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chromium-webrtc-autoroll a94a4cc197 Roll chromium_revision 2951ce9ba1..8d359ae542 (889518:889625) 
Change log: 2951ce9ba1..8d359ae542
Full diff: 2951ce9ba1..8d359ae542

Changed dependencies
* src/base: 47c75183d7..1b195da248
* src/build: 5f88cb3c2c..e353b02625
* src/ios: 76595faf30..e470857626
* src/testing: 0231deaeb8..8dc98b062a
* src/third_party: 672883196a..07fdd542df
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/84decbd7a4..8bf7a0b3c8
* src/third_party/depot_tools: 3ba11d965c..a5b6b2f8b7
* src/tools: 1e3438b50c..0ccc146add
DEPS diff: 2951ce9ba1..8d359ae542/DEPS

Clang version changed llvmorg-13-init-11649-g4d788fb8:llvmorg-13-init-11999-g50c0aaed
Details: 2951ce9ba1..8d359ae542/tools/clang/scripts/update.py

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I37495c771784dad3cf7c71a9a80fa937015a5033
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221460
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#34236}
2021-06-06 22:42:28 +00:00
api
Add a function to check if the packet in a PacketResult has been received.
2021-06-03 12:42:49 +00:00
audio
AudioSendStream: s/worker_queue_/rtp_transport_queue_/g
2021-06-01 08:00:22 +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 (2021-06-06T04:02:33).
2021-06-06 05:17:07 +00:00
common_audio
Avoid undefined behavior in a division operation.
2021-04-23 07:49:24 +00:00
common_video
Update BitBuffer methods to style guide
2021-05-18 11:10:27 +00:00
data
…
docs
Create a VideoFrameTrackingId RTP header extension.
2021-03-25 17:25:18 +00:00
examples
Added PeerConnectionObserverJni::OnRemoveTrack()
2021-06-03 19:24:55 +00:00
g3doc
doc: document rtp payload type mapping behaviour
2021-06-04 06:23:32 +00:00
logging
Add documentation for RTC event log
2021-06-03 09:03:18 +00:00
media
Make VideoSendStream::UpdateActiveSimulcastLayers not block.
2021-06-04 12:32:24 +00:00
modules
Avoid using legacy rtp header parser in the rtp_to_text tool
2021-06-04 16:41:23 +00:00
net/dcsctp
dcsctp: Stay in stream if not producing fragment
2021-06-02 09:45:59 +00:00
p2p
Remove cricket::DtlsTransportState.
2021-05-21 21:45:29 +00:00
pc
doc: document rtp payload type mapping behaviour
2021-06-04 06:23:32 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Proxy: solve event tracing with compile time strings.
2021-05-28 06:53:29 +00:00
rtc_tools
Convert to NTP time using the real clock.
2021-05-21 19:29:32 +00:00
sdk
Move AV1X-AV1 mapping to VideoCodecTypeMime
2021-06-04 15:38:48 +00:00
stats
Populate qualityLimitationDurations stats for outbound RTP streams
2021-05-31 21:39:37 +00:00
system_wrappers
Make Clock::ConvertTimestampToNtpTime pure virtual
2021-05-21 09:55:14 +00:00
test
Reduce usage of RtpHeaderParser::CreateForTest in favor of RtpPacket
2021-06-03 12:29:09 +00:00
tools_webrtc
Make generate_license.py compatible with Python 3.
2021-05-31 07:05:46 +00:00
video
Handle encoder_ == nullptr in VideoStreamEncoder::EncodeVideoFrame.
2021-06-04 21:06:36 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Switch from check_targets to no_check_targets in .gn
2021-05-20 10:42:21 +00:00
.vpython
Update six library version
2021-04-26 16:39:07 +00:00
AUTHORS
Added PeerConnectionObserverJni::OnRemoveTrack()
2021-06-03 19:24:55 +00:00
BUILD.gn
Switch from check_targets to no_check_targets in .gn
2021-05-20 10:42:21 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
DEPS
Roll chromium_revision 2951ce9ba1..8d359ae542 (889518:889625)
2021-06-06 22:42:28 +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
Make the remote_bitrate_estimator build target private
2020-11-26 12:21:22 +00:00
OWNERS
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
PATENTS
…
presubmit_test_mocks.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
presubmit_test.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
PRESUBMIT.py
sctp: Rename SctpTransport to UsrSctpTransport
2021-04-12 10:40:34 +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
…
README.md
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
WATCHLISTS
Add hta@ to rtc_base/ and api/ WATCHLISTS.
2021-01-06 09:43:34 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Turn on the RTC_ENABLE_WIN_WGC build flag.
2021-05-10 20:16:52 +00:00
whitespace.txt
Reland "Triggering CI."
2021-03-22 11:57:23 +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

Description
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%