4591fbd09f
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information. This CL was reviewed and approved in pieces in the following CLs: https://webrtc-codereview.appspot.com/24209004/ https://webrtc-codereview.appspot.com/24229004/ https://webrtc-codereview.appspot.com/24259004/ https://webrtc-codereview.appspot.com/25109004/ https://webrtc-codereview.appspot.com/26099004/ https://webrtc-codereview.appspot.com/27069004/ https://webrtc-codereview.appspot.com/27969004/ https://webrtc-codereview.appspot.com/27989004/ https://webrtc-codereview.appspot.com/29009004/ https://webrtc-codereview.appspot.com/30929004/ https://webrtc-codereview.appspot.com/30939004/ https://webrtc-codereview.appspot.com/31999004/ Committing as TBR to the original reviewers. BUG=chromium:81439 TEST=none TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom Review URL: https://webrtc-codereview.appspot.com/23129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
53 lines
1.8 KiB
C++
53 lines
1.8 KiB
C++
/*
|
|
* 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
|
|
// 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 "webrtc/system_wrappers/interface/file_wrapper.h"
|
|
#include "webrtc/typedefs.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 int32_t Start(const char* fileNameUTF8) = 0;
|
|
|
|
// Close the existing file. No more packets will be recorded.
|
|
virtual int32_t 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 int32_t DumpPacket(const uint8_t* packet,
|
|
size_t packetLength) = 0;
|
|
|
|
protected:
|
|
virtual ~RtpDump();
|
|
};
|
|
} // namespace webrtc
|
|
#endif // WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
|