git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
niklase@google.com
466
modules/video_coding/main/test/rtp_player.cc
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466
modules/video_coding/main/test/rtp_player.cc
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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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#include "rtp_player.h"
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#include "../source/internal_defines.h"
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#include "rtp_rtcp.h"
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#include "tick_time.h"
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#include <cstdlib>
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#ifdef WIN32
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#include <windows.h>
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#include <Winsock2.h>
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#else
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#include <arpa/inet.h>
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#endif
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using namespace webrtc;
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RawRtpPacket::RawRtpPacket(WebRtc_UWord8* data, WebRtc_UWord16 len)
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:
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rtpData(), rtpLen(len), resendTimeMs(-1)
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{
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rtpData = new WebRtc_UWord8[rtpLen];
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memcpy(rtpData, data, rtpLen);
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}
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RawRtpPacket::~RawRtpPacket()
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{
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delete [] rtpData;
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}
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LostPackets::LostPackets()
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:
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_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
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_lossCount(0),
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ListWrapper(),
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_debugFile(NULL)
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{
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_debugFile = fopen("PacketLossDebug.txt", "w");
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}
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LostPackets::~LostPackets()
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{
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if (_debugFile)
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{
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fclose(_debugFile);
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}
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ListItem* item = First();
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while (item != NULL)
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{
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RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
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if (packet != NULL)
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{
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delete packet;
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}
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Erase(item);
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item = First();
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}
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delete &_critSect;
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}
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WebRtc_UWord32 LostPackets::AddPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen)
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{
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CriticalSectionScoped cs(_critSect);
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RawRtpPacket* packet = new RawRtpPacket(rtpData, rtpLen);
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ListItem* newItem = new ListItem(packet);
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InsertBefore(First(), newItem);
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const WebRtc_UWord16 seqNo = (rtpData[2] << 8) + rtpData[3];
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if (_debugFile != NULL)
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{
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fprintf(_debugFile, "%u Lost packet: %u\n", _lossCount, seqNo);
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}
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_lossCount++;
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return 0;
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}
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WebRtc_UWord32 LostPackets::SetResendTime(WebRtc_UWord16 sequenceNumber, WebRtc_Word64 resendTime)
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{
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CriticalSectionScoped cs(_critSect);
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ListItem* item = First();
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while (item != NULL)
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{
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RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
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const WebRtc_UWord16 seqNo = (packet->rtpData[2] << 8) + packet->rtpData[3];
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const WebRtc_Word64 nowMs = VCMTickTime::MillisecondTimestamp();
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if (sequenceNumber == seqNo && packet->resendTimeMs + 10 < nowMs)
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{
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if (_debugFile != NULL)
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{
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fprintf(_debugFile, "Resend %u at %u\n", seqNo, MaskWord64ToUWord32(resendTime));
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}
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packet->resendTimeMs = resendTime;
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return 0;
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}
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item = Next(item);
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}
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fprintf(_debugFile, "Packet not lost %u\n", sequenceNumber);
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return -1;
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}
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WebRtc_UWord32 LostPackets::NumberOfPacketsToResend() const
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{
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CriticalSectionScoped cs(_critSect);
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WebRtc_UWord32 count = 0;
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ListItem* item = First();
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while (item != NULL)
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{
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RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
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if (packet->resendTimeMs >= 0)
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{
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count++;
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}
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item = Next(item);
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}
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return count;
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}
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void LostPackets::ResentPacket(WebRtc_UWord16 seqNo)
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{
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CriticalSectionScoped cs(_critSect);
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if (_debugFile != NULL)
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{
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fprintf(_debugFile, "Resent %u at %u\n", seqNo,
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MaskWord64ToUWord32(VCMTickTime::MillisecondTimestamp()));
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}
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}
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RTPPlayer::RTPPlayer(const char* filename, RtpData* callback)
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:
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_rtpModule(*RtpRtcp::CreateRtpRtcp(1, false)),
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_nextRtpTime(0),
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_dataCallback(callback),
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_firstPacket(true),
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_lossRate(0.0f),
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_nackEnabled(false),
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_resendPacketCount(0),
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_noLossStartup(100),
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_endOfFile(false),
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_rttMs(0),
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_firstPacketRtpTime(0),
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_firstPacketTimeMs(0),
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_reorderBuffer(NULL),
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_reordering(false),
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_nextPacket(),
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_nextPacketLength(0),
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_randVec(),
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_randVecPos(0)
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{
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_rtpFile = fopen(filename, "rb");
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memset(_nextPacket, 0, sizeof(_nextPacket));
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}
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RTPPlayer::~RTPPlayer()
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{
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RtpRtcp::DestroyRtpRtcp(&_rtpModule);
