| e2976c87f7ba627fa1e1246f0ccfb34b4b9f3a73 |
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04-Jan-2016 |
Peter Boström <pbos@webrtc.org> |
Remove DISABLED_ON_ macros.
Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.
This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.
The change also removes gtest_disable.h as an unused include from many
other files.
BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1547343002 .
Cr-Commit-Position: refs/heads/master@{#11150}
/external/webrtc/webrtc/base/virtualsocket_unittest.cc
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| 5237aaf243d29732f59557361b7a993c0a18cf0e |
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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/webrtc/base/virtualsocket_unittest.cc
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| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
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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/webrtc/base/virtualsocket_unittest.cc
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| 38f8893235f3b80ae9ca89db66d62ca819b51c01 |
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14-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
WebRTC Bug 4865
Bug 4865: even without STUN/TURN, as long as the peer is on the open internet, the connectivity should work. This is actually a regression even for hangouts.
We need to issue the 0.0.0.0 candidate into Port::candidates_ and filter it out later. The reason is that when we create connection, we need a local candidate to match the remote candidate.
The same connection later will be updated with the prflx local candidate once the STUN ping response is received.
BUG=webrtc:4865
R=juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1274013002 .
Cr-Commit-Position: refs/heads/master@{#9708}
/external/webrtc/webrtc/base/virtualsocket_unittest.cc
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| f358aea7bff8091c608e1afd8cf395ec2702ff76 |
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18-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Fix WebRTC IP leaks.
WebRTC binds to individual NICs and listens for incoming Stun packets. Sending stun through this specific NIC binding could make OS route the packet differently hence exposing non-VPN public IP.
The fix here is
1. to bind to any address (0:0:0:0) instead. This way, the routing will be the same as how chrome/http is.
2. also, remove the any all 0s addresses which happens when we bind to all 0s.
BUG=4276
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39129004
Cr-Commit-Position: refs/heads/master@{#8418}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8418 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocket_unittest.cc
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| d3b453be17d6f91c4e1f9a5544b7b2d52d448f81 |
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14-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Remove the incremental IP address behavior from virtualsocketserver
VirtualSocketServer, when binding to any address (all 0s), will assign a unique IP address by incrementing the IP address, resulted in 0.0.0.1. However, this breaks the testing of 4276 where we bind to all 0s and expect the local address should remain all 0s.
BUG=4276
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35189004
Cr-Commit-Position: refs/heads/master@{#8370}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8370 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocket_unittest.cc
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| ec499beaf53409bbdc67dcda48cfc29ade2afa32 |
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07-Feb-2015 |
jlmiller@webrtc.org <jlmiller@webrtc.org> |
Increase testclient timeout from 1 to 5 seconds
BUG=4182
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38839004
Cr-Commit-Position: refs/heads/master@{#8285}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8285 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocket_unittest.cc
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| c732a3e5113bd64c85eeefa7a2ed3a5076e3db87 |
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10-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Re-enable allmost all base tests.
BUG=3836
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/22989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7416 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocket_unittest.cc
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| fded02c164ea4cc3d28d7f30ac9ce9d94d76ef7a |
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19-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
base: disabled several base tests on Mac so that rtc_unittests can be turned back on
BUG=N/A
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7240 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocket_unittest.cc
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| f048872e915a3ee229044ec4bc541f6cbf9e4de1 |
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13-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a modified copy of talk/base to webrtc/base. It is the first step in
migrating talk/base to webrtc/base.
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocket_unittest.cc
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| e9a604accd54ab14dbf98f99ccdcf3ae1c54d27c |
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13-May-2014 |
perkj@webrtc.org <perkj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."
This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.
http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457
> Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
>
> BUG=N/A
> R=andrew@webrtc.org, wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/12199004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocket_unittest.cc
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| 2c7d1b39b9374d2bc9bda4755fd4813db66a135c |
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12-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocket_unittest.cc
|