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webrtc_m130/talk/session/media
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solenberg 5629a1dba2 Fix flaky test TestSrtpError, introduced in https://codereview.webrtc.org/1362913004. 
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1380103002

Cr-Commit-Position: refs/heads/master@{#10137}
2015-10-01 15:46:05 +00:00
..
audiomonitor.cc
(Auto)update libjingle 73222930-> 73226398
2014-08-13 17:26:08 +00:00
audiomonitor.h
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
bundlefilter_unittest.cc
(Auto)update libjingle 73222930-> 73226398
2014-08-13 17:26:08 +00:00
bundlefilter.cc
Change GetStreamBySsrc to not copy StreamParams.
2015-01-22 23:00:41 +00:00
bundlefilter.h
(Auto)update libjingle 73222930-> 73226398
2014-08-13 17:26:08 +00:00
channel_unittest.cc
Fix flaky test TestSrtpError, introduced in https://codereview.webrtc.org/1362913004.
2015-10-01 15:46:05 +00:00
channel.cc
Remove unused SignalMediaError and infrastructure.
2015-10-01 11:10:40 +00:00
channel.h
Remove unused SignalMediaError and infrastructure.
2015-10-01 11:10:40 +00:00
channelmanager_unittest.cc
Remove use of DeviceManager from ChannelManager.
2015-09-24 07:41:59 +00:00
channelmanager.cc
Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'.
2015-10-01 09:32:41 +00:00
channelmanager.h
Remove SetAudioDelayOffset() and friends.
2015-09-24 10:53:14 +00:00
currentspeakermonitor_unittest.cc
Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository.
2014-12-19 22:29:55 +00:00
currentspeakermonitor.cc
Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository.
2014-12-16 21:09:08 +00:00
currentspeakermonitor.h
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
2014-12-18 20:31:29 +00:00
externalhmac.cc
Update libsrtp includes in preparation of roll into Chromium.
2015-03-23 22:12:19 +00:00
externalhmac.h
Update libsrtp includes in preparation of roll into Chromium.
2015-03-23 22:12:19 +00:00
mediamonitor.cc
(Auto)update libjingle 73222930-> 73226398
2014-08-13 17:26:08 +00:00
mediamonitor.h
Thread annotation of rtc::CriticalSection.
2014-09-24 07:10:57 +00:00
mediasession_unittest.cc
Reland Change WebRTC SslCipher to be exposed as number only
2015-10-01 04:49:02 +00:00
mediasession.cc
Reland Change WebRTC SslCipher to be exposed as number only
2015-10-01 04:49:02 +00:00
mediasession.h
Reland Change WebRTC SslCipher to be exposed as number only
2015-10-01 04:49:02 +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
Roll chromium_revision 8af41b3..dcb0929 (324854:325030)
2015-04-15 15:22:19 +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 Change WebRTC SslCipher to be exposed as number only
2015-10-01 04:49:02 +00:00
srtpfilter.cc
Reland Change WebRTC SslCipher to be exposed as number only
2015-10-01 04:49:02 +00:00
srtpfilter.h
Reland Change WebRTC SslCipher to be exposed as number only
2015-10-01 04:49:02 +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
(Auto)update libjingle 73222930-> 73226398
2014-08-13 17:26:08 +00:00