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chromium-webrtc-autoroll de21bf4cd3 Roll chromium_revision 925c16d3e7..abb1ee24a4 (690586:690691) 
Change log: 925c16d3e7..abb1ee24a4
Full diff: 925c16d3e7..abb1ee24a4

Changed dependencies
* src/base: 256225bdc9..f283357afd
* src/build: 3f22131f84..2250dc6f7c
* src/ios: f7415575d2..98c3dac131
* src/testing: edc35efc46..0f6cf75689
* src/third_party: 4c85cff6ab..796dd67cab
* src/third_party/depot_tools: 0e5fff1a88..59099ed6ee
* src/tools: 936903eeec..b872f472fb
DEPS diff: 925c16d3e7..abb1ee24a4/DEPS

No update to Clang.

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

Change-Id: Ie89f536e7add447e861d21887989d56ce7339580
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150621
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@{#28972}
2019-08-27 12:59:22 +00:00
api
Add TURN_LOGGING_ID
2019-08-27 07:18:00 +00:00
audio
Removes deprecated InsertPacket/TimeToSendPacket/TimeToSendPadding
2019-08-26 11:55:55 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Fix frames dropped statistics
2019-08-27 07:43:01 +00:00
common_audio
Remove rtc_use_lto GN arg.
2019-08-20 14:00:49 +00:00
common_video
Add new FrameRateEstimator utility class for more precis FPS estimation.
2019-08-14 12:15:06 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Implement JNI and objc implementation for Ice Candidate Pair Change event surfacing
2019-08-12 23:58:50 +00:00
logging
Delete deprecated rtc_event_log header
2019-08-07 10:58:17 +00:00
media
Fix the bug and reland "Make min video target bitrate configurable."
2019-08-27 11:12:12 +00:00
modules
Fix frames dropped statistics
2019-08-27 07:43:01 +00:00
p2p
Add TURN_LOGGING_ID
2019-08-27 07:18:00 +00:00
pc
Fix frames dropped statistics
2019-08-27 07:43:01 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
BalancedDegradationSettings: add min bitrate configuration for resolution.
2019-08-27 11:05:10 +00:00
rtc_tools
Remove rtcbot.
2019-08-19 09:26:08 +00:00
sdk
Add ObjC interface wrapping new GetImplementations method.
2019-08-23 12:06:36 +00:00
stats
Add implemented-but-missing members to RTCMediaStreamTrackStats::Members
2019-08-21 10:59:08 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Add helper functions to convert between integer milliseconds and fixed-point seconds.
2019-08-16 14:49:46 +00:00
test
Fix deadlock in VideoSendStream tests, cause of flake on some bots.
2019-08-27 10:05:07 +00:00
tools_webrtc
Sync download_tools.py with changes in gclient_utils module.
2019-08-02 08:15:15 +00:00
video
BalancedDegradationSettings: add min bitrate configuration for resolution.
2019-08-27 11:05:10 +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 Visual Studio Code project folder to gitignore file.
2019-01-21 18:42:33 +00:00
.gn
Remove last mention of ortc from the codebase.
2019-05-25 07:28:05 +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
Allowing buffering a LNTF (loss notification) feedback message in RTCPSender
2019-06-03 16:28:34 +00:00
AUTHORS
Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly
2019-06-28 19:12:14 +00:00
BUILD.gn
Migrate WebRTC test infra to ABSL_FLAG.
2019-07-19 06:54:04 +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
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision 925c16d3e7..abb1ee24a4 (690586:690691)
2019-08-27 12:59:22 +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
Add juberti@ to webrtc root owners
2019-05-17 18:11:58 +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
Disable RunPythonTests on rtc_tools.
2019-08-06 12:48:33 +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
Remove myself from OWNERS in a few places.
2019-06-10 07:57:46 +00:00
webrtc.gni
Remove rtc_use_lto GN arg.
2019-08-20 14:00:49 +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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C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%