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platform-external-webrtc/webrtc
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zhihuang d3501adf17 Create the SrtpTransportInterface. 
Create the SrtpTransportInterface, a subclass of RtpTransportInterface, which
allows the user to set the send and receive keys. The functionalities are
implemented inside the RtpTransportAdapters on top of BaseChannel.

BUG=webrtc:7013

Review-Url: https://codereview.webrtc.org/2714813004
Cr-Commit-Position: refs/heads/master@{#17023}
2017-03-03 22:39:06 +00:00
..
api
Create the SrtpTransportInterface.
2017-03-03 22:39:06 +00:00
audio
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
2017-03-02 01:02:23 +00:00
base
Fixing race between CallbackCanceled and CancelCallback in AsyncInvoker.
2017-03-03 18:33:18 +00:00
call
Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule.
2017-03-03 11:20:24 +00:00
common_audio
Enable GN check for webrtc/common_audio
2017-03-01 15:07:10 +00:00
common_video
Add performance tracing for PlatformThread and parts of the video code.
2017-03-03 15:21:18 +00:00
examples
Add Metal video view in AppRTCMobile and Metal availability macro.
2017-03-03 14:11:10 +00:00
logging
Remove unused include from rtc_event_log_parser.cc
2017-02-28 09:46:19 +00:00
media
Add performance tracing for PlatformThread and parts of the video code.
2017-03-03 15:21:18 +00:00
modules
APM configuration files generator.
2017-03-03 16:20:19 +00:00
ortc
Create the SrtpTransportInterface.
2017-03-03 22:39:06 +00:00
p2p
Removing HTTPS and SOCKS proxy server code.
2017-03-03 18:23:08 +00:00
pc
Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.
2017-03-01 23:34:36 +00:00
sdk
Add unit tests for RTCMTLVideoView.
2017-03-03 16:24:06 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Provide an inline implementation of IsEnabled.
2017-02-28 17:26:22 +00:00
test
Packet Loss Tracker - Stream Separation
2017-03-03 18:51:35 +00:00
tools
Add probe logging to RtcEventLog.
2017-02-27 10:18:46 +00:00
video
Add performance tracing for PlatformThread and parts of the video code.
2017-03-03 15:21:18 +00:00
voice_engine
Modify TransportFeedbackPacketLossTrackerTest to use parameterization
2017-03-03 19:11:06 +00:00
.gitignore
…
BUILD.gn
Revert of Removing unused audio device tests (patchset #1 id:1 of https://codereview.webrtc.org/2716843002/ )
2017-02-26 15:12:50 +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
Remove the Windows Wave audio device implementation.
2017-02-17 22:48:07 +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
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +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
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +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
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_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Setting is_component_build to false by default
2017-03-03 19:36:37 +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.