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chromium-webrtc-autoroll eedb0a1f17 Roll chromium_revision 23b4d2134b..aa7b61fdc4 (631425:631597) 
Change log: 23b4d2134b..aa7b61fdc4
Full diff: 23b4d2134b..aa7b61fdc4

Changed dependencies
* src/base: 9d3da50502..025e09f3ca
* src/build: 855557d13b..e1e623809c
* src/ios: 03c63aa0f0..f500efe6ba
* src/testing: d9cbcc099c..0704a2aafd
* src/third_party: 21818efd84..abd88f1df6
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/40bc713e02..ed0a2fef24
* src/third_party/depot_tools: 5a52587030..ae2acf7bf9
* src/tools: e366afe657..8edbc736f4
DEPS diff: 23b4d2134b..aa7b61fdc4/DEPS

No update to Clang.

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

Change-Id: I9747fe76776216b912dfd03c99f09afa300ffcca
Reviewed-on: https://webrtc-review.googlesource.com/c/122760
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@{#26656}
2019-02-13 08:41:40 +00:00
api
AEC3: Fix delay hysteresis validation
2019-02-12 12:05:20 +00:00
audio
Adds audio priority bitrate field trial parameter.
2019-02-12 16:03:22 +00:00
build_overrides
Disable hermetic toolchain when building on macOS 10.14.
2018-12-07 13:57:28 +00:00
call
Adds audio priority bitrate field trial parameter.
2019-02-12 16:03:22 +00:00
common_audio
Stop using Googletest legacy APIs.
2019-01-31 13:23:33 +00:00
common_video
Always add/rewrite VUI and set max_num_reorder_frames to 0.
2019-02-04 09:18:46 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Remove unused file.
2019-02-06 10:24:07 +00:00
logging
Stop hard-coding default IDs for RTP extensions
2019-02-09 01:04:35 +00:00
media
Delete video source proxying in WebRtcVideoSendStream
2019-02-11 12:43:31 +00:00
modules
Update H264EncoderImpl to use EncodedImage::Allocate
2019-02-12 13:33:23 +00:00
p2p
Remove TCPPort incoming_only_ member.
2019-02-04 19:46:30 +00:00
pc
Delete obsolete usage of FakeConstraints
2019-02-12 12:27:04 +00:00
resources
Reland of Opus multistream.
2019-01-29 12:16:19 +00:00
rtc_base
Add Foundation.framework to cocoa_threading target
2019-02-12 22:07:20 +00:00
rtc_tools
Do not build rtp_generator in Chromium builds.
2019-02-04 14:48:58 +00:00
sdk
Delete unused media constraints
2019-02-08 14:45:00 +00:00
stats
Expose video freeze metrics in GetStats.
2019-02-04 09:58:08 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Makes Clock interface fully mutable.
2019-01-31 08:12:05 +00:00
test
Introduce cross traffic for emulated network layer.
2019-02-12 13:18:33 +00:00
tools_webrtc
Delete obsolete usage of FakeConstraints
2019-02-12 12:27:04 +00:00
video
Adds audio priority bitrate field trial parameter.
2019-02-12 16:03:22 +00:00
.clang-format
Tune ObjC clang-format configuration
2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore format commit.
2018-06-20 09:26:44 +00:00
.gitignore
Add Visual Studio Code project folder to gitignore file.
2019-01-21 18:42:33 +00:00
.gn
Re-enable gtest absl pretty printers.
2018-08-13 13:54:05 +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
Allow use of functions in absl/algorithms
2019-01-28 18:42:16 +00:00
AUTHORS
Add ',' between elements in RTCStatsReport::ToJson
2019-01-08 16:05:23 +00:00
BUILD.gn
Add gn configs to remove the dependency to audio and video codecs.
2019-02-12 20:41:25 +00:00
CODE_OF_CONDUCT.md
Add code of conduct to WebRTC repo
2017-05-16 12:09:13 +00:00
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
Delete constant RTP_PAYLOAD_NAME_SIZE
2019-01-25 07:59:52 +00:00
DEPS
Roll chromium_revision 23b4d2134b..aa7b61fdc4 (631425:631597)
2019-02-13 08:41:40 +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
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
Update template to follow chromium copyright style
2013-04-24 01:01:28 +00:00
native-api.md
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Clean up root OWNERS.
2018-11-09 14:23:59 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
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
Add PRESUBMIT to enforce usage of new Googletest APIs.
2019-02-06 11:06:02 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
README.md
Tell users where they can find the native API headers
2017-11-14 10:36:46 +00:00
style-guide.md
Remove rule that discourages passing optional by const reference
2019-02-05 11:58:05 +00:00
WATCHLISTS
Modify pc/ WATCHLISTS definition
2019-01-31 22:09:16 +00:00
webrtc.gni
Add gn configs to remove the dependency to audio and video codecs.
2019-02-12 20:41:25 +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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