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Taylor Brandstetter 21c80320ca Expose enableDscp in Obj-C API.
network_priority was already exposed, but without the ability to set
enable_dscp, it wasn't actually doing anything.

Bug: webrtc:5658
Change-Id: I092bc3dd46e3e7be363313203428bccfccccf3c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171641
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30951}
2020-03-31 19:58:15 +00:00
api Introduce TransformableFrameInterface. 2020-03-30 13:35:26 +00:00
audio Insert audio frame transformer between encoder and packetizer. 2020-03-31 11:14:00 +00:00
build_overrides Purge phoglund from most OWNERS files. 2020-03-09 14:08:30 +00:00
call [InsertableStreams] Set video frame transformer if RTP stream already started. 2020-03-31 14:07:29 +00:00
common_audio Changed fft4g to be built as C++ 2020-03-20 12:10:16 +00:00
common_video Remove wildcard ownership for build files. 2020-02-19 14:05:46 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Rename index.md to README.md to make it automatically show up 2020-03-06 10:43:51 +00:00
examples Chromium refactor: Replace "resources_dirs" with "sources" 2020-03-25 19:32:15 +00:00
logging Replace DataSize and DataRate factories with newer versions 2020-02-18 16:09:50 +00:00
media Regression test for SCTP transport. 2020-03-31 19:53:13 +00:00
modules Fixes TaskQueuePacedSender padding while only sending non-paced audio. 2020-03-31 14:48:15 +00:00
p2p Enforce "comprehension-required" STUN rules. 2020-03-28 02:07:49 +00:00
pc Reland "Distinguish between send and receive codecs" 2020-03-29 21:03:27 +00:00
resources iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00
rtc_base Add test for relay bandwidth capping. 2020-03-30 13:02:46 +00:00
rtc_tools Add packet rate plots to event_log_visualizer. 2020-03-30 14:46:41 +00:00
sdk Expose enableDscp in Obj-C API. 2020-03-31 19:58:15 +00:00
stats Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API 2020-03-11 12:08:32 +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 RTC_EXPORT to webrtc::Clock 2020-03-10 23:00:21 +00:00
test Remove the histogram flag and all Chart JSON code. 2020-03-28 13:44:43 +00:00
tools_webrtc Revert "Trigger CI bots." 2020-03-28 22:37:03 +00:00
video [InsertableStreams] Set video frame transformer if RTP stream already started. 2020-03-31 14:07:29 +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 .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Reenable libaom decoder by default 2020-03-18 18:04:41 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Fix typo in abseil-in-webrtc.md. 2019-12-18 14:27:34 +00:00
AUTHORS Fix bad frees in error paths of WebRtcAecm_Create 2020-03-21 23:01:47 +00:00
BUILD.gn Audio egress implementation for initial voip api in api/voip. 2020-03-27 18:45:43 +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 Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery" 2020-02-07 08:23:58 +00:00
DEPS Roll chromium_revision a0d7df3386..4d555ede52 (754491:754603) 2020-03-30 21:15:24 +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 Remove phoglund as root owner. 2020-03-30 12:15:56 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Fix public_deps presubmit and gn format fighting each other. 2020-01-30 11:22:46 +00:00
presubmit_test.py Use source_sets in component builds and static_library in release builds. 2019-10-17 21:17:18 +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 Add guidance to style guide how to reference a bug in a TODO 2019-12-11 11:55:52 +00:00
WATCHLISTS Remove benwright@webrtc.org from WATCHLISTS 2020-01-31 18:46:52 +00:00
webrtc.gni Reformat GN files. 2020-01-21 12:13:11 +00:00
webrtc_lib_link_test.cc Rewrite the lib link test to just be a binary. 2019-10-18 07:42:20 +00:00
whitespace.txt Whitespace change 2020-01-28 07:53:15 +00:00

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