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Sebastian Jansson 0a5ed896e2 Adds remote estimates to rtc event log. 
Bug: webrtc:10742
Change-Id: I0db998a05492603fcdeedca780d9ee3d64aa00d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151651
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29233}
2019-09-19 09:22:37 +00:00
api
Adds remote estimates to rtc event log.
2019-09-19 09:22:37 +00:00
audio
Using struct for bitrate allocation limits.
2019-09-18 17:03:59 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Adds remote estimates to rtc event log.
2019-09-19 09:22:37 +00:00
common_audio
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
common_video
Cleaning up C++14 move into lambda TODOs.
2019-09-17 19:18:26 +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
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
logging
Adds remote estimates to rtc event log.
2019-09-19 09:22:37 +00:00
media
Only handle each RTCP once.
2019-09-18 16:54:39 +00:00
modules
Implement Dependency Descriptor writer
2019-09-19 08:51:40 +00:00
p2p
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
pc
Only handle each RTCP once.
2019-09-18 16:54:39 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
Introduce integer division helpers with non-default rounding
2019-09-18 15:08:39 +00:00
rtc_tools
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
sdk
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +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
Only handle each RTCP once.
2019-09-18 16:54:39 +00:00
tools_webrtc
Stop explicitly setting use_prebuilt_instrumented_libraries on msan bots.
2019-09-12 18:21:38 +00:00
video
Revert "Fix minor regression caused by a8336d3"
2019-09-18 14:38:15 +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
Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5."
2019-09-06 05:36:23 +00:00
BUILD.gn
Introduce api/crypto/BUILD.gn.
2019-09-13 17:21:47 +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 eae7ecf757..afdb2e7a8b (697744:697871)
2019-09-19 02:33:30 +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
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Readme 255 MiB
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Java 3%
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