2dd3134e50
Add stats for duplicate sent and received NACK requests.
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R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7559 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 12:42:30 +00:00
2f4b14e3f3
Make RTCP sender report send media bytes.
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r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 15:25:39 +00:00
8098e07478
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
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Add counter to RTCP sender and RTCP receiver.
Add video api GetRtcpPacketTypes().
BUG=2638
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 11:59:02 +00:00
a6ad6e5b58
Add callbacks for send channel rtcp statistics
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BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5220 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 09:48:44 +00:00
766154aa1d
Removed unused code.
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R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 08:35:50 +00:00
7d6bd22019
Propagate estimated RTT from receivers to rtt observer.
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BUG=1613
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 12:14:34 +00:00
8469f7b328
Added support for sending and receiving RTCP XR packets:
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- Receiver reference time report block
- DLRR report block (RFC3611).
BUG=1613
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2196010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4898 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 13:15:34 +00:00
28a331eede
Add support for multiple report blocks.
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Use a weighted average of fraction loss for bandwidth estimation.
TEST=trybots and vie_auto_test --automated
BUG=1811
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2198004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4762 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-17 07:49:56 +00:00
822fbd8b68
Update talk to 50918584.
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Together with Stefan's http://review.webrtc.org/1960004/ .
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2048004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 23:38:54 +00:00
aa4d96a134
Revert r4301
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R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +00:00
66b2e5c05a
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
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rtp_rtcp implementation.
This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.
With this change the dead-or-alive and packet timeout APIs are removed.
TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org , pbos@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1745004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-05 14:30:48 +00:00
d900e8bea8
Proper spacing for end-of-namespace comments.
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BUG=
R=mflodman@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1760006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-03 15:12:26 +00:00
a048d7cb0a
Include files from webrtc/.. paths in rtp_rtcp/
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BUG=1662
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1557004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 14:27:38 +00:00
8ca8a71de2
Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode."
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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
ccd4b2aec8
Add a default RTT to CallStats and use different values for buffered/real-time mode.
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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
2f44673d66
WebRtc_Word32 => int32_t for rtp_rtcp/
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BUG=314
Review URL: https://webrtc-codereview.appspot.com/1279007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-08 11:08:41 +00:00
b586507986
Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages.
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Also make sure RTT is computed independently of whether it's time to send RTCP messages or not.
BUG=1298
Review URL: https://webrtc-codereview.appspot.com/1060005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 14:33:42 +00:00
a678a3baee
Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
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TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1044004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-21 07:42:11 +00:00
20ed36dada
Break out RtpClock to system_wrappers and make it more generic.
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The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.
Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.
TEST=vie_auto_test, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1041004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 14:01:20 +00:00
2f225cadde
Add logs when no RTCP RR has been received for three regular RTCP intervals.
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BUG=1267
TEST=Unittest added.
Review URL: https://webrtc-codereview.appspot.com/1019006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 13:54:43 +00:00
14b43beb7c
Move src/ -> webrtc/
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TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00
