webrtc/modules/video_coding/main/test/rtp_player.cc

467 lines
12 KiB
C++

/*
* Copyright (c) 2011 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.
*/
#include "rtp_player.h"
#include "../source/internal_defines.h"
#include "rtp_rtcp.h"
#include "tick_time.h"
#include <cstdlib>
#ifdef WIN32
#include <windows.h>
#include <Winsock2.h>
#else
#include <arpa/inet.h>
#endif
using namespace webrtc;
RawRtpPacket::RawRtpPacket(WebRtc_UWord8* data, WebRtc_UWord16 len)
:
rtpData(), rtpLen(len), resendTimeMs(-1)
{
rtpData = new WebRtc_UWord8[rtpLen];
memcpy(rtpData, data, rtpLen);
}
RawRtpPacket::~RawRtpPacket()
{
delete [] rtpData;
}
LostPackets::LostPackets()
:
_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
_lossCount(0),
ListWrapper(),
_debugFile(NULL)
{
_debugFile = fopen("PacketLossDebug.txt", "w");
}
LostPackets::~LostPackets()
{
if (_debugFile)
{
fclose(_debugFile);
}
ListItem* item = First();
while (item != NULL)
{
RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
if (packet != NULL)
{
delete packet;
}
Erase(item);
item = First();
}
delete &_critSect;
}
WebRtc_UWord32 LostPackets::AddPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen)
{
CriticalSectionScoped cs(_critSect);
RawRtpPacket* packet = new RawRtpPacket(rtpData, rtpLen);
ListItem* newItem = new ListItem(packet);
InsertBefore(First(), newItem);
const WebRtc_UWord16 seqNo = (rtpData[2] << 8) + rtpData[3];
if (_debugFile != NULL)
{
fprintf(_debugFile, "%u Lost packet: %u\n", _lossCount, seqNo);
}
_lossCount++;
return 0;
}
WebRtc_UWord32 LostPackets::SetResendTime(WebRtc_UWord16 sequenceNumber, WebRtc_Word64 resendTime)
{
CriticalSectionScoped cs(_critSect);
ListItem* item = First();
while (item != NULL)
{
RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
const WebRtc_UWord16 seqNo = (packet->rtpData[2] << 8) + packet->rtpData[3];
const WebRtc_Word64 nowMs = VCMTickTime::MillisecondTimestamp();
if (sequenceNumber == seqNo && packet->resendTimeMs + 10 < nowMs)
{
if (_debugFile != NULL)
{
fprintf(_debugFile, "Resend %u at %u\n", seqNo, MaskWord64ToUWord32(resendTime));
}
packet->resendTimeMs = resendTime;
return 0;
}
item = Next(item);
}
fprintf(_debugFile, "Packet not lost %u\n", sequenceNumber);
return -1;
}
WebRtc_UWord32 LostPackets::NumberOfPacketsToResend() const
{
CriticalSectionScoped cs(_critSect);
WebRtc_UWord32 count = 0;
ListItem* item = First();
while (item != NULL)
{
RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
if (packet->resendTimeMs >= 0)
{
count++;
}
item = Next(item);
}
return count;
}
void LostPackets::ResentPacket(WebRtc_UWord16 seqNo)
{
CriticalSectionScoped cs(_critSect);
if (_debugFile != NULL)
{
fprintf(_debugFile, "Resent %u at %u\n", seqNo,
MaskWord64ToUWord32(VCMTickTime::MillisecondTimestamp()));
}
}
RTPPlayer::RTPPlayer(const char* filename, RtpData* callback)
:
_rtpModule(*RtpRtcp::CreateRtpRtcp(1, false)),
_nextRtpTime(0),
_dataCallback(callback),
_firstPacket(true),
_lossRate(0.0f),
_nackEnabled(false),
_resendPacketCount(0),
_noLossStartup(100),
_endOfFile(false),
_rttMs(0),
_firstPacketRtpTime(0),
_firstPacketTimeMs(0),
_reorderBuffer(NULL),
_reordering(false),
_nextPacket(),
_nextPacketLength(0),
_randVec(),
_randVecPos(0)
{
_rtpFile = fopen(filename, "rb");
memset(_nextPacket, 0, sizeof(_nextPacket));
}
RTPPlayer::~RTPPlayer()
{
RtpRtcp::DestroyRtpRtcp(&_rtpModule);
if (_rtpFile != NULL)
