

Format almost everything.

2019-07-08 13:45:15 +00:00

DEPS

Roll chromium_revision a42eacf137..78591f12ff (692288:692389)

2019-09-02 02:37:14 +00:00

ENG_REVIEW_OWNERS

Enforce LGTM from owners of depends-on paths in DEPS via presubmit.

2018-09-28 12:49:54 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

native-api.md

Delete unused I420 "codec"

2018-12-18 12:30:58 +00:00

OWNERS

Add juberti@ to webrtc root owners

2019-05-17 18:11:58 +00:00

PATENTS

Moving src/webrtc into src/.

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

presubmit_test_mocks.py

Reland: Add presubmit check for changes in 3pp

2018-05-22 13:11:18 +00:00

presubmit_test.py

Fixing py lint errors

2018-07-23 15:28:48 +00:00

PRESUBMIT.py

Disable RunPythonTests on rtc_tools.

2019-08-06 12:48:33 +00:00

pylintrc

Fixing py lint errors

2018-07-23 15:28:48 +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

Remove rule that discourages passing optional by const reference

2019-02-05 11:58:05 +00:00

WATCHLISTS

Remove myself from OWNERS in a few places.

2019-06-10 07:57:46 +00:00

webrtc.gni

Remove rtc_use_lto GN arg.

2019-08-20 14:00:49 +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%