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chromium-webrtc-autoroll df5f864055 Roll chromium_revision c814c493d7..20cffe6848 (814002:814194) 
Change log: c814c493d7..20cffe6848
Full diff: c814c493d7..20cffe6848

Changed dependencies
* src/base: ddaf6eb849..f3eb7b12e0
* src/build: d2516aa452..1bf0c35110
* src/ios: dffd8a6463..a89fc48a60
* src/testing: 2fbb1eacdb..1c37217884
* src/third_party: ec67470d20..8629fc0c0e
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/83c2cb1060..532e564f62
* src/third_party/depot_tools: d0b4e1976c..9929780af3
* src/tools: 8d11a3bd1a..34b77b3991
DEPS diff: c814c493d7..20cffe6848/DEPS

No update to Clang.

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

Change-Id: I36ae1ced365253c11412c18d6b1b15eac752c69f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186924
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@{#32328}
2020-10-06 14:51:03 +00:00
api
Add experimental extension RtpVideoLayersAllocation
2020-10-05 13:38:13 +00:00
audio
Async audio processing API
2020-10-02 12:33:34 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Cleans up code related to video packetization overhead.
2020-10-05 13:22:33 +00:00
common_audio
Reland "Activating AVX2 support by default"
2020-10-03 22:37:13 +00:00
common_video
Remove deprecated i420 buffer pool
2020-09-23 12:22:40 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Adding documentation to become a WebRTC committer.
2020-09-02 07:49:04 +00:00
examples
Fix libyuv includes for libyuv roll
2020-10-02 08:03:41 +00:00
logging
Allow RTCP packets longer than 1500 bytes in RTC event log.
2020-10-05 15:26:54 +00:00
media
Take max bitrate into account for target bitrate decision when min bitrate is empty
2020-10-05 11:11:51 +00:00
modules
Refactor NetEq delay manager logic.
2020-10-06 13:22:45 +00:00
p2p
Revert "Replace sigslot usages with robocaller library."
2020-10-06 11:40:43 +00:00
pc
Update RTC_DCHECK_IS_ON checks in SdpOfferAnswerHandler
2020-10-06 14:10:36 +00:00
resources
Reland "Activating AVX2 support by default"
2020-10-03 22:37:13 +00:00
rtc_base
Update RTC_DCHECK_IS_ON checks in SdpOfferAnswerHandler
2020-10-06 14:10:36 +00:00
rtc_tools
Add method to extract triage alerts from RTC event log analyzer.
2020-10-02 13:41:30 +00:00
sdk
Delete AllowBlockingCalls and android_media_codec_common.h
2020-10-01 16:38:09 +00:00
stats
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
style-guide
…
system_wrappers
Reland "Activating AVX2 support by default"
2020-10-03 22:37:13 +00:00
test
Revert "Replace sigslot usages with robocaller library."
2020-10-06 11:40:43 +00:00
tools_webrtc
Create isolated output directory when creating the output file.
2020-09-29 15:38:52 +00:00
video
Cleans up code related to video packetization overhead.
2020-10-05 13:22:33 +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
Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
2020-09-07 08:37:14 +00:00
.vpython
Add 'requests' to .vpython.
2020-09-09 14:36:03 +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
Fix bug with the sps-pps-idr-in-keyframe fmtp parameter.
2020-09-19 09:23:39 +00:00
BUILD.gn
Rename CancerStickCastle to RoboCaller.
2020-09-21 19:50:39 +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
DEPS
Roll chromium_revision c814c493d7..20cffe6848 (814002:814194)
2020-10-06 14:51:03 +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
…
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
…
presubmit_test_mocks.py
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +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 PRESUBMIT.py warnings.
2020-09-07 08:34:44 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +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
webrtc: Replace is_desktop_linux with is_linux
2020-10-05 17:29:54 +00:00
whitespace.txt
Trigger CI bots.
2020-09-06 10:13:23 +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
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Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%