pbos@webrtc.org
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38344ed280
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Move thread_annotations.h to webrtc/base/.
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-09-24 06:05:00 +00:00 |
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stefan@webrtc.org
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168f23faa5
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Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-07-11 13:44:02 +00:00 |
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stefan@webrtc.org
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4ef438e2de
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Remove the send-side cname getter APIs from voice and video engine.
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-07-11 09:55:30 +00:00 |
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pbos@webrtc.org
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de1429e9ad
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Add thread annotations to Call API.
Also constified a lot of pointers and reordered members to make
protected members more grouped together.
R=kjellander@webrtc.org, stefan@webrtc.org
BUG=2770
Review URL: https://webrtc-codereview.appspot.com/15399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-04-28 13:00:21 +00:00 |
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henrik.lundin@webrtc.org
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b10363f3b6
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Re-landing "Routing SuspendChange to VideoSendStream::Stats"
This was originally committed as r5687, but reverted due to a flaky
test.
BUG=3040
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5695 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-03-13 13:31:21 +00:00 |
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henrik.lundin@webrtc.org
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be39470203
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Revert "Routing SuspendChange to VideoSendStream::Stats"
The test VideoSendStreamTest.SuspendBelowMinBitrate seems flaky.
Reverting and investigating.
BUG=3040
Review URL: https://webrtc-codereview.appspot.com/9799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5681 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-03-11 17:13:14 +00:00 |
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henrik.lundin@webrtc.org
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1598b80f52
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Routing SuspendChange to VideoSendStream::Stats
Also checking that the statistics are properly updated in
VideoSendStreamTest.SuspendBelowMinBitrate.
Adding a test to SendStatisticsProxyTest.
Checking callback status in rampup test, too.
BUG=2457
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5678 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-03-11 14:57:35 +00:00 |
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sprang@webrtc.org
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09315705b9
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Wire up statistics in video receive stream of new API
This CL includes Call tests that test both send and receive sides.
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-02-07 12:06:29 +00:00 |
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sprang@webrtc.org
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ccd42840bc
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Wire up statistics in video send stream of new video engine api
Note, this CL does not contain any tests. Those are implemeted as call
tests and will be submitted when the receive stream is wired up as well.
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5559006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-01-07 09:54:34 +00:00 |
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