71f055fb41336316324942f828e022e2f7d93ec7 - src

commit	71f055fb41336316324942f828e022e2f7d93ec7	[log] [tgz]
author	sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>	Wed Dec 04 15:09:27 2013
committer	sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>	Wed Dec 04 15:09:27 2013
tree	60a642576e0d9a4f5c4ea3f2bc232383f53b3ea5
parent	9e5b0342f6bca12926a749d86befe461dfa0465f [diff]

Add send frame rate statistics callback

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

Review URL: https://webrtc-codereview.appspot.com/4479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5213 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/common_types.h[diff]
webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h[diff]
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h[diff]
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc[diff]
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h[diff]
webrtc/modules/rtp_rtcp/source/rtp_sender.cc[diff]
webrtc/modules/rtp_rtcp/source/rtp_sender.h[diff]
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc[diff]
webrtc/modules/video_coding/main/source/encoded_frame.cc[diff]
webrtc/modules/video_coding/main/source/jitter_buffer.cc[diff]
webrtc/modules/video_coding/main/source/jitter_buffer.h[diff]
webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc[diff]
webrtc/modules/video_coding/main/source/media_optimization.cc[diff]
webrtc/modules/video_coding/main/source/receiver.cc[diff]
webrtc/video_engine/vie_channel.cc[diff]
webrtc/video_engine/vie_channel.h[diff]
webrtc/video_engine/vie_rtp_rtcp_impl.cc[diff]

17 files changed

tree: 60a642576e0d9a4f5c4ea3f2bc232383f53b3ea5