Commit Graph

10 Commits

Author SHA1 Message Date
pbos@webrtc.org
3c10758b3b Check before send/receive rtp header extensions.
BUG=1788
R=pbos@webrtc.org, tommi@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13949004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6744 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-20 15:27:35 +00:00
stefan@webrtc.org
168f23faa5 Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.

R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 13:44:02 +00:00
stefan@webrtc.org
4ef438e2de Remove the send-side cname getter APIs from voice and video engine.
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 09:55:30 +00:00
pbos@webrtc.org
1e92b0a93d Add ToString() to VideoSendStream::Config.
Adds ToString() to subsequent parts as well as a common.gyp to define
ToString() methods for config.h. VideoStream is also moved to config.h.

BUG=3171
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6170 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 09:35:06 +00:00
sprang@webrtc.org
09315705b9 Wire up statistics in video receive stream of new API
This CL includes Call tests that test both send and receive sides.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 12:06:29 +00:00
pbos@webrtc.org
c279a5d72c Wire up RTX in VideoReceiveStream.
Also adds a test to make sure that a retransmitted frame is actually
received and decoded on the remote side. The previous NACK test checked
retransmission, but not that the receiver actually takes care of the
retransmitted packet.

BUG=2399
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 09:30:53 +00:00
pbos@webrtc.org
e02d47515f Set up receiver RTX config using a std::map.
This change removes video_payload_type from RtxConfig as it can be
inferred from the map key or config otherwise. Wiring up this config is
part of issue 2399.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5402 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-20 14:43:55 +00:00
sprang@webrtc.org
ccd42840bc Wire up statistics in video send stream of new video engine api
Note, this CL does not contain any tests. Those are implemeted as call
tests and will be submitted when the receive stream is wired up as well.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5559006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 09:54:34 +00:00
pbos@webrtc.org
ce90eff345 Rename RTP-extension constants.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5137 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 11:48:56 +00:00
pbos@webrtc.org
16e03b7bd8 Separate Call API/build files from video_engine/.
BUG=2535
R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 16:32:01 +00:00