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webrtc_m130/webrtc
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pbos e6fe2a9a7f Simplify default temporal layers. 
Moves towards separating which layers may be referenced instead of
referencing libvpx flags directly. This will make strategies easier to
extract and usable from hardware encoders (RTCVideoEncoder, for
instance).

BUG=chromium:702017, webrtc:7349
R=brandtr@webrtc.org, marpan@webrtc.org. sprang@webrtc.org

Review-Url: https://codereview.webrtc.org/2747123005
Cr-Commit-Position: refs/heads/master@{#17349}
2017-03-22 20:44:15 +00:00
..
api
Make lint errors fatal in presubmit and fix files in whitelisted paths
2017-03-21 14:52:41 +00:00
audio
Reland of Delete class MockCongestionController. (patchset #1 id:1 of https://codereview.webrtc.org/2762133003/ )
2017-03-22 10:28:54 +00:00
base
Add a missing DCHECK to PlatformThread::SetPriority.
2017-03-20 10:47:17 +00:00
call
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
2017-03-21 20:26:06 +00:00
common_audio
iSAC fix entropy coder: Recently added DCHECK could in fact trigger
2017-03-13 12:28:47 +00:00
common_video
Add FullStack test for simulcast screenshare mode.
2017-03-09 17:23:30 +00:00
examples
Add video codec setting to AppRTCMobile on iOS.
2017-03-17 16:01:59 +00:00
logging
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +00:00
media
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
2017-03-21 20:26:06 +00:00
modules
Simplify default temporal layers.
2017-03-22 20:44:15 +00:00
ortc
Add the function ToRtpParameters.
2017-03-09 01:15:06 +00:00
p2p
Parse the connection data in SDP (c= line).
2017-03-21 18:04:53 +00:00
pc
Parse the connection data in SDP (c= line).
2017-03-21 18:04:53 +00:00
sdk
Better handling of error condition in MediaCodecVideoEncoder.
2017-03-15 12:53:14 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Delete utf_util_win.h.
2017-03-14 08:32:50 +00:00
test
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +00:00
tools
Improve error handling for ffmpeg operations
2017-03-15 15:27:31 +00:00
video
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +00:00
voice_engine
TransportFeedbackPacketLossTracker to receive std::vector<PacketFeedback> in place of the entire feedback
2017-03-21 14:31:35 +00:00
.gitignore
…
BUILD.gn
Add bwe_simulations_tests to default build.
2017-03-22 14:48:11 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Delete unused constant kConfigParameterSize.
2017-03-22 15:19:07 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Use RTC_UNUSED instead of conditional compilation in BWE simulator tool.
2017-03-17 13:34:47 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream.h
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
2017-03-21 20:26:06 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.