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chromium-webrtc-autoroll 71e9acb97c Roll chromium_revision 8ffd72401d..d8b1c2f5b2 (768712:770095) 
Change log: 8ffd72401d..d8b1c2f5b2
Full diff: 8ffd72401d..d8b1c2f5b2

Changed dependencies
* src/base: e6c0c5b9ad..9cea79a086
* src/build: f70e3b9685..13e29491b1
* src/buildtools: 204a35a2a6..64ebbe62e1
* src/buildtools/linux64: git_revision:5ed3c9cc67b090d5e311e4bd2aba072173e82db9..git_revision:ab32747ae7a399c57b04280f38e49b8fdf237a8a
* src/buildtools/mac: git_revision:5ed3c9cc67b090d5e311e4bd2aba072173e82db9..git_revision:ab32747ae7a399c57b04280f38e49b8fdf237a8a
* src/buildtools/win: git_revision:5ed3c9cc67b090d5e311e4bd2aba072173e82db9..git_revision:ab32747ae7a399c57b04280f38e49b8fdf237a8a
* src/ios: 7b694bd936..98854a7c6f
* src/testing: 5a5fb44e80..04bfc27905
* src/third_party: 57686d64cb..7f6ed0a92c
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/a810d82575..78b3337a10
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/087cffcba4..503f81b8fe
* src/third_party/depot_tools: 30ef5cb437..5a7be3da21
* src/third_party/ffmpeg: 587a3f4849..be66dc5fd0
* src/third_party/freetype/src: 3f70e6d20c..62fea391fa
* src/third_party/libvpx/source/libvpx: 77960f37b3..1243d2fc27
* src/tools: b64de32dc9..2414245236
* src/tools/swarming_client: 160b445a44..7e8636295f
DEPS diff: 8ffd72401d..d8b1c2f5b2/DEPS

No update to Clang.

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

Change-Id: I88dde47fb99213d2007b0dd1e6ee332bab53b584
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175720
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@{#31326}
2020-05-19 14:38:12 +00:00
api
Add ability for audioproc_f to operate on any AudioProcessing object.
2020-05-19 11:37:18 +00:00
audio
Rename several more tests that use EXPECT_DEATH to *DeathTest.
2020-05-18 16:10:04 +00:00
build_overrides
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
call
Rename several more tests that use EXPECT_DEATH to *DeathTest.
2020-05-18 16:10:04 +00:00
common_audio
Rename several more tests that use EXPECT_DEATH to *DeathTest.
2020-05-18 16:10:04 +00:00
common_video
In GenericFrameInfo do not set frame_id and frame_diff
2020-05-15 09:43:34 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
docs: Improve Android testing docs
2020-05-14 08:05:37 +00:00
examples
Remove android_manifest_for_lint from BUILD.gn files
2020-05-19 13:23:25 +00:00
logging
Reduce alert spam in rtc_event_log_visualizer
2020-05-19 13:22:20 +00:00
media
VideoAdapter: remove lock recursions.
2020-05-19 13:09:51 +00:00
modules
Log decoder implementation name
2020-05-19 12:23:30 +00:00
p2p
Clear address:port in icecandidateerror for tcp servers with private IP
2020-05-15 11:30:20 +00:00
pc
FakeAudioCaptureModule: remove lock recursions.
2020-05-18 18:01:58 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Add support for injecting Clock to RepeatingTaskHandle.
2020-05-18 14:11:44 +00:00
rtc_tools
Remove android_manifest_for_lint from BUILD.gn files
2020-05-19 13:23:25 +00:00
sdk
Pass datachannel priority in DC open messages
2020-05-17 10:57:27 +00:00
stats
Reland "Improve outbound-rtp statistics for simulcast"
2020-05-05 20:22:19 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Do not trigger -Wimplicit-int-float-conversion
2020-04-09 11:43:14 +00:00
test
FakeEncoder: remove lock recursions.
2020-05-19 08:57:15 +00:00
tools_webrtc
Fix the name of the file generated by generate_sslroots.py.
2020-04-25 13:37:30 +00:00
video
Reland "RtpVideoStreamReceiver::RtcpFeedbackBuffer: remove lock recursions."
2020-05-19 13:04:20 +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
Fix typo in abseil-in-webrtc.md.
2019-12-18 14:27:34 +00:00
AUTHORS
docs: Improve Android testing docs
2020-05-14 08:05:37 +00:00
BUILD.gn
VoIP API implementation on top of AudioIngress/Egress
2020-05-05 19:55:29 +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
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
DEPS
Roll chromium_revision 8ffd72401d..d8b1c2f5b2 (768712:770095)
2020-05-19 14:38:12 +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
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
Fix public_deps presubmit and gn format fighting each other.
2020-01-30 11:22:46 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
README.md
Fix link in documentation. (take 2)
2020-04-16 11:08:43 +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
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
Remove android_manifest_for_lint from BUILD.gn files
2020-05-19 13:23:25 +00:00
whitespace.txt
Trigger bots again.
2020-05-11 05:38:59 +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

Description
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%