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Jianhui Dai 7d6b5b878f Add StopOutputOnWriteFailure to RtcEventLogImplTest 
Bug: chromium:1288710
Change-Id: Ib913fb392077512d43ea408c95115129d3ff3425
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295800
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39509}
2023-03-09 08:50:12 +00:00
api stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats 2023-03-07 14:27:47 +00:00
audio Add a clone method to the audio frame transformer API. 2023-03-06 08:22:25 +00:00
build_overrides Use default values provided by PartitionAlloc instead of hard-coded ones 2022-12-07 09:11:35 +00:00
call Stop Posting tasks when we don't need to. 2023-03-06 15:13:39 +00:00
common_audio Check FMA3 support before use it in SincResampler 2023-01-31 17:28:55 +00:00
common_video Fix AVC PPS parser unit test. 2023-03-01 12:00:49 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update documentation links in docs/native-code/development/index.md 2023-02-20 16:35:23 +00:00
examples Remove mentions of already deleted field trials 2023-03-01 15:53:37 +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 Add a new test binary to check if split-channel works 2023-03-07 11:14:31 +00:00
logging Add StopOutputOnWriteFailure to RtcEventLogImplTest 2023-03-09 08:50:12 +00:00
media Reland "Use two MediaChannels for 2 directions." 2023-03-07 12:57:35 +00:00
modules Skip calling rtcp callback on packets containing invalid rtcp message 2023-03-08 14:19:54 +00:00
net/dcsctp Implement support for Chrome task origin tracing. #3.5/4 2023-03-01 11:11:37 +00:00
p2p Replace use of test-only connections() with P2PTransportChannel member. 2023-02-27 16:49:05 +00:00
pc Update SetHeaderExtensionsToNegotiate to match specification 2023-03-08 08:46:35 +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 Prevent warnings from timestamp aligner used in AudioDeviceBuffer 2023-03-06 15:47:51 +00:00
rtc_tools Handling NetEqSetMinimumDelay events in neteq_rtpplay. 2023-02-09 09:39:29 +00:00
sdk Remove resolution alignment requirement (part 2) 2023-03-07 17:18:55 +00:00
stats stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats 2023-03-07 14:27:47 +00:00
system_wrappers Add option to log a warning for unregistered field trials 2023-02-28 15:43:18 +00:00
test stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats 2023-03-07 14:27:47 +00:00
tools_webrtc Noop change to trigger bots 2023-02-13 10:30:38 +00:00
video Use CodecTypeToPayloadString 2023-03-08 14:38:06 +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 Set Fuchsia Api level + update SDK version 2022-09-14 08:49:56 +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 Add a new test binary to check if split-channel works 2023-03-07 11:14:31 +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 115884e3e5..9931d87c4c (1114800:1114909) 2023-03-09 02:51:50 +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 Add option to log a warning for unregistered field trials 2023-02-28 15:43:18 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +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