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Autoroller c54f706993 Roll chromium_revision ecf8a6133e..a88423acf9 (569618:571410) 
Change log: ecf8a6133e..a88423acf9
Full diff: ecf8a6133e..a88423acf9

Changed dependencies:
* src/base: f7595e419a..311c937b26
* src/build: 69593eb8fa..c9333f9faf
* src/buildtools: 5941c1b3df..9c9fd97928
* src/ios: 181b18c878..34302909a8
* src/testing: 8354b28f74..b47e929d27
* src/third_party: 46683344d7..b77d94a9b3
* src/third_party/android_tools: https://chromium.googlesource.com/android_tools.git/+log/3545ab5b98..130499e252
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/6ff2ba80b7..fec83fc78d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/87eefd4f11..e2d6bbca62
* src/third_party/depot_tools: c5a26a769e..ae1f03388f
* src/third_party/freetype/src: 7915fd51f1..a632fb547e
* src/third_party/libvpx/source/libvpx: 8648a64c83..583859d739
* src/third_party/libyuv: bc383e76d6..4d67b3e851
* src/third_party/r8: 1.0.30..1.2.28-cr0
* src/third_party/usrsctp/usrsctplib: 159d060dce..7a8bc9a90c
* src/tools: 592ddd1d14..6ff0d88db8
DEPS diff: ecf8a6133e..a88423acf9/DEPS

Clang version changed 334100:335608
Details: ecf8a6133e..a88423acf9/tools/clang/scripts/update.py

TBR=buildbot@webrtc.org,marpan@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: If3229d875265bca1bffffd01a793098ad2106f9f
Reviewed-on: https://webrtc-review.googlesource.com/86240
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23779}
2018-06-29 07:33:16 +00:00
api
Add PeerConnection GetRtpSender/ReceiverCapabilities
2018-06-28 12:40:07 +00:00
audio
Remove StreamStatistician::IsPacketInOrder
2018-06-28 08:44:40 +00:00
build_overrides
Add phoglund@ to various OWNERS and remove kjellander@
2017-10-19 09:21:12 +00:00
call
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
2018-06-27 10:33:40 +00:00
common_audio
Extract fft4g into separate build target
2018-06-26 13:39:25 +00:00
common_video
Add bit depth information to PlanarYuvBuffer
2018-06-26 20:23:37 +00:00
data
…
examples
Replace rtc::Optional with absl::optional
2018-06-21 09:32:56 +00:00
infra
Flip luci.webrtc.try to production
2018-05-30 08:30:00 +00:00
logging
Port RtcEventLog encoder unittests to the new parser API.
2018-06-28 15:31:23 +00:00
media
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
2018-06-27 10:33:40 +00:00
modules
AEC3: Allow filter adaptation even though the estimated echo is saturated
2018-06-28 22:45:18 +00:00
ortc
Add PeerConnection GetRtpSender/ReceiverCapabilities
2018-06-28 12:40:07 +00:00
p2p
Remove the flag PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS.
2018-06-27 00:43:09 +00:00
pc
Enable PeerConnectionEndToEndTest.CallWithLegacySdp on ASan.
2018-06-29 05:00:46 +00:00
resources
AGC2 RNN VAD: Polishing.
2018-05-15 16:41:02 +00:00
rtc_base
Use field trial parser for BBR Experiment.
2018-06-28 07:52:58 +00:00
rtc_tools
Adding NetEq lifetime stats to event log visualizer.
2018-06-26 11:27:09 +00:00
sdk
Fixed crash when PCF is destroyed before DataChannel in ObjC
2018-06-28 12:54:22 +00:00
stats
Adding "is_standardized" flag to RTCStatsMember.
2018-06-28 00:43:46 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Use the sparse histogram in RTC_HISTOGRAM_ENUMERATION_SPARSE.
2018-06-27 00:41:29 +00:00
test
Revert "Add Timestamp accessor methods to the EncodedImage class."
2018-06-26 11:52:45 +00:00
tools_webrtc
Roll back checking in the third_party directory
2018-06-27 13:04:08 +00:00
video
Remove StreamStatistician::IsPacketInOrder
2018-06-28 08:44:40 +00:00
.clang-format
…
.git-blame-ignore-revs
Let git-hyper-blame ignore format commit.
2018-06-20 09:26:44 +00:00
.gitignore
Roll back checking in the third_party directory
2018-06-27 13:04:08 +00:00
.gn
Opt out of "Migrate the Android Support Lib to android_deps".
2018-04-05 13:40:53 +00:00
.vpython
Add vpython dependencies needed to run presubmit tests on LUCI
2018-05-18 08:10:25 +00:00
AUTHORS
Generalize SimulcastEncoderAdapter, use for H264 & VP8.
2018-06-21 15:57:43 +00:00
BUILD.gn
A new PeerConnection level perf test.
2018-06-27 23:19:05 +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
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
DEPS
Roll chromium_revision ecf8a6133e..a88423acf9 (569618:571410)
2018-06-29 07:33:16 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
…
LICENSE_THIRD_PARTY
Remove third party dependecies that are not more in the source code
2018-06-21 11:33:41 +00:00
native-api.md
Remove legacy VoiceEngine.
2018-01-12 11:31:52 +00:00
OWNERS
Add mbonadei@ to build configs OWNERS.
2018-06-20 12:39:11 +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
Roll chromium_revision 95336cb92b..191d55580e (557816:557824)
2018-05-11 11:17:05 +00:00
PRESUBMIT.py
Roll back checking in the third_party directory
2018-06-27 13:04:08 +00:00
pylintrc
Removing invalid-name from disabled pylint checks.
2017-10-11 08:06:49 +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 style guidance about forward declarations.
2018-03-28 20:58:27 +00:00
typedefs.h
Remove typedefs.h from webrtc/ root (part 1)
2018-05-23 12:07:10 +00:00
WATCHLISTS
Adding mbonadei@ to build_files WATCHLIST.
2018-06-20 12:38:06 +00:00
webrtc.gni
Don't call deprecated FFmpeg API.
2018-06-26 13:57:35 +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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