107 lines
3.5 KiB
C
107 lines
3.5 KiB
C
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/*
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* Copyright (c) 2011 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_VIDEO_CODING_TEST_RTP_PLAYER_H_
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#define WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
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#include "typedefs.h"
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#include "rtp_rtcp.h"
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#include "list_wrapper.h"
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#include "critical_section_wrapper.h"
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#include "video_coding_defines.h"
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#include <stdio.h>
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#include <string>
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#define HDR_SIZE 8 // rtpplay packet header size in bytes
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#define FIRSTLINELEN 40
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#define RAND_VEC_LENGTH 4096
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struct RawRtpPacket
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{
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public:
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RawRtpPacket(WebRtc_UWord8* data, WebRtc_UWord16 len);
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~RawRtpPacket();
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WebRtc_UWord8* rtpData;
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WebRtc_UWord16 rtpLen;
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WebRtc_Word64 resendTimeMs;
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};
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class LostPackets : public webrtc::ListWrapper
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{
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public:
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LostPackets();
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~LostPackets();
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WebRtc_UWord32 AddPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen);
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WebRtc_UWord32 SetResendTime(WebRtc_UWord16 sequenceNumber, WebRtc_Word64 resendTime);
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WebRtc_UWord32 TotalNumberOfLosses() const { return _lossCount; };
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WebRtc_UWord32 NumberOfPacketsToResend() const;
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void ResentPacket(WebRtc_UWord16 seqNo);
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void Lock() {_critSect.Enter();};
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void Unlock() {_critSect.Leave();};
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private:
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webrtc::CriticalSectionWrapper& _critSect;
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WebRtc_UWord32 _lossCount;
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FILE* _debugFile;
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};
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struct PayloadCodecTuple
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{
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PayloadCodecTuple(WebRtc_UWord8 plType, std::string codecName, webrtc::VideoCodecType type) :
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name(codecName), payloadType(plType), codecType(type) {};
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const std::string name;
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const WebRtc_UWord8 payloadType;
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const webrtc::VideoCodecType codecType;
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};
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class RTPPlayer : public webrtc::VCMPacketRequestCallback
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{
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public:
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RTPPlayer(const char* filename, webrtc::RtpData* callback);
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virtual ~RTPPlayer();
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WebRtc_Word32 Initialize(const webrtc::ListWrapper& payloadList);
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WebRtc_Word32 NextPacket(const WebRtc_Word64 timeNow);
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WebRtc_UWord32 TimeUntilNextPacket() const;
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WebRtc_Word32 SimulatePacketLoss(float lossRate, bool enableNack = false, WebRtc_UWord32 rttMs = 0);
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WebRtc_Word32 SetReordering(bool enabled);
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WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers, WebRtc_UWord16 length);
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void Print() const;
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private:
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WebRtc_Word32 SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen);
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WebRtc_Word32 ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset);
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WebRtc_Word32 ReadHeader();
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FILE* _rtpFile;
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webrtc::RtpRtcp& _rtpModule;
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WebRtc_UWord32 _nextRtpTime;
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webrtc::RtpData* _dataCallback;
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bool _firstPacket;
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float _lossRate;
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bool _nackEnabled;
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LostPackets _lostPackets;
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WebRtc_UWord32 _resendPacketCount;
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WebRtc_Word32 _noLossStartup;
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bool _endOfFile;
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WebRtc_UWord32 _rttMs;
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WebRtc_Word64 _firstPacketRtpTime;
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WebRtc_Word64 _firstPacketTimeMs;
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RawRtpPacket* _reorderBuffer;
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bool _reordering;
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WebRtc_Word16 _nextPacket[8000];
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WebRtc_Word32 _nextPacketLength;
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int _randVec[RAND_VEC_LENGTH];
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int _randVecPos;
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};
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#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
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