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chromium-webrtc-autoroll 3588394b26 Roll chromium_revision d026ac796d..103665932a (631722:631883) 
Change log: d026ac796d..103665932a
Full diff: d026ac796d..103665932a

Changed dependencies
* src/base: 354a74e0aa..4c05ce85b3
* src/build: 2030c341b0..46a607ce3a
* src/ios: b963feb787..dd5c3fa929
* src/testing: c84f277b30..11c391f12e
* src/third_party: e249a3f77a..341e254d24
* src/third_party/libvpx/source/libvpx: ce4336c2ab..e2381829e9
* src/tools: 4bf6f3fed8..65d01f0881
DEPS diff: d026ac796d..103665932a/DEPS

No update to Clang.

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

Change-Id: Ibf379a1a606c08828bfca800c7e2fc67c424547a
Reviewed-on: https://webrtc-review.googlesource.com/c/122930
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@{#26671}
2019-02-13 22:01:33 +00:00
api
IceTransportWithTransportChannel: Initialize |thread_checker_| in declaration
2019-02-13 12:45:12 +00:00
audio
Adds CallEncoder to ChannelSend.
2019-02-13 15:01:53 +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
Parse XR, FIR and PLI in rtc_event_log_parser.cc
2019-02-13 19:11:42 +00:00
media
Add settings to turn off VP8 base layer qp limit
2019-02-13 11:54:19 +00:00
modules
Parse XR, FIR and PLI in rtc_event_log_parser.cc
2019-02-13 19:11:42 +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
Handle HKDF key derivation when building with OpenSSL.
2019-02-13 17:44:02 +00:00
rtc_tools
Fix -Wunused-result warnings
2019-02-13 09:24:08 +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
Ensure TestPeers are destroyed at the end of Run.
2019-02-13 12:44:02 +00:00
tools_webrtc
Delete obsolete usage of FakeConstraints
2019-02-12 12:27:04 +00:00
video
Update EncodedFrameForMediaTransport to use Retain() rather than set_buffer + memcpy.
2019-02-13 15:08:53 +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 d026ac796d..103665932a (631722:631883)
2019-02-13 22:01:33 +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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