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a5cb98cbbd
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Breaking out RTP header parsing from the RTP module.
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video.
Moving bandwidth estimation before the RTP module is also required for RTX.
TEST=vie_auto_test, voe_auto_test, trybots.
BUG=1811
R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1545004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-05-29 12:12:51 +00:00 |
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7ebbea14a9
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Add handling of the absolute send time header extension to the rtp_rtcp module.
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1480004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4041 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-05-16 11:10:31 +00:00 |
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7bc465bd21
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Fix issues with incorrect wrap checks when having big buffers and high bitrate.
Introduces shared functions for timestamp and sequence number wrap checks.
BUG=1607
TESTS=trybots
Review URL: https://webrtc-codereview.appspot.com/1291005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3833 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-11 17:48:02 +00:00 |
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ab9202b673
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Removing remaining WebRtc_Word32 not in typedefs.h
BUG=
Review URL: https://webrtc-codereview.appspot.com/1306006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3813 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-10 17:59:17 +00:00 |
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03e3117d87
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Removed redundant VP8 width/height and made sure the generic width/height is set.
Review URL: https://webrtc-codereview.appspot.com/1158005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3656 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-03-12 09:59:27 +00:00 |
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aea96d36e3
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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
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2013-02-19 22:09:36 +00:00 |
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b586507986
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Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages.
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
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2013-02-01 14:33:42 +00:00 |
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ae1a58bba4
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Replace AudioFrame's operator= with CopyFrom().
Enforce DISALLOW_COPY_AND_ASSIGN to catch offenders.
Review URL: https://webrtc-codereview.appspot.com/1031007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3395 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-01-22 04:44:30 +00:00 |
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418443c531
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Remove operator overloading from RTPFragmentationHeader.
Instead supply a CopyFrom() method.
TEST=vie_auto_test
Review URL: https://webrtc-codereview.appspot.com/972004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3158 4adac7df-926f-26a2-2b94-8c16560cd09d
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2012-11-23 19:17:23 +00:00 |
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14b43beb7c
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Move src/ -> webrtc/
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
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2012-10-22 18:19:23 +00:00 |
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