zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
30,929 Commits 1 Branch 0 Tags
bc633d714deee8d7bf579c10d5322546e32213dd
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 bc633d714d Roll chromium_revision a69b9c614c..f0f2118569 (786202:787142) 
Change log: a69b9c614c..f0f2118569
Full diff: a69b9c614c..f0f2118569

Changed dependencies
* src/base: 9183ce544b..16546fe468
* src/build: 18e5f25087..992194ef50
* src/ios: 2ea69251b8..90035757d7
* src/testing: 778d425fe0..5c2a63f2d6
* src/third_party: 5d8b539958..992830e84d
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/430a742303..eda849d2e6
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/7cbda52a2d..e4221b4636
* src/third_party/depot_tools: e62496bafc..9af33fa208
* src/third_party/freetype/src: 0f35b042e0..96fb73efd5
* src/third_party/perfetto: faef2531ec..af56531347
* src/tools: af8d53c7e3..5b541bb823
DEPS diff: a69b9c614c..f0f2118569/DEPS

No update to Clang.

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

Change-Id: I075e13fb84c0c487f49f4cb8fdeb02579083f2ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179021
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@{#31697}
2020-07-10 10:39:08 +00:00
api
Reland "Complete migration from "track" to "inbound-rtp" stats"
2020-07-10 10:17:50 +00:00
audio
webrtc::AudioSendStream: Add lock annotation to audio_level_
2020-07-06 17:05:25 +00:00
build_overrides
Rename sanitizers suppression files.
2020-07-03 18:29:58 +00:00
call
[Adaptation] Move AdaptationConstraints to VideoStreamAdapter
2020-07-09 13:06:56 +00:00
common_audio
Replace slave -> helper, master -> reference
2020-06-29 12:18:05 +00:00
common_video
Migrate common_video/ and examples/ to webrtc::Mutex.
2020-07-07 13:33:27 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
examples
Add missing dependencies to java targets.
2020-07-10 09:17:00 +00:00
logging
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
media
Migrate leftovers in media/ and modules/ to webrtc::Mutex.
2020-07-10 08:27:45 +00:00
modules
Migrate leftovers in media/ and modules/ to webrtc::Mutex.
2020-07-10 08:27:45 +00:00
p2p
Revert "Implement packets_(sent | received) for RTCTransportStats"
2020-07-08 09:42:41 +00:00
pc
Reland "Complete migration from "track" to "inbound-rtp" stats"
2020-07-10 10:17:50 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Migrate rtc_base to webrtc::Mutex.
2020-07-08 20:38:54 +00:00
rtc_tools
Migrate rtc_tools/, sdk/, and system_wrappers/ to webrtc::Mutex.
2020-07-10 08:02:05 +00:00
sdk
Add missing dependencies to java targets.
2020-07-10 09:17:00 +00:00
stats
Reland "Complete migration from "track" to "inbound-rtp" stats"
2020-07-10 10:17:50 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Migrate rtc_tools/, sdk/, and system_wrappers/ to webrtc::Mutex.
2020-07-10 08:02:05 +00:00
test
Migrate a leftover in test/ to webrtc::Mutex.
2020-07-09 14:28:21 +00:00
tools_webrtc
Fix 'Perf Android32 (M Nexus5)' config in MB.
2020-07-08 15:04:10 +00:00
video
[Adaptation] Move AdaptationConstraints to VideoStreamAdapter
2020-07-09 13:06:56 +00:00
.clang-format
Tune ObjC clang-format configuration
2017-05-11 09:14:18 +00:00
.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
Reenable libaom decoder by default
2020-03-18 18:04:41 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md
Use absl_deps in order to preapre to the Abseil component build release.
2020-06-08 12:59:40 +00:00
AUTHORS
Relanding: Fix data channel message integrity violation
2020-07-07 03:06:24 +00:00
BUILD.gn
Convert GN libs lists to frameworks
2020-07-06 10:08:09 +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
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
DEPS
Roll chromium_revision a69b9c614c..f0f2118569 (786202:787142)
2020-07-10 10:39:08 +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
Make transient suppression optionally excludable via defines
2020-04-02 11:44:07 +00:00
OWNERS
Remove phoglund as root owner.
2020-03-30 12:15:56 +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
Inclusive language in PRESUBMIT.py.
2020-06-30 14:03:10 +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
Fix link in documentation. (take 2)
2020-04-16 11:08:43 +00:00
style-guide.md
C++ style: We don't allow designated initializers
2020-06-03 09:11:09 +00:00
WATCHLISTS
Remove benwright@webrtc.org from WATCHLISTS
2020-01-31 18:46:52 +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
Revert "Build: Disable the iLBC audio codec by default"
2020-07-09 11:27:06 +00:00
whitespace.txt
Trigger bots.
2020-06-15 08:40:44 +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%