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chromium-webrtc-autoroll 783278a310 Roll chromium_revision 211a9ad899..6ae5319dfd (841507:841669) 
Change log: 211a9ad899..6ae5319dfd
Full diff: 211a9ad899..6ae5319dfd

Changed dependencies
* src/base: 1ab8a47ade..f211390a6d
* src/ios: b62dfbee82..d539e9cef3
* src/testing: 067c94e136..662c78e0ae
* src/third_party: 6560227876..2ccfc92076
* src/third_party/depot_tools: 60b433cf0b..0e4e5ae594
* src/third_party/freetype/src: 0901a6f74c..3431553836
* src/third_party/perfetto: afc78f5e6a..6112541431
* src/tools: 5f45dce5dc..9e0b107f46
DEPS diff: 211a9ad899..6ae5319dfd/DEPS

No update to Clang.

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

Change-Id: Icb7c241746259c8064f8c6f6650393ea583aac58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200927
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@{#32927}
2021-01-08 23:13:08 +00:00
api
Add ability to specify random seed when creating built it network emulation
2021-01-07 16:40:50 +00:00
audio
Using absl::optional for round trip time return type handling.
2020-12-15 08:38:12 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Update WebRTC code version (2021-01-08T04:02:26).
2021-01-08 05:48:34 +00:00
common_audio
Organize iLBC headers as per style guide
2020-12-31 20:57:18 +00:00
common_video
Reland "Rename FATAL() into RTC_FATAL()."
2020-11-18 20:49:08 +00:00
data
…
docs
doc: update working-with-native-branches information
2020-12-16 09:02:46 +00:00
examples
Enforcing return type handling on VoIP API.
2020-12-11 20:38:15 +00:00
logging
Add static constexpr type to RTC event log events.
2021-01-05 13:48:41 +00:00
media
Wire scalability_mode when simulcast is not in use (i.e. streams==1)
2021-01-08 14:32:28 +00:00
modules
SSE2 and NEON kill switches for AGC2
2021-01-07 11:41:28 +00:00
p2p
Optimize calls to std::string::find() and friends for a single char.
2020-12-16 09:01:44 +00:00
pc
Make the JsepSessionDesription clone() method copy candidates.
2021-01-05 12:55:24 +00:00
resources
Erle metric: avoid the decreasing of the metric when there is no render activity.
2020-12-09 11:48:11 +00:00
rtc_base
Add ability to specify random seed when creating built it network emulation
2021-01-07 16:40:50 +00:00
rtc_tools
Use a task queue for call interaction in video_replay tool
2020-12-14 15:25:43 +00:00
sdk
Delete unneeded dependencies on the Module abstraction
2020-12-21 09:09:57 +00:00
stats
stats: add transportId to codec stats
2020-11-17 12:34:39 +00:00
style-guide
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
system_wrappers
Delete use of RWLockWrapper from SimulatedClock
2020-11-04 08:01:08 +00:00
test
Add ability to specify random seed when creating built it network emulation
2021-01-07 16:40:50 +00:00
tools_webrtc
Make RTC_OBJC_EXPORT respect is_component_build.
2020-12-17 15:52:46 +00:00
video
Move setting of encoder bitrate allocation callback type to VideoSendStream
2021-01-07 09:29:05 +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
Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
2020-09-07 08:37:14 +00:00
.vpython
Reland "Add protobuf-py2_py3 3.13.0 to .vpython."
2020-11-20 07:52:26 +00:00
abseil-in-webrtc.md
Polish the "Using Abseil in WebRTC" docs
2020-10-16 13:42:00 +00:00
AUTHORS
Fix "control reaches end of non-void function" warnings
2020-10-27 10:22:23 +00:00
BUILD.gn
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
DEPS
Roll chromium_revision 211a9ad899..6ae5319dfd (841507:841669)
2021-01-08 23:13:08 +00:00
ENG_REVIEW_OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Make the remote_bitrate_estimator build target private
2020-11-26 12:21:22 +00:00
OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
PATENTS
…
presubmit_test_mocks.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
presubmit_test.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
PRESUBMIT.py
Fix Authors Check presubmit.
2020-12-15 11:59:03 +00:00
pylintrc
Undo enforcing of PEP-8 pylint changes for method and function names.
2020-11-10 18:26:25 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
doc: move bug reporting instructions to the repository
2020-10-21 14:47:49 +00:00
style-guide.md
C++ style: We don't allow designated initializers
2020-06-03 09:11:09 +00:00
WATCHLISTS
Add hta@ to rtc_base/ and api/ WATCHLISTS.
2021-01-06 09:43:34 +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
Get rid of recently added rtc_openssl_is_boringssl build argument.
2020-12-15 08:34:42 +00:00
whitespace.txt
Trigger bots for FYI waterfall.
2020-11-20 09:35:54 +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 here 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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