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chromium-webrtc-autoroll 2f4354e903 Roll chromium_revision bdc89d87bf..fbf280c2d2 (706176:706299) 
Change log: bdc89d87bf..fbf280c2d2
Full diff: bdc89d87bf..fbf280c2d2

Changed dependencies
* src/base: a11f823e69..27a911ca78
* src/build: 33811bfefc..082f11b299
* src/ios: 32861351b8..b5d54be944
* src/testing: 6805ea7896..29537de4e5
* src/third_party: dff4de38f7..1f7528c826
* src/tools: a617ae5842..18ed58d5b7
DEPS diff: bdc89d87bf..fbf280c2d2/DEPS

No update to Clang.

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

Change-Id: Ibf4b64f934470ceae7177f4d5e005c8030e5e8e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157180
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@{#29497}
2019-10-16 04:41:52 +00:00
api
AEC3: Add support for logging warnings on delay buffer changes
2019-10-15 12:16:51 +00:00
audio
Cleanup: Replacing set extension status bool with CHECK.
2019-10-15 12:55:46 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender
2019-10-15 14:40:48 +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.
2019-09-30 12:52:07 +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
Cleanup includes in modules/include/module_common_types.h
2019-10-07 16:06:26 +00:00
media
Update the header extensions capabilities with mid, rid and rrid
2019-10-15 14:45:58 +00:00
modules
Replace RtpPacketizerH264::Fragment struct with rtc::ArrayView
2019-10-15 16:14:21 +00:00
p2p
Add a turn port prune policy to keep the first ready turn port.
2019-10-14 19:08:23 +00:00
pc
Update the header extensions capabilities with mid, rid and rrid
2019-10-15 14:45:58 +00:00
resources
Delete voice_detection() pointer to submodule
2019-10-07 13:06:05 +00:00
rtc_base
Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest
2019-10-15 09:17:36 +00:00
rtc_tools
Plot delay feedback in RTCP arrival order.
2019-10-11 13:52:10 +00:00
sdk
Add a turn port prune policy to keep the first ready turn port.
2019-10-14 19:08:23 +00:00
stats
Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
2019-10-15 10:43:59 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Move code related to VideoCodingModule to its own build target
2019-09-10 12:34:38 +00:00
test
Put the resources_dir flag into its own target.
2019-10-15 11:45:20 +00:00
tools_webrtc
Stop using goma for MSVC bots.
2019-10-08 15:19:17 +00:00
video
Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender
2019-10-15 14:40:48 +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
Add Ramprakash Jelari to AUTHORS.
2019-10-15 08:07:26 +00:00
BUILD.gn
Add missing dependencies to the static library.
2019-10-10 08:22:59 +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
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision bdc89d87bf..fbf280c2d2 (706176:706299)
2019-10-16 04:41:52 +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 #COMPONENT to WebRTC.
2019-10-08 12:20:39 +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%