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chromium-webrtc-autoroll ad0be281a4 Roll chromium_revision e9fe59463e..78ea2c1bec (841903:842019) 
Change log: e9fe59463e..78ea2c1bec
Full diff: e9fe59463e..78ea2c1bec

Changed dependencies
* src/base: 319011ddce..096849d4ea
* src/build: d1a74631f6..8874a0a063
* src/ios: 852ea28117..280ea2c55b
* src/testing: a7771bfba5..b6a5362054
* src/third_party: e8fdf5d0a4..6144d06a12
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e174329bab..12607ec3c9
* src/third_party/perfetto: 35f78f4a45..eddc7d0df5
* src/tools: 5410766644..2bc54343d8
DEPS diff: e9fe59463e..78ea2c1bec/DEPS

No update to Clang.

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

Change-Id: I4407b1ee4ea72aede90ecf466557679cae2f2775
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201136
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@{#32937}
2021-01-11 15:13:01 +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 (2021-01-11T04:01:55).
2021-01-11 06:01:03 +00:00
common_audio
Organize iLBC headers as per style guide
2020-12-31 20:57:18 +00:00
common_video
Delete deprecated H264BitstreamParser methods
2021-01-11 09:31:54 +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
Inclusive language: Remove a couple of occurences of "whitelist"
2021-01-11 07:53:03 +00:00
media
Use rtc::CopyOnWriteBuffer::MutableData through webrtc
2021-01-11 11:31:33 +00:00
modules
Use rtc::CopyOnWriteBuffer::MutableData through webrtc
2021-01-11 11:31:33 +00:00
p2p
Use rtc::CopyOnWriteBuffer::MutableData through webrtc
2021-01-11 11:31:33 +00:00
pc
Use rtc::CopyOnWriteBuffer::MutableData through webrtc
2021-01-11 11:31:33 +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
Inclusive language: Remove a couple of occurences of "whitelist"
2021-01-11 07:53:03 +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 deprecated H264BitstreamParser methods
2021-01-11 09:31:54 +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
Use rtc::CopyOnWriteBuffer::MutableData through webrtc
2021-01-11 11:31:33 +00:00
tools_webrtc
Make RTC_OBJC_EXPORT respect is_component_build.
2020-12-17 15:52:46 +00:00
video
Do not proxy VideoSendStreamImpl::OnVideoLayersAllocationUpdated
2021-01-11 11:23:13 +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
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 e9fe59463e..78ea2c1bec (841903:842019)
2021-01-11 15:13:01 +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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C 2.2%
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