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chromium-webrtc-autoroll a7dad03fb1 Roll chromium_revision da6a45d3d1..3bc28c823c (749241:749350) 
Change log: da6a45d3d1..3bc28c823c
Full diff: da6a45d3d1..3bc28c823c

Changed dependencies
* src/base: 802ab61871..097bc4098e
* src/build: 455b1d9ff3..339716a5f0
* src/ios: 8487262865..fb85a1e63a
* src/testing: b8d4bc4dea..524f3fe2f4
* src/third_party: 14cc07ffe1..3055d05fd8
* src/third_party/android_build_tools/bundletool: P0-ZY8wc-hAu5TZYFH7bId8H9Ucy7mNGCg1IPzXuZpEC..Yyhy8FicC3R0ATRzWqGNh4ffsCLz_0nu_BjmNPAdhvIC
* src/third_party/android_deps/libs/com_google_android_gms_play_services_fido: version:17.0.0-cr0..version:18.1.0-cr0
* src/third_party/depot_tools: 96c9784d1a..53effe844c
* src/third_party/ffmpeg: 545152f302..d2350e1645
* src/tools: 3f5d4a663b..bda209202b
DEPS diff: da6a45d3d1..3bc28c823c/DEPS

No update to Clang.

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

Change-Id: I6c3c417387a8318fe045a82581348d00db7ffcf6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170300
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@{#30768}
2020-03-11 22:43:49 +00:00
api
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
audio
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
build_overrides
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
call
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
common_audio
Optimizations and refactoring of the APM 3-band split filter
2020-02-24 13:19:14 +00:00
common_video
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Rename index.md to README.md to make it automatically show up
2020-03-06 10:43:51 +00:00
examples
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
logging
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
media
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
modules
Cleanup RtcpReceiver tests
2020-03-11 15:15:14 +00:00
p2p
Make Connection::id() const
2020-03-05 09:30:18 +00:00
pc
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Allow TimestampAligner to translate timestamp without new observation of system clock.
2020-03-10 17:22:54 +00:00
rtc_tools
Fix 'all' build on Android platforms.
2020-03-11 08:29:06 +00:00
sdk
RtpTransceiver: add kStopped enumeration value.
2020-03-11 11:19:51 +00:00
stats
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +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 RTC_EXPORT to webrtc::Clock
2020-03-10 23:00:21 +00:00
test
Fix: don't use recovered packets in UlpFEC recovery
2020-03-11 12:49:11 +00:00
tools_webrtc
Revert "Reland "Whitespace change to kick bots.""
2020-03-11 14:48:46 +00:00
video
Remove deprecated RtpVideoStreamReceiver constructor.
2020-03-11 17:38:34 +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
Add Av1 Decoder wrapper behind a build flag
2020-03-11 11:20:56 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md
Fix typo in abseil-in-webrtc.md.
2019-12-18 14:27:34 +00:00
AUTHORS
Reland "Reland "Reland "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."""""
2020-02-26 20:35:54 +00:00
BUILD.gn
Fix 'all' build on non Android platforms.
2020-03-11 08:24:27 +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
Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"
2020-02-07 08:23:58 +00:00
DEPS
Roll chromium_revision da6a45d3d1..3bc28c823c (749241:749350)
2020-03-11 22:43:49 +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
Fix public_deps presubmit and gn format fighting each other.
2020-01-30 11:22:46 +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
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
Reformat GN files.
2020-01-21 12:13:11 +00:00
whitespace.txt
Whitespace change
2020-01-28 07:53:15 +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%