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Danil Chapovalov 02b17a5507 Add helper to calculate frame dependencies based on encoder buffer usage 
Bug: webrtc:10342
Change-Id: I1d856d060c2defcd10310f0d8639ce8a9554fff3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168194
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30458}
2020-02-05 16:19:10 +00:00
api Adds simulated time controller API. 2020-02-03 10:19:08 +00:00
audio Revert "Inlines NullAudioPoller functionality into AudioState class." 2020-01-30 18:14:11 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Move initial quality experiment to adaptation module 2020-02-05 10:10:22 +00:00
common_audio Add floating point support for writing and reading wav files 2020-01-30 13:38:19 +00:00
common_video Add helper to calculate frame dependencies based on encoder buffer usage 2020-02-05 16:19:10 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Fix links 2020-02-04 14:33:46 +00:00
examples Reformat GN files. 2020-01-21 12:13:11 +00:00
logging Reformat GN files. 2020-01-21 12:13:11 +00:00
media Allow software fallback on lowest simulcast stream for temporal support 2020-01-31 16:44:47 +00:00
modules Add helper to calculate frame dependencies based on encoder buffer usage 2020-02-05 16:19:10 +00:00
p2p AsyncTCPSocket: try sending outgoing data until EWOULDBLOCK 2020-02-03 21:19:57 +00:00
pc disallow pairing ICE-TCP with a local ip address 2020-02-05 12:20:35 +00:00
resources Reformat GN files. 2020-01-21 12:13:11 +00:00
rtc_base Define rtc::BufferT::const_iterator 2020-02-04 14:47:12 +00:00
rtc_tools Reformat GN files. 2020-01-21 12:13:11 +00:00
sdk Hold a reference to AndroidVideoTrackSource while calling onFrameCaptured. 2020-02-04 15:00:05 +00:00
stats [Stats] Include fecPackets[Reeceived/Discarded] in Members() 2020-01-28 11:22:09 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Reformat GN files. 2020-01-21 12:13:11 +00:00
test Accept undecoded frame pairs in VideoLayerAnalyzer 2020-02-03 09:46:55 +00:00
tools_webrtc Make the dashboard upload script read protos instead of JSON. 2020-01-30 10:25:47 +00:00
video Move initial quality experiment to adaptation module 2020-02-05 10:10:22 +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 Make the dashboard upload script read protos instead of JSON. 2020-01-30 10:25:47 +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 fa85f826d0..dd5a54c29b (736081:736224) 2020-01-29 04:51:45 +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 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

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