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chromium-webrtc-autoroll 4854b9f804 Roll chromium_revision 230cc8f7e4..5cbf4ebd59 (698466:698593) 
Change log: 230cc8f7e4..5cbf4ebd59
Full diff: 230cc8f7e4..5cbf4ebd59

Changed dependencies
* src/base: b70d15bab8..7390b2c043
* src/build: ceed4a086a..0bf2b96dc4
* src/ios: 1ccb60e1f4..162f8951a2
* src/testing: bf89d559c0..ae682ebd48
* src/third_party: e6a4066287..6fbd6866c4
* src/tools: 9ee4313e8c..f2bc23dbbd
DEPS diff: 230cc8f7e4..5cbf4ebd59/DEPS

No update to Clang.

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

Change-Id: I46c846483750f7a0d711774e5db66ec0763c5a18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154020
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@{#29264}
2019-09-20 20:31:44 +00:00
api
Change DatagramTransportInterface methods to pure virtual.
2019-09-20 20:11:24 +00:00
audio
Propagating TargetRate struct to BitrateAllocator.
2019-09-19 14:03:04 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Propagating TargetRate struct to BitrateAllocator.
2019-09-19 14:03:04 +00:00
common_audio
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
common_video
Cleaning up C++14 move into lambda TODOs.
2019-09-17 19:18:26 +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 remote estimates to rtc event log.
2019-09-19 09:22:37 +00:00
media
Remove obsolete todo comment in simulcast.h
2019-09-20 14:17:20 +00:00
modules
Delete isac GetBandwidthInfo/SetBandwidthInfo
2019-09-20 13:53:52 +00:00
p2p
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
pc
Cleanup deprecated monitoring of MediaTransport state.
2019-09-20 19:44:20 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
Improve field trial error message.
2019-09-19 09:38:49 +00:00
rtc_tools
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
sdk
Adds logging of audio sessions status on the recording side in ADM for Android.
2019-09-19 11:35:10 +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
Cap h264 fuzzer input to 200k.
2019-09-20 18:15:00 +00:00
tools_webrtc
Stop explicitly setting use_prebuilt_instrumented_libraries on msan bots.
2019-09-12 18:21:38 +00:00
video
Cleanup passing rtp packet to ulpfec receiver.
2019-09-20 11:09:11 +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
Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5."
2019-09-06 05:36:23 +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 230cc8f7e4..5cbf4ebd59 (698466:698593)
2019-09-20 20:31:44 +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
Tell users where they can find the native API headers
2017-11-14 10:36:46 +00:00
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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