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chromium-webrtc-autoroll 26d0876814 Roll chromium_revision 339f6a582b..92e6bfe90b (630696:630806) 
Change log: 339f6a582b..92e6bfe90b
Full diff: 339f6a582b..92e6bfe90b

Changed dependencies
* src/base: a9b919bb6e..a563b8b714
* src/build: c7f76dd385..a4ea4a2675
* src/ios: cf21a08042..73d4a7373f
* src/testing: 2c5dd7a4c5..99f2bdd254
* src/third_party: 03a9db4381..5895ca7d37
* src/tools: a21b4c33be..071d2064f5
DEPS diff: 339f6a582b..92e6bfe90b/DEPS

No update to Clang.

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

Change-Id: If668945cb8e1c370e9a3e87e2484aafa24209435
Reviewed-on: https://webrtc-review.googlesource.com/c/122340
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@{#26639}
2019-02-11 17:35:08 +00:00
api
Revert "Reland "Partial frame capture API part 1""
2019-02-11 14:20:37 +00:00
audio
Propagate base minimum delay to audio_receiver_stream
2019-02-06 11:07:42 +00:00
build_overrides
Disable hermetic toolchain when building on macOS 10.14.
2018-12-07 13:57:28 +00:00
call
H.264 temporal layers w/frame marking (PART 2/3)
2019-02-09 16:47:09 +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
Remove crit_render_ lock from webrtc::GainControlImpl
2019-02-11 14:14:40 +00:00
p2p
Remove TCPPort incoming_only_ member.
2019-02-04 19:46:30 +00:00
pc
Split rtp_receiver for readability.
2019-02-11 12:47:51 +00:00
resources
Reland of Opus multistream.
2019-01-29 12:16:19 +00:00
rtc_base
Make some new rtc_base targets publicly visible
2019-02-11 08:58:58 +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
Remove crit_render_ lock from webrtc::GainControlImpl
2019-02-11 14:14:40 +00:00
tools_webrtc
Resume rolling buildtools, now as chromium/src/buildtools
2019-02-08 10:36:39 +00:00
video
Revert "Partial frame capture API part 3"
2019-02-11 12:13:17 +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
Set WEBRTC_USE_H264 in common_config
2019-02-08 10:17:04 +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 339f6a582b..92e6bfe90b (630696:630806)
2019-02-11 17:35:08 +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
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
Override default manifest from Chromium in WebRTC.
2019-01-24 13:00:49 +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
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Readme 255 MiB
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