solenberg@webrtc.org
8215106371
Framework for testing bandwidth estimation.
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BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2317004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5008 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 14:23:26 +00:00
pbos@webrtc.org
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
solenberg@webrtc.org
91811e2b04
Remove unused multi stream bandwidth estimator.
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BUG=
R=mflodman@webrtc.org , stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1712004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4264 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-25 20:36:14 +00:00
stefan@webrtc.org
de98478965
Update the remote bitrate estimator before passing the packet to the RTP module.
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This solves the problem of reconstructed packets biasing the bandwidth estimate.
TEST=vie_auto_test --automated, trybots
R=mflodman@webrtc.org , solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1594005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4171 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 12:15:40 +00:00
solenberg@webrtc.org
a6db54d4c9
- Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup.
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- Changed implementation of SetReceiveAbsoluteSendTimeStatus API so the RBE instance is changed when at least one channel in a group has the extension enabled.
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1553005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4113 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27 16:02:56 +00:00
pbos@webrtc.org
47ce120efb
Include files from webrtc/.. paths in remote_bitrate_estimator/
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BUG=1662
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1552004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4108 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27 12:41:33 +00:00
solenberg@webrtc.org
561990fd73
- Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp.
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- Changed RemoteBitrateObserver::OnReceivedBitrateChanged() to use a const & instead of non-const *, to avoid unnecessary copying.
- Refactored RemoteBitrateEstimatorTest so it can be instantiated for both single and multi stream BWE (first using a parameterized test, but then as a standard test fixture and a few helper functions).
- Refactored some tests in RemoteBitrateEstimatorTest into a common function CapacityDropTestHelper().
BUG=
R=andresp@webrtc.org , mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1521004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4086 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-22 19:04:19 +00:00
andresp@webrtc.org
29b2219914
Adding a factory to remote bitrate estimator and allow it to be set via config.
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Additionally:
- clean api to set remote bitrate estimator mode.
- clean api to set over use detector options.
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1448006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 12:10:58 +00:00
mikhal@webrtc.org
a73d52ca52
revert r3871
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TBR= solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1331004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3872 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-18 20:26:28 +00:00
solenberg@webrtc.org
9756017717
- Replace the BWE_MIN and BWE_MAX macros with std::min and std::max
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- Add 'virtual' to a bunch of overridden methods of RemoteBitrateEstimatorMultiStream and RemoteBitrateEstimatorSingleStream.
BUG=
Review URL: https://webrtc-codereview.appspot.com/1324005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3871 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-18 19:12:42 +00:00
pbos@webrtc.org
ff7e1303e8
WebRtc_Word32 => int32_t remote_bitrate_estimator/
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BUG=314
Review URL: https://webrtc-codereview.appspot.com/1275009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3775 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-08 10:04:37 +00:00
fischman@webrtc.org
aea96d36e3
Rename webrtc::StatsObserver to webrtc::CallStatsObserver
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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
stefan@webrtc.org
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
stefan@webrtc.org
4100b0402e
Move SSRC list to RemoteBitrateEstimator.
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BUG=1105
Review URL: https://webrtc-codereview.appspot.com/965027
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3130 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-19 10:09:20 +00:00
stefan@webrtc.org
42aa10eba7
Clarifies the bandwidth estimation interfaces.
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BUG=
Review URL: https://webrtc-codereview.appspot.com/965019
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3087 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-13 15:02:13 +00:00
andrew@webrtc.org
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