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kthelgason b12a3e3427 Synchronize task queue operations in QualityScaler tests. 
These tests were disabled due to flakiness when running on the bots.
Hopefully synchronizing all operations that run on Task Queue will
fix this.

BUG=webrtc:6799

Review-Url: https://codereview.webrtc.org/2774643002
Cr-Commit-Position: refs/heads/master@{#17463}
2017-03-30 08:04:55 +00:00
..
api
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
audio
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
2017-03-30 06:57:43 +00:00
base
Add Darwin thread.h implementation.
2017-03-29 09:32:36 +00:00
call
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +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
iOS:Add loopback launch argument functionality in AppRTCMobile.
2017-03-28 15:27:41 +00:00
logging
Visualize events related to probing in the total bitrate graph.
2017-03-29 14:28:54 +00:00
media
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
2017-03-30 06:57:43 +00:00
modules
Synchronize task queue operations in QualityScaler tests.
2017-03-30 08:04:55 +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
Minor Cleanup of RTCAudioSource.
2017-03-28 11:35:58 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
test
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
2017-03-30 06:57:43 +00:00
tools
Visualize events related to probing in the total bitrate graph.
2017-03-29 14:28:54 +00:00
video
Add back sender_time and receiver_time metrics to full stack tests.
2017-03-30 08:02:15 +00:00
voice_engine
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
.gitignore
…
BUILD.gn
Add Darwin thread.h implementation.
2017-03-29 09:32:36 +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.