platform-external-webrtc

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chromium-webrtc-autoroll 20f3f220bc Roll chromium_revision 6cc8efac3b..0c4572c29e (800536:801101)

Change log: 6cc8efac3b..0c4572c29e
Full diff: 6cc8efac3b..0c4572c29e

Changed dependencies
* src/base: c1f9e3cf41..8eb5a08461
* src/build: acef2c50d7..5e95547410
* src/buildtools: b00ad0af63..ff93f3ea1a
* src/buildtools/linux64: git_revision:e327ffdc503815916db2543ec000226a8df45163..git_revision:6f13aaac55a977e1948910942675c69f2b4f7a94
* src/buildtools/mac: git_revision:e327ffdc503815916db2543ec000226a8df45163..git_revision:6f13aaac55a977e1948910942675c69f2b4f7a94
* src/buildtools/win: git_revision:e327ffdc503815916db2543ec000226a8df45163..git_revision:6f13aaac55a977e1948910942675c69f2b4f7a94
* src/ios: 4156807865..a959a9d317
* src/testing: 86a61ed9c7..e86ac19ff1
* src/third_party: b6744f889c..e548090601
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/74161f485b..c947efabcb
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/7a2e3f8396..a54f10f750
* src/third_party/depot_tools: bfb423d266..490961030b
* src/third_party/freetype/src: f9f6adb625..cdc009c24a
* src/third_party/perfetto: 9869f187b5..746445da75
* src/third_party/r8: vvymFSkKtWKWNmfz0PL_0H8MD8V40P--A9aUfxfpF6QC..N9LppKV-9lFkp7JQtmcLHhm7xHqFv0SPa6aDPtgNCdwC
* src/tools: 32d15ee98f..fbc54ad6ff
DEPS diff: 6cc8efac3b..0c4572c29e/DEPS

No update to Clang.

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

Change-Id: I821072cc2094d9e9ca9b5c8b38dbd44b88bc4c74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182403
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@{#31984}

2020-08-24 20:23:06 +00:00

api

Delete deprecated version of EncodedImageCallback::OnEncodedImage

2020-08-24 11:00:19 +00:00

audio

DTMF Event Sub-API on VoIP API

2020-08-20 17:10:02 +00:00

build_overrides

set perfetto flag to default value of false

2020-07-22 10:14:53 +00:00

call

Revert "[Adaptation] Remove QualityScalerResource when disabled."

2020-08-17 13:46:30 +00:00

common_audio

Support AVX2/FMA intrinsics in audio FIR filter

2020-08-19 10:21:31 +00:00

common_video

Move function PrintVideoFrame to the test file where it is used.

2020-08-06 15:20:19 +00:00

data

Remove old data files.

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

docs

Updated IOS documentation with correct build_ios_libs.py path

2020-08-17 09:37:59 +00:00

examples

Moved Asynchronicity From Java to C++ for AndroidVoip Demo App

2020-08-18 17:54:56 +00:00

logging

Group decoded frame events by SSRC when compressing RTC event log.

2020-08-19 09:47:20 +00:00

media

Removing call to deprecated SetExtraOptions method

2020-08-20 16:13:12 +00:00

modules

Passing the estimated capture clock offset to SendVideo.

2020-08-24 13:31:42 +00:00

p2p

p2p_transport_channel: Add estimated disconnected time to CandidatePairChangeEvent

2020-08-20 11:40:01 +00:00

Transition ICE gathering state to "new" once all transports go away

2020-08-20 18:55:52 +00:00

resources

iSAC API wrapper unit test fix

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

rtc_base

Implement network monitor for iOS.

2020-08-20 21:46:18 +00:00

rtc_tools

Use non-deprecated EncodedImageCallback::OnEncodedImage in rtc_tools

2020-08-12 09:17:15 +00:00

sdk

Delete deprecated version of EncodedImageCallback::OnEncodedImage

2020-08-24 11:00:19 +00:00

stats

Reland "Implement packets_(sent | received) for RTCTransportStats"

2020-07-10 11:50:59 +00:00

style-guide

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

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

system_wrappers

Reland "Support AVX2/FMA intrinsics in Audio Resampler module"

2020-08-17 10:40:44 +00:00

test

Reland "Call OnReceivedOverhead after audio network adaptor is created."

2020-08-20 16:07:41 +00:00

tools_webrtc

Fix clang revision regexp in chromium autoroller to match new format

2020-08-07 09:22:03 +00:00

video

Enable Vp8VariableFramerateScreenshare by default

2020-08-21 15:03:58 +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

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

Use absl_deps in order to preapre to the Abseil component build release.

2020-06-08 12:59:40 +00:00

AUTHORS

Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder

2020-08-05 09:41:49 +00:00

BUILD.gn

Reland "Support AVX2/FMA intrinsics in Audio Resampler module"

2020-08-17 10:40:44 +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

Move definition of SpatialLayer to api/video_codecs/spatial_layer.h

2020-08-17 09:45:19 +00:00

DEPS

Roll chromium_revision 6cc8efac3b..0c4572c29e (800536:801101)

2020-08-24 20:23:06 +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

Make transient suppression optionally excludable via defines

2020-04-02 11:44:07 +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

Inclusive language in PRESUBMIT.py.

2020-07-22 10:01:23 +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

Inclusive language in PRESUBMIT.py.

2020-07-22 10:01:23 +00:00

pylintrc

Fixing py lint errors

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

README.chromium

Add CPEPrefix.

2020-07-13 11:42:07 +00:00

README.md

Fix link in documentation. (take 2)

2020-04-16 11:08:43 +00:00

style-guide.md

C++ style: We don't allow designated initializers

2020-06-03 09:11:09 +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

Reland "Support AVX2/FMA intrinsics in Audio Resampler module"

2020-08-17 10:40:44 +00:00

whitespace.txt

Trigger CI bots.

2020-07-15 17:50:55 +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 here 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%