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Daniel Cheng 0e1d3c5675 Add reference, pointer, and co type aliases for rtc::ArrayView. 
Many STL containers define these type aliases, and they are easier to
work with than add_const_t<add_lvalue_reference_t<value_type>>.

In a followup, `WTF::Vector` in Blink's conversion constructor from
other containers will be SFINAE-guarded using these type aliases.

Bug: chromium:1408442
Change-Id: I7790e6f462a32e7e49bc6468afeda6b2e6d4b631
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300180
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Daniel Cheng <dcheng@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39771}
2023-04-05 20:18:57 +00:00
api Add reference, pointer, and co type aliases for rtc::ArrayView. 2023-04-05 20:18:57 +00:00
audio Remove AudioConfig::Mode. 2023-04-04 08:44:23 +00:00
build_overrides Always check out google_benchmark, part 3. 2023-03-14 12:14:51 +00:00
call Update WebRTC code version (2023-04-03T04:03:14). 2023-04-03 05:23:52 +00:00
common_audio Check FMA3 support before use it in SincResampler 2023-01-31 17:28:55 +00:00
common_video Changed sps parser and sps parser unit test case for h264, and it is working 2023-03-14 12:15:54 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs doc: rename index.md to README.md 2023-03-13 13:16:22 +00:00
examples Remove dependency to android_support_test_runner in webrtc 2023-04-05 08:40:19 +00:00
experiments Add tool for generating field trial registry header 2022-10-18 07:25:43 +00:00
g3doc Add docs about adding a new test binary. 2023-03-07 11:12:33 +00:00
infra Run Fuchsia Perf Testers on stronger machines. 2023-03-27 09:06:45 +00:00
logging Ensure correct destruction order in RtcEventLogImplTest 2023-03-23 11:50:41 +00:00
media Unnest VideoEncoderFactoryTemplate in webrtc_video_engine_unittest.cc 2023-04-05 15:13:36 +00:00
modules Cleanup ReceiveSideCongestionController api from OnReceivedPacket taking legacy RTPHeader 2023-04-05 17:07:27 +00:00
net/dcsctp dcsctp: Support zero checksum packets 2023-04-02 21:38:00 +00:00
p2p Print discovered network interfaces for debugging ICE_NEVER_CONNECTED endcause. 2023-03-29 11:58:22 +00:00
pc Close PC before test ends to reduce risk of flake. 2023-04-05 15:28:37 +00:00
resources Clarify and extend test support for certain sample rates in audio processing 2022-08-03 14:26:36 +00:00
rtc_base Print discovered network interfaces for debugging ICE_NEVER_CONNECTED endcause. 2023-03-29 11:58:22 +00:00
rtc_tools In rtc event log visualizer use modern api of the ReceiveSideCongestionController 2023-04-04 10:30:44 +00:00
sdk Remove dependency to android_support_test_runner in webrtc 2023-04-05 08:40:19 +00:00
stats Remove deprecated RTCStatsReport(int64) and timestamp_us 2023-03-22 08:00:53 +00:00
system_wrappers Add option to log a warning for unregistered field trials 2023-02-28 15:43:18 +00:00
test Remove AudioConfig::Mode. 2023-04-04 08:44:23 +00:00
tools_webrtc Noop change to trigger bots 2023-02-13 10:30:38 +00:00
video Remove IsSinglecastOrAllNonFirstLayersInactive() helper. 2023-04-04 13:59:07 +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 android32_ndk_api_level to 21. 2023-03-13 12:37:57 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.style.yapf Fix mb.py presubmit issues. 2021-12-08 08:53:00 +00:00
.vpython Remove unused script webrtc_dashboard_upload.py 2022-03-21 12:54:42 +00:00
.vpython3 Add python grpc to .vpython3 for ios test runner 2022-09-16 12:26:48 +00:00
AUTHORS Changed OutputToDebug to create a CFString at compile-time, rather than runtime 2023-02-19 22:41:59 +00:00
BUILD.gn Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
CODE_OF_CONDUCT.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +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 289bd8077e..5e8209d152 (1126466:1126606) 2023-04-05 16:32:38 +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 phoglund from ENG_REVIEW_OWNERS 2021-10-08 08:29:42 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt
native-api.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
OWNERS Add infra owners file 2022-12-02 09:21:47 +00:00
OWNERS_INFRA Add infra owners file 2022-12-02 09:21:47 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Update portaudio to the latest 2022-05-13 09:01:34 +00:00
presubmit_test.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
presubmit_test_mocks.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
pylintrc tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +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 Remove xooglers from WATCHLISTS and OWNERS 2022-11-30 15:33:25 +00:00
webrtc.gni Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
webrtc_lib_link_test.cc Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
whitespace.txt Trigger bots 2022-11-17 21:29:53 +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