{
fclose(_rtpFile);
}
if (_reorderBuffer != NULL)
{
delete _reorderBuffer;
_reorderBuffer = NULL;
}
}
WebRtc_Word32 RTPPlayer::Initialize(const ListWrapper& payloadList)
{
std::srand(321);
for (int i=0; i < RAND_VEC_LENGTH; i++)
{
_randVec[i] = rand();
}
_randVecPos = 0;
WebRtc_Word32 ret = _rtpModule.SetNACKStatus(kNackOff);
if (ret < 0)
{
return -1;
}
ret = _rtpModule.InitReceiver();
if (ret < 0)
{
return -1;
}
_rtpModule.InitSender();
_rtpModule.SetRTCPStatus(kRtcpNonCompound);
_rtpModule.SetTMMBRStatus(true);
ret = _rtpModule.RegisterIncomingDataCallback(_dataCallback);
if (ret < 0)
{
return -1;
}
// Register payload types
ListItem* item = payloadList.First();
while (item != NULL)
{
PayloadCodecTuple* payloadType = static_cast<PayloadCodecTuple*>(item->GetItem());
if (payloadType != NULL)
{
if (_rtpModule.RegisterReceivePayload(payloadType->name.c_str(), payloadType->payloadType) < 0)
{
return -1;
}
}
item = payloadList.Next(item);
}
if (ReadHeader() < 0)
{
return -1;
}
memset(_nextPacket, 0, sizeof(_nextPacket));
_nextPacketLength = ReadPacket(_nextPacket, &_nextRtpTime);
return 0;
}
WebRtc_Word32 RTPPlayer::ReadHeader()
{
char firstline[FIRSTLINELEN];
if (_rtpFile == NULL)
{
return -1;
}
fgets(firstline, FIRSTLINELEN, _rtpFile);
if(strncmp(firstline,"#!rtpplay",9) == 0) {
if(strncmp(firstline,"#!rtpplay1.0",12) != 0){
printf("ERROR: wrong rtpplay version, must be 1.0\n");
return -1;
}
}
else if (strncmp(firstline,"#!RTPencode",11) == 0) {
if(strncmp(firstline,"#!RTPencode1.0",14) != 0){
printf("ERROR: wrong RTPencode version, must be 1.0\n");
return -1;
}
}
else {
printf("ERROR: wrong file format of input file\n");
return -1;
}
WebRtc_UWord32 start_sec;
WebRtc_UWord32 start_usec;
WebRtc_UWord32 source;
WebRtc_UWord16 port;
WebRtc_UWord16 padding;
fread(&start_sec, 4, 1, _rtpFile);
start_sec=ntohl(start_sec);
fread(&start_usec, 4, 1, _rtpFile);
start_usec=ntohl(start_usec);
fread(&source, 4, 1, _rtpFile);
source=ntohl(source);
fread(&port, 2, 1, _rtpFile);
port=ntohs(port);
fread(&padding, 2, 1, _rtpFile);
padding=ntohs(padding);
return 0;
}
WebRtc_UWord32 RTPPlayer::TimeUntilNextPacket() const
{
WebRtc_Word64 timeLeft = (_nextRtpTime - _firstPacketRtpTime) - (VCMTickTime::MillisecondTimestamp() - _firstPacketTimeMs);
if (timeLeft < 0)
{
return 0;
}
return static_cast<WebRtc_UWord32>(timeLeft);
}
WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
{
// Send any packets ready to be resent
_lostPackets.Lock();
ListItem* item = _lostPackets.First();
_lostPackets.Unlock();
while (item != NULL)
{
_lostPackets.Lock();
RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
_lostPackets.Unlock();
if (timeNow >= packet->resendTimeMs && packet->resendTimeMs != -1)
{
const WebRtc_UWord16 seqNo = (packet->rtpData[2] << 8) + packet->rtpData[3];
printf("Resend: %u\n", seqNo);
WebRtc_Word32 ret = SendPacket(packet->rtpData, packet->rtpLen);
ListItem* itemToRemove = item;
_lostPackets.Lock();
item = _lostPackets.Next(item);
_lostPackets.Erase(itemToRemove);
delete packet;
_lostPackets.Unlock();
_resendPacketCount++;
if (ret > 0)
{
_lostPackets.ResentPacket(seqNo);
}
else if (ret < 0)
{
return ret;
}
}
else
{
_lostPackets.Lock();
item = _lostPackets.Next(item);
_lostPackets.Unlock();
}
}
// Send any packets from rtp file
if (!_endOfFile && (TimeUntilNextPacket() == 0 || _firstPacket))
{
_rtpModule.Process();
