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chromium-webrtc-autoroll b3fb339c44 Roll chromium_revision 65c32b57c2..2b82941a12 (719106:719656) 
Change log: 65c32b57c2..2b82941a12
Full diff: 65c32b57c2..2b82941a12

Changed dependencies
* src/base: db1805cd24..9186496962
* src/build: d3aa45deb2..555ec37684
* src/ios: a20c344fa8..53ea13c180
* src/testing: bed6128e4e..72e16daaa5
* src/third_party: 9353bfefce..4481c01a7a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e705827b36..c9d4943dc8
* src/third_party/depot_tools: 83aafc979f..34267f83de
* src/third_party/libvpx/source/libvpx: 9b73e21c0d..b8549ed889
* src/tools: c6edd6506e..66d55b7c94
DEPS diff: 65c32b57c2..2b82941a12/DEPS

No update to Clang.

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

Change-Id: Icba28e8dc4a326ef1d24b42119eb2bbaac373bf9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160903
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@{#29941}
2019-11-27 18:59:14 +00:00
api
Removes caching SimulcastEncoderAdapter::GetEncoderInfo()
2019-11-27 17:06:54 +00:00
audio
Delete media transport integration.
2019-11-26 19:19:36 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Add string<->VideoCodecType conversion for all codec types.
2019-11-27 14:15:07 +00:00
common_audio
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
common_video
Revert "VideoFrame: Store a reference to an encoded frame"
2019-11-21 14:55:21 +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
Fix errorprone issues preventing Chromium Roll.
2019-11-27 12:52:48 +00:00
logging
Make rtc::Thread a TaskQueue
2019-11-25 11:38:27 +00:00
media
Removes caching SimulcastEncoderAdapter::GetEncoderInfo()
2019-11-27 17:06:54 +00:00
modules
Revert "Add support for RtpEncodingParameters::max_framerate"
2019-11-27 14:01:53 +00:00
p2p
make Connection::port() protected
2019-11-27 11:13:48 +00:00
pc
Revert "Add support for RtpEncodingParameters::max_framerate"
2019-11-27 14:01:53 +00:00
resources
Changed the digital AGC1 gain to properly support multichannel
2019-11-23 08:42:59 +00:00
rtc_base
Fix errorprone issues preventing Chromium Roll.
2019-11-27 12:52:48 +00:00
rtc_tools
EventLogVisualizer pacer delay plot fix.
2019-11-19 10:23:31 +00:00
sdk
Fix errorprone issues preventing Chromium Roll.
2019-11-27 12:52:48 +00:00
stats
Add totalInterFrameDelay to RTCInboundRTPStreamStats
2019-11-25 10:50:37 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Revert "Reland "Make webrtc_fuzzer_main depend on webrtc_component in Chromium.""
2019-11-23 15:10:47 +00:00
test
Restore tests that were accidently deleted during refactoring
2019-11-27 13:42:58 +00:00
tools_webrtc
Revert "Remove temporary workaround for generate_licenses."
2019-11-27 17:56:45 +00:00
video
Propagate absolute capture time through video receiving side.
2019-11-27 17:07:59 +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
Switch to compiling WebRTC -std=c++14 by default
2019-09-09 19:24:16 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md
Update style guide for absl::make_unique.
2019-09-18 06:10:58 +00:00
AUTHORS
Force alignment of generated JVM called functions.
2019-11-21 12:34:35 +00:00
BUILD.gn
Add ability to strip out logging messages from the binary
2019-11-26 16:29:50 +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 65c32b57c2..2b82941a12 (719106:719656)
2019-11-27 18:59:14 +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
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
PRESUBMIT.py
Add possibility to skip check_includes presubmit check.
2019-10-22 19:35:31 +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_lib_link_test.cc
Rewrite the lib link test to just be a binary.
2019-10-18 07:42:20 +00:00
webrtc.gni
Add ability to strip out logging messages from the binary
2019-11-26 16:29:50 +00:00
whitespace.txt
Revert "Whitespace change"
2019-11-11 14:58:20 +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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