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chromium-webrtc-autoroll 0a55c2f6cd Roll chromium_revision 43c8771f89..7100fe3f3e (774929:775078) 
Change log: 43c8771f89..7100fe3f3e
Full diff: 43c8771f89..7100fe3f3e

Changed dependencies
* src/base: cf5969c394..7ca560de63
* src/build: 932b84b103..439d91cf49
* src/ios: 03956cbc88..0efbc367ac
* src/testing: 0ffd841e81..32baa7e2f5
* src/third_party: 4a37c7d618..e7b7d70ddf
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/15d5f65efa..92c6c3ec3b
* src/third_party/depot_tools: 84620d395a..e65444f91d
* src/third_party/google_benchmark/src: 6746c65bcf..367119482f
* src/tools: b561482b1c..f9938a9591
DEPS diff: 43c8771f89..7100fe3f3e/DEPS

No update to Clang.

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

Change-Id: Iab0b338eaeb869641001922da6cb2bef26479c09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176482
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@{#31446}
2020-06-04 12:49:53 +00:00
api
Introduce absl_deps in rtc_* C++ templates.
2020-06-04 08:41:32 +00:00
audio
Remove dependency from RtpRtcp on the Module interface.
2020-06-04 08:11:21 +00:00
build_overrides
Introduces rtc_base/synchronization/mutex.h.
2020-06-04 09:55:12 +00:00
call
Remove dependency from RtpRtcp on the Module interface.
2020-06-04 08:11:21 +00:00
common_audio
Add peah as OWNER to common_audio
2020-06-02 11:59:28 +00:00
common_video
Calculate chain_diff for DependencyDescriptor from GenericFrameInfo
2020-05-20 10:30:30 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
docs: Improve Android testing docs
2020-05-14 08:05:37 +00:00
examples
Add missing deps for android libs to unblock chromium roll
2020-06-03 15:26:21 +00:00
logging
Replace mock macros with unified MOCK_METHOD macro
2020-05-28 14:11:53 +00:00
media
[Stats] Don't attempt to aggregate empty VideoSenderInfos.
2020-06-04 09:03:52 +00:00
modules
Introduces rtc_base/synchronization/mutex.h.
2020-06-04 09:55:12 +00:00
p2p
Disable remote ICE candidate DNS lookups when the IceTransportPolicy is Relay or None
2020-05-29 16:10:54 +00:00
pc
Reland "Added mid to error messages reported during SDP apply."
2020-06-04 11:16:46 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Introduces rtc_base/synchronization/mutex.h.
2020-06-04 09:55:12 +00:00
rtc_tools
Remove dependency from RtpRtcp on the Module interface.
2020-06-04 08:11:21 +00:00
sdk
Add missing deps for android libs to unblock chromium roll
2020-06-03 15:26:21 +00:00
stats
Reland "Improve outbound-rtp statistics for simulcast"
2020-05-05 20:22:19 +00:00
style-guide
…
system_wrappers
Rename more death test to *DeathTest
2020-05-26 20:27:34 +00:00
test
Add multi head queue implementation.
2020-06-04 10:37:05 +00:00
tools_webrtc
Roll chromium_revision 0cd53b97fa..e728aee5f1 (770251:774589)
2020-06-03 21:19:55 +00:00
video
Remove dependency from RtpRtcp on the Module interface.
2020-06-04 08:11:21 +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
Introduces rtc_base/synchronization/mutex.h.
2020-06-04 09:55:12 +00:00
AUTHORS
Disable remote ICE candidate DNS lookups when the IceTransportPolicy is Relay or None
2020-05-29 16:10:54 +00:00
BUILD.gn
Introduces rtc_base/synchronization/mutex.h.
2020-06-04 09:55:12 +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
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
DEPS
Roll chromium_revision 43c8771f89..7100fe3f3e (774929:775078)
2020-06-04 12:49:53 +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
…
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
Remove runtime/references check from cpplint config.
2020-05-30 16:07:24 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
…
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
Introduce absl_deps in rtc_* C++ templates.
2020-06-04 08:41:32 +00:00
whitespace.txt
Trigger bots
2020-05-23 08:30:10 +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
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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