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Joe Mason 9cd326c0cf Update missed link to chromium C++11 styleguide 
Follow-up to https://webrtc-review.googlesource.com/c/src/+/232260:
there's a second link to the obsolete chromium-cpp.appspot.com in the
same file that was missed.

Also updates the link to be more precise: the new markdown file has
anchors on individual entries, not just the section headers.

R=danilchap@webrtc.org

Bug: chromium:1243839
Change-Id: I17918d155aacf3465a46fd674a598139a0870165
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232560
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35043}
2021-09-20 21:05:39 +00:00
api New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. 2021-09-15 09:57:29 +00:00
audio Deduplicate set of the rtp header extension uri constants 2021-09-14 13:38:44 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Update WebRTC code version (2021-09-19T04:02:28). 2021-09-19 05:46:52 +00:00
common_audio Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
common_video Fix integer overflow in h264 pps parser 2021-09-20 11:28:36 +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 AV1X with AV1 2021-09-14 08:29:02 +00:00
g3doc Update missed link to chromium C++11 styleguide 2021-09-20 21:05:39 +00:00
logging Migrate rtc event log from rtc::BitBuffer to BitstreamReader 2021-09-07 14:03:27 +00:00
media APM: remove webrtc::Config 2021-09-17 11:21:32 +00:00
modules Default the safetyped calculation of packet interarrival times. 2021-09-17 13:58:55 +00:00
net/dcsctp dcsctp: move HandoverReadinessStatus::ToString definition to build target public:socket 2021-09-20 12:25:15 +00:00
p2p Delete BasicPacketSocketFactory default constructor 2021-09-03 10:46:29 +00:00
pc Return proxied object in OnTransceiver 2021-09-16 09:40:52 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Fix integer underflow in BitstreamReader::ConsumeBits 2021-09-20 19:37:49 +00:00
rtc_tools Improve points visualization in metrics_plotter. 2021-09-10 10:59:58 +00:00
sdk Verify that locks are handled correctly on RTCAudioSession 2021-09-17 01:41:42 +00:00
stats Reland "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-06 14:26:55 +00:00
system_wrappers Use GTEST_SKIP() instead of early return. 2021-08-12 15:24:13 +00:00
test Fix integer underflow in BitstreamReader::ConsumeBits 2021-09-20 19:37:49 +00:00
tools_webrtc rename use_x11 to ozone_platform_x11 2021-09-20 15:13:55 +00:00
video New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. 2021-09-15 09:57:29 +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
.vpython3 Add .vpython3 file to webrtc 2021-09-15 16:56:30 +00:00
AUTHORS rename use_x11 to ozone_platform_x11 2021-09-20 15:13:55 +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 Roll chromium_revision d2004ccb64..35a3026fcf (922852:922957) 2021-09-20 18:31:12 +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 Add .vpython3 file to webrtc 2021-09-15 16:56:30 +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 rename use_x11 to ozone_platform_x11 2021-09-20 15:13:55 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger bots 2021-09-08 18:30:03 +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