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Java 2.9%
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Per Kjellander ee153c92fe Send rtcp::RemoteEstimate and rtcp::TransportFeedback in one packet 
Change-Id: I53912f4e82a9fd795f8886d6b2cdb313bde08c4d
BUG: webrtc:10742
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156380
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29437}
2019-10-10 16:40:39 +00:00
api Enable capturing from camera in PC framework 2019-10-10 13:06:39 +00:00
audio Don't allocate audio if we have no transport sequence number. 2019-10-10 13:20:50 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Fix handling of large packets in RtxReceiveStream 2019-10-10 08:39:46 +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 Cleanup includes in modules/include/module_common_types.h 2019-10-07 16:06:26 +00:00
media Add/remove receive streams with SSRC 0 from media channels 2019-10-07 23:01:28 +00:00
modules Send rtcp::RemoteEstimate and rtcp::TransportFeedback in one packet 2019-10-10 16:40:39 +00:00
p2p Update RTC_LOGs in DtlsTransport to be able to distinguish errors. 2019-10-04 12:13:52 +00:00
pc Revert "Implement rollback for setRemoteDescription" 2019-10-10 09:09:14 +00:00
resources Delete voice_detection() pointer to submodule 2019-10-07 13:06:05 +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 Fix build errors of RTCAudioDeviceTests 2019-10-08 15:28:33 +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 Enable capturing from camera in PC framework 2019-10-10 13:06:39 +00:00
tools_webrtc Stop using goma for MSVC bots. 2019-10-08 15:19:17 +00:00
video Count disabled due to low bw streams or layers as bw limited quality in GetStats 2019-10-09 16:58:34 +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 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 SetStreams API of RtpSender wrapped for iOS and Android 2019-10-08 13:51:19 +00:00
BUILD.gn Add missing dependencies to the static library. 2019-10-10 08:22:59 +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 651f5a2987..a1c9c88904 (704530:704650) 2019-10-10 16:37:30 +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 Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Delete unused I420 "codec" 2018-12-18 12:30:58 +00:00
OWNERS Add #COMPONENT to WebRTC. 2019-10-08 12:20:39 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py absl::make_unique presubmit check. 2019-09-17 17:47:31 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +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