void H264MediaSubsession::startStream(unsigned clientSessionId,
void* streamToken,
TaskFunc* rtcpRRHandler,
void* rtcpRRHandlerClientData,
unsigned short& rtpSeqNum,
unsigned& rtpTimestamp,
ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
void* serverRequestAlternativeByteHandlerClientData) {
Log("-----[%d] %s .... calling\n", gettid(), __func__);
DestinationTable::iterator it = fDestinationsHashTable.find(clientSessionId);
if (it==fDestinationsHashTable.end())
{
Log("Rtsp session %u not found\n", clientSessionId);
return;
}
MyDestinations* destinations = it->second;
char* dest_addr = NULL;
if(destinations != NULL)
{
dest_addr = inet_ntoa(destinations->addr);
Log("-----Session destionation is %s:%d\n", dest_addr, ntohs(destinations->rtpPort.num()));
}
VideoCodec::RTPMap rtpMap;
rtpMap[rtpPayloadType] = m_Watcher->GetVideoCodec();
//video.StartSending(clientSessionId,dest_addr,ntohs(destinations->rtpPort.num()),rtpMap);
RTPSession *rtp = m_Watcher->StartSendingVideo(clientSessionId,dest_addr,ntohs(destinations->rtpPort.num()),rtpMap);
rtp->SetRtcpRRHandler(rtcpRRHandler, rtcpRRHandlerClientData);
}
RTPSession *AudioStream::AddAudioWatcher(unsigned clientSessionId,char *sendAudioIp,int sendAudioPort,AudioCodec::RTPMap& rtpMap, AudioCodec::Type watcherCodec)
{
Log(">Add Audio Watcher[%u, %s,%d]\n",clientSessionId, sendAudioIp,sendAudioPort);
//Si tenemos video
if (sendAudioPort==0) {
Error("No video\n");
return NULL;
}
Rtps::iterator it = rtps.find(clientSessionId);
RTPSession* rtp = NULL;
//Si no esta
if (it == rtps.end()) {
Error("Rtp not init\n");
return NULL;
}
rtp = (*it).second;
if(!rtp->Init())
{
delete rtp;
Error("Rtp init error\n");
return NULL;
}
//Iniciamos las sesiones rtp de envio
if(!rtp->SetRemotePort(sendAudioIp,sendAudioPort))
{
delete rtp;
Error("Rtp SetRemotePort error\n");
return NULL;
}
rtp->SetSendingAudioRTPMap(rtpMap);
//Set video codec
if(!rtp->SetSendingAudioCodec(watcherCodec))
{
delete rtp;
//Error
Error("%s video codec not supported by peer\n",AudioCodec::GetNameFor(watcherCodec));
return NULL;
}
rtps[clientSessionId] = rtp;
//LOgeamos
Log("<Add Audio Watcher success [%u]\n",clientSessionId);
return rtp;
}
void CSenderDlg::OnSend()
{
CFileDialog dlg(TRUE, NULL, NULL, OFN_HIDEREADONLY | OFN_OVERWRITEPROMPT,
"(*.*)|*.*|", NULL);
if (dlg.DoModal() == IDOK)
{
m_SourceFile = dlg.GetPathName();
}
source=fopen(m_SourceFile,"rb");
if ( source == NULL )
{
AfxMessageBox("can't access the file");
}
//#ifdef win32
//#endif
size_t bufRead = 0;
Mark=true;
BYTE buffer[bufsize];
int t;
t=sess.Create(5800);
if(t!=0)AfxMessageBox("can't create");
if(t==0){sen=true;}
sess.SetMulticastTTL(1); //設定multicast封包TTL
sess.AddDestination(ntohl(inet_addr("127.0.0.1")),6900); //指定傳送目的端
