platform-external-webrtc

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Björn Terelius 81bbd7199a Roll chromium_revision 88f89d7a7d..4d0815dcb4 (793251:793545)

Remove (unused) androidx dependency

Change log: 88f89d7a7d..4d0815dcb4
Full diff: 88f89d7a7d..4d0815dcb4

Changed dependencies
* src/base: c560562564..143f2bf322
* src/build: f1439400c6..af38c5df81
* src/buildtools: 9e121212d4..42e9461958
* src/ios: 6f6045fc51..52cafcd064
* src/testing: c0f4b658c2..5501fd11e6
* src/third_party: 280bd1b343..cbac85ebcf
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/70fee17204..74161f485b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/0138877b75..26d2e8aa28
* src/third_party/perfetto: 23b75a4d65..f6c424bb59
* src/tools: 63b96f52fa..1e881a2fda
DEPS diff: 88f89d7a7d..4d0815dcb4/DEPS

No update to Clang.

BUG=None

Change-Id: I65b2f55d56dd20a06e1889325591994aab6996fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180660
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31819}

2020-07-31 10:44:13 +00:00

api

Revert "Pass NetworkMonitorFactory through PeerConnectionFactory."

2020-07-31 07:34:42 +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 "Removed VideoDecoderFactory::LegacyCreateVideoDecoder and VideoReceiveStream::Config::stream_id."

2020-07-27 13:55:00 +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

Reland "sdp: parse and serialize b=TIAS"

2020-07-30 21:16:08 +00:00

modules

Removed FrameBuffer::Start function.

2020-07-30 11:15:56 +00:00

p2p

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

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

Revert "Pass NetworkMonitorFactory through PeerConnectionFactory."

2020-07-31 07:34:42 +00:00

resources

iSAC API wrapper unit test fix

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

rtc_base

Revert "Pass NetworkMonitorFactory through PeerConnectionFactory."

2020-07-31 07:34:42 +00:00

rtc_tools

Migrate rtc_tools/, sdk/, and system_wrappers/ to webrtc::Mutex.

2020-07-10 08:02:05 +00:00

sdk

Revert "Pass NetworkMonitorFactory through PeerConnectionFactory."

2020-07-31 07:34:42 +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

Add ability to get network stats from endpoint instance

2020-07-29 08:18:04 +00:00

tools_webrtc

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

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

video

Removed FrameBuffer::Start function.

2020-07-30 11:15:56 +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

Explicitly set presentation style to full screen, because,

2020-07-17 07:15:36 +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 88f89d7a7d..4d0815dcb4 (793251:793545)

2020-07-31 10:44:13 +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++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%