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chromium-webrtc-autoroll eee1a49fe1 Roll chromium_revision 0ec8ef3c9f..e0dd604cfd (718517:718623) 
Change log: 0ec8ef3c9f..e0dd604cfd
Full diff: 0ec8ef3c9f..e0dd604cfd

Changed dependencies
* src/build: 7875528379..0ca62e351e
* src/third_party: 00544f5296..b2aa7c13f5
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/5459c3891e..56b1f53bcd
* src/tools: 528bef253e..6f634b9fc1
DEPS diff: 0ec8ef3c9f..e0dd604cfd/DEPS

No update to Clang.

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

Change-Id: I1e9e0fff278ce6c9356aefe8df249948a1618c1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160470
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@{#29904}
2019-11-25 14:46:35 +00:00
api
Fix VideoStreamEncoder to not reference encoded data from the RunPostEncode task.
2019-11-25 14:41:55 +00:00
audio
Adding VoIP specific channel adjustments
2019-11-22 15:25:05 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Makes all of RtpVideoSenderTest use simulated time
2019-11-25 12:13:38 +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
Revert "VideoFrame: Store a reference to an encoded frame"
2019-11-21 14:55:21 +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
Add @Nullable annotations to quiet errorprone.
2019-11-19 12:50:30 +00:00
logging
Make rtc::Thread a TaskQueue
2019-11-25 11:38:27 +00:00
media
Add totalInterFrameDelay to RTCInboundRTPStreamStats
2019-11-25 10:50:37 +00:00
modules
Add IvfVideoFrameGenerator
2019-11-25 13:38:09 +00:00
p2p
Make IceController injectable
2019-11-22 19:45:32 +00:00
pc
Add totalInterFrameDelay to RTCInboundRTPStreamStats
2019-11-25 10:50:37 +00:00
resources
Changed the digital AGC1 gain to properly support multichannel
2019-11-23 08:42:59 +00:00
rtc_base
Add class for ExponentialMovingAverage
2019-11-25 13:17:59 +00:00
rtc_tools
EventLogVisualizer pacer delay plot fix.
2019-11-19 10:23:31 +00:00
sdk
Make it possible to inject a custom NetEqFactory from the java interface.
2019-11-25 12:39:08 +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
Revert "Reland "Make webrtc_fuzzer_main depend on webrtc_component in Chromium.""
2019-11-23 15:10:47 +00:00
test
Add IvfVideoFrameGenerator
2019-11-25 13:38:09 +00:00
tools_webrtc
Remove temporary workaround for generate_licenses.
2019-11-25 10:23:25 +00:00
video
Fix VideoStreamEncoder to not reference encoded data from the RunPostEncode task.
2019-11-25 14:41:55 +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 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
Force alignment of generated JVM called functions.
2019-11-21 12:34:35 +00:00
BUILD.gn
Move stun.h to api/.
2019-11-18 16:11:27 +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
common_types.h
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision 0ec8ef3c9f..e0dd604cfd (718517:718623)
2019-11-25 14:46:35 +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
Manual chromium roll: Compile using JDK 11
2019-11-20 12:13:43 +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

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%