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Florent Castelli a8336d3cf4 Connect the stable target rate to the video encoders
The stable target rate is used to make smarter choices in the rate
to chose which layers to enable in SVC or simulcast modes.
the addition of hysteresis, we can improve a call quality by reducing
the amount of resolution switch.


Bug: webrtc:10126
Change-Id: I04d0df9e6bbe247e2f2a668207ff74d475e2464c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150642
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29112}
2019-09-09 15:06:51 +00:00
api Connect the stable target rate to the video encoders 2019-09-09 15:06:51 +00:00
audio Deprecate SingleThreadedTaskQueueForTesting class. 2019-09-03 10:31:30 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Deprecate SingleThreadedTaskQueueForTesting class. 2019-09-03 10:31:30 +00:00
common_audio Remove rtc_use_lto GN arg. 2019-08-20 14:00:49 +00:00
common_video Add new FrameRateEstimator utility class for more precis FPS estimation. 2019-08-14 12:15:06 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM 2019-09-09 08:10:41 +00:00
logging Delete deprecated rtc_event_log_factory_interface.h 2019-09-05 08:57:36 +00:00
media Reland "New build target p2p:stun_types" 2019-09-06 10:14:38 +00:00
modules Check input parameter in RemoteEstimatorProxy::IncomingPacket without lock 2019-09-09 11:51:12 +00:00
p2p Use obfuscated IPs in logging in p2p/ and pc/. 2019-09-06 21:51:56 +00:00
pc Make payload type demux conditional on media direction 2019-09-07 00:08:51 +00:00
resources Use the AEC3 high-pass filter for the whole APM 2019-08-23 20:04:10 +00:00
rtc_base Implement stable rate support in SimulcastRateAllocator 2019-09-06 11:17:30 +00:00
rtc_tools Revert "Always create output_dir in setup_apprtc.py." 2019-09-04 18:19:12 +00:00
sdk Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM 2019-09-09 08:10:41 +00:00
stats Add support for RTCTransportStats.selectedCandidatePairChanges 2019-08-28 13:22:08 +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 helper functions to convert between integer milliseconds and fixed-point seconds. 2019-08-16 14:49:46 +00:00
test Add support of displaying video during the PC level test 2019-09-09 14:22:50 +00:00
tools_webrtc Add license for android_ndk 2019-08-28 14:57:28 +00:00
video Connect the stable target rate to the video encoders 2019-09-09 15:06:51 +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 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 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 Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5." 2019-09-06 05:36:23 +00:00
BUILD.gn Make the RtpHeaderParserImpl available to tests and tools only. 2019-08-29 15:56:40 +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 82910f739a..75cf3925c2 (694601:694706) 2019-09-09 12:32:32 +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_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +00:00
PRESUBMIT.py Disable RunPythonTests on rtc_tools. 2019-08-06 12:48:33 +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 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

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