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chromium-webrtc-autoroll 7c3b10013c Roll chromium_revision 3fcb948181..3d7980bda8 (704895:705004) 
Change log: 3fcb948181..3d7980bda8
Full diff: 3fcb948181..3d7980bda8

Changed dependencies
* src/build: 2eecbbd8a6..406278e59f
* src/testing: 2ffaeb2ec9..06df520050
* src/third_party: 68fd1214ab..0152ca9a8b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/383a4e801d..284b452883
* src/third_party/depot_tools: 6a1d77869d..8e57b4bc55
* src/third_party/icu: 93a34f0ec1..5005010d69
* src/tools: f64a624030..de3668c6d6
DEPS diff: 3fcb948181..3d7980bda8/DEPS

No update to Clang.

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

Change-Id: Iec43eb7410e0cb2873eee8e19cb21bb1d502d2cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156483
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@{#29442}
2019-10-11 06:46:41 +00:00
api
Enable capturing from camera in PC framework
2019-10-10 13:06:39 +00:00
audio
Don't allocate audio if we have no transport sequence number.
2019-10-10 13:20:50 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Fix handling of large packets in RtxReceiveStream
2019-10-10 08:39:46 +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.
2019-09-30 12:52:07 +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
Cleanup includes in modules/include/module_common_types.h
2019-10-07 16:06:26 +00:00
media
Add/remove receive streams with SSRC 0 from media channels
2019-10-07 23:01:28 +00:00
modules
Send rtcp::RemoteEstimate and rtcp::TransportFeedback in one packet
2019-10-10 16:40:39 +00:00
p2p
Update RTC_LOGs in DtlsTransport to be able to distinguish errors.
2019-10-04 12:13:52 +00:00
pc
Remove RTCP tests from channel_unittest.
2019-10-10 18:33:07 +00:00
resources
Delete voice_detection() pointer to submodule
2019-10-07 13:06:05 +00:00
rtc_base
Removing AudioAllocationSettings moving functionality to AudioSendStream.
2019-10-03 10:52:16 +00:00
rtc_tools
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
sdk
Fix build errors of RTCAudioDeviceTests
2019-10-08 15:28:33 +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
Use fake clock for replay fuzzing
2019-10-10 19:03:47 +00:00
tools_webrtc
Stop using goma for MSVC bots.
2019-10-08 15:19:17 +00:00
video
Count disabled due to low bw streams or layers as bw limited quality in GetStats
2019-10-09 16:58:34 +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
SetStreams API of RtpSender wrapped for iOS and Android
2019-10-08 13:51:19 +00:00
BUILD.gn
Add missing dependencies to the static library.
2019-10-10 08:22:59 +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 3fcb948181..3d7980bda8 (704895:705004)
2019-10-11 06:46:41 +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 #COMPONENT to WebRTC.
2019-10-08 12:20:39 +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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