admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/talk/session/media
History
Nico Weber dbe5bd9ad5 Delete unused function SetSessionError. 
https://webrtc-codereview.appspot.com/47589004/ remove the use.

BUG=505316
Originally reviewed at https://codereview.webrtc.org/1296103002/
TBR=sergeyu@chromium.org

Review URL: https://codereview.webrtc.org/1299703002 .

Cr-Commit-Position: refs/heads/master@{#9719}
2015-08-17 18:14:26 +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
Support generation of EC keys using P256 curve and support ECDSA certs.
2015-08-17 12:09:10 +00:00
channel.cc
Delete unused function SetSessionError.
2015-08-17 18:14:26 +00:00
channel.h
Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well.
2015-08-07 23:05:42 +00:00
channelmanager_unittest.cc
VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
2015-05-29 13:05:52 +00:00
channelmanager.cc
Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it.
2015-08-05 10:26:01 +00:00
channelmanager.h
VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
2015-05-29 13:05:52 +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
Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
2015-08-11 11:13:00 +00:00
mediasession.cc
Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
2015-08-11 11:13:00 +00:00
mediasession.h
Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
2015-08-11 11:13:00 +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
soundclip.cc
Remove Soundclip handling from libjingle.
2015-05-19 09:37:39 +00:00
soundclip.h
Remove Soundclip handling from libjingle.
2015-05-19 09:37:39 +00:00
srtpfilter_unittest.cc
Use WebRTC API to convert byteorder in srtpfilter.
2015-03-31 22:02:50 +00:00
srtpfilter.cc
Protect access to shared list of SRTP sessions.
2015-05-28 23:21:00 +00:00
srtpfilter.h
Prevent potential double-free if srtp_create fails.
2015-05-27 21:41:52 +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