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chromium-webrtc-autoroll 331b00bdb8 Roll chromium_revision 26d0995106..a9c1e4afb9 (720053:720171) 
Change log: 26d0995106..a9c1e4afb9
Full diff: 26d0995106..a9c1e4afb9

Changed dependencies
* src/base: c24bb21404..0759871ba8
* src/build: 586d50f811..2fc048cf25
* src/ios: b3026b1a11..0c6f3cc26c
* src/testing: a33380acd4..7fc8295e7d
* src/third_party: 2b7c0d1b0e..d4ae047bf6
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b97d4ceb42..bcfcc04c53
* src/third_party/depot_tools: 1bfe890bd8..7c62ed63ac
* src/tools: 1e52da6374..0f56390d56
DEPS diff: 26d0995106..a9c1e4afb9/DEPS

No update to Clang.

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

Change-Id: I44661670da19f07c83182277b5de99fe683997a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161067
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@{#29959}
2019-11-29 12:47:17 +00:00
api
Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_.
2019-11-29 09:45:50 +00:00
audio
Delete media transport integration.
2019-11-26 19:19:36 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Implements a task-queue based PacedSender, wires it up for field trials
2019-11-28 12:13:53 +00:00
common_audio
Add ability to disable detailed error message in RTC_CHECKs
2019-11-28 17:51:00 +00:00
common_video
Revert "VideoFrame: Store a reference to an encoded frame"
2019-11-21 14:55:21 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Fixing some typos.
2019-09-10 10:03:50 +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
Removes caching SimulcastEncoderAdapter::GetEncoderInfo()
2019-11-27 17:06:54 +00:00
modules
In RtpDepacketizerAV1 use aggregation header to detect key frames
2019-11-29 10:14:20 +00:00
p2p
Add rtt estimate EventBasedExponentialMovingAverage to Connection
2019-11-28 09:24:44 +00:00
pc
Add directive to make TRACE_EVENT macros optional.
2019-11-28 15:58:24 +00:00
resources
Changed the digital AGC1 gain to properly support multichannel
2019-11-23 08:42:59 +00:00
rtc_base
Add ability to disable detailed error message in RTC_CHECKs
2019-11-28 17:51:00 +00:00
rtc_tools
Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_.
2019-11-29 09:45:50 +00:00
sdk
Fix ErrorProne MultiVariableDeclaration.
2019-11-28 18:49:20 +00:00
stats
Add totalInterFrameDelay to RTCInboundRTPStreamStats
2019-11-25 10:50:37 +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 ability to disable detailed error message in RTC_CHECKs
2019-11-28 17:51:00 +00:00
test
Add ability to disable detailed error message in RTC_CHECKs
2019-11-28 17:51:00 +00:00
tools_webrtc
Revert "Remove temporary workaround for generate_licenses."
2019-11-27 17:56:45 +00:00
video
Propagate absolute capture time through video receiving side.
2019-11-27 17:07:59 +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 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
Update style guide for absl::make_unique.
2019-09-18 06:10:58 +00:00
AUTHORS
Force alignment of generated JVM called functions.
2019-11-21 12:34:35 +00:00
BUILD.gn
Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_.
2019-11-29 09:45:50 +00:00
CODE_OF_CONDUCT.md
Add code of conduct to WebRTC repo
2017-05-16 12:09:13 +00:00
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 26d0995106..a9c1e4afb9 (720053:720171)
2019-11-29 12:47:17 +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
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
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
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
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
Update WebRTC's C++ style guide to reflect the switch to C++14.
2019-09-16 11:45:35 +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
Add ability to disable detailed error message in RTC_CHECKs
2019-11-28 17:51:00 +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
Languages
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C 2.2%
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Python 1.3%
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