zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
33,556 Commits 1 Branch 0 Tags
11b92cfa6542e329ddf8924279849b2712c90d81
Go to file
Clone
Open with VS Code Open with VSCodium Open with Intellij IDEA
Download ZIP Download TAR.GZ Download BUNDLE
Harald Alvestrand 11b92cfa65 Refactoring: Move groups-by-mid into Bundle manager 
Bug: webrtc:12837
Change-Id: I2431dbbf8cc291b5f3848d81cf290fd3e97ec15d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222614
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34324}
2021-06-17 14:35:33 +00:00
api
Mark VideoSendTiming flags as invalid by default.
2021-06-17 12:39:34 +00:00
audio
Update the sync_group id without recreating audio receive streams.
2021-06-16 19:34:18 +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 (2021-06-17T04:05:50).
2021-06-17 05:28:35 +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
Remove old data files.
2018-10-05 14:40:21 +00:00
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
Reland "Deprecate microsecond timestamps in RTC event log."
2021-06-17 12:08:54 +00:00
media
Make WebRtcAudioReceiveStream::stream_ const.
2021-06-17 08:39:44 +00:00
modules
Add timestamp to log message in generic_decoder.cc
2021-06-17 10:14:14 +00:00
net/dcsctp
dcsctp: Do explicit bounds checking in bounded IO
2021-06-16 13:02:32 +00:00
p2p
Sanitize hostname literals when mDNS obfuscation is on.
2021-06-15 14:41:46 +00:00
pc
Refactoring: Move groups-by-mid into Bundle manager
2021-06-17 14:35:33 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
LOG DTLS (failed) handshake retransmission
2021-06-16 13:13:52 +00:00
rtc_tools
Reland "Correctly handle retransmissions/padding in early loss detection."
2021-06-16 08:14:27 +00:00
sdk
Change YuvConverter.convert to catch GLExceptions and return null.
2021-06-17 11:33:54 +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
Mark VideoSendTiming flags as invalid by default.
2021-06-17 12:39:34 +00:00
tools_webrtc
Remove unnused build configs for M1 builder
2021-06-17 09:37:50 +00:00
video
Revert "Avoid video stream allocation on configuration change after timeout."
2021-06-17 08:33:24 +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
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
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 6ade74989a..6f7025c98c (893176:893293)
2021-06-17 05:00:35 +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
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
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%