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chromium-webrtc-autoroll 8bbf9e2c6e Roll chromium_revision 002d8b5c6a..c1f96a7b93 (705236:705365) 
Change log: 002d8b5c6a..c1f96a7b93
Full diff: 002d8b5c6a..c1f96a7b93

Changed dependencies
* src/base: 59af27da20..a6334dd4f2
* src/build: bf0e194f45..549a40a4c5
* src/ios: dfd113bfb8..08c81f4c2b
* src/testing: d1271cbde1..8cd66881d6
* src/third_party: 28d3d24709..de05005b10
* src/third_party/depot_tools: 12265a8f7d..2cbae8a8b9
* src/third_party/r8: VYsSPB6QlnJQH-2vJBhK8T6r0mexvbb9klMKwnK22GoC..W_6e81yiWSvg-zaPh6Ho1HEnaAi3_tApuT7jChFwtFMC
* src/tools: 24cea76dc1..0bd6a1cb82
DEPS diff: 002d8b5c6a..c1f96a7b93/DEPS

No update to Clang.

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

Change-Id: I9149c767aefb2d4820847845935190ff63a6be95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156606
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@{#29457}
2019-10-12 02:34:08 +00:00
api
AEC3: Add support for multiple channels to the reverb modelling
2019-10-11 21:49:58 +00:00
audio
NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate
2019-10-11 08:34:53 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Reduce number of RTPVideoSender::SendVideo parameters
2019-10-11 10:59:21 +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
Extend WebRTC-Video-MinVideoBitrate to experiment per-codec
2019-10-11 15:34:33 +00:00
modules
AEC3: Add support for multiple channels to the reverb modelling
2019-10-11 21:49:58 +00:00
p2p
Update RTC_LOGs in DtlsTransport to be able to distinguish errors.
2019-10-04 12:13:52 +00:00
pc
Clarify expectation on GlobalLock
2019-10-11 13:11:11 +00:00
resources
Delete voice_detection() pointer to submodule
2019-10-07 13:06:05 +00:00
rtc_base
Extend WebRTC-Video-MinVideoBitrate to experiment per-codec
2019-10-11 15:34:33 +00:00
rtc_tools
Plot delay feedback in RTCP arrival order.
2019-10-11 13:52:10 +00:00
sdk
Clarify expectation on GlobalLock
2019-10-11 13:11:11 +00:00
stats
Add qualityLimitationResolutionChanges stat
2019-09-09 15:22:57 +00:00
style-guide
…
system_wrappers
Move code related to VideoCodingModule to its own build target
2019-09-10 12:34:38 +00:00
test
Remove AudioProcessing::level_estimator() getter
2019-10-11 18:08:17 +00:00
tools_webrtc
Stop using goma for MSVC bots.
2019-10-08 15:19:17 +00:00
video
Fix WebRTC-Video-MinVideoBitrate for VP9
2019-10-11 17:56:51 +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
…
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 002d8b5c6a..c1f96a7b93 (705236:705365)
2019-10-12 02:34:08 +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
…

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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