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
50 lines
1.7 KiB
C++
50 lines
1.7 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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#ifndef WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
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#define WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
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#include "webrtc/modules/utility/interface/rtp_dump.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class FileWrapper;
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class RtpDumpImpl : public RtpDump
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{
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public:
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RtpDumpImpl();
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virtual ~RtpDumpImpl();
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virtual int32_t Start(const char* fileNameUTF8) OVERRIDE;
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virtual int32_t Stop() OVERRIDE;
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virtual bool IsActive() const OVERRIDE;
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virtual int32_t DumpPacket(const uint8_t* packet,
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size_t packetLength) OVERRIDE;
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private:
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// Return the system time in ms.
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inline uint32_t GetTimeInMS() const;
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// Return x in network byte order (big endian).
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inline uint32_t RtpDumpHtonl(uint32_t x) const;
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// Return x in network byte order (big endian).
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inline uint16_t RtpDumpHtons(uint16_t x) const;
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// Return true if the packet starts with a valid RTCP header.
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// Note: See RtpUtility::RtpHeaderParser::RTCP() for details on how
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// to determine if the packet is an RTCP packet.
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bool RTCP(const uint8_t* packet) const;
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private:
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CriticalSectionWrapper* _critSect;
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FileWrapper& _file;
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uint32_t _startTime;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
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