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Johnny Lee c53850675b Enable H.264 temporal scalability in simulcast. 
Bug: webrtc:10651
Change-Id: I58372186930ce33e925f85edb0f308657dbfe273
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142840
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28381}
2019-06-26 09:21:32 +00:00
api Reland "Cleanup of RTP references in GoogCC implementation." 2019-06-24 09:10:52 +00:00
audio Avoid triggering a false error logging when using encryptor and sending DTX. 2019-06-24 10:55:06 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Switch from RtpPacketSender to RtpPacketPacer interface usage. 2019-06-24 10:46:06 +00:00
common_audio Refactor WavWriter to use FileWrapper rather than PlatformFile 2019-06-14 10:18:28 +00:00
common_video Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker. 2019-06-20 10:24:29 +00:00
crypto Adding new top-level directory crypto/ 2019-03-08 00:35:05 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Mass refactoring: Change JNI #includes to use full paths (webrtc/). 2019-06-26 08:23:14 +00:00
logging Minor rtc_event_log_impl cleanup. 2019-06-20 11:57:54 +00:00
media Enable H.264 temporal scalability in simulcast. 2019-06-26 09:21:32 +00:00
modules Enable H.264 temporal scalability in simulcast. 2019-06-26 09:21:32 +00:00
p2p Remove flags include from p2p/base/datagram_dtls_adaptor.cc. 2019-06-25 19:27:01 +00:00
pc Also fail CreateOffer and CreateAnswer if there is a session error 2019-06-25 18:20:31 +00:00
resources Cleanup of resources from removed remote bitrate estimate test framework. 2019-06-18 10:22:01 +00:00
rtc_base BalancedDegradationSettings: Add option to configure QP thresholds. 2019-06-24 09:32:51 +00:00
rtc_tools Remove deprecated flags from compare_videos.py. 2019-06-25 19:39:01 +00:00
sdk Mass refactoring: Change JNI #includes to use full paths (webrtc/). 2019-06-26 08:23:14 +00:00
stats Implement QualityLimitationReasonTracker and expose "reason". 2019-05-28 16:23:55 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Sanity-check field trial string at initialization. 2019-06-11 14:11:06 +00:00
test Improvements to scenario video stats for scenario tests. 2019-06-24 15:42:42 +00:00
tools_webrtc Roll chromium_revision 6ae0f0cd4c..bf62d746a4 (669703:669828) + fix AndroidManifest 2019-06-18 17:10:06 +00:00
video BalancedDegradationSettings: Add option to configure QP thresholds. 2019-06-24 09:32:51 +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 format commit. 2018-06-20 09:26:44 +00:00
.gitignore Add Visual Studio Code project folder to gitignore file. 2019-01-21 18:42:33 +00:00
.gn Remove last mention of ortc from the codebase. 2019-05-25 07:28:05 +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 Allowing buffering a LNTF (loss notification) feedback message in RTCPSender 2019-06-03 16:28:34 +00:00
AUTHORS Import proto_library.gni when rtc_enable_protobuf is true 2019-02-27 09:56:42 +00:00
BUILD.gn Cleanup of resources from removed remote bitrate estimate test framework. 2019-06-18 10:22:01 +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 Move kRtpCsrcSize from common_types.h to rtp_headers.h 2019-05-10 09:43:39 +00:00
DEPS Roll chromium_revision 42482d4f53..c30c3a10ff (672061:672280) 2019-06-26 08:19:53 +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 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.py Removes legacy bitrate controller. 2019-06-11 13:16:05 +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 Remove rule that discourages passing optional by const reference 2019-02-05 11:58:05 +00:00
WATCHLISTS Remove myself from OWNERS in a few places. 2019-06-10 07:57:46 +00:00
webrtc.gni Specify min_sdk_version for unittest apks also in GN configs 2019-06-26 07:18:15 +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