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chromium-webrtc-autoroll 2225c061f4 Roll chromium_revision 9f21b695c1..ae812cd84f (699240:699366) 
Change log: 9f21b695c1..ae812cd84f
Full diff: 9f21b695c1..ae812cd84f

Changed dependencies
* src/base: f0ff2fdcdc..48123426be
* src/build: 8162badf95..2885ef8046
* src/ios: 29dff84a5e..0969b164cd
* src/testing: 808431ec17..817467ec61
* src/third_party: 2ace733e40..d5c12817e0
* src/tools: c5b43503fa..8144dc3f1d
DEPS diff: 9f21b695c1..ae812cd84f/DEPS

No update to Clang.

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

Change-Id: I535b470073518c952ae3001bfdc8c872cdd12be1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154362
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@{#29289}
2019-09-24 16:42:32 +00:00
api
Replacing bandwidth adaptation trial with stable target in Opus encoder.
2019-09-24 16:35:02 +00:00
audio
Propagating TargetRate struct to BitrateAllocator.
2019-09-19 14:03:04 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Replacing bandwidth adaptation trial with stable target in Opus encoder.
2019-09-24 16:35:02 +00:00
common_audio
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
common_video
Always pass arguments to INSTANTIATE_TEST_SUITE_P.
2019-09-24 08:56:24 +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
New build target api:media_interface
2019-09-19 09:32:27 +00:00
logging
Adds abs-send-time and size field outputs to event log parser.
2019-09-24 16:31:22 +00:00
media
Remove obsolete todo comment in simulcast.h
2019-09-20 14:17:20 +00:00
modules
Replacing bandwidth adaptation trial with stable target in Opus encoder.
2019-09-24 16:35:02 +00:00
p2p
Delete the BasicPortAllocator constructor that enables gturn
2019-09-24 07:46:28 +00:00
pc
Always pass arguments to INSTANTIATE_TEST_SUITE_P.
2019-09-24 08:56:24 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
Handle macro _M_ARM64 for MSVC build
2019-09-24 08:34:04 +00:00
rtc_tools
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
sdk
Add release callback and reference count to java EncodedImage class
2019-09-24 12:26:09 +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
Replacing bandwidth adaptation trial with stable target in Opus encoder.
2019-09-24 16:35:02 +00:00
tools_webrtc
Stop explicitly setting use_prebuilt_instrumented_libraries on msan bots.
2019-09-12 18:21:38 +00:00
video
Always pass arguments to INSTANTIATE_TEST_SUITE_P.
2019-09-24 08:56:24 +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 9f21b695c1..ae812cd84f (699240:699366)
2019-09-24 16:42:32 +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%