signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
29,305 Commits 2 Branches 174 Tags 441 MiB
C++ 91.4%
Java 2.8%
Objective-C++ 2%
C 1.4%
Python 1.2%
Other 1%
6286bb0b2c
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Yves Gerey 6286bb0b2c Roll chromium_revision ac00cfc7cd..da78695105 (723527:724157) 
Manual tweak:
 * do not roll src/testing (iossim update).
 * do not roll src/ios

Change log: ac00cfc7cd..da78695105
Full diff: ac00cfc7cd..da78695105

Changed dependencies
* src/base: 7ef8e2b029..74d23bf118
* src/build: 4da2a5b127..dfe5662014
* src/third_party: f5a239ae12..4ca87a167e
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/8953fbe6c5..4082c9147b
* src/third_party/depot_tools: e9aeaee522..8b34eb4d1e
* src/tools: 25869a7ef1..6bcb3d7f48
DEPS diff: ac00cfc7cd..da78695105/DEPS

No update to Clang.

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

Change-Id: Ic345277fc9881362a1e6c09c4958774851d13698
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161984
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30073}
2019-12-12 14:22:54 +00:00
api Adds queue length setter to simulated network node builder. 2019-12-12 09:15:21 +00:00
audio Minor fixes to ChannelSend. 2019-12-02 09:30:51 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Initial version of ResourceAdaptationProcessor and friends. 2019-12-10 15:31:43 +00:00
common_audio Remove potential real-time reallocation in PushResampler 2019-12-11 13:16:37 +00:00
common_video Optimize FindNaluIndices 2019-12-11 15:28:06 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Refresh some links in the docs folder. 2019-12-10 15:32:48 +00:00
examples Fix errorprone issues preventing Chromium Roll. 2019-11-27 12:52:48 +00:00
logging Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_. 2019-11-29 09:45:50 +00:00
media Execute cached video encoder switching request if encoder switching is allowed after the switch request was made. 2019-12-11 17:16:04 +00:00
modules Makes padding prefer video SSRCs instead of audio. 2019-12-11 16:32:14 +00:00
p2p STUN PING request 2019-12-11 14:23:30 +00:00
pc Revert "Distinguish between send and receive video codecs" 2019-12-09 14:48:55 +00:00
resources Changed the digital AGC1 gain to properly support multichannel 2019-11-23 08:42:59 +00:00
rtc_base Add rtc::Thread::PostDelayedTask 2019-12-11 20:42:43 +00:00
rtc_tools Move audioproc_f to rtc_tools. 2019-12-10 12:08:56 +00:00
sdk Delete transitional method EncodedImage.maybeRetain 2019-12-12 14:11:14 +00:00
stats Add totalInterFrameDelay to RTCInboundRTPStreamStats 2019-11-25 10:50:37 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Add directive to make webrtc metrics optional. 2019-12-09 13:55:50 +00:00
test Migrate tests from RtpDepacketizer to VideoRtpDepacketizer interface 2019-12-10 17:37:46 +00:00
tools_webrtc [UBSan] Remove suppression for opus. 2019-12-10 08:59:30 +00:00
video Make struct SynchronizationDelays more general. 2019-12-11 13:55:16 +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 Switch to compiling WebRTC -std=c++14 by default 2019-09-09 19:24:16 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Update style guide for absl::make_unique. 2019-09-18 06:10:58 +00:00
AUTHORS Revert "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""" 2019-12-09 11:26:20 +00:00
BUILD.gn Initial version of ResourceAdaptationProcessor and friends. 2019-12-10 15:31:43 +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 ac00cfc7cd..da78695105 (723527:724157) 2019-12-12 14:22:54 +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 Add guidance to style guide how to reference a bug in a TODO 2019-12-11 11:55:52 +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 Do not disable metrics by default. 2019-12-11 08:08:58 +00:00
whitespace.txt Revert "Whitespace change" 2019-11-11 14:58:20 +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