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Jan Grulich 65a9a515b5 Add additional checks in case of early portal error
In case ScreenCast portal fails right at the beginning, we need to check
the response before trying to get session handle to avoid accessing
non-existing portal data.

Also on early failure do not continue making source request if we failed
before and don't have session handle.

Bug: webrtc:13429
Change-Id: I2bfbd2c6e96e3cda1e62aa9dc07f66d4c7496b53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272400
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37872}
2022-08-22 21:30:59 +00:00
api [PCLF] Add ToString method to VideoDumpOptions 2022-08-22 12:47:37 +00:00
audio Update rtc::Event::Wait call sites to use TimeDelta. 2022-08-19 10:07:28 +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 (2022-08-22T04:02:37). 2022-08-22 05:12:29 +00:00
common_audio Reenable WebRTC PushResampler format checks on Windows clang debug builds 2022-08-05 11:03:08 +00:00
common_video Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs [ios] Remove the support for bitcode 2022-07-04 09:01:52 +00:00
examples Reenable some iOS tests. 2022-08-19 13:10:17 +00:00
g3doc Clarify how to reference WebRTC bugs in TODOs 2022-07-01 08:03:34 +00:00
infra Add Weetbix realm ACL 2022-08-22 06:39:39 +00:00
logging Encode remote link capacity estimates in legacy RTC event log format 2022-08-11 12:57:02 +00:00
media Remove sigslot usage from SctpTransportInternal 2022-08-22 13:51:17 +00:00
modules Add additional checks in case of early portal error 2022-08-22 21:30:59 +00:00
net/dcsctp dcsctp: Add handover state for stream counts 2022-08-22 11:04:31 +00:00
p2p Replace RTCCertificateGeneratorCallback interface with an AnyInvocable 2022-08-22 16:53:14 +00:00
pc Replace RTCCertificateGeneratorCallback interface with an AnyInvocable 2022-08-22 16:53:14 +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 Replace RTCCertificateGeneratorCallback interface with an AnyInvocable 2022-08-22 16:53:14 +00:00
rtc_tools Update rtc::Event::Wait call sites to use TimeDelta. 2022-08-19 10:07:28 +00:00
sdk rtc::Event: Finalize migration to TimeDelta. 2022-08-19 13:44:57 +00:00
stats stats: implement outbound-rtp.active 2022-07-28 13:35:40 +00:00
system_wrappers Delete nisse@webrtc.org from OWNERS files 2022-07-28 08:47:38 +00:00
test Remove sigslot usage from SctpTransportInternal 2022-08-22 13:51:17 +00:00
tools_webrtc Migrate mb script to python3. 2022-08-17 13:50:43 +00:00
video Move decoder instance ownership to VideoReceiver2 2022-08-22 13:42:47 +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 Prevent jsoncpp from hiding deprecated declarations in WebRTC 2022-04-11 12:33:47 +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 Update protobuf-py2_py3 wheel. 2022-07-01 15:17:36 +00:00
AUTHORS Enable Multithreaded H264 Encoding For OpenH264 2022-08-19 10:30:37 +00:00
BUILD.gn Delete QueuedTask and ToQueuedTask as no longer needed 2022-08-09 11:11:26 +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 137733c440..bd812e5c8f (1037767:1037901) 2022-08-22 20:46:38 +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
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 add some eng prod owners to PRESUBMIT.py. 2022-03-18 13:19:07 +00:00
PATENTS
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 fhernqvist from watchlists 2022-08-11 14:44:52 +00:00
webrtc.gni [Cast Convergence] Replace is_chromecast with new args 2022-06-16 00:50:08 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger CI. 2022-08-12 11:03:03 +00:00

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