//sess.AddDestination(ntohl(inet_addr("192.168.25.79")),6900);
if(t!=0)AfxMessageBox("can't add destination");
(void) fseek( source, 0, SEEK_SET );
while(!feof(source))
{
bufRead = fread(buffer, sizeof(BYTE), bufsize, source );
nPT=rand()%64+1;
nTSTAMP=rand()%1000+1;
//g_ACode.EncodeAudioData ((char*)buffer,bufsize,m_cBufOut,&iOut);
//t=sess.SendPacket(m_cBufOut,iOut,nPT,Mark,nTSTAMP);
t=sess.SendPacket(buffer,bufsize,nPT,Mark,nTSTAMP);
if(t!=0)AfxMessageBox("can't sending");
Sleep(5);
Mark=false;
}
(void) fflush( source );
(void) fclose( source );
AfxMessageBox("send success");
WSACleanup();
}
void* startsound(void *_ca)
{
FILE *source=fopen("c://12pcm.wav","rb");
if ( source == NULL )
{
cout<<"can't access the file"<<endl;
}
//#ifdef win32
//#endif
size_t bufRead = 0;
bool Mark=true;
#define bufsize 1024
BYTE buffer[bufsize];
int t;
RTPSession sess;
t=sess.Create(5800);
if(t!=0)cout<<"can't create"<<endl;
bool sen;
if(t==0){sen=true;}
int nPT ;
int nTSTAMP;
sess.SetMulticastTTL(1); //設定multicast封包TTL
sess.AddDestination(ntohl(inet_addr("192.168.3.190")),portt); //指定傳送目的端
//sess.AddDestination(ntohl(inet_addr("192.168.25.79")),6900);
if(t!=0) cout<<"can't add destination"<<endl;
for (int jj=0;j<1000;j++)
{
(void) fseek( source, 0, SEEK_SET );
while(!feof(source))
{
bufRead = fread(buffer, sizeof(BYTE), bufsize, source );
nPT=rand()%64+1;
nTSTAMP=rand()%1000+1;
t=sess.SendPacket(buffer,bufsize,nPT,Mark,nTSTAMP);
if(t!=0)cout<<"can't sending"<<endl;
Sleep(5);
Mark=false;
}
}
(void) fflush( source );
(void) fclose( source );
cout<<("send success")<<endl;
WSACleanup();
return NULL;
}
/***********************************************************************************************************
**函数:Rtp_Lock
**功能:
**输入参数:
**返回值:
***********************************************************************************************************/
static int RtpSetup( uint16_t portbase)
{
int status;
RTPUDPv4TransmissionParams transparams;
RTPSessionParams sessparams;
sessparams.SetOwnTimestampUnit(1.0/90000.0);
sessparams.SetMaximumPacketSize(1200);
transparams.SetPortbase(portbase);
sess.SetDefaultPayloadType(PLOAD_TYPE);
sess.SetDefaultMark(false);
sess.SetDefaultTimestampIncrement(DefaultTimestampIncrement);
status = sess.Create(sessparams,&transparams);
checkerror(status);
return status;
}
/***********************************************************************************************************
**函数:Rtp_Lock
**功能:
**输入参数:
**返回值:
***********************************************************************************************************/
static int SimpleRtpIsActive(void)
{
int ret=false;
Rtp_Lock();
ret=sess.IsActive();
Rtp_UnLock();
return ret;
}
// Create a new RTP session. If multicast is not being used then multicastIP
// should be set to an empty string.