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if (_rtpFile != NULL)
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{
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fclose(_rtpFile);
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}
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if (_reorderBuffer != NULL)
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{
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delete _reorderBuffer;
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_reorderBuffer = NULL;
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}
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}
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WebRtc_Word32 RTPPlayer::Initialize(const ListWrapper& payloadList)
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{
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std::srand(321);
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for (int i=0; i < RAND_VEC_LENGTH; i++)
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{
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_randVec[i] = rand();
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}
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_randVecPos = 0;
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WebRtc_Word32 ret = _rtpModule.SetNACKStatus(kNackOff);
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if (ret < 0)
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{
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return -1;
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}
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ret = _rtpModule.InitReceiver();
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if (ret < 0)
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{
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return -1;
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}
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_rtpModule.InitSender();
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_rtpModule.SetRTCPStatus(kRtcpNonCompound);
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_rtpModule.SetTMMBRStatus(true);
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ret = _rtpModule.RegisterIncomingDataCallback(_dataCallback);
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if (ret < 0)
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{
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return -1;
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}
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// Register payload types
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ListItem* item = payloadList.First();
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while (item != NULL)
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{
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PayloadCodecTuple* payloadType = static_cast<PayloadCodecTuple*>(item->GetItem());
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if (payloadType != NULL)
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{
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if (_rtpModule.RegisterReceivePayload(payloadType->name.c_str(), payloadType->payloadType) < 0)
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{
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return -1;
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}
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}
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item = payloadList.Next(item);
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}
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if (ReadHeader() < 0)
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{
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return -1;
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}
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memset(_nextPacket, 0, sizeof(_nextPacket));
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_nextPacketLength = ReadPacket(_nextPacket, &_nextRtpTime);
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return 0;
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}
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WebRtc_Word32 RTPPlayer::ReadHeader()
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{
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char firstline[FIRSTLINELEN];
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if (_rtpFile == NULL)
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{
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return -1;
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}
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fgets(firstline, FIRSTLINELEN, _rtpFile);
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if(strncmp(firstline,"#!rtpplay",9) == 0) {
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if(strncmp(firstline,"#!rtpplay1.0",12) != 0){
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printf("ERROR: wrong rtpplay version, must be 1.0\n");
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return -1;
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}
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}
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else if (strncmp(firstline,"#!RTPencode",11) == 0) {
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if(strncmp(firstline,"#!RTPencode1.0",14) != 0){
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printf("ERROR: wrong RTPencode version, must be 1.0\n");
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return -1;
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}
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}
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else {
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printf("ERROR: wrong file format of input file\n");
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return -1;
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}
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WebRtc_UWord32 start_sec;
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WebRtc_UWord32 start_usec;
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WebRtc_UWord32 source;
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WebRtc_UWord16 port;
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WebRtc_UWord16 padding;
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fread(&start_sec, 4, 1, _rtpFile);
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start_sec=ntohl(start_sec);
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fread(&start_usec, 4, 1, _rtpFile);
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start_usec=ntohl(start_usec);
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fread(&source, 4, 1, _rtpFile);
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source=ntohl(source);
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fread(&port, 2, 1, _rtpFile);
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port=ntohs(port);
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fread(&padding, 2, 1, _rtpFile);
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padding=ntohs(padding);
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return 0;
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}
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WebRtc_UWord32 RTPPlayer::TimeUntilNextPacket() const
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{
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WebRtc_Word64 timeLeft = (_nextRtpTime - _firstPacketRtpTime) - (VCMTickTime::MillisecondTimestamp() - _firstPacketTimeMs);
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if (timeLeft < 0)
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{
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return 0;
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}
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return static_cast<WebRtc_UWord32>(timeLeft);
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}
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WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
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{
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// Send any packets ready to be resent
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_lostPackets.Lock();
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ListItem* item = _lostPackets.First();
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_lostPackets.Unlock();
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while (item != NULL)
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{
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_lostPackets.Lock();
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RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
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_lostPackets.Unlock();
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if (timeNow >= packet->resendTimeMs && packet->resendTimeMs != -1)
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{
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const WebRtc_UWord16 seqNo = (packet->rtpData[2] << 8) + packet->rtpData[3];
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printf("Resend: %u\n", seqNo);
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WebRtc_Word32 ret = SendPacket(packet->rtpData, packet->rtpLen);
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ListItem* itemToRemove = item;
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_lostPackets.Lock();
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item = _lostPackets.Next(item);
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_lostPackets.Erase(itemToRemove);
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delete packet;
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_lostPackets.Unlock();
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_resendPacketCount++;
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if (ret > 0)
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{
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_lostPackets.ResentPacket(seqNo);
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}
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else if (ret < 0)
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{
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return ret;
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}
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}
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else
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{
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_lostPackets.Lock();
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item = _lostPackets.Next(item);
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_lostPackets.Unlock();
