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chromium-webrtc-autoroll 9142cfdfde Roll chromium_revision b133b38aa4..ada5164b5b (794854:794977)

Change log: b133b38aa4..ada5164b5b
Full diff: b133b38aa4..ada5164b5b

Changed dependencies
* src/build: 2e781420a9..725c95091a
* src/testing: 165367ff94..36e4d4e9a5
* src/third_party: a4b288cba7..c36638882d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ac60992d41..d84b21c3f0
* src/third_party/perfetto: f8ccdc63a1..23c5b63d1c
* src/tools: 4def294a52..16f8e073d7
* src/tools/luci-go: git_revision:56ae79476e3caf14da59d75118408aa778637936..git_revision:b022173f8069cf8001d4cf2a87ce7c5f0eae220f
* src/tools/luci-go: git_revision:56ae79476e3caf14da59d75118408aa778637936..git_revision:b022173f8069cf8001d4cf2a87ce7c5f0eae220f
* src/tools/luci-go: git_revision:56ae79476e3caf14da59d75118408aa778637936..git_revision:b022173f8069cf8001d4cf2a87ce7c5f0eae220f
DEPS diff: b133b38aa4..ada5164b5b/DEPS

No update to Clang.

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

Change-Id: I527bc580053fa80ffbfa090aad7d40caf2fa8dab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181020
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@{#31859}

2020-08-05 13:28:56 +00:00

api

Reland "Add EncodedImageCallback::OnEncodedImage without RTPFragmentationHeader"

2020-08-04 12:17:06 +00:00

audio

Remove unused critical section includes.

2020-07-16 13:52:28 +00:00

build_overrides

set perfetto flag to default value of false

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

call

Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."

2020-08-03 15:45:41 +00:00

common_audio

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

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

common_video

Ignore RTPFragmentationHeader when rewriting H264 SPS

2020-07-27 09:42:25 +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

Implemented Android Demo Application for VoIP API

2020-07-21 16:34:22 +00:00

logging

Search and replace gendered terms according to style guide:

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

media

Reducing threshold for usrsctp "buffer low" callback.

2020-08-04 20:08:06 +00:00

modules

Delete deprecated RTPSenderVideo::SendVideo function

2020-08-05 13:10:36 +00:00

p2p

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

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

sdp: reject sdp with malformed b= lines

2020-08-03 18:21:14 +00:00

resources

iSAC API wrapper unit test fix

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

rtc_base

Delete old name ProcessAllMessageQueues

2020-08-03 11:24:17 +00:00

rtc_tools

Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."

2020-08-03 15:45:41 +00:00

sdk

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

stats

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

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

style-guide

…

system_wrappers

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

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

test

Add networks stats collector from PeerConnection GetStats API

2020-08-04 09:05:14 +00:00

tools_webrtc

Fix "Assignment had no effect." error during chromium roll.

2020-07-30 12:57:20 +00:00

video

Reland "Only enable conference mode simulcast allocations with flag enabled"

2020-08-04 10:30:08 +00:00

.clang-format

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.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

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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

Replaces OverheadObserver with simple getter.

2020-05-07 17:33:45 +00:00

DEPS

Roll chromium_revision b133b38aa4..ada5164b5b (794854:794977)

2020-08-05 13:28:56 +00:00

ENG_REVIEW_OWNERS

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LICENSE

…

license_template.txt

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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

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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

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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%