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brandtr 43c31e7afe Make configuration logic harsher in FlexfecReceiveStream. 
Before this change, the configuration logic in FlexfecReceiveStream
tried to make unsupported configurations work, e.g., by dropping the
protection of some media streams when multiple media streams were
protected by a single FlexFEC stream. This CL makes the configuration logic
return more errors on such unsupported configurations.
This harmonizes the logic with the new configuration logic in
VideoSendStream, for the FlexfecSender.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2499963002
Cr-Commit-Position: refs/heads/master@{#15083}
2016-11-15 13:26:51 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed].
2016-11-14 09:41:56 +00:00
audio
Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
2016-11-14 19:30:16 +00:00
base
Ping the premier connection on each network with higher priority.
2016-11-09 04:04:14 +00:00
build
MB: Add new Win8 and Win10 bots.
2016-11-11 07:07:22 +00:00
call
Make configuration logic harsher in FlexfecReceiveStream.
2016-11-15 13:26:51 +00:00
common_audio
Avoid left-shifting negative values in a number of places
2016-11-15 09:45:59 +00:00
common_video
Negotiate H264 profiles in SDP
2016-11-12 17:53:08 +00:00
examples
Add ARDSettingsModelTests to apprtcmobile_test target.
2016-11-15 08:41:31 +00:00
libjingle
GN: Move xmllite_xmpp_unittests into webrtc/libjingle
2016-11-08 11:18:43 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Rename the adapt audio bitrate experiment.
2016-11-15 13:23:35 +00:00
modules
Wire up FlexfecSender in RTP module and VideoSendStream.
2016-11-15 13:25:44 +00:00
p2p
Re-enable the P2PTransportChannelMultihomedTest.TestBasic
2016-11-10 21:59:50 +00:00
pc
Negotiate H264 profiles in SDP
2016-11-12 17:53:08 +00:00
sdk
Negotiate H264 profiles in SDP
2016-11-12 17:53:08 +00:00
stats
RTCMediaStream[Track]Stats added.
2016-11-08 14:29:26 +00:00
system_wrappers
Add stats for frequency offset when converting RTP timestamp to NTP time.
2016-11-02 14:14:10 +00:00
test
Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
2016-11-14 19:30:16 +00:00
tools
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
video
Wire up FlexfecSender in RTP module and VideoSendStream.
2016-11-15 13:25:44 +00:00
voice_engine
Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
2016-11-14 19:30:16 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
First step in providing a UdpTransportChannel.
2016-11-09 21:44:13 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Revert of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2494683006/ )
2016-11-11 14:03:26 +00:00
common_types.h
Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ )
2016-11-12 16:54:50 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Revert of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2494683006/ )
2016-11-11 14:03:26 +00:00
config.h
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Revert of Issue 2434073003: Extract bitrate allocation ... (patchset #4 id:60001 of https://codereview.webrtc.org/2488833004/ )
2016-11-10 16:30:39 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Remove RED/RTX workaround from sender/receiver and VideoEngine2.
2016-11-11 11:28:38 +00:00
video_send_stream.h
Wire up FlexfecSender in RTP module and VideoSendStream.
2016-11-15 13:25:44 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Moving stun_prober target from webrtc/p2p to webrtc/examples
2016-11-01 10:09:19 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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.