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chromium-webrtc-autoroll f2dc05978f Roll chromium_revision 6f7e5e79ce..d814fc7ea9 (727038:727531) 
Change log: 6f7e5e79ce..d814fc7ea9
Full diff: 6f7e5e79ce..d814fc7ea9

Changed dependencies
* src/base: 96567c8f45..de91707f3e
* src/build: 8b4f17ac5f..69e421a3a6
* src/ios: 108bcc40ba..f230393662
* src/testing: 5a36a5534b..240d660c68
* src/third_party: 9056dff38d..533ae73635
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/efb804d60c..189074525f
* src/tools: 41fb035d80..c83a8ed9c4
DEPS diff: 6f7e5e79ce..d814fc7ea9/DEPS

No update to Clang.

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

Change-Id: I23a84892cc04d70b24a7906c8e1c61f5fb5667c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163960
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@{#30126}
2019-12-27 04:39:05 +00:00
api
Revert "Reland "Reland "Distinguish between send and receive video codecs"""
2019-12-20 13:57:12 +00:00
audio
Minor fixes to ChannelSend.
2019-12-02 09:30:51 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Revert "Extracts ssrc based feedback tracking from feedback adapter."
2019-12-13 14:47:48 +00:00
common_audio
Remove potential real-time reallocation in PushResampler
2019-12-11 13:16:37 +00:00
common_video
Add ability to resize buffers pool in decoder and use it in IVF generator
2019-12-16 14:51:16 +00:00
data
…
docs
Refresh some links in the docs folder.
2019-12-10 15:32:48 +00:00
examples
Fix errorprone issues preventing Chromium Roll.
2019-11-27 12:52:48 +00:00
logging
Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_.
2019-11-29 09:45:50 +00:00
media
Revert "Reland "Reland "Distinguish between send and receive video codecs"""
2019-12-20 13:57:12 +00:00
modules
Remove unused parameter in RtpFragmentize
2019-12-20 11:22:33 +00:00
p2p
Stop an SCTP connection when the DTLS transport closes.
2019-12-16 14:54:56 +00:00
pc
Revert "Reland "Reland "Distinguish between send and receive video codecs"""
2019-12-20 13:57:12 +00:00
resources
Make the high-pass filter operate in full-band
2019-12-18 16:01:24 +00:00
rtc_base
Use a fake clock for rtc::Thread::PostDelayedTask test
2019-12-16 15:46:48 +00:00
rtc_tools
Move audioproc_f to rtc_tools.
2019-12-10 12:08:56 +00:00
sdk
Add video codec AV1 to the deprecated android decoder/encoder wrappers
2019-12-23 09:29:46 +00:00
stats
Add totalInterFrameDelay to RTCInboundRTPStreamStats
2019-11-25 10:50:37 +00:00
style-guide
…
system_wrappers
Add directive to make webrtc metrics optional.
2019-12-09 13:55:50 +00:00
test
Don't allow creation of sockets for wild card IPs in emulated networks.
2019-12-17 10:16:59 +00:00
tools_webrtc
[UBSan] Remove suppression for opus.
2019-12-10 08:59:30 +00:00
video
Signal requested resolution alignment requirements from sinks to sources.
2019-12-19 10:39:04 +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
Add .clangd to .gitignore
2019-10-28 12:27:50 +00:00
.gn
Switch to compiling WebRTC -std=c++14 by default
2019-09-09 19:24:16 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md
Fix typo in abseil-in-webrtc.md.
2019-12-18 14:27:34 +00:00
AUTHORS
Fixed crash on iOS13, methods beginGeneratingDeviceOrientationNotifications and endGeneratingDeviceOrientationNotifications.
2019-12-20 12:24:46 +00:00
BUILD.gn
Initial version of ResourceAdaptationProcessor and friends.
2019-12-10 15:31:43 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
common_types.h
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision 6f7e5e79ce..d814fc7ea9 (727038:727531)
2019-12-27 04:39:05 +00:00
ENG_REVIEW_OWNERS
…
LICENSE
…
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
…
presubmit_test_mocks.py
…
presubmit_test.py
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
PRESUBMIT.py
Add possibility to skip check_includes presubmit check.
2019-10-22 19:35:31 +00:00
pylintrc
…
README.chromium
…
README.md
…
style-guide.md
Add guidance to style guide how to reference a bug in a TODO
2019-12-11 11:55:52 +00:00
WATCHLISTS
Add saza to audio watchlists
2019-09-03 14:55:43 +00:00
webrtc_lib_link_test.cc
Rewrite the lib link test to just be a binary.
2019-10-18 07:42:20 +00:00
webrtc.gni
Do not disable metrics by default.
2019-12-11 08:08:58 +00:00
whitespace.txt
Revert "Whitespace change"
2019-11-11 14:58:20 +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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