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chromium-webrtc-autoroll da76ad3c49 Roll chromium_revision 7a46078a3f..1f82a0280b (768198:768300) 
Change log: 7a46078a3f..1f82a0280b
Full diff: 7a46078a3f..1f82a0280b

Changed dependencies
* src/base: a4ea6a9188..d3e2c5e998
* src/build: 995e5d1e59..665b41e921
* src/ios: 9e7aab0d66..56d162893a
* src/third_party: b289306719..23a952e0a7
* src/third_party/depot_tools: 8280ae5e0a..30ef5cb437
* src/tools: 2205024628..b4bbcec8d9
DEPS diff: 7a46078a3f..1f82a0280b/DEPS

No update to Clang.

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

Change-Id: I07c92a0f49904725fdbc42a7a68d625204149300
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174985
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@{#31245}
2020-05-13 16:52:07 +00:00
api
Separate capturing device index from VideoConfig
2020-05-13 09:16:40 +00:00
audio
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
build_overrides
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
call
Don't await adaptation after deg-pref change
2020-05-13 12:14:36 +00:00
common_audio
Merge OOURA BUILD.gn files.
2020-04-21 09:06:37 +00:00
common_video
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
data
…
docs
Fix link to prerequisite SW for Android development.
2020-04-17 08:32:06 +00:00
examples
Wrap WebRTC OBJC API types with RTC_OBJC_TYPE.
2020-05-04 15:01:26 +00:00
logging
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
media
VideoTrackSourceInterface: make some newly introduced methods pure virtual.
2020-05-11 12:28:32 +00:00
modules
Reset frame queue in ScreenCapturerX11::SelectSource to fix issues with different sized monitors.
2020-05-12 22:45:05 +00:00
p2p
Add field trial to let idle connection live longer than 30s
2020-05-13 13:20:09 +00:00
pc
reorder sdes suites to not prefer gcm
2020-05-12 14:43:43 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Mark construction time members of PhysicalSocketServer as const
2020-05-13 14:37:35 +00:00
rtc_tools
Exclude frame_analyzer_host build on iOS.
2020-04-20 21:53:54 +00:00
sdk
Revert "Add an optional override for AudioRecord device"
2020-05-13 13:15:29 +00:00
stats
Reland "Improve outbound-rtp statistics for simulcast"
2020-05-05 20:22:19 +00:00
style-guide
…
system_wrappers
Do not trigger -Wimplicit-int-float-conversion
2020-04-09 11:43:14 +00:00
test
Separate capturing device index from VideoConfig
2020-05-13 09:16:40 +00:00
tools_webrtc
Fix the name of the file generated by generate_sslroots.py.
2020-04-25 13:37:30 +00:00
video
VideoStreamEncoderTest: Wait for QP usage handled callback event.
2020-05-13 15:23:25 +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
Fix typo in abseil-in-webrtc.md.
2019-12-18 14:27:34 +00:00
AUTHORS
IWYU: uint32_t is defined in cstdint
2020-05-07 17:04:15 +00:00
BUILD.gn
VoIP API implementation on top of AudioIngress/Egress
2020-05-05 19:55:29 +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 7a46078a3f..1f82a0280b (768198:768300)
2020-05-13 16:52:07 +00:00
ENG_REVIEW_OWNERS
…
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
Fix public_deps presubmit and gn format fighting each other.
2020-01-30 11:22:46 +00:00
pylintrc
…
README.chromium
…
README.md
Fix link in documentation. (take 2)
2020-04-16 11:08:43 +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
Android: Remove min_sdk_version in GN (reland)
2020-05-13 15:46:45 +00:00
whitespace.txt
Trigger bots again.
2020-05-11 05:38:59 +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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