commit 1c8391205e9caf05337dbbe071944b7ba11a6cec
author henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Aug 11 12:29:38 2014
committer henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Aug 11 12:29:38 2014
tree 07f7bea523d9cb78e69359fdf8210f5d3cf55b04
parent 96d8b0e69f34f9ab29248eefbb55c4d698752144

Use test::Packet test::PacketSource classes in neteq_rtpplay

This change replaces the old NETEQTEST_RTPpacket and
NETEQTEST_DummyRTPpacket with the new test::Packet class. Note that the
Packet class automatically handles "dummy" packets (i.e., packets for
which only the header and a length field was stored to file)
automatically. There is no need to explicitly signal this to the
application any longer. The RTP input file is now handled as a
test::PacketSource object.

Also adding a new ConvertHeader method to the Packet class. This is
needed to extract the header information as an alternative data type.

Finally, some dead code was deleted from rtp_analyze.cc (unrelated to
the reset of this change).

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6862 4adac7df-926f-26a2-2b94-8c16560cd09d