Taylor Brandstetter f475d365a2 Properly handle different transports having different SSL roles. ...

This meant splitting "transport_options" into audio/video/data options,
for when creating the answer, and giving "GetSslRole" a "transport_name"
parameter so we can retrieve the current role on a per-transport basis.

BUG=webrtc:4525
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1516993002 .

Cr-Commit-Position: refs/heads/master@{#11192}

2016-01-08 23:36:06 +00:00

..

audiomonitor.cc

Use suffixed {uint,int}{8,16,32,64}_t types.

2015-10-07 10:23:32 +00:00

audiomonitor.h

Use suffixed {uint,int}{8,16,32,64}_t types.

2015-10-07 10:23:32 +00:00

bundlefilter_unittest.cc

Remove BundleFilter filtering of RTCP.

2015-11-16 18:20:03 +00:00

bundlefilter.cc

Remove BundleFilter filtering of RTCP.

2015-11-16 18:20:03 +00:00

bundlefilter.h

Remove BundleFilter filtering of RTCP.

2015-11-16 18:20:03 +00:00

channel_unittest.cc

Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector

2015-12-17 11:04:24 +00:00

channel.cc

Deleted VideoCapturer::screencast_max_pixels, together with

2015-12-21 21:18:18 +00:00

channel.h

Deleted VideoCapturer::screencast_max_pixels, together with

2015-12-21 21:18:18 +00:00

channelmanager_unittest.cc

Remove cricket::VideoEncoderConfig.

2015-12-11 18:54:46 +00:00

channelmanager.cc

Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )

2015-12-19 18:14:18 +00:00

channelmanager.h

Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )

2015-12-19 18:14:18 +00:00

currentspeakermonitor_unittest.cc

Merging BaseSession code into WebRtcSession.

2015-10-14 22:02:50 +00:00

currentspeakermonitor.cc

Merging BaseSession code into WebRtcSession.

2015-10-14 22:02:50 +00:00

currentspeakermonitor.h

Merging BaseSession code into WebRtcSession.

2015-10-14 22:02:50 +00:00

externalhmac.cc

Update libsrtp includes in preparation of roll into Chromium.

2015-03-23 22:12:19 +00:00

externalhmac.h

Use suffixed {uint,int}{8,16,32,64}_t types.

2015-10-07 10:23:32 +00:00

mediamonitor.cc

Use suffixed {uint,int}{8,16,32,64}_t types.

2015-10-07 10:23:32 +00:00

mediamonitor.h

Use suffixed {uint,int}{8,16,32,64}_t types.

2015-10-07 10:23:32 +00:00

mediasession_unittest.cc

Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector

2015-12-17 11:04:24 +00:00

mediasession.cc

Properly handle different transports having different SSL roles.

2016-01-08 23:36:06 +00:00

mediasession.h

Properly handle different transports having different SSL roles.

2016-01-08 23:36:06 +00:00

mediasink.h

Adds trunk/talk folder of revision 359 from libjingles google code to

2013-07-10 00:45:36 +00:00

planarfunctions_unittest.cc

Use suffixed {uint,int}{8,16,32,64}_t types.

2015-10-07 10:23:32 +00:00

rtcpmuxfilter_unittest.cc

Add RtcpMuxPolicy support to PeerConnection.

2015-05-21 14:48:19 +00:00

rtcpmuxfilter.cc

Add RtcpMuxPolicy support to PeerConnection.

2015-05-21 14:48:19 +00:00

rtcpmuxfilter.h

Add RtcpMuxPolicy support to PeerConnection.

2015-05-21 14:48:19 +00:00

srtpfilter_unittest.cc

Reland Convert internal representation of Srtp cryptos from string to int

2015-11-19 03:42:00 +00:00

srtpfilter.cc

Fix build break in google3 import caused by https://codereview.webrtc.org/1532543003

2015-12-18 02:02:35 +00:00

srtpfilter.h

Allow remote fingerprint update during a call

2015-12-05 18:00:04 +00:00

typewrapping.h.pump

(Auto)update libjingle 72097588-> 72159069

2014-07-29 17:36:52 +00:00

voicechannel.h

Adds trunk/talk folder of revision 359 from libjingles google code to

2013-07-10 00:45:36 +00:00

yuvscaler_unittest.cc

Use suffixed {uint,int}{8,16,32,64}_t types.

2015-10-07 10:23:32 +00:00