zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
33,291 Commits 1 Branch 0 Tags
6c96611069fc3e2d608ed4469c4cc3007b22642c
Go to file
Clone
Open with VS Code Open with VSCodium Open with Intellij IDEA
Download ZIP Download TAR.GZ Download BUNDLE
chromium-webrtc-autoroll 6c96611069 Roll chromium_revision 3bdbd47d23..b8d2317c2e (884422:884575) 
Change log: 3bdbd47d23..b8d2317c2e
Full diff: 3bdbd47d23..b8d2317c2e

Changed dependencies
* src/base: e34cf0158f..62a9dbcb7f
* src/build: 4e5721997a..d0db4e6f41
* src/ios: e0e60851e1..f6d02ed75e
* src/testing: 1bebb81f26..a6baa962fe
* src/third_party: 721e99a5ba..290861ba4e
* src/third_party/freetype/src: 1bc801b0d6..06e21ffedf
* src/third_party/perfetto: dac610dd85..9a1689eeaa
* src/tools: 04d8fb2093..93f86305be
* src/tools/luci-go: git_revision:22d464e2f8f3bd2bd33f69fe819326d63f881008..git_revision:9cd9603def7a649cd3e29517180d6622be2fa898
* src/tools/luci-go: git_revision:22d464e2f8f3bd2bd33f69fe819326d63f881008..git_revision:9cd9603def7a649cd3e29517180d6622be2fa898
* src/tools/luci-go: git_revision:22d464e2f8f3bd2bd33f69fe819326d63f881008..git_revision:9cd9603def7a649cd3e29517180d6622be2fa898
DEPS diff: 3bdbd47d23..b8d2317c2e/DEPS

No update to Clang.

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

Change-Id: I31569661a7a962e619a1573e7a6189d7ac865c64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219502
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@{#34059}
2021-05-19 19:10:56 +00:00
api
AEC3: Change adaptation speed of the matched filter after a delay is found
2021-05-19 13:37:13 +00:00
audio
Use Timestamp to represent packet receive timestamps
2021-05-04 13:16:54 +00:00
build_overrides
Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492)
2021-04-12 18:25:58 +00:00
call
Update WebRTC code version (2021-05-19T04:03:01).
2021-05-19 05:37:13 +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
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
g3doc
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
logging
Update BitBuffer methods to style guide
2021-05-18 11:10:27 +00:00
media
sctp: Reduce logging level for common calls
2021-05-19 13:22:43 +00:00
modules
Add an UlpFec test to RtpRtcp unit tests.
2021-05-19 15:26:24 +00:00
net/dcsctp
dcsctp: Report duplicate TSNs
2021-05-19 12:57:03 +00:00
p2p
Define cricket::DtlsTransportState in terms of webrtc::DtlsTransportState.
2021-05-12 07:06:38 +00:00
pc
Add test for many non-dropped packets in unreliable mode.
2021-05-19 17:30:46 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Update BitBuffer methods to style guide
2021-05-18 11:10:27 +00:00
rtc_tools
Remove REMB throttling funcionality from PacketRouter
2021-05-12 11:24:58 +00:00
sdk
Set nativeObserver to 0 to avoid double release.
2021-05-17 19:53:43 +00:00
stats
Simplify reference counting implementation of PendingTaskSafetyFlag.
2021-04-21 07:04:01 +00:00
system_wrappers
Consolidate the different NTP clocks into one.
2021-04-08 13:54:04 +00:00
test
dcsctp: Add socket fuzzer
2021-05-14 06:41:10 +00:00
tools_webrtc
crc32c: Point the licensing script to the LICENSE file
2021-05-03 16:46:30 +00:00
video
Remove redundant VideoSendStream::rtcp_stats field
2021-05-18 13:37:51 +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
Enable GN check on //net.
2021-05-03 14:23:09 +00:00
.vpython
Update six library version
2021-04-26 16:39:07 +00:00
AUTHORS
Revert "Ensure method which updates UI is called in main thread"
2021-04-30 09:26:03 +00:00
BUILD.gn
Disable range-loop-analysis when use_xcode_clang=true.
2021-05-07 15:24:28 +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 3bdbd47d23..b8d2317c2e (884422:884575)
2021-05-19 19:10:56 +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
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
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
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
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%