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chromium-webrtc-autoroll d21db5d67a Roll chromium_revision e2b55cc552..35431c5114 (703005:703133) 
Change log: e2b55cc552..35431c5114
Full diff: e2b55cc552..35431c5114

Changed dependencies
* src/base: 69fdc2eb93..90b97acc04
* src/build: 408555bc71..cf8d1d9646
* src/ios: b31c7692b8..755d7028c1
* src/testing: f458a3f95d..33de9cd815
* src/third_party: 5b4e412a1e..7656f5d8ad
* src/third_party/depot_tools: aa4d8a7560..d696f20129
* src/tools: 18ad419fc8..344b646ecf
DEPS diff: e2b55cc552..35431c5114/DEPS

No update to Clang.

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

Change-Id: I637d0c738997a9fcd48f58b4ac80c22f378f3da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155780
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@{#29388}
2019-10-05 04:34:00 +00:00
api
Delete VCMEncodedFrame::VerifyAndAllocate
2019-10-04 14:16:49 +00:00
audio
Adds support for passing a vector of packets to the paced sender.
2019-10-04 08:56:11 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Adds RTPSenderVideo::Config struct with red/ulpfec config
2019-10-04 14:19:49 +00:00
common_audio
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
common_video
Always pass arguments to INSTANTIATE_TEST_SUITE_P.
2019-09-30 12:52:07 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Fixing some typos.
2019-09-10 10:03:50 +00:00
examples
New build target api:media_interface
2019-09-19 09:32:27 +00:00
logging
Adds abs-send-time and size field outputs to event log parser.
2019-09-24 16:31:22 +00:00
media
SimulcastEncoderAdapter: support per layer fallback and single encoder proxying
2019-10-01 17:31:44 +00:00
modules
Fixed issue with H264 packet buffer where it was not detecting presence of sps/pps for idr frames
2019-10-04 14:56:05 +00:00
p2p
Update RTC_LOGs in DtlsTransport to be able to distinguish errors.
2019-10-04 12:13:52 +00:00
pc
Delete old placeholders for moved api/ header files
2019-10-01 12:23:19 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
Removing AudioAllocationSettings moving functionality to AudioSendStream.
2019-10-03 10:52:16 +00:00
rtc_tools
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
sdk
Avoid memcpy in JavaToNativeEncodedImage
2019-10-01 12:55:44 +00:00
stats
Add qualityLimitationResolutionChanges stat
2019-09-09 15:22:57 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Move code related to VideoCodingModule to its own build target
2019-09-10 12:34:38 +00:00
test
Account for IP and UDP headers in emulated network
2019-10-04 12:32:02 +00:00
tools_webrtc
Replacing /target:target with /target in BUILD autofix.
2019-10-04 10:07:42 +00:00
video
Reset QP sum when QP is not reported on decoded frame.
2019-10-03 16:17:00 +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
Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH.
2019-09-04 18:49:28 +00:00
.gn
Switch to compiling WebRTC -std=c++14 by default
2019-09-09 19:24:16 +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
Update style guide for absl::make_unique.
2019-09-18 06:10:58 +00:00
AUTHORS
pipewire: handle deleting the capturer while a D-Bus call is in progress
2019-09-26 18:58:56 +00:00
BUILD.gn
Cleanup of simple TODO(srte) comments.
2019-10-04 07:57:16 +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 e2b55cc552..35431c5114 (703005:703133)
2019-10-05 04:34:00 +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
absl::make_unique presubmit check.
2019-09-17 17:47:31 +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
Update WebRTC's C++ style guide to reflect the switch to C++14.
2019-09-16 11:45:35 +00:00
WATCHLISTS
Add saza to audio watchlists
2019-09-03 14:55:43 +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
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%