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webrtc/talk/session/media
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Jiayang Liu 4b3c0d6f34 Use WebRTC API to convert byteorder in srtpfilter. 
This CL uses WebRTC API to convert 64bit from big-endian to host-endian,
so the internal "be64_to_cpu" of libsrtp is not used. The code path of
"be64_to_cpu" in newer versions of libsrtp depends on compile-time
defines that are not available in WebRTC.

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

Review URL: https://webrtc-codereview.appspot.com/46749004

Cr-Commit-Position: refs/heads/master@{#8904}
2015-03-31 22:02:50 +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
rtc::Buffer: Rename length to size, for conformance with the STL
2015-03-24 09:20:19 +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
Delete NullVideoRenderer
2015-03-24 15:11:24 +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
Use WebRTC API to convert byteorder in srtpfilter.
2015-03-31 22:02:50 +00:00
srtpfilter.cc
Use WebRTC API to convert byteorder in srtpfilter.
2015-03-31 22:02:50 +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