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13 Commits

Author SHA1 Message Date
mflodman@webrtc.org
7a4eb2837a Calculate the available bandwidth before sending a TMMBR 
Also changed the way TMMBR was processed since it did not match the new bandwidth estimator.

Review URL: http://webrtc-codereview.appspot.com/270003
Patch from pwestin1 <pwestin@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@925 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 12:54:46 +00:00
stefan@webrtc.org
fbea4e555d Solves two bandwidth estimation issues and measures the sent video bitrate. 
Issues solved:
1. Possible overflow when reducing the bandwidth estimate at the send-side
2. A burst of loss reports could make us reduce the rate way too far since
   we reduced the rate relative the current estimate and not the actual
   rate sent.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/244011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:08:29 +00:00
stefan@webrtc.org
5eb64f06be Fix BitrateSent() API when having a default RTP module. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/242004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 13:42:50 +00:00
stefan@webrtc.org
d0bdab0128 Adding API to get sent total bitrate, FEC bitrate and NACK bitrate. 
Also adding tests for this in vie_auto_test.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/199001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 14:24:54 +00:00
pwestin@webrtc.org
1da1ce0da5 First implementation of simulcast, adds VP8 simulcast to video engine. 
Changed API to RTP module
Expanded Auto test with a test for simulcast
Made the video codec tests compile
Added the vp8_simulcast files to this cl
Added missing auto test file
Review URL: http://webrtc-codereview.appspot.com/188001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 15:19:55 +00:00
pwestin@webrtc.org
741da942ec Added support for new RTCP message REMB (remote estimated max bitrate) 
Review URL: http://webrtc-codereview.appspot.com/149001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@628 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 13:52:04 +00:00
stefan@webrtc.org
269f8a14c6 Undoing change committed in r514 since it broke bandwidth estimation 
Review URL: http://webrtc-codereview.appspot.com/132011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@531 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 09:51:59 +00:00
andrew@webrtc.org
4d905f88c6 Fix clang warnings in rtp. 
Review URL: http://webrtc-codereview.appspot.com/132006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 19:22:27 +00:00
pwestin@webrtc.org
e9f0e2eb20 Moved _rtpReceiver to protected 
Review URL: http://webrtc-codereview.appspot.com/132005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 13:16:52 +00:00
perkj@google.com
12f1fc4fe5 Fix initialization defect in constructor webrtc::ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(WebRtc_Word32, bool) initialization list. 
Review URL: http://webrtc-codereview.appspot.com/125002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 14:26:33 +00:00
hellner@google.com
977c2966fc Review URL: http://webrtc-codereview.appspot.com/109006 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 17:30:30 +00:00
marpan@google.com
80c5d7a80e Allow the setting of FEC-UEP feature on/off to be done in media_opt(VCM). 
Review URL: http://webrtc-codereview.appspot.com/71004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-15 21:32:40 +00:00
niklase@google.com
470e71d364 git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00