git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/utility/interface/rtp_dump.h b/modules/utility/interface/rtp_dump.h
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+/*
+ * 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_