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chromium-webrtc-autoroll 101750ac5b Roll chromium_revision 5f083e73a6..fa6a961223 (785559:785707) 
Change log: 5f083e73a6..fa6a961223
Full diff: 5f083e73a6..fa6a961223

Changed dependencies
* src/base: 8aeffef0ee..92109190e6
* src/build: 729634b87d..3bd002c3db
* src/ios: d3245aa303..d40932a520
* src/testing: a711b11325..8933687c5e
* src/third_party: f8c6d72cc0..f955149bca
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9ef36d0c0f..1b843e7903
* src/third_party/depot_tools: 81923d6d2c..ce1375c5f0
* src/third_party/ffmpeg: 388ba367aa..15f6cfec63
* src/tools: ee10cd4775..d09ab91c2a
DEPS diff: 5f083e73a6..fa6a961223/DEPS

No update to Clang.

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

Change-Id: Ia073e02e55a7980c154adc7ff12784c929d43746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178772
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@{#31650}
2020-07-07 15:27:46 +00:00
api
Implement packets_(sent | received) for RTCTransportStats
2020-07-07 10:45:05 +00:00
audio
webrtc::AudioSendStream: Add lock annotation to audio_level_
2020-07-06 17:05:25 +00:00
build_overrides
Rename sanitizers suppression files.
2020-07-03 18:29:58 +00:00
call
[Adaptation] Remove deg.prefs from ResourceAdaptationProcessor
2020-07-07 08:37:05 +00:00
common_audio
Replace slave -> helper, master -> reference
2020-06-29 12:18:05 +00:00
common_video
Migrate common_video/ and examples/ to webrtc::Mutex.
2020-07-07 13:33:27 +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
Migrate common_video/ and examples/ to webrtc::Mutex.
2020-07-07 13:33:27 +00:00
logging
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
media
Migrate media/ to webrtc::Mutex.
2020-07-07 13:46:47 +00:00
modules
Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex.
2020-07-07 14:35:58 +00:00
p2p
Implement packets_(sent | received) for RTCTransportStats
2020-07-07 10:45:05 +00:00
pc
Implement packets_(sent | received) for RTCTransportStats
2020-07-07 10:45:05 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Inclusive language in //rtc_base.
2020-07-06 10:39:39 +00:00
rtc_tools
Roll chromium_revision 4d95e6c77b..71a0e1904e (776481:782339)
2020-06-26 05:33:14 +00:00
sdk
Convert GN libs lists to frameworks
2020-07-06 10:08:09 +00:00
stats
Implement packets_(sent | received) for RTCTransportStats
2020-07-07 10:45:05 +00:00
style-guide
…
system_wrappers
Use absl_deps in order to preapre to the Abseil component build release.
2020-06-08 12:59:40 +00:00
test
Add simulated time support for PC level test.
2020-07-06 21:18:00 +00:00
tools_webrtc
Rename sanitizers suppression files.
2020-07-03 18:29:58 +00:00
video
[Adaptation] Remove deg.prefs from ResourceAdaptationProcessor
2020-07-07 08:37:05 +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
Relanding: Fix data channel message integrity violation
2020-07-07 03:06:24 +00:00
BUILD.gn
Convert GN libs lists to frameworks
2020-07-06 10:08:09 +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 5f083e73a6..fa6a961223 (785559:785707)
2020-07-07 15:27:46 +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
…
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
…
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
Inclusive language in PRESUBMIT.py.
2020-06-30 14:03:10 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
…
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
Build: Disable the iLBC audio codec by default
2020-07-01 08:52:35 +00:00
whitespace.txt
Trigger bots.
2020-06-15 08:40:44 +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%