static RTPSession createRtpSession(const std::string& multicastIP,
unsigned int port)
{
RTPSession session;
//if (setenv("LOGNAME", "video", 0) != 0) {
// throw std::runtime_error("Error setting LOGNAME environment variable");
//}
// Set up session params
RTPSessionParams sessionparams;
sessionparams.SetUsePollThread(false);
sessionparams.SetMaximumPacketSize(1400);
sessionparams.SetAcceptOwnPackets(true);
sessionparams.SetOwnTimestampUnit(1.0f/900000.0f);
sessionparams.SetResolveLocalHostname(false);
sessionparams.SetSessionBandwidth(9000000);
sessionparams.SetReceiveMode(RTPTransmitter::AcceptAll);
RTPUDPv4TransmissionParams transparams;
transparams.SetPortbase(port);
int status = session.Create(sessionparams, &transparams);
if (status < 0) {
throw std::runtime_error("Error creating RTP session");
}
// Join multicast groups if they are specified
if (multicastIP.size() > 0) {
if (!session.SupportsMulticasting()) {
throw std::runtime_error("Multicast not supported!");
} else {
int joinip = ntohl(inet_addr(multicastIP.c_str()));
RTPIPv4Address joinaddr(joinip, port);
int jstatus = session.JoinMulticastGroup(joinaddr);
if (jstatus < 0) {
throw std::runtime_error("Unable to join multicast group");
}
}
}
return session;
}
void* QTSSCallbacks::Easy_GetRTSPPushSessions()
{
OSRefTable* reflectorSessionMap = QTSServerInterface::GetServer()->GetReflectorSessionMap();
EasyMsgSCRTSPPushSessionListACK ack;
ack.SetHeaderValue(EASY_TAG_VERSION, "1.0");
ack.SetHeaderValue(EASY_TAG_CSEQ, "1");
UInt32 uIndex= 0;
OSMutexLocker locker(reflectorSessionMap->GetMutex());
for (OSRefHashTableIter theIter(reflectorSessionMap->GetHashTable()); !theIter.IsDone(); theIter.Next())
{
OSRef* theRef = theIter.GetCurrent();
ReflectorSession* theSession = (ReflectorSession*)theRef->GetObject();
EasyDarwinRTSPSession session;
session.index = uIndex;
char* fullRequestURL = NULL;
RTPSession* clientSession = (RTPSession*) theSession->GetBroadcasterSession();
if(clientSession == NULL) continue;
clientSession->GetValueAsString(qtssCliSesFullURL,0,&fullRequestURL);
session.Url = fullRequestURL;
session.Name = theSession->GetSessionName();
session.numOutputs = theSession->GetNumOutputs();
ack.AddSession(session);
uIndex++;
}
char count[16] = { 0 };
sprintf(count,"%d", uIndex);
ack.SetBodyValue(EASY_TAG_SESSION_COUNT, count);
string msg = ack.GetMsg();
UInt32 theMsgLen = strlen(msg.c_str());
char* retMsg = new char[theMsgLen+1];
retMsg[theMsgLen] = '\0';
strncpy(retMsg, msg.c_str(), strlen(msg.c_str()));
return (void*)retMsg;
}
void CSenderDlg::OnClose()
{
if(sen)
{sess.Destroy();
sen=false;
}
WSACleanup();
OnOK();
}
void H264MediaSubsession::getStreamParameters(unsigned clientSessionId,
netAddressBits clientAddress,
Port const& clientRTPPort,
Port const& clientRTCPPort,
int tcpSocketNum,
unsigned char rtpChannelId,
unsigned char rtcpChannelId,
netAddressBits& destinationAddress,
u_int8_t& destinationTTL,
Boolean& isMulticast,
Port& serverRTPPort,
Port& serverRTCPPort,
void*& streamToken) {
MyOnDemandMediaSubsession::getStreamParameters(clientSessionId,clientAddress,clientRTPPort,clientRTCPPort,tcpSocketNum,rtpChannelId,rtcpChannelId,destinationAddress,destinationTTL,isMulticast,serverRTPPort,serverRTCPPort,streamToken);
RTPSession *rtp = m_Watcher->StartRecvingVideoRtcp(clientSessionId);
serverRTPPort = rtp->GetLocalPort();
serverRTCPPort = rtp->GetLocalPort() + 1;
Log("-----%u:[%d] %s .... calling\n", clientSessionId, gettid(), __func__);
}
/***********************************************************************************************************
