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* Copyright (c) 2012 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
// Notes: supported platforms are Windows, Linux and Mac OSX
#include "typedefs.h"
#include "file_wrapper.h"
namespace webrtc {
class RtpDump
// 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 char* 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;
virtual ~RtpDump();
} // namespace webrtc