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chromium-webrtc-autoroll 2534e2cbf3 Roll chromium_revision 54b8c1d099..82995245b5 (797178:797326)

Change log: 54b8c1d099..82995245b5
Full diff: 54b8c1d099..82995245b5

Changed dependencies
* src/base: d08d95a8bd..fab17b7040
* src/build: 960c227c19..21db6d80b0
* src/ios: b0b7693530..f76eb471ee
* src/testing: e3ac1706db..3930b5745c
* src/third_party: fb8c70e76a..b2981bba93
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d56f379356..9f3ce85bab
* src/third_party/depot_tools: 65eb98d6ed..0e99b9be0a
* src/third_party/perfetto: c4451adcce..dfa133b7b4
* src/tools: ae2d09af5d..dfb7d15e35
DEPS diff: 54b8c1d099..82995245b5/DEPS

No update to Clang.

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

Change-Id: Ie959612ddf5c63878f73e4224cdc73fb26fed675
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181449
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@{#31923}

2020-08-12 18:40:23 +00:00

api

Delete legacy MockEncodedImageCallback::OnEncodedFrame

2020-08-12 14:41:00 +00:00

audio

Added Error Checking in Ingress/Egress and Extra Unit Tests

2020-08-06 20:48:13 +00:00

build_overrides

set perfetto flag to default value of false

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

call

Stop mentioning RTPFragmentationHeader in call/

2020-08-12 13:59:38 +00:00

common_audio

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

2020-07-30 17:35:30 +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

Search and replace gendered terms according to style guide:

2020-06-12 14:12:54 +00:00

examples

Disable switch camera and route change buttons when processing

2020-08-11 13:18:39 +00:00

logging

Add missing tests for DTLS state logging in RTC event log.

2020-08-07 11:49:43 +00:00

media

Move H264::Profile to h264_profile_level_id.h

2020-08-12 12:10:24 +00:00

modules

Delete legacy MockEncodedImageCallback::OnEncodedFrame

2020-08-12 14:41:00 +00:00

p2p

Delete left-over TODO comment

2020-08-07 11:36:51 +00:00

Reland "peerconnection: prefer spec names for signaling state"

2020-08-11 15:44:00 +00:00

resources

iSAC API wrapper unit test fix

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

rtc_base

Update settings for balanced degradation.

2020-08-10 10:59:17 +00:00

rtc_tools

Use non-deprecated EncodedImageCallback::OnEncodedImage in rtc_tools

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

sdk

Reland "Implement transceiver.stop()"

2020-08-11 10:46:23 +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

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

2020-07-30 17:35:30 +00:00

test

Prepare for deleting VideoCodec::plType

2020-08-11 14:20:59 +00:00

tools_webrtc

Fix clang revision regexp in chromium autoroller to match new format

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

video

[Adaptation] Don't toggle EncoderUsageResource on/off

2020-08-11 15:05:10 +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

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

2020-07-30 17:35:30 +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

Move H264::Profile to h264_profile_level_id.h

2020-08-12 12:10:24 +00:00

DEPS

Roll chromium_revision 54b8c1d099..82995245b5 (797178:797326)

2020-08-12 18:40:23 +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

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

2020-07-30 17:35:30 +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++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%