zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
28,865 Commits 1 Branch 0 Tags
6f5b9e01bc9373b3574c0d8a0c7918284e75af79
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 6f5b9e01bc Roll chromium_revision d68d92fb45..9109135db0 (709806:709913) 
Change log: d68d92fb45..9109135db0
Full diff: d68d92fb45..9109135db0

Changed dependencies
* src/base: d3a63c2892..fca6c87102
* src/build: 2b40e7bc8b..114fe8652e
* src/ios: 010771296a..2db5ae2009
* src/testing: 85152663b9..3a2c7fc05f
* src/third_party: 42240c35cd..30cdf5b0ac
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/7c5920efc2..e44b5225db
* src/tools: 2c37082a08..efa9758841
DEPS diff: d68d92fb45..9109135db0/DEPS

No update to Clang.

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

Change-Id: Iaa22a8a4a7c72711c9ac1ca0b8bef2579757dcf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158582
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@{#29633}
2019-10-28 16:33:10 +00:00
api
Implement crypto stats on DTLS transport
2019-10-28 11:30:23 +00:00
audio
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Calls OnPacketsAcknowledged on RtpRtcp instead of RTPSender directly.
2019-10-24 12:13:56 +00:00
common_audio
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
common_video
Adding missing RTC_EXPORT for component build.
2019-10-18 09:17:56 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Fixing some typos.
2019-09-10 10:03:50 +00:00
examples
Fix gtk color-space conversion in peerconnection_client
2019-10-23 11:25:06 +00:00
logging
Move rtc_base/ignore_wundef.h to its own target.
2019-10-19 10:50:36 +00:00
media
Added usrsctp reliablitiy stress test.
2019-10-25 16:51:38 +00:00
modules
Make AudioFrame member instead of raw pointer in APM test fixture
2019-10-28 15:34:29 +00:00
p2p
Implement crypto stats on DTLS transport
2019-10-28 11:30:23 +00:00
pc
Reland "[PeerConnection] Use an OperationsChain in PeerConnection for async ops."
2019-10-28 12:35:50 +00:00
resources
Delete voice_detection() pointer to submodule
2019-10-07 13:06:05 +00:00
rtc_base
Implement crypto stats on DTLS transport
2019-10-28 11:30:23 +00:00
rtc_tools
Move rtc_base/ignore_wundef.h to its own target.
2019-10-19 10:50:36 +00:00
sdk
Added allow_codec_switching parameter to RTCConfig.
2019-10-25 11:06:31 +00:00
stats
Implement crypto stats on DTLS transport
2019-10-28 11:30:23 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
test
Fix MemoryLogWriter so that it always writes the full data.
2019-10-25 12:07:32 +00:00
tools_webrtc
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
video
change PacketBuffer to return it's result rather that use callback
2019-10-23 16:50:57 +00:00
.clang-format
…
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .clangd to .gitignore
2019-10-28 12:27:50 +00:00
.gn
Switch to compiling WebRTC -std=c++14 by default
2019-09-09 19:24:16 +00:00
.vpython
Add vpython dependencies needed to run presubmit tests on LUCI
2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md
Update style guide for absl::make_unique.
2019-09-18 06:10:58 +00:00
AUTHORS
Add RTCStatisticsReport.h to WebRTC.framework.
2019-10-21 11:16:10 +00:00
BUILD.gn
Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
2019-10-25 17:54:32 +00:00
CODE_OF_CONDUCT.md
Add code of conduct to WebRTC repo
2017-05-16 12:09:13 +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
common_types.h
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision d68d92fb45..9109135db0 (709806:709913)
2019-10-28 16:33:10 +00:00
ENG_REVIEW_OWNERS
Enforce LGTM from owners of depends-on paths in DEPS via presubmit.
2018-09-28 12:49:54 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
…
native-api.md
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Add #COMPONENT to WebRTC.
2019-10-08 12:20:39 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +00:00
presubmit_test.py
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
PRESUBMIT.py
Add possibility to skip check_includes presubmit check.
2019-10-22 19:35:31 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
README.md
Tell users where they can find the native API headers
2017-11-14 10:36:46 +00:00
style-guide.md
Update WebRTC's C++ style guide to reflect the switch to C++14.
2019-09-16 11:45:35 +00:00
WATCHLISTS
Add saza to audio watchlists
2019-09-03 14:55:43 +00:00
webrtc_lib_link_test.cc
Rewrite the lib link test to just be a binary.
2019-10-18 07:42:20 +00:00
webrtc.gni
Make rtc_json poisonous in WebRTC
2019-10-25 13:46:03 +00:00
whitespace.txt
Whitespace change
2018-02-23 10:34:16 +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 http://www.webrtc.org/native-code/development 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%