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chromium-webrtc-autoroll 52c3688f08 Roll chromium_revision 7c60285921..201c5e601d (733079:733179) 
Change log: 7c60285921..201c5e601d
Full diff: 7c60285921..201c5e601d

Changed dependencies
* src/base: ddbea9e33a..9d612519b6
* src/build: 119626479c..ddc21b674f
* src/testing: e3cdbcefa1..bf24920780
* src/third_party: b386f4a2e0..e76d79501f
* src/third_party/depot_tools: 98cfac11bd..b24ca5ac74
* src/tools: 0ada6e217f..a1ac6cc4e1
DEPS diff: 7c60285921..201c5e601d/DEPS

No update to Clang.

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

Change-Id: If88108d79132622697e2fbbba774ec21193f4f82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166558
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@{#30313}
2020-01-19 02:18:09 +00:00
api
Move Offset constants from VideoSendTiming value to VideoTimingExtension class
2020-01-17 15:57:38 +00:00
audio
Cleanup: Using RtpRtcp directly from AudioSendStream
2020-01-15 19:01:50 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
ResourceAdaptationModule HasInputVideo+DegradationPreference setters.
2020-01-17 10:43:02 +00:00
common_audio
Remove potential real-time reallocation in PushResampler
2019-12-11 13:16:37 +00:00
common_video
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Update Linux documentation links
2020-01-10 07:51:46 +00:00
examples
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
logging
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
media
Revert "Distinguish between send and receive codecs"
2020-01-17 02:47:23 +00:00
modules
Make MouseCursorMonitor optional for DesktopAndCursorComposer.
2020-01-18 01:28:22 +00:00
p2p
Improve DTLS logging.
2020-01-16 12:28:11 +00:00
pc
Fix race condition around rtc::ScopedFakeClock.
2020-01-17 18:55:29 +00:00
resources
Make the high-pass filter operate in full-band
2019-12-18 16:01:24 +00:00
rtc_base
Do not allow sending tasks to a thread that is quitting.
2020-01-17 13:49:27 +00:00
rtc_tools
Revert "Enable using a custom NetEqFactory in simulations"
2020-01-16 22:56:21 +00:00
sdk
Update RTCAudioSession isInterrupted state when audio session is activated while interrupted.
2020-01-17 16:12:28 +00:00
stats
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Add directive to make webrtc metrics optional.
2019-12-09 13:55:50 +00:00
test
Adds current thread to yielders in SimulatedThread::SendTask.
2020-01-17 15:08:53 +00:00
tools_webrtc
Repurpose upload script to read natively generated histogram json.
2020-01-17 13:19:07 +00:00
video
ResourceAdaptationModule HasInputVideo+DegradationPreference setters.
2020-01-17 10:43:02 +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
Update Android camera switch API to allow specifying a name
2020-01-09 16:04:09 +00:00
BUILD.gn
Suppress C5041 constexpr warning for MSVC 2019
2020-01-14 07:44:35 +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 7c60285921..201c5e601d (733079:733179)
2020-01-19 02:18:09 +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
…
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
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +00:00
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
Extract an interface from the perf results logger.
2020-01-14 06:05:02 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
…
README.md
Tell users where they can find the native API headers
2017-11-14 10:36:46 +00:00
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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