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chromium-webrtc-autoroll 4c1086a36a Roll chromium_revision 374f209d46..7356cb8842 (732255:732376) 
Change log: 374f209d46..7356cb8842
Full diff: 374f209d46..7356cb8842

Changed dependencies
* src/build: 1bee638a8c..ac1042c847
* src/buildtools: 1f38b432e5..40fcea44ce
* src/ios: 6630b3ea39..2ec14af0a4
* src/testing: d9f629bf24..beaff39b06
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/f7d73bb520..e5b0598dee
* src/third_party/depot_tools: 2a04803267..e9730d75a0
* src/third_party/googletest/src: 5395345ca4..306f3754a7
* src/tools: 8c1a06706f..362b864204
DEPS diff: 374f209d46..7356cb8842/DEPS

No update to Clang.

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

Change-Id: I8eb66419ade3aeddac5c28c81e5fb0555b3ac875
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166280
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@{#30282}
2020-01-16 14:48:25 +00:00
api
Add processing time to VideoFrame
2020-01-16 14:11:15 +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
Remove higher_spatial_layers from RTPVideoHeader structure as unused.
2020-01-16 11:11:39 +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
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
modules
Add processing time to VideoFrame
2020-01-16 14:11:15 +00:00
p2p
Improve DTLS logging.
2020-01-16 12:28:11 +00:00
pc
Revert "Replaces SynchronousMethodCall with rtc::Thread::Invoke."
2020-01-15 12:34:35 +00:00
resources
Make the high-pass filter operate in full-band
2019-12-18 16:01:24 +00:00
rtc_base
Change rate stats classes to use int64_t not size_t
2020-01-15 13:46:38 +00:00
rtc_tools
Add data dependency to event_log_visualizer.
2020-01-15 09:21:10 +00:00
sdk
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
stats
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
style-guide
…
system_wrappers
Add directive to make webrtc metrics optional.
2019-12-09 13:55:50 +00:00
test
Make it possible to inject IceTransport in pc quality test fixture
2020-01-16 11:56:50 +00:00
tools_webrtc
[UBSan] Remove suppression for opus.
2019-12-10 08:59:30 +00:00
video
Move DegradationPreference logic to the encoder queue.
2020-01-15 11:58:04 +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
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 374f209d46..7356cb8842 (732255:732376)
2020-01-16 14:48:25 +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
Extract an interface from the perf results logger.
2020-01-14 06:05:02 +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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