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Jaehyun Ko 75b0f5575e Replace legacy getStats with standard getStats in the Android example 
Bug: webrtc:12688
Change-Id: I7e2e10ab1b1ce994bbfbcfad377a77b39239d3d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221760
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34870}
2021-08-30 10:45:15 +00:00
api
Remove media/base/h264_profile_level_id.* and media/base/vp9_profile.h
2021-08-30 10:31:08 +00:00
audio
Wire up non-sender RTT for audio, and implement related standardized stats.
2021-08-30 09:03:50 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
2021-08-30 10:20:55 +00:00
common_audio
Use backticks not vertical bars to denote variables in comments
2021-08-10 10:40:03 +00:00
common_video
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
2021-08-30 10:20:55 +00:00
data
…
docs
Update Mac prerequisites
2021-08-23 19:52:17 +00:00
examples
Replace legacy getStats with standard getStats in the Android example
2021-08-30 10:45:15 +00:00
g3doc
Use absl instead of self-made function for low-level bit counting
2021-08-26 08:56:37 +00:00
logging
Use backticks not vertical bars to denote variables in comments
2021-08-10 10:40:03 +00:00
media
Remove media/base/h264_profile_level_id.* and media/base/vp9_profile.h
2021-08-30 10:31:08 +00:00
modules
Remove media/base/h264_profile_level_id.* and media/base/vp9_profile.h
2021-08-30 10:31:08 +00:00
net/dcsctp
dcsctp: Avoid bundling FORWARD-TSN and DATA chunks
2021-08-19 11:28:40 +00:00
p2p
Add support for manually configuring subnets as VPN
2021-08-25 14:49:11 +00:00
pc
Wire up non-sender RTT for audio, and implement related standardized stats.
2021-08-30 09:03:50 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Use absl instead of self-made function for low-level bit counting
2021-08-26 08:56:37 +00:00
rtc_tools
Delete legacy rtp header parser as no longer used
2021-08-09 12:14:52 +00:00
sdk
Remove media/base/h264_profile_level_id.* and media/base/vp9_profile.h
2021-08-30 10:31:08 +00:00
stats
Wire up non-sender RTT for audio, and implement related standardized stats.
2021-08-30 09:03:50 +00:00
system_wrappers
Use GTEST_SKIP() instead of early return.
2021-08-12 15:24:13 +00:00
test
[PCLF] Add support for dumping video with multiple receivers
2021-08-30 10:09:05 +00:00
tools_webrtc
remove reference to swarming_client
2021-08-24 07:02:25 +00:00
video
frame transformer: expose payload type
2021-08-25 08:33:20 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.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
Increase iOS deployment target from 10 to 12.
2021-07-02 17:02:27 +00:00
.vpython
Update links to point at main branch
2021-07-22 16:41:26 +00:00
AUTHORS
Fix _hRecThread,_hPlayThread RTC_DCHECK reverse bug.
2021-08-18 11:36:46 +00:00
BUILD.gn
Allow export of Obj-C symbols without C++ ones.
2021-07-30 22:54:59 +00:00
CODE_OF_CONDUCT.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
codereview.settings
…
DEPS
Use absl instead of self-made function for low-level bit counting
2021-08-26 08:56:37 +00:00
DIR_METADATA
Move metadata in OWNERS files to DIR_METADATA files.
2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
g3doc.lua
Improve webrtc documentation infra. Preview at:
2021-03-30 10:29:30 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
OWNERS
Fix OWNERS according to recent changes to path expressions.
2021-07-30 13:36:20 +00:00
PATENTS
…
presubmit_test_mocks.py
Add presubmit check to guard against assert() usage.
2021-07-22 17:08:26 +00:00
presubmit_test.py
Add presubmit check to guard against assert() usage.
2021-07-22 17:08:26 +00:00
PRESUBMIT.py
fix some typos
2021-08-12 18:37:10 +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: add g3doc sitemap to toplevel readme
2021-07-23 07:55:17 +00:00
WATCHLISTS
Update WATCHLISTS
2021-08-23 13:37:55 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Drop support for PipeWire 0.2
2021-08-16 09:54:27 +00:00
whitespace.txt
Trigger bots.
2021-08-23 15:29:25 +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
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%