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chromium-webrtc-autoroll e72a78e4b2 Roll chromium_revision 42b9d99dab..e76bf0b809 (700503:700610) 
Change log: 42b9d99dab..e76bf0b809
Full diff: 42b9d99dab..e76bf0b809

Changed dependencies
* src/base: 85620fb3e5..c9cbf78733
* src/ios: eaabefdb4a..f1a036ffe0
* src/testing: 7fe65418f8..7e847b739d
* src/third_party: 87925dcbf7..be719c373a
* src/third_party/depot_tools: e656e98831..2b82936829
* src/tools: 0c38ca1e46..b4efeff828
DEPS diff: 42b9d99dab..e76bf0b809/DEPS

No update to Clang.

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

Change-Id: If7b2055753e9a42e1ec5aa8b18c57770d3a096d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154662
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@{#29329}
2019-09-27 08:39:28 +00:00
api
Allow receive-only use of datagram transport for data channels.
2019-09-26 20:01:06 +00:00
audio
Moves contents of bitrate_controller to goog_cc
2019-09-25 08:43:24 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Converts const methods in BitrateAllocator to non-member functions.
2019-09-25 11:55:13 +00:00
common_audio
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
common_video
Always pass arguments to INSTANTIATE_TEST_SUITE_P.
2019-09-24 08:56:24 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Fixing some typos.
2019-09-10 10:03:50 +00:00
examples
New build target api:media_interface
2019-09-19 09:32:27 +00:00
logging
Adds abs-send-time and size field outputs to event log parser.
2019-09-24 16:31:22 +00:00
media
Remove obsolete todo comment in simulcast.h
2019-09-20 14:17:20 +00:00
modules
pipewire: handle deleting the capturer while a D-Bus call is in progress
2019-09-26 18:58:56 +00:00
p2p
Delete the BasicPortAllocator constructor that enables gturn
2019-09-24 07:46:28 +00:00
pc
Allow receive-only use of datagram transport for data channels.
2019-09-26 20:01:06 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
Add Slice method to CopyOnWriteBuffer and use it in FEC code.
2019-09-26 09:48:07 +00:00
rtc_tools
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
sdk
Avoids update of WebRTC.Audio.SourceMatchesRecordingSession for Android < N
2019-09-26 14:59:12 +00:00
stats
Add qualityLimitationResolutionChanges stat
2019-09-09 15:22:57 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Move code related to VideoCodingModule to its own build target
2019-09-10 12:34:38 +00:00
test
Report payload byte counts in PC-level quality tests
2019-09-26 14:56:47 +00:00
tools_webrtc
Stop explicitly setting use_prebuilt_instrumented_libraries on msan bots.
2019-09-12 18:21:38 +00:00
video
VideoStreamEncoder: Don't crop and scales frames for external encoders
2019-09-26 11:49:09 +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
Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH.
2019-09-04 18:49:28 +00:00
.gn
Switch to compiling WebRTC -std=c++14 by default
2019-09-09 19:24:16 +00:00
.vpython
Add vpython dependencies needed to run presubmit tests on LUCI
2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md
Update style guide for absl::make_unique.
2019-09-18 06:10:58 +00:00
AUTHORS
pipewire: handle deleting the capturer while a D-Bus call is in progress
2019-09-26 18:58:56 +00:00
BUILD.gn
Introduce api/crypto/BUILD.gn.
2019-09-13 17:21:47 +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
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision 42b9d99dab..e76bf0b809 (700503:700610)
2019-09-27 08:39:28 +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
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Add juberti@ to webrtc root owners
2019-05-17 18:11:58 +00:00
PATENTS
…
presubmit_test_mocks.py
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +00:00
presubmit_test.py
Fixing py lint errors
2018-07-23 15:28:48 +00:00
PRESUBMIT.py
absl::make_unique presubmit check.
2019-09-17 17:47:31 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
…
README.md
…
style-guide.md
Update WebRTC's C++ style guide to reflect the switch to C++14.
2019-09-16 11:45:35 +00:00
WATCHLISTS
Add saza to audio watchlists
2019-09-03 14:55:43 +00:00
webrtc.gni
Remove rtc_use_lto GN arg.
2019-08-20 14:00:49 +00:00
whitespace.txt
Whitespace change
2018-02-23 10:34:16 +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
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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