**函数:SimpleGetRtpPoll
**功能:
**输入参数:
**返回值:
***********************************************************************************************************/
int SimpleGetRtpPoll(void)
{
int status;
if(SimpleRtpIsActive()!=true)
{
return -1;
}
status = sess.Poll();
checkerror(status);
return status;
}
/***********************************************************************************************************
**函数:SimpleSetDefaultPayloadType
**功能:
**输入参数:
**返回值:
***********************************************************************************************************/
int SimpleSetDefaultPayloadType(uint8_t pt)
{
if(SimpleRtpIsActive()!=true)
{
printf("rtp Session is error,%s ,%d\n",__FUNCTION__, __LINE__);
return -1;
}
Rtp_Lock();
sess.SetDefaultPayloadType(pt);
Rtp_UnLock();
return 0;
}
/***********************************************************************************************************
**函数:Rtp_Lock
**功能:
**输入参数:
**返回值:
***********************************************************************************************************/
int SimpleDeleteDestination(uint32_t ipaddr,uint16_t port)
{
if(SimpleRtpIsActive()!=true)
{
printf("rtp Session is error,%s ,%d\n",__FUNCTION__, __LINE__);
return -1;
}
Rtp_Lock();
sess.DeleteDestination(ipaddr,port);
Rtp_UnLock();
return 0;
}
/***********************************************************************************************************
**函数:SimpleClearDestinations
**功能:
**输入参数:
**返回值:
***********************************************************************************************************/
int SimpleClearDestinations(void)
{
if(SimpleRtpIsActive()!=true)
{
printf("rtp Session is error,%s ,%d\n",__FUNCTION__, __LINE__);
return -1;
}
Rtp_Lock();
sess.ClearDestinations();
Rtp_UnLock();
return 0;
}
/***********************************************************************************************************
**函数:SimpleIncrementTimestampDefault
**功能:
**输入参数:
**返回值:
***********************************************************************************************************/
int SimpleIncrementTimestampDefault()
{
if(SimpleRtpIsActive()!=true)
{
printf("rtp Session is error,%s ,%d\n",__FUNCTION__, __LINE__);
return -1;
}
Rtp_Lock();
sess.IncrementTimestampDefault();
Rtp_UnLock();
return 0;
}
int AudioStream::DeleteAudioWatcher(unsigned clientSessionId)
{
Log(">DeleteAudioWatcher [%u]\n",clientSessionId);
//El iterator
Rtps::iterator it = rtps.find(clientSessionId);
//Si no esta
if (it == rtps.end())
return Error("Rtps not found\n");
RTPSession* rtp = (*it).second;
rtps.erase(it);
rtp->End();
delete rtp;
Log(">DeleteAudioWatcher [%u] success\n",clientSessionId);
return 1;
}
/***********************************************************************************************************
**函数:Rtp_Lock
**功能:
**输入参数:
**返回值:
***********************************************************************************************************/
int SimpleAddDestination(uint32_t ipaddr, uint16_t destport)
{
int status;
if(SimpleRtpIsActive()!=true)
{
printf("rtp Session is error,%s ,%d\n",__FUNCTION__, __LINE__);
return -1;
}
Rtp_Lock();
RTPIPv4Address addr(ipaddr,destport);
status = sess.AddDestination(addr);
checkerror(status);
Rtp_UnLock();
return status;
}
void Java_cn_nickwar_MainActivity_nativeWorker(JNIEnv* env, jobject obj) {
uint16_t portbase=8000,destport=9000;
std::string ipstr="192.168.1.102";
uint32_t destip=inet_addr(ipstr.c_str());
int status,i,num;
RTPSession session;
RTPSessionParams sessionparams;
RTPUDPv4TransmissionParams transparams;
RTPIPv4Address addr;
