webrtc/talk/session/media
jiayl@webrtc.org a197a5eed6 Update libsrtp includes in preparation of roll into Chromium.
This CL is in preparation to roll the libsrtp update which landed in
https://codereview.chromium.org/936663005/ into Chromium.

BUG=https://code.google.com/p/chromium/issues/detail?id=328475
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40209004

Cr-Commit-Position: refs/heads/master@{#8838}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8838 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-23 22:12:19 +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 Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. 2015-02-02 23:54:40 +00:00
channel.cc Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession. 2015-03-16 21:16:23 +00:00
channel.h Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession. 2015-03-16 21:16:23 +00:00
channelmanager_unittest.cc Refactor how the TransportChannels are set in the BaseChannel to rely lesson Session, so that in the future we can rely on Transport instead, and also be able to change Transports on the fly for BUNDLE. 2015-03-16 20:19:42 +00:00
channelmanager.cc Add thread checks to the CaptureManager. 2015-02-25 10:09:45 +00:00
channelmanager.h Add thread checks to the CaptureManager. 2015-02-25 10:09:45 +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
mediarecorder_unittest.cc Use a NULL session in unit tests that don't actually use the session. 2015-03-13 20:05:46 +00:00
mediarecorder.cc (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
mediarecorder.h (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
mediasession_unittest.cc Check associated payload type when negotiate RTX codecs. 2015-03-16 04:15:23 +00:00
mediasession.cc Check associated payload type when negotiate RTX codecs. 2015-03-16 04:15:23 +00:00
mediasession.h After another round of reviews. 2015-02-24 20:20:19 +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 Update libjingle license statements at top of talk files for consistency 2015-01-20 21:36:13 +00:00
rtcpmuxfilter_unittest.cc Update libjingle license statements at top of talk files for consistency 2015-01-20 21:36:13 +00:00
rtcpmuxfilter.cc (Auto)update libjingle 72097588-> 72159069 2014-07-29 17:36:52 +00:00
rtcpmuxfilter.h move xmpp and p2p to webrtc 2014-10-28 22:20:11 +00:00
soundclip.cc (Auto)update libjingle 72097588-> 72159069 2014-07-29 17:36:52 +00:00
soundclip.h (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
srtpfilter_unittest.cc Update libsrtp includes in preparation of roll into Chromium. 2015-03-23 22:12:19 +00:00
srtpfilter.cc Update libsrtp includes in preparation of roll into Chromium. 2015-03-23 22:12:19 +00:00
srtpfilter.h Remove or rename typedefs with _t prefixes. 2014-12-17 13:43:55 +00:00
typewrapping.h.pump (Auto)update libjingle 72097588-> 72159069 2014-07-29 17:36:52 +00:00
typingmonitor_unittest.cc move xmpp and p2p to webrtc 2014-10-28 22:20:11 +00:00
typingmonitor.cc (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
typingmonitor.h (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +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