8b06200802
BUG=1662 R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1786004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4336 4adac7df-926f-26a2-2b94-8c16560cd09d
53 lines
1.8 KiB
C++
53 lines
1.8 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This file implements a class that writes a stream of RTP and RTCP packets
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// to a file according to the format specified by rtpplay. See
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// http://www.cs.columbia.edu/irt/software/rtptools/.
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// Notes: supported platforms are Windows, Linux and Mac OSX
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#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
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#define WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
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#include "webrtc/system_wrappers/interface/file_wrapper.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class RtpDump
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{
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public:
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// Factory method.
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static RtpDump* CreateRtpDump();
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// Delete function. Destructor disabled.
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static void DestroyRtpDump(RtpDump* object);
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// Open the file fileNameUTF8 for writing RTP/RTCP packets.
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// Note: this API also adds the rtpplay header.
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virtual int32_t Start(const char* fileNameUTF8) = 0;
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// Close the existing file. No more packets will be recorded.
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virtual int32_t Stop() = 0;
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// Return true if a file is open for recording RTP/RTCP packets.
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virtual bool IsActive() const = 0;
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// Writes the RTP/RTCP packet in packet with length packetLength in bytes.
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// Note: packet should contain the RTP/RTCP part of the packet. I.e. the
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// first bytes of packet should be the RTP/RTCP header.
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virtual int32_t DumpPacket(const uint8_t* packet,
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uint16_t packetLength) = 0;
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protected:
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virtual ~RtpDump();
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
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