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chromium-webrtc-autoroll 1d7a3fe6ed Roll chromium_revision 7a8fca54e0..4591d36de2 (800031:800155) 
Change log: 7a8fca54e0..4591d36de2
Full diff: 7a8fca54e0..4591d36de2

Changed dependencies
* src/base: b8d3ffc3b8..ec18ec11ef
* src/build: 183d29ca63..579e98a211
* src/ios: cabfd90bbe..6cd7533818
* src/testing: 855f6eb5b6..f284de851c
* src/third_party: 54c9cadd81..e52741fa71
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c244e33ead..7f34313548
* src/third_party/depot_tools: 25f1303ed8..81f4ba7e7d
* src/third_party/perfetto: ef0fb4c270..56d9df4fa5
* src/tools: fe43defcbf..241a11d7d7
DEPS diff: 7a8fca54e0..4591d36de2/DEPS

No update to Clang.

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

Change-Id: I54c6b2091d160b3eb0a730cc81b06bfe250d603c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182102
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@{#31975}
2020-08-20 17:15:43 +00:00
api
DTMF Event Sub-API on VoIP API
2020-08-20 17:10:02 +00:00
audio
DTMF Event Sub-API on VoIP API
2020-08-20 17:10:02 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Revert "[Adaptation] Remove QualityScalerResource when disabled."
2020-08-17 13:46:30 +00:00
common_audio
Support AVX2/FMA intrinsics in audio FIR filter
2020-08-19 10:21:31 +00:00
common_video
Move function PrintVideoFrame to the test file where it is used.
2020-08-06 15:20:19 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Updated IOS documentation with correct build_ios_libs.py path
2020-08-17 09:37:59 +00:00
examples
Moved Asynchronicity From Java to C++ for AndroidVoip Demo App
2020-08-18 17:54:56 +00:00
logging
Group decoded frame events by SSRC when compressing RTC event log.
2020-08-19 09:47:20 +00:00
media
Removing call to deprecated SetExtraOptions method
2020-08-20 16:13:12 +00:00
modules
Send estimated capture clock offset when sending Abs-capture-time RTP header extension.
2020-08-20 16:23:22 +00:00
p2p
p2p_transport_channel: Add estimated disconnected time to CandidatePairChangeEvent
2020-08-20 11:40:01 +00:00
pc
Make Objective-C interface use SetDirectionWithError
2020-08-17 10:01:49 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Removing NetworkMonitorFactory static global.
2020-08-13 20:50:23 +00:00
rtc_tools
Use non-deprecated EncodedImageCallback::OnEncodedImage in rtc_tools
2020-08-12 09:17:15 +00:00
sdk
p2p_transport_channel: Add estimated disconnected time to CandidatePairChangeEvent
2020-08-20 11:40:01 +00:00
stats
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Reland "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-08-17 10:40:44 +00:00
test
Reland "Call OnReceivedOverhead after audio network adaptor is created."
2020-08-20 16:07:41 +00:00
tools_webrtc
Fix clang revision regexp in chromium autoroller to match new format
2020-08-07 09:22:03 +00:00
video
Remove RTPFragmentationHeader creation and propagation through webrtc
2020-08-17 16:37:33 +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
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
Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder
2020-08-05 09:41:49 +00:00
BUILD.gn
Reland "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-08-17 10:40:44 +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
Move definition of SpatialLayer to api/video_codecs/spatial_layer.h
2020-08-17 09:45:19 +00:00
DEPS
Roll chromium_revision 7a8fca54e0..4591d36de2 (800031:800155)
2020-08-20 17:15:43 +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
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
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +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
Reland "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-08-17 10:40:44 +00:00
whitespace.txt
Trigger CI bots.
2020-07-15 17:50:55 +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%