if (destip == INADDR_NONE) {
__android_log_print(ANDROID_LOG_DEBUG, "pspm.native", "Bad IP address specified");
}
destip = ntohl(destip);
num = 40;
sessionparams.SetOwnTimestampUnit(1.0/10.0);
sessionparams.SetAcceptOwnPackets(true);
transparams.SetPortbase(portbase);
addr.SetIP(destip);
addr.SetPort(destport);
status = session.Create(sessionparams,&transparams);
if (status<0) {
std::string tmp = "Create:";
__android_log_print(ANDROID_LOG_DEBUG, "pspm.native", (tmp+RTPGetErrorString(status)).c_str());
}
status = session.AddDestination(addr);
if (status<0) {
std::string tmp = "AddDestination:";
__android_log_print(ANDROID_LOG_DEBUG, "pspm.native", (tmp+RTPGetErrorString(status)).c_str());
}
while(!m_bExitApp)
{
session.BeginDataAccess();
unsigned char *buff = NULL;
if (session.GotoFirstSourceWithData())
{
do
{
RTPPacket *pack;
while((pack = session.GetNextPacket()) !=NULL)
{
__android_log_print(ANDROID_LOG_DEBUG, "pspm.native", "got packet!\n");
char message[26];
sprintf(message, "got packet");
jstring messageString = env->NewStringUTF(message);
env->CallVoidMethod(obj, rtpresultFromJNI, messageString);
if (NULL != env->ExceptionOccurred()) {
// break;
continue;
}
if (pack->GetPayloadLength()>0) {
buff = pack->GetPayloadData();
__android_log_print(ANDROID_LOG_DEBUG, "pspm.native", "packt data:%s",buff);
}
session.DeletePacket(pack);
}
}
while(session.GotoNextSourceWithData());
}
session.EndDataAccess();
//
#ifndef RTP_SUPPORT_THREAD
status = sess.Poll();
if (status<0) {
session.Destroy();
return;
}
#endif
RTPTime::Wait(RTPTime(0,5000));
}
session.Destroy();
return;
}
int main(int argc, char ** argv)
{
uint16_t portbase,destport;
uint32_t destip;
std:string ipstr; //get the destiny ip from the std input stream
// NALU_t *n;
printf("Please Enter:./test framenum(500)\n");
if(argc!=2)
{
printf("Please Enter:./test framenum\n");
exit(EXIT_FAILURE);
}
//dev_name=argv[1];
frame_num=atoi(argv[1]); //convert a string to an integer
if(!frame_num)
{
printf("frame_num should be over 0 frame\n");
exit(EXIT_FAILURE);
}
// @0506 here ,there is no need to test YUYV | YUV420 | tmpH264 , for Module Test
//openFiles();
//*****************************************************************************
//****************** the capturing process *****************start*********
printf("capture video begin\n");
open_device();
init_device();
start_capturing();
//fp = fopen(filename, "wa+"); //yuv422
// ***********
//alloc_image();
// ***********
// ***********
//open_x264file("test420.yuv");
init_encoder();
init_picture(yuv_size_2);
// ***********
portbase = 33334;
destport = 9000;
//ipstr = "222.197.174.76";
// @ 1204 change the ip to the LAN IP
//ipstr = "202.115.11.128";
ipstr = "192.168.5.155";
destip = inet_addr(ipstr.c_str());
//destip = ntohl(destip); ///************** very important **************
destip = htonl(destip);
//********************** get the rtp parameter from std input stream ****above ***
// @2 Setting basic parameter ----------------------------------
RTPUDPv4TransmissionParams transparams;
transparams.SetPortbase(portbase);
RTPSessionParams sessparams;
sessparams.SetOwnTimestampUnit(1.0/90000.0);
status=sess.Create(sessparams,&transparams);
checkerror(status);
RTPIPv4Address addr(destip,destport);
status = sess.AddDestination(addr);
checkerror(status);
sess.SetDefaultPayloadType(96);
sess.SetDefaultMark(false);
sess.SetDefaultTimestampIncrement(90000.0/10.0); // frame rate control 10 / 25
RTPTime delay(0.040);//RTPTime delay(0.040)
//RTPTime::Wait(delay);
RTPTime starttime=RTPTime::CurrentTime();
