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b1a897680d5bb2b0a6fa8f350083bc22b38162df
platform-external-webrtc/webrtc
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aleloi b1a897680d Disable AudioCodingModuleTest.TestPacketLossStereo for iOS. 
It failed twice in succession after https://codereview.webrtc.org/2752233002.

TBR=ossu@webrtc.org
NOTRY=True
BUG=webrtc:7057

Review-Url: https://codereview.webrtc.org/2778983003
Cr-Commit-Position: refs/heads/master@{#17410}
2017-03-28 09:28:08 +00:00
..
api
iOS: Add new RTCVideoSource interface
2017-03-28 08:56:41 +00:00
audio
Fix UT failure by temporarily uncommenting
2017-03-27 15:53:11 +00:00
base
Add MakeUnique from chromium and change StunMessage::AddAttribute to take a unique_ptr.
2017-03-27 23:17:19 +00:00
call
Let Call register ReceiveSideCongestionController as CallStatsObserver.
2017-03-28 08:16:25 +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
Fix cpplint errors in locations that are already being checked
2017-03-24 10:22:49 +00:00
examples
Pass settings model to ARDAppClient instead of individual settings.
2017-03-28 08:22:48 +00:00
logging
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +00:00
media
Set max bitrate for audio send stream based on RtpParameters.
2017-03-27 20:04:25 +00:00
modules
Disable AudioCodingModuleTest.TestPacketLossStereo for iOS.
2017-03-28 09:28:08 +00:00
ortc
Add the function ToRtpParameters.
2017-03-09 01:15:06 +00:00
p2p
Add MakeUnique from chromium and change StunMessage::AddAttribute to take a unique_ptr.
2017-03-27 23:17:19 +00:00
pc
Pass ownership of candidate to PeerConnection::OnIceCandidate
2017-03-25 15:31:12 +00:00
sdk
iOS: Add new RTCVideoSource interface
2017-03-28 08:56:41 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
remove more CriticalSectionWrappers.
2017-03-27 14:24:57 +00:00
test
remove more CriticalSectionWrappers.
2017-03-27 14:24:57 +00:00
tools
Attach TransportFeedbackPacketLossTracker to ANA (PLR only)
2017-03-23 18:04:48 +00:00
video
Define RtpTransportControllerSendInterface.
2017-03-27 12:36:15 +00:00
voice_engine
Fix UT failure by temporarily uncommenting
2017-03-27 15:53:11 +00:00
.gitignore
…
BUILD.gn
Enable complete_static_lib on Mac and iOS
2017-03-23 09:49:15 +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
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.