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webrtc-version-updater 0a03ed8492 Update WebRTC code version (2021-01-26T04:02:59). 
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I013252952edd3ab54a1d8947b824bec795096c6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203920
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#33070}
2021-01-26 05:34:42 +00:00
api
Add option to attach custom object to an rtp packet
2021-01-25 18:31:34 +00:00
audio
ChannelStatistics RTT test case around remote SSRC change.
2021-01-19 19:07:47 +00:00
build_overrides
[build] Remove obsolete gn flag
2021-01-11 17:57:44 +00:00
call
Update WebRTC code version (2021-01-26T04:02:59).
2021-01-26 05:34:42 +00:00
common_audio
Implement a Neon optimized function to find the argmax element in an array.
2021-01-21 13:42:34 +00:00
common_video
Delete unused VideoType enum values
2021-01-14 11:45:37 +00:00
data
…
docs
doc: update working-with-native-branches information
2020-12-16 09:02:46 +00:00
examples
Reland "Refactor rtc_base build targets."
2021-01-15 17:00:05 +00:00
logging
Cleanup FakeRtcEventLog from thread awareness
2021-01-20 14:06:47 +00:00
media
Add resolution_bitrate_limits to EncoderInfo field trial.
2021-01-21 07:53:57 +00:00
modules
Add option to attach custom object to an rtp packet
2021-01-25 18:31:34 +00:00
p2p
Reland "Refactor rtc_base build targets."
2021-01-15 17:00:05 +00:00
pc
Monitor the "concealed samples" stat for the audio during negotiation.
2021-01-22 13:18:05 +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
Enable RTC_NO_UNIQUE_ADDRESS on MSan builds.
2021-01-23 14:07:28 +00:00
rtc_tools
Reland "Refactor rtc_base build targets."
2021-01-15 17:00:05 +00:00
sdk
Reset IO thread checker when iOS audio unit stops
2021-01-23 10:22:58 +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
Introduce network emulated endpoint optional name for better logging
2021-01-21 18:48:04 +00:00
tools_webrtc
mb: Fully remove references to 'masters' in favor of 'builder_groups'.
2021-01-14 22:41:08 +00:00
video
Restrict usage of resolution bitrate limits to singlecast
2021-01-22 21:41:00 +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 .cache to .gitignore.
2021-01-20 15:01:07 +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
Prefix HAVE_SCTP macro with WEBRTC_.
2021-01-20 10:51:07 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
DEPS
Roll chromium_revision da24822732..18311e2720 (844357:844473)
2021-01-18 11:18:49 +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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