if (_firstPacket)
{
_firstPacketRtpTime = static_cast<WebRtc_Word64>(_nextRtpTime);
_firstPacketTimeMs = VCMTickTime::MillisecondTimestamp();
}
if (_reordering && _reorderBuffer == NULL)
{
_reorderBuffer = new RawRtpPacket(reinterpret_cast<WebRtc_UWord8*>(_nextPacket), static_cast<WebRtc_UWord16>(_nextPacketLength));
return 0;
}
WebRtc_Word32 ret = SendPacket(reinterpret_cast<WebRtc_UWord8*>(_nextPacket), static_cast<WebRtc_UWord16>(_nextPacketLength));
if (_reordering && _reorderBuffer != NULL)
{
RawRtpPacket* rtpPacket = _reorderBuffer;
_reorderBuffer = NULL;
SendPacket(rtpPacket->rtpData, rtpPacket->rtpLen);
delete rtpPacket;
}
_firstPacket = false;
if (ret < 0)
{
return ret;
}
_nextPacketLength = ReadPacket(_nextPacket, &_nextRtpTime);
if (_nextPacketLength < 0)
{
_endOfFile = true;
return 0;
}
else if (_nextPacketLength == 0)
{
return 0;
}
}
if (_endOfFile && _lostPackets.NumberOfPacketsToResend() == 0)
{
return 1;
}
return 0;
}
WebRtc_Word32 RTPPlayer::SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen)
{
if ((_randVec[(_randVecPos++) % RAND_VEC_LENGTH] + 1.0)/(RAND_MAX + 1.0) < _lossRate &&
_noLossStartup < 0)
{
if (_nackEnabled)
{
const WebRtc_UWord16 seqNo = (rtpData[2] << 8) + rtpData[3];
printf("Throw: %u\n", seqNo);
_lostPackets.AddPacket(rtpData, rtpLen);
return 0;
}
}
else
{
WebRtc_Word32 ret = _rtpModule.IncomingPacket(rtpData, rtpLen);
if (ret < 0)
{
return -1;
}
}
if (_noLossStartup >= 0)
{
_noLossStartup--;
}
return 1;
}
WebRtc_Word32 RTPPlayer::ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset)
{
WebRtc_UWord16 length, plen;
if (fread(&length,2,1,_rtpFile)==0)
return(-1);
length=ntohs(length);
if (fread(&plen,2,1,_rtpFile)==0)
return(-1);
plen=ntohs(plen);
if (fread(offset,4,1,_rtpFile)==0)
return(-1);
*offset=ntohl(*offset);
// Use length here because a plen of 0 specifies rtcp
length = (WebRtc_UWord16) (length - HDR_SIZE);
if (fread((unsigned short *) rtpdata,1,length,_rtpFile) != length)
return(-1);
#ifdef JUNK_DATA
// destroy the RTP payload with random data
if (plen > 12) { // ensure that we have more than just a header
for ( int ix = 12; ix < plen; ix=ix+2 ) {
rtpdata[ix>>1] = (short) (rtpdata[ix>>1] + (short) rand());
}
}
#endif
return plen;
}
WebRtc_Word32 RTPPlayer::SimulatePacketLoss(float lossRate, bool enableNack, WebRtc_UWord32 rttMs)
{
_nackEnabled = enableNack;
_lossRate = lossRate;
_rttMs = rttMs;
return 0;
}
WebRtc_Word32 RTPPlayer::SetReordering(bool enabled)
{
_reordering = enabled;
return 0;
}
WebRtc_Word32 RTPPlayer::ResendPackets(const WebRtc_UWord16* sequenceNumbers, WebRtc_UWord16 length)
{
if (sequenceNumbers == NULL)
{
return 0;
}
for (int i=0; i < length; i++)
{
_lostPackets.SetResendTime(sequenceNumbers[i], VCMTickTime::MillisecondTimestamp() + _rttMs);
}
return 0;
}
void RTPPlayer::Print() const
{
printf("Lost packets: %u, resent packets: %u\n", _lostPackets.TotalNumberOfLosses(), _resendPacketCount);
printf("Packets still lost: %u\n", _lostPackets.GetSize());
printf("Packets waiting to be resent: %u\n", _lostPackets.NumberOfPacketsToResend());
printf("Sequence numbers:\n");
ListItem* item = _lostPackets.First();
while (item != NULL)
{
RawRtpPacket* packet = static_cast<RawRtpPacket*>(item->GetItem());
const WebRtc_UWord16 seqNo = (packet->rtpData[2] << 8) + packet->rtpData[3];
printf("%u, ", seqNo);
item = _lostPackets.Next(item);
}
printf("\n");
}