1c8391205e
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
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.