History log of /external/webrtc/talk/media/base/streamparams_unittest.cc
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
5237aaf243d29732f59557361b7a993c0a18cf0e 11-Nov-2015 tfarina <tfarina@chromium.org> Convert usage of ARRAY_SIZE to arraysize.

ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.

BUG=None
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1405023016

Cr-Commit-Position: refs/heads/master@{#10594}
/external/webrtc/talk/media/base/streamparams_unittest.cc
0c4e06b4c6107a1b94f764e279e4fb4161e905b0 07-Oct-2015 Peter Boström <pbos@webrtc.org> Use suffixed {uint,int}{8,16,32,64}_t types.

Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/talk/media/base/streamparams_unittest.cc
a09a99950ec40aef6421e4ba35eee7196b7a6e68 13-Aug-2014 buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> (Auto)update libjingle 73222930-> 73226398

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/streamparams_unittest.cc
d4e598d57aed714a599444a7eab5e8fdde52a950 29-Jul-2014 buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> (Auto)update libjingle 72097588-> 72159069

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/streamparams_unittest.cc
5301b0f1fce9478dfa56476e174332a1d67b053a 17-Jul-2014 pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Move additional state into WebRtcVideoSendStream.

Prevents having two places where codecs etc. are set up and allows us to
avoid creating the underlying VideoSendStream before send codecs are
set up.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6716 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/streamparams_unittest.cc
5bc25c41fc7880545052770dbcfe67f233c9b0c0 05-Dec-2013 sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Update libjingle to 57692857

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/streamparams_unittest.cc
28e20752806a492f5a6a5d343c02f9556f39b1cd 10-Jul-2013 henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/streamparams_unittest.cc