This is useful for debugging h264 input when we don't have an h264 decoder, as the resulting file should be possible to play back using mplayer. It is also often convenient to dump rtp packets in an interleaved format where the size of a packet is inserted before the actual payload.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42139004
Cr-Commit-Position: refs/heads/master@{#8558}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8558 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding original_length to the Packet struct. This is populated with
the plen value from the RTP dump file. In the case of reading a
pcap file, original_length will be equal to length.
Also increasing the maximum packet size to 3500 bytes. This is to
accomodate some test files that contain PCM16b audio encoding.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7333 4adac7df-926f-26a2-2b94-8c16560cd09d
Plays back rtpdump files from Wireshark in realtime as well as save the
resulting raw video to file. Unlike the RTP playback tool it doesn't
support faster-than-realtime playback/rendering, but it instead utilizes
the same path as production code and also contains support for playing
back FEC.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6838 4adac7df-926f-26a2-2b94-8c16560cd09d
