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webrtc_m130/webrtc
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denicija 5db450d244 iOS:Add loopback launch argument functionality in AppRTCMobile. 
It will enable us to start immediate loopback just by passing
the "loopback" string, as argument when launching.
Usefull for testing and command line profiling.

BUG=NONE

Review-Url: https://codereview.webrtc.org/2777983004
Cr-Commit-Position: refs/heads/master@{#17426}
2017-03-28 15:27:41 +00:00
..
api
iOS: Add new RTCVideoSource interface
2017-03-28 08:56:41 +00:00
audio
Don't hardcode MediaType::ANY in FakeNetworkPipe.
2017-03-28 11:59:41 +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
Don't hardcode MediaType::ANY in FakeNetworkPipe.
2017-03-28 11:59:41 +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
Remove VoECodec from FakeWebRtcVoiceEngine.
2017-03-28 12:07:15 +00:00
modules
Revert of Log created probe clusters to RtcEventLog. (patchset #1 id:1 of https://codereview.chromium.org/2783693002/ )
2017-03-28 14:59:51 +00:00
ortc
Add the function ToRtpParameters.
2017-03-09 01:15:06 +00:00
p2p
Disable possibly flaky AudioCodingModuleTest.TestPacketLossStereo for Linux.
2017-03-28 10:00:07 +00:00
pc
Pass ownership of candidate to PeerConnection::OnIceCandidate
2017-03-25 15:31:12 +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
Don't hardcode MediaType::ANY in FakeNetworkPipe.
2017-03-28 11:59:41 +00:00
tools
Attach TransportFeedbackPacketLossTracker to ANA (PLR only)
2017-03-23 18:04:48 +00:00
video
Don't hardcode MediaType::ANY in FakeNetworkPipe.
2017-03-28 11:59:41 +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
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.