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

modules/utility/interface/rtp_dump.h

@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file implements a class that writes a stream of RTP and RTCP packets
+// to a file according to the format specified by rtpplay. See
+// http://www.cs.columbia.edu/irt/software/rtptools/.
+// Notes: supported platforms are Windows, Linux and Mac OSX
+
+#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
+#define WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
+
+#include "typedefs.h"
+#include "file_wrapper.h"
+
+namespace webrtc {
+class RtpDump
+{
+public:
+    // Factory method.
+    static RtpDump* CreateRtpDump();
+
+    // Delete function. Destructor disabled.
+    static void DestroyRtpDump(RtpDump* object);
+
+    // Open the file fileNameUTF8 for writing RTP/RTCP packets.
+    // Note: this API also adds the rtpplay header.
+    virtual WebRtc_Word32 Start(const WebRtc_Word8* fileNameUTF8) = 0;
+
+    // Close the existing file. No more packets will be recorded.
+    virtual WebRtc_Word32 Stop() = 0;
+
+    // Return true if a file is open for recording RTP/RTCP packets.
+    virtual bool IsActive() const = 0;
+
+    // Writes the RTP/RTCP packet in packet with length packetLength in bytes.
+    // Note: packet should contain the RTP/RTCP part of the packet. I.e. the
+    // first bytes of packet should be the RTP/RTCP header.
+    virtual WebRtc_Word32 DumpPacket(const WebRtc_UWord8* packet,
+                                     WebRtc_UWord16 packetLength) = 0;
+
+protected:
+    virtual ~RtpDump();
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_