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 |
|