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Java 2.9%
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Danil Chapovalov c6f81a71e5 Remove higher_spatial_layers from RTPVideoHeader structure as unused. 
The idea to communicate spatial dependencies with spatial layers bitmask
wasn't fully implemented and was dropped in later version of the descriptor.

Bug: webrtc:10342
Change-Id: I1ed191c3a2a9d2e1e9ddf313f781ca8257c34dfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166165
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30278}
2020-01-16 11:11:39 +00:00
api GOOG_PING: improve handshake 2020-01-15 16:09:38 +00:00
audio Cleanup: Using RtpRtcp directly from AudioSendStream 2020-01-15 19:01:50 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Remove higher_spatial_layers from RTPVideoHeader structure as unused. 2020-01-16 11:11:39 +00:00
common_audio Remove potential real-time reallocation in PushResampler 2019-12-11 13:16:37 +00:00
common_video Concatenate string literals at compile time. 2020-01-14 14:47:48 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update Linux documentation links 2020-01-10 07:51:46 +00:00
examples Concatenate string literals at compile time. 2020-01-14 14:47:48 +00:00
logging Concatenate string literals at compile time. 2020-01-14 14:47:48 +00:00
media Concatenate string literals at compile time. 2020-01-14 14:47:48 +00:00
modules Remove higher_spatial_layers from RTPVideoHeader structure as unused. 2020-01-16 11:11:39 +00:00
p2p GOOG_PING: improve handshake 2020-01-15 16:09:38 +00:00
pc Revert "Replaces SynchronousMethodCall with rtc::Thread::Invoke." 2020-01-15 12:34:35 +00:00
resources Make the high-pass filter operate in full-band 2019-12-18 16:01:24 +00:00
rtc_base Change rate stats classes to use int64_t not size_t 2020-01-15 13:46:38 +00:00
rtc_tools Add data dependency to event_log_visualizer. 2020-01-15 09:21:10 +00:00
sdk Concatenate string literals at compile time. 2020-01-14 14:47:48 +00:00
stats Concatenate string literals at compile time. 2020-01-14 14:47:48 +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 directive to make webrtc metrics optional. 2019-12-09 13:55:50 +00:00
test Change H264 depacketizer to implement VideoRtpDepacketizer interface 2020-01-15 12:26:55 +00:00
tools_webrtc [UBSan] Remove suppression for opus. 2019-12-10 08:59:30 +00:00
video Move DegradationPreference logic to the encoder queue. 2020-01-15 11:58:04 +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 Add .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Switch to compiling WebRTC -std=c++14 by default 2019-09-09 19:24:16 +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 Update Android camera switch API to allow specifying a name 2020-01-09 16:04:09 +00:00
BUILD.gn Suppress C5041 constexpr warning for MSVC 2019 2020-01-14 07:44:35 +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 1d541bc5a0..374f209d46 (732155:732255) 2020-01-16 04:48:29 +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 #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 Extract an interface from the perf results logger. 2020-01-14 06:05:02 +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 Add saza to audio watchlists 2019-09-03 14:55:43 +00:00
webrtc.gni Do not disable metrics by default. 2019-12-11 08:08:58 +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 Revert "Whitespace change" 2019-11-11 14:58:20 +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