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chromium-webrtc-autoroll 22997d66b8 Roll chromium_revision 2ad52fb2a4..d1f68eb66e (632357:632477) 
Change log: 2ad52fb2a4..d1f68eb66e
Full diff: 2ad52fb2a4..d1f68eb66e

Changed dependencies
* src/base: 61c98912a6..32407e1dea
* src/build: f7f21da60d..383614544b
* src/ios: f9421ecdbc..877c8752bb
* src/testing: 3a16217f2b..b5e8334f08
* src/third_party: 0b6da2cdde..4063eb822e
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/754352edf2..778eee0740
* src/third_party/depot_tools: 452a49f4b6..2b14855de9
* src/tools: 7df6c9fa94..0d447c418e
DEPS diff: 2ad52fb2a4..d1f68eb66e/DEPS

No update to Clang.

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

Change-Id: Ibc0e2b3aca2255bab886c326ef4a9b7f4ea00d28
Reviewed-on: https://webrtc-review.googlesource.com/c/123203
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@{#26700}
2019-02-15 03:32:52 +00:00
api
Update test code to use EncodedImage::Allocate
2019-02-14 15:50:45 +00:00
audio
Always offer transport sequence number header extension for audio
2019-02-14 15:28:07 +00:00
build_overrides
Disable hermetic toolchain when building on macOS 10.14.
2018-12-07 13:57:28 +00:00
call
Update test code to use EncodedImage::Allocate
2019-02-14 15:50:45 +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
Add generic packet rates to event_log_visualizer.
2019-02-14 22:45:37 +00:00
media
Always offer transport sequence number header extension for audio
2019-02-14 15:28:07 +00:00
modules
Update test code to use EncodedImage::Allocate
2019-02-14 15:50:45 +00:00
p2p
Remove TCPPort incoming_only_ member.
2019-02-04 19:46:30 +00:00
pc
Replace replace_substrs with Abseil
2019-02-15 01:22:54 +00:00
resources
Reland of Opus multistream.
2019-01-29 12:16:19 +00:00
rtc_base
Replace replace_substrs with Abseil
2019-02-15 01:22:54 +00:00
rtc_tools
Add generic packet rates to event_log_visualizer.
2019-02-14 22:45:37 +00:00
sdk
Move MediaConstraintsInterface to sdk/, and make it a concrete class
2019-02-14 12:07:07 +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
Add support of fast media sending in peer connection e2e test
2019-02-15 00:51:03 +00:00
tools_webrtc
Delete obsolete usage of FakeConstraints
2019-02-12 12:27:04 +00:00
video
Always offer transport sequence number header extension for audio
2019-02-14 15:28:07 +00:00
.clang-format
…
.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
Replace replace_substrs with Abseil
2019-02-15 01:22:54 +00:00
AUTHORS
Add ',' between elements in RTCStatsReport::ToJson
2019-01-08 16:05:23 +00:00
BUILD.gn
Move MediaConstraintsInterface to sdk/, and make it a concrete class
2019-02-14 12:07:07 +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 2ad52fb2a4..d1f68eb66e (632357:632477)
2019-02-15 03:32:52 +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
…
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
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Readme 255 MiB
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