806dc3b0e62ec68f594e9aadab601b2db7e6c6d5 - src

commit	806dc3b0e62ec68f594e9aadab601b2db7e6c6d5	[log] [tgz]
author	hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d>	Tue Apr 09 19:54:10 2013
committer	hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d>	Tue Apr 09 19:54:10 2013
tree	d3dfcd5a2bb4142d0ef0ba1182180bd75b02f074
parent	4d2f5de67a8548a295bb2a8b69f030e294c2b8c3 [diff]

More trace events

The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.

BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc[diff]
webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc[diff]
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc[diff]
webrtc/modules/rtp_rtcp/source/rtp_receiver.cc[diff]
webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc[diff]
webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc[diff]
webrtc/modules/rtp_rtcp/source/rtp_sender.cc[diff]
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc[diff]
webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc[diff]
webrtc/modules/rtp_rtcp/source/rtp_sender_video.h[diff]
webrtc/modules/video_capture/video_capture_impl.cc[diff]
webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc[diff]
webrtc/modules/video_coding/main/source/generic_decoder.cc[diff]
webrtc/modules/video_coding/main/source/generic_encoder.cc[diff]
webrtc/modules/video_coding/main/source/jitter_buffer.cc[diff]
webrtc/modules/video_coding/main/source/receiver.cc[diff]
webrtc/modules/video_coding/main/source/video_coding_impl.cc[diff]
webrtc/video_engine/vie_capturer.cc[diff]
webrtc/video_engine/vie_encoder.cc[diff]
webrtc/video_engine/vie_sync_module.cc[diff]

20 files changed

tree: d3dfcd5a2bb4142d0ef0ba1182180bd75b02f074