c569a49a3d
Unit tests for SSLAdapter
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R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17309004
Patch from Manish Jethani <manish.jethani@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7269 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 05:56:44 +00:00
88772874da
Disabled several rtc_unittests so the tests can be turned on in the waterfall
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BUG=3836
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7254 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 07:30:48 +00:00
b2efb6771c
Put base tests in webrtc_tests.gyp
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BUG=N/A
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7140 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 17:28:19 +00:00
66a3582170
Create a copy of talk/sound under webrtc/sound.
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BUG=3379
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6986 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 22:04:04 +00:00
74aaf29a0f
Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
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The filter is an exponential filter borrowed from video coding module.
The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic.
BUG=
R=henrika@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 21:28:26 +00:00
37b4e1bbcb
webrtc/base: add dependent setting for gtest include directory that was missed when creating base_tests.gyp. Same as https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle_tests.gyp?r=6484#39
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BUG=N/A
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6574 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 16:39:17 +00:00
1e3c5c248a
Importing ThreadChecker class from Chromium
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The ThreadChecker class is imported/re-implemented from Chromium.
The implementation is changed to depend on WebRTC primitives.
R=andrew@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6446 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 11:34:44 +00:00
2bae3211b1
Add missing sources to webrtc/base/base.gyp
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During my work on setting up a GN build for
WebRTC, I discovered that the base.gyp is trying
to remove source files (for the Chromium build)
that are not added in the initial set.
I assume these files should be listed and that
GYP just doesn't complain when it's trying to
remove a file that is not present in the sources
list.
natserver_main.cc is also removed, since it's not used anywhere.
There are also a couple of other header files that are
used in other code that probably also should be listed in
base.gyp (please do this in another CL):
* compile_assert.h
* dscp.h
* move.h
* template_util.h
BUG=None
TEST=Trybots passing clobber compile step.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6438 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 07:11:19 +00:00
f048872e91
Adds a modified copy of talk/base to webrtc/base. It is the first step in
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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
2014-05-13 18:00:26 +00:00
e9a604accd
Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."
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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
2014-05-13 08:15:48 +00:00
2c7d1b39b9
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
2014-05-12 18:03:09 +00:00
