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chromium-webrtc-autoroll bd24260791 Roll chromium_revision eae7ecf757..afdb2e7a8b (697744:697871) 
Change log: eae7ecf757..afdb2e7a8b
Full diff: eae7ecf757..afdb2e7a8b

Changed dependencies
* src/base: 01355d2940..3fe4a418db
* src/build: 71ed08eea6..050608ea95
* src/ios: e12f797b06..797d97f55d
* src/testing: 74e29479c8..c223ceca68
* src/third_party: a939fa9d82..d7037a0728
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/cbc960899a..6a17fd7fb7
* src/third_party/depot_tools: 3f79763629..c6be56eced
* src/tools: 8c0c370d66..efcee32728
DEPS diff: eae7ecf757..afdb2e7a8b/DEPS

No update to Clang.

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

Change-Id: Ia0e447ae02cf1a0a9a1e1a49deca4762b486fd81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153620
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@{#29229}
2019-09-19 02:33:30 +00:00
api
Using struct for bitrate allocation limits.
2019-09-18 17:03:59 +00:00
audio
Using struct for bitrate allocation limits.
2019-09-18 17:03:59 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Using struct for bitrate allocation limits.
2019-09-18 17:03:59 +00:00
common_audio
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
common_video
Cleaning up C++14 move into lambda TODOs.
2019-09-17 19:18:26 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Fixing some typos.
2019-09-10 10:03:50 +00:00
examples
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
logging
Cleaning up C++14 move into lambda TODOs.
2019-09-17 19:18:26 +00:00
media
Only handle each RTCP once.
2019-09-18 16:54:39 +00:00
modules
Removed unused include from PacketBuffer.
2019-09-18 15:35:59 +00:00
p2p
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
pc
Only handle each RTCP once.
2019-09-18 16:54:39 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
Introduce integer division helpers with non-default rounding
2019-09-18 15:08:39 +00:00
rtc_tools
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
sdk
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
stats
Add qualityLimitationResolutionChanges stat
2019-09-09 15:22:57 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Move code related to VideoCodingModule to its own build target
2019-09-10 12:34:38 +00:00
test
Only handle each RTCP once.
2019-09-18 16:54:39 +00:00
tools_webrtc
Stop explicitly setting use_prebuilt_instrumented_libraries on msan bots.
2019-09-12 18:21:38 +00:00
video
Revert "Fix minor regression caused by a8336d3"
2019-09-18 14:38:15 +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
Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH.
2019-09-04 18:49:28 +00:00
.gn
Switch to compiling WebRTC -std=c++14 by default
2019-09-09 19:24:16 +00:00
.vpython
Add vpython dependencies needed to run presubmit tests on LUCI
2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md
Update style guide for absl::make_unique.
2019-09-18 06:10:58 +00:00
AUTHORS
Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5."
2019-09-06 05:36:23 +00:00
BUILD.gn
Introduce api/crypto/BUILD.gn.
2019-09-13 17:21:47 +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
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision eae7ecf757..afdb2e7a8b (697744:697871)
2019-09-19 02:33:30 +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
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Add juberti@ to webrtc root owners
2019-05-17 18:11:58 +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
Fixing py lint errors
2018-07-23 15:28:48 +00:00
PRESUBMIT.py
absl::make_unique presubmit check.
2019-09-17 17:47:31 +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
Tell users where they can find the native API headers
2017-11-14 10:36:46 +00:00
style-guide.md
Update WebRTC's C++ style guide to reflect the switch to C++14.
2019-09-16 11:45:35 +00:00
WATCHLISTS
Add saza to audio watchlists
2019-09-03 14:55:43 +00:00
webrtc.gni
Remove rtc_use_lto GN arg.
2019-08-20 14:00:49 +00:00
whitespace.txt
Whitespace change
2018-02-23 10:34:16 +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 http://www.webrtc.org/native-code/development 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%