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nisse fdbfdc9786 Let PacketRouter separate send and receive modules. 
This is in preparation for merging the ViERemb logic in packet_router,
to send REMB feedback as sender reports if possible, otherwise as
receiver reports.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2774623006
Cr-Commit-Position: refs/heads/master@{#17489}
2017-03-31 12:44:52 +00:00
..
api
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
audio
Let PacketRouter separate send and receive modules.
2017-03-31 12:44:52 +00:00
base
Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
2017-03-31 10:08:07 +00:00
call
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +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
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
MacOS: Add metal renderer and view.
2017-03-31 09:47:29 +00:00
logging
Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
2017-03-31 10:08:07 +00:00
media
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
modules
Let PacketRouter separate send and receive modules.
2017-03-31 12:44:52 +00:00
ortc
Disable the ORTC integration tests on TSan.
2017-03-28 20:45:18 +00:00
p2p
Disable possibly flaky AudioCodingModuleTest.TestPacketLossStereo for Linux.
2017-03-28 10:00:07 +00:00
pc
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
sdk
New RTCCameraVideoCapturer.
2017-03-31 12:12:12 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Remove ALL usage of CriticalSectionWrapper.
2017-03-31 09:03:55 +00:00
test
Let PacketRouter separate send and receive modules.
2017-03-31 12:44:52 +00:00
tools
Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
2017-03-31 10:08:07 +00:00
video
Let PacketRouter separate send and receive modules.
2017-03-31 12:44:52 +00:00
voice_engine
Let PacketRouter separate send and receive modules.
2017-03-31 12:44:52 +00:00
.gitignore
…
BUILD.gn
Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
2017-03-31 10:08:07 +00:00
call.h
…
codereview.settings
…
common_types.cc
…
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
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
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
…
video_frame.h
…
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 #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +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.