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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 Remove old data files. 2018-10-05 14:40:21 +00:00
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 Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
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.py fix some typos 2021-08-12 18:37:10 +00:00
presubmit_test.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +00:00
presubmit_test_mocks.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +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.gni Drop support for PipeWire 0.2 2021-08-16 09:54:27 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +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