5 Commits

Author SHA1 Message Date
pbos@webrtc.org
025f4f152b Stats+Config moved into VideoSend/ReceiveStreams.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1561006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4182 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-05 11:33:21 +00:00
stefan@webrtc.org
a5cb98cbbd 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
2013-05-29 12:12:51 +00:00
pbos@webrtc.org
1ecee9a15a Break video_engine/new_include/common.h into smaller parts.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1571005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4128 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 11:34:32 +00:00
pbos@webrtc.org
f5d4cb1958 Include files from webrtc/.. paths in video_engine/
BUG=1662
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1492004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4056 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17 13:44:48 +00:00
pbos@webrtc.org
29d5839233 New VideoEngine API implementation on top of old one, first steps.
BUG=1668
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1360004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4044 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 12:08:03 +00:00