kwiberg@webrtc.org
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00b8f6b364
|
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
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2015-02-26 14:43:50 +00:00 |
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pkasting@chromium.org
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16825b1a82
|
Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
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2015-01-12 21:51:21 +00:00 |
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asapersson@webrtc.org
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8084f9500f
|
Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval.
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7863 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-12-10 11:04:13 +00:00 |
|
asapersson@webrtc.org
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1ae1d0c471
|
Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module).
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2383004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5139 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-11-20 12:46:11 +00:00 |
|
pbos@webrtc.org
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f5d4cb1958
|
Include files from webrtc/.. paths in video_engine/
BUG=1662
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1492004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4056 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-05-17 13:44:48 +00:00 |
|
stefan@webrtc.org
|
8ca8a71de2
|
Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode."
This reverts commit aae26db1da5803482b094357c546b8454ab1c26d.
BUG=1613
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1327008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3890 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-04-23 16:48:32 +00:00 |
|
stefan@webrtc.org
|
ccd4b2aec8
|
Add a default RTT to CallStats and use different values for buffered/real-time mode.
BUG=1613
Review URL: https://webrtc-codereview.appspot.com/1326007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-04-23 15:58:23 +00:00 |
|
fischman@webrtc.org
|
aea96d36e3
|
Rename webrtc::StatsObserver to webrtc::CallStatsObserver
to avoid ODR violations with peerconnectioninterface.h in libjingle.
Conflict introduced in
https://webrtc-codereview.appspot.com/1060005/diff/14010/webrtc/modules/interface/module_common_types.h#newcode326
TEST=none
BUG=none
Review URL: https://webrtc-codereview.appspot.com/1105011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3540 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-02-19 22:09:36 +00:00 |
|
mflodman@webrtc.org
|
b2f474e8bb
|
Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled.
This CL will be followed by another CL connecting the dots.
BUG=769
TEST=New unittest added.
Review URL: https://webrtc-codereview.appspot.com/968006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3117 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2012-11-16 13:57:26 +00:00 |
|