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chromium-webrtc-autoroll bb6677709b Roll chromium_revision b581de5b1b..210e790756 (730447:730612)

Change log: b581de5b1b..210e790756
Full diff: b581de5b1b..210e790756

Changed dependencies
* src/buildtools/linux64: git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5..git_revision:0c5557d173ce217cea095086a9c9610068123503
* src/buildtools/mac: git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5..git_revision:0c5557d173ce217cea095086a9c9610068123503
* src/buildtools/win: git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5..git_revision:0c5557d173ce217cea095086a9c9610068123503
* src/third_party/depot_tools: 7a8bf94894..5e96ad12ac
DEPS diff: b581de5b1b..210e790756/DEPS

No update to Clang.

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

Change-Id: I39e364b2aec00a902f8d665716c36e1fd48385da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165820
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@{#30239}

2020-01-13 18:47:00 +00:00

api

Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs""""

2020-01-13 09:03:37 +00:00

audio

Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api

2020-01-13 18:31:30 +00:00

build_overrides

Remove crbug.com/904400 workaround.

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

call

Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api

2020-01-13 18:31:30 +00:00

common_audio

Remove potential real-time reallocation in PushResampler

2019-12-11 13:16:37 +00:00

common_video

Add ability to resize buffers pool in decoder and use it in IVF generator

2019-12-16 14:51:16 +00:00

data

Remove old data files.

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

docs

Update Linux documentation links

2020-01-10 07:51:46 +00:00

examples

Cleanup: Replace MessageQueue pointers with Thread pointers.

2020-01-10 19:03:12 +00:00

logging

Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_.

2019-11-29 09:45:50 +00:00

media

Add saza@ and peah@ to OWNERS of some audio files

2020-01-13 12:31:21 +00:00

modules

Delete RtpGenericDepacketizer as no longer used

2020-01-13 13:45:37 +00:00

p2p

Avoid [[nodiscard]] warning C4834 with MSVC 2019

2020-01-13 10:07:35 +00:00

Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs""""

2020-01-13 09:03:37 +00:00

resources

Make the high-pass filter operate in full-band

2019-12-18 16:01:24 +00:00

rtc_base

Cleanup: Merges Thread and MessageQueue.

2020-01-13 13:53:20 +00:00

rtc_tools

[Android] Replace java_files with sources

2020-01-02 20:08:20 +00:00

sdk

[iOS] Reset VT session when H264 decoder malfunction error happen

2020-01-13 14:57:36 +00:00

stats

Implement RTCOutboundRtpStreamStats::remoteId.

2020-01-07 17:26:01 +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 directive to make webrtc metrics optional.

2019-12-09 13:55:50 +00:00

test

Delete RtpGenericDepacketizer as no longer used

2020-01-13 13:45:37 +00:00

tools_webrtc

[UBSan] Remove suppression for opus.

2019-12-10 08:59:30 +00:00

video

Move DegradationPreference logic out of VideoSourceSinkController.

2020-01-13 17:24:48 +00:00

.clang-format

…

.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

Switch to compiling WebRTC -std=c++14 by default

2019-09-09 19:24:16 +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

Update Android camera switch API to allow specifying a name

2020-01-09 16:04:09 +00:00

BUILD.gn

[Android] Replace java_files with sources

2020-01-02 20:08:20 +00:00

CODE_OF_CONDUCT.md

…

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 b581de5b1b..210e790756 (730447:730612)

2020-01-13 18:47:00 +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

…

license_template.txt

…

native-api.md

Delete unused I420 "codec"

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

OWNERS

Add #COMPONENT to WebRTC.

2019-10-08 12:20:39 +00:00

PATENTS

…

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

Make RTCAudioSession accessible to Swift.

2020-01-08 09:15:25 +00:00

pylintrc

Fixing py lint errors

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

README.chromium

…

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

Add saza to audio watchlists

2019-09-03 14:55:43 +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

Do not disable metrics by default.

2019-12-11 08:08:58 +00:00

whitespace.txt

Revert "Whitespace change"

2019-11-11 14:58:20 +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%