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Autoroller 9eeff94860 Roll chromium_revision d54862fccc..11cc0bafaf (595183:595285) 
Change log: d54862fccc..11cc0bafaf
Full diff: d54862fccc..11cc0bafaf

Changed dependencies
* src/base: 54ecd85c67..1c9cf1a7fb
* src/build: 943188ae3c..e76ff65158
* src/ios: 8cf6659a93..76ae1c23d5
* src/testing: 021d90ae91..2f8676e0c3
* src/third_party: fa102cd369..e93b379280
* src/tools: c2a94531bf..030931ce4d
DEPS diff: d54862fccc..11cc0bafaf/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None

Change-Id: Iff60da8e365350106681c6358944f98ce46d6bd1
Reviewed-on: https://webrtc-review.googlesource.com/102767
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24900}
2018-09-29 02:20:34 +00:00
api
Parse two-byte header extensions.
2018-09-28 08:32:17 +00:00
audio
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
2018-09-28 12:00:28 +00:00
build_overrides
Add phoglund@ to various OWNERS and remove kjellander@
2017-10-19 09:21:12 +00:00
call
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
2018-09-28 12:00:28 +00:00
common_audio
Remove simple stringstream usages.
2018-09-06 12:53:19 +00:00
common_video
Avoid overflow when parsing H.264 SPS.
2018-09-27 11:50:10 +00:00
data
WebRTC: Replace ProjectRootPath by ResourcePath
2016-11-22 18:43:05 +00:00
examples
Remove deprecated field_trial_default and metrics_default.
2018-09-28 07:21:07 +00:00
infra
Reland "Use new internal trybot in commit queue"
2018-09-25 11:02:03 +00:00
logging
Parse two-byte header extensions.
2018-09-28 08:32:17 +00:00
media
Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
2018-09-28 08:48:02 +00:00
modules
Revert "Remove APM-internal usage of EchoControlMobile"
2018-09-28 13:39:19 +00:00
ortc
Remove deprecated field_trial_default and metrics_default.
2018-09-28 07:21:07 +00:00
p2p
Remove deprecated field_trial_default and metrics_default.
2018-09-28 07:21:07 +00:00
pc
Parse two-byte header extensions.
2018-09-28 08:32:17 +00:00
resources
Drop legacy AEC metrics interface from ApmTest.Process
2018-09-14 08:16:43 +00:00
rtc_base
Delete unused HTTP server code
2018-09-28 09:48:47 +00:00
rtc_tools
Reland "Add option to call VMAF in compare_videos.py."
2018-09-28 14:08:10 +00:00
sdk
Throw exception if MediaStreamTrack is constructed with a null native track.
2018-09-28 15:01:00 +00:00
stats
Remove deprecated field_trial_default and metrics_default.
2018-09-28 07:21:07 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Remove deprecated field_trial_default and metrics_default.
2018-09-28 07:21:07 +00:00
test
Adds simulated time scenario client.
2018-09-28 12:30:44 +00:00
tools_webrtc
Autoroller: switch back to old-style "=" tags for TBR to work
2018-09-28 13:20:43 +00:00
video
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
2018-09-28 12:00:28 +00:00
.clang-format
Tune ObjC clang-format configuration
2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore format commit.
2018-06-20 09:26:44 +00:00
.gitignore
Reland "Compile frame analyzer for the host machine on perf tests."
2018-09-18 09:51:19 +00:00
.gn
Re-enable gtest absl pretty printers.
2018-08-13 13:54:05 +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
Add a list of allowed and disallowed Abseil things
2018-08-27 08:37:20 +00:00
AUTHORS
Remove unused var in device info bits from video capture module for Linux
2018-09-12 10:36:33 +00:00
BUILD.gn
Remove deprecated field_trial_default and metrics_default.
2018-09-28 07:21:07 +00:00
CODE_OF_CONDUCT.md
Add code of conduct to WebRTC repo
2017-05-16 12:09:13 +00:00
codereview.settings
Make Gerrit the default for WebRTC changes
2017-09-29 01:38:07 +00:00
common_types.h
Remove double declaration of 2 conversion functions.
2018-09-11 06:00:05 +00:00
DEPS
Roll chromium_revision d54862fccc..11cc0bafaf (595183:595285)
2018-09-29 02:20:34 +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
Add documentation about field_trial/metrics custom impl.
2018-09-18 11:27:59 +00:00
OWNERS
Add a list of allowed and disallowed Abseil things
2018-08-27 08:37:20 +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
Enforce LGTM from owners of depends-on paths in DEPS via presubmit.
2018-09-28 12:49:54 +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
Add a list of allowed and disallowed Abseil things
2018-08-27 08:37:20 +00:00
WATCHLISTS
Reland "Remove steveanton from api/ watchlist"
2018-09-05 11:45:33 +00:00
webrtc.gni
Reland "Add RTC_EXPORT macro to export WebRTC symbols."
2018-09-18 12:07:58 +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
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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