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chromium-webrtc-autoroll 9c63ef91da Roll chromium_revision c87d451712..49b7dc5e37 (840164:840329) 
Change log: c87d451712..49b7dc5e37
Full diff: c87d451712..49b7dc5e37

Changed dependencies
* src/base: bbb32a2f11..0907756181
* src/build: 656d17b1d7..505140eae0
* src/ios: 0f85ceb84f..380661c264
* src/testing: af3b7c2f1b..213e8ef125
* src/third_party: 711835b5a0..90ef10a78c
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2c541cdf00..de6e3a9d85
* src/third_party/perfetto: 88ee6a6370..f9a1b53d28
* src/tools: bb2a7cbfcd..6b0e60c77e
DEPS diff: c87d451712..49b7dc5e37/DEPS

No update to Clang.

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

Change-Id: Ief71791f58564fa71e7226873a06db9183fdc1b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200666
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@{#32907}
2021-01-05 23:20:49 +00:00
api
Make the JsepSessionDesription clone() method copy candidates.
2021-01-05 12:55:24 +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 (2021-01-05T04:04:14).
2021-01-05 06:04:58 +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
Remove old data files.
2018-10-05 14:40:21 +00:00
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
Use the correct function name in the RTC log output.
2020-12-30 11:48:31 +00:00
modules
Organize iLBC headers as per style guide
2020-12-31 20:57:18 +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 preprocessor support for additional architectures
2020-12-31 21:14:27 +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 a field trial to skip REMB modification of BWE internal state.
2020-12-21 10:49:06 +00:00
tools_webrtc
Make RTC_OBJC_EXPORT respect is_component_build.
2020-12-17 15:52:46 +00:00
video
Add support for VideoLayersAllocation for Vp9 scv/ksvc and none scalable
2020-12-30 16:45:03 +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
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
DEPS
Roll chromium_revision c87d451712..49b7dc5e37 (840164:840329)
2021-01-05 23:20:49 +00:00
ENG_REVIEW_OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
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
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
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
Remove benwright@webrtc.org from WATCHLISTS
2020-01-31 18:46:52 +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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