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}
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}
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// Send any packets from rtp file
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if (!_endOfFile && (TimeUntilNextPacket() == 0 || _firstPacket))
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{
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_rtpModule.Process();
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if (_firstPacket)
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{
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_firstPacketRtpTime = static_cast<WebRtc_Word64>(_nextRtpTime);
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_firstPacketTimeMs = VCMTickTime::MillisecondTimestamp();
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}
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if (_reordering && _reorderBuffer == NULL)
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{
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_reorderBuffer = new RawRtpPacket(reinterpret_cast<WebRtc_UWord8*>(_nextPacket), static_cast<WebRtc_UWord16>(_nextPacketLength));
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return 0;
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}
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WebRtc_Word32 ret = SendPacket(reinterpret_cast<WebRtc_UWord8*>(_nextPacket), static_cast<WebRtc_UWord16>(_nextPacketLength));
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if (_reordering && _reorderBuffer != NULL)
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{
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RawRtpPacket* rtpPacket = _reorderBuffer;
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_reorderBuffer = NULL;
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SendPacket(rtpPacket->rtpData, rtpPacket->rtpLen);
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delete rtpPacket;
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}
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_firstPacket = false;
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if (ret < 0)
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{
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return ret;
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}
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_nextPacketLength = ReadPacket(_nextPacket, &_nextRtpTime);
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if (_nextPacketLength < 0)
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{
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_endOfFile = true;
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return 0;
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}
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else if (_nextPacketLength == 0)
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{
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return 0;
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}
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}
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if (_endOfFile && _lostPackets.NumberOfPacketsToResend() == 0)
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{
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return 1;
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}
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return 0;
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}
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WebRtc_Word32 RTPPlayer::SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen)
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{
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if ((_randVec[(_randVecPos++) % RAND_VEC_LENGTH] + 1.0)/(RAND_MAX + 1.0) < _lossRate &&
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_noLossStartup < 0)
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{
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if (_nackEnabled)
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{
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const WebRtc_UWord16 seqNo = (rtpData[2] << 8) + rtpData[3];
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printf("Throw: %u\n", seqNo);
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_lostPackets.AddPacket(rtpData, rtpLen);
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return 0;
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}
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}
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else
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{
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WebRtc_Word32 ret = _rtpModule.IncomingPacket(rtpData, rtpLen);
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if (ret < 0)
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{
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return -1;
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}
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}
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if (_noLossStartup >= 0)
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{
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_noLossStartup--;
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}
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return 1;
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}
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WebRtc_Word32 RTPPlayer::ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset)
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{
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WebRtc_UWord16 length, plen;
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if (fread(&length,2,1,_rtpFile)==0)
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return(-1);
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length=ntohs(length);
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if (fread(&plen,2,1,_rtpFile)==0)
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return(-1);
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plen=ntohs(plen);
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if (fread(offset,4,1,_rtpFile)==0)
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return(-1);
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*offset=ntohl(*offset);
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// Use length here because a plen of 0 specifies rtcp
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length = (WebRtc_UWord16) (length - HDR_SIZE);
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if (fread((unsigned short *) rtpdata,1,length,_rtpFile) != length)
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return(-1);
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#ifdef JUNK_DATA
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// destroy the RTP payload with random data
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if (plen > 12) { // ensure that we have more than just a header
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for ( int ix = 12; ix < plen; ix=ix+2 ) {
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rtpdata[ix>>1] = (short) (rtpdata[ix>>1] + (short) rand());
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}
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}
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#endif
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return plen;
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}
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WebRtc_Word32 RTPPlayer::SimulatePacketLoss(float lossRate, bool enableNack, WebRtc_UWord32 rttMs)
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{
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_nackEnabled = enableNack;
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_lossRate = lossRate;
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_rttMs = rttMs;
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return 0;
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}
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WebRtc_Word32 RTPPlayer::SetReordering(bool enabled)
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{
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_reordering = enabled;
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return 0;
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}
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WebRtc_Word32 RTPPlayer::ResendPackets(const WebRtc_UWord16* sequenceNumbers, WebRtc_UWord16 length)
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{
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if (sequenceNumbers == NULL)
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{
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return 0;
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}
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for (int i=0; i < length; i++)
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{
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_lostPackets.SetResendTime(sequenceNumbers[i], VCMTickTime::MillisecondTimestamp() + _rttMs);
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}
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return 0;
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}
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void RTPPlayer::Print() const
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{
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printf("Lost packets: %u, resent packets: %u\n", _lostPackets.TotalNumberOfLosses(), _resendPacketCount);
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printf("Packets still lost: %u\n", _lostPackets.GetSize());
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printf("Packets waiting to be resent: %u\n", _lostPackets.NumberOfPacketsToResend());
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printf("Sequence numbers:\n");
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ListItem* item = _lostPackets.First();
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while (item != NULL)
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{
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RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
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const WebRtc_UWord16 seqNo = (packet->rtpData[2] << 8) + packet->rtpData[3];
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printf("%u, ", seqNo);
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item = _lostPackets.Next(item);
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}
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printf("\n");
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}
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