RTP video playback tool using Call APIs.

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

webrtc/modules/video_coding/main/test/rtp_player.h

@@ -44,20 +44,6 @@ class PayloadCodecTuple {
 typedef std::vector<PayloadCodecTuple> PayloadTypes;
 typedef std::vector<PayloadCodecTuple>::const_iterator PayloadTypesIterator;
 
-// Implemented by something that can provide RTP packets, for instance a file
-// format parser such as the rtp_file_reader or the pcap_file_reader.
-class RtpPacketSourceInterface {
- public:
-  virtual ~RtpPacketSourceInterface() {}
-
-  // Read next RTP packet into buffer pointed to by rtp_data. On call, 'length'
-  // field must be filled in with the size of the buffer. The actual size of
-  // the packet is available in 'length' upon returning. Time in milliseconds
-  // from start of stream is returned in 'time_ms'.
-  virtual int NextPacket(uint8_t* rtp_data, uint32_t* length,
-                         uint32_t* time_ms) = 0;
-};
-
 // Implemented by RtpPlayer and given to client as a means to retrieve
 // information about a specific RTP stream.
 class RtpStreamInterface {