webrtc_m130

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chromium-webrtc-autoroll 1251e4a69a Roll chromium_revision 6bc2e35371..b23796061a (755432:755766)

Change log: 6bc2e35371..b23796061a
Full diff: 6bc2e35371..b23796061a

Changed dependencies
* src/base: fb5957fcc5..2f0acc1163
* src/build: d01d1b6d2f..37c7abd9fc
* src/ios: 31d9e19a8d..435d839123
* src/testing: 0f0c80137e..80630cdbeb
* src/third_party: 1cf8966df1..0d8a8640d3
* src/third_party/depot_tools: 4f30cf0179..9db428f4f7
* src/third_party/harfbuzz-ng/src: 558f922788..014e038b2c
* src/third_party/icu: d7aff76cf6..13cfcd5874
* src/third_party/libvpx/source/libvpx: 5532775efe..667138e1f0
* src/tools: 6cd5df917b..a6d086006b
DEPS diff: 6bc2e35371..b23796061a/DEPS

No update to Clang.

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

Change-Id: I2c3d1a76aeb869224b6fdfc1b12d9a8cf9d1b477
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172704
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@{#30975}

2020-04-02 11:10:58 +00:00

api

Transform received audio frames in ChannelReceive.

2020-04-01 11:23:00 +00:00

audio

Transform received audio frames in ChannelReceive.

2020-04-01 11:23:00 +00:00

build_overrides

Purge phoglund from most OWNERS files.

2020-03-09 14:08:30 +00:00

call

Insert audio frame transformer between depacketizer and decoder.

2020-04-01 08:15:53 +00:00

common_audio

Changed fft4g to be built as C++

2020-03-20 12:10:16 +00:00

common_video

Remove wildcard ownership for build files.

2020-02-19 14:05:46 +00:00

data

Remove old data files.

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

docs

doc: describe native turnserver scope more clearly

2020-04-01 19:48:20 +00:00

examples

Chromium refactor: Replace "resources_dirs" with "sources"

2020-03-25 19:32:15 +00:00

logging

Replace DataSize and DataRate factories with newer versions

2020-02-18 16:09:50 +00:00

media

Add AV1 to default video encoder factory

2020-04-02 10:13:22 +00:00

modules

Delete deprecated RtpFrameObject constructor

2020-04-02 10:50:57 +00:00

p2p

Extend rtc::AdapterType with 2g, 3G, 4G & 5G enum values.

2020-04-02 07:48:36 +00:00

Extend rtc::AdapterType with 2g, 3G, 4G & 5G enum values.

2020-04-02 07:48:36 +00:00

resources

iSAC API wrapper unit test fix

2020-02-27 14:27:23 +00:00

rtc_base

Fix errors C2238, C2248 and C2059 on MSVC bots.

2020-04-02 09:54:27 +00:00

rtc_tools

Total packet rate plots for event_log_visualizer.

2020-04-01 13:14:24 +00:00

sdk

Add AV1 to default video encoder factory

2020-04-02 10:13:22 +00:00

stats

Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API

2020-03-11 12:08:32 +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 RTC_EXPORT to webrtc::Clock

2020-03-10 23:00:21 +00:00

test

Remove the histogram flag and all Chart JSON code.

2020-03-28 13:44:43 +00:00

tools_webrtc

Fix android_arm_rel on Chromium Roll.

2020-04-01 17:50:18 +00:00

video

[InsertableStreams] Set video frame transformer if RTP stream already started.

2020-03-31 14:07:29 +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 .clangd to .gitignore

2019-10-28 12:27:50 +00:00

.gn

Reenable libaom decoder by default

2020-03-18 18:04:41 +00:00

.vpython

Add source-side perf upload script for WebRTC.

2019-11-18 14:37:01 +00:00

abseil-in-webrtc.md

Fix typo in abseil-in-webrtc.md.

2019-12-18 14:27:34 +00:00

AUTHORS

Fix bad frees in error paths of WebRtcAecm_Create

2020-03-21 23:01:47 +00:00

BUILD.gn

Audio egress implementation for initial voip api in api/voip.

2020-03-27 18:45:43 +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

Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"

2020-02-07 08:23:58 +00:00

DEPS

Roll chromium_revision 6bc2e35371..b23796061a (755432:755766)

2020-04-02 11:10:58 +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

Remove phoglund as root owner.

2020-03-30 12:15:56 +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

Use source_sets in component builds and static_library in release builds.

2019-10-17 21:17:18 +00:00

PRESUBMIT.py

Fix public_deps presubmit and gn format fighting each other.

2020-01-30 11:22:46 +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

Add guidance to style guide how to reference a bug in a TODO

2019-12-11 11:55:52 +00:00

WATCHLISTS

Remove benwright@webrtc.org from WATCHLISTS

2020-01-31 18:46:52 +00:00

webrtc_lib_link_test.cc

Rewrite the lib link test to just be a binary.

2019-10-18 07:42:20 +00:00

webrtc.gni

Reformat GN files.

2020-01-21 12:13:11 +00:00

whitespace.txt

Whitespace change

2020-01-28 07:53:15 +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%