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webrtc_m130/webrtc
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jianj cb5d1150db Enable vp9 row-based multithreading. 
Improves current multi-threading performance (for the same number of threads) by:
5%-8% speedup for 2 threads, ~15% for 4 threads on Linux/Mac
with standalone libvpx.
~6% speedup on AppRTC on Linux/Mac with 4 threads.

BUG=None

Review-Url: https://codereview.webrtc.org/2776803002
Cr-Commit-Position: refs/heads/master@{#17432}
2017-03-29 06:56:08 +00:00
..
api
iOS: Add new RTCVideoSource interface
2017-03-28 08:56:41 +00:00
audio
Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
2017-03-28 16:40:59 +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
Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
2017-03-28 16:40:59 +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
iOS:Add loopback launch argument functionality in AppRTCMobile.
2017-03-28 15:27:41 +00:00
logging
Delete support for receiving RTCP RPSI and SLI message.
2017-03-22 14:15:09 +00:00
media
Change minimum DTMF event duration to be 40 milliseconds
2017-03-28 18:18:32 +00:00
modules
Enable vp9 row-based multithreading.
2017-03-29 06:56:08 +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
Change minimum DTMF event duration to be 40 milliseconds
2017-03-28 18:18:32 +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
remove more CriticalSectionWrappers.
2017-03-27 14:24:57 +00:00
test
Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
2017-03-28 16:40:59 +00:00
tools
Attach TransportFeedbackPacketLossTracker to ANA (PLR only)
2017-03-23 18:04:48 +00:00
video
Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
2017-03-28 16:40:59 +00:00
voice_engine
Experiment-driven configuration of PLR/RPLR-based FecController
2017-03-28 12:03:55 +00:00
.gitignore
…
BUILD.gn
Enable complete_static_lib on Mac and iOS
2017-03-23 09:49:15 +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
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
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
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.