// // @3.0 Preparation for sending NALU package----------------------------------------------------
// char sendbuf[1500];
// char *nalu_payload;
// int size=0;
// unsigned int timestamp_increase=0,ts_current=0;
// OpenBitstreamFile("test.h264");
// n=AllocNALU(8000000);// alloc memory for the <struct NALU> n->maxSize = 8_000_000 = 8 MB
// int start=false;
// //****************************************************************
mainloop(frame_num); // **** capture frame *****
// fclose(fp);
stop_capturing();
uninit_device();
close_device();
close_encoder();
printf("capture video is over\n");
//****************** the capturing process ********************end ******
//****************** x264 encode ******************************* start ***
// open_yuvfile(filename);
// alloc_image();
// swscale_start(); // ****** yuv422 -> I420 *******
// swscale_close();
// open_x264file(dst_filename);
// init_encoder();
// init_picture(yuv_size);
// encode_frame(); // **** encode x264 format ***
// close_encoder();
//****************** x264 encode ******************************* end ***
/////---------------------------------------------------------------------------------------------------
printf("encode x264 video is over\n");
printf("Wait to jrtpsend....\n");
//****************** RTP send *********************** start **********
// @1 Getting basic parameter -------------------------------
// std::cout<<"Enter local portbase:"<<std::endl;
// std::cin>>portbase;
// std::cout<<std::endl;
// std::cout << "Enter the destination IP address" << std::endl;
// std::cin >> ipstr;
// destip = inet_addr(ipstr.c_str());
// if (destip == INADDR_NONE)
// {
// std::cerr << "Bad IP address specified" << std::endl;
// return -1;
// }
// destip = ntohl(destip);
// std::cout << "Enter the destination port" << std::endl;
// std::cin >> destport;
// portbase = 33333;
// destport = 55555;
// ipstr = "222.197.174.76";
// destip = inet_addr(ipstr.c_str());
// //********************** get the rtp parameter from std input stream ****above ***
// // @2 Setting basic parameter ----------------------------------
// RTPUDPv4TransmissionParams transparams;
// transparams.SetPortbase(portbase);
// RTPSessionParams sessparams;
// sessparams.SetOwnTimestampUnit(1.0/90000.0);
// status=sess.Create(sessparams,&transparams);
// checkerror(status);
// RTPIPv4Address addr(destip,destport);
// status = sess.AddDestination(addr);
// checkerror(status);
// sess.SetDefaultPayloadType(96);
// sess.SetDefaultMark(false);
// sess.SetDefaultTimestampIncrement(90000.0/10.0);
// RTPTime delay(0.030);//RTPTime delay(0.040)
// //RTPTime::Wait(delay);
// RTPTime starttime=RTPTime::CurrentTime();
// // @3.0 Preparation for sending NALU package----------------------------------------------------
// char sendbuf[1500];
// char *nalu_payload;
// int size=0;
// unsigned int timestamp_increase=0,ts_current=0;
// OpenBitstreamFile("test.h264");
// n=AllocNALU(8000000);// alloc memory for the <struct NALU> n->maxSize = 8_000_000 = 8 MB
// int start=false;
// NALU packages sending loop -----------------------------------
// while(!feof(bits))
// {
// size=GetAnnexbNALU(n); // having original souce for this function
// if(size<4)
// {
// printf("get nalu error!\n");
// continue;
// }
// printf("size:%d\n",size);
// dump(n);
// if(!start)
// {
// if(n->nal_unit_type==1||n->nal_unit_type==5||n->nal_unit_type==6||n->nal_unit_type==7)
// {
// printf("begin\n");
// start=true;
// }
// }
// if(n->len <= MAX_RTP_PKT_LENGTH)
// {
// nalu_hdr=(NALU_HEADER *)&sendbuf[0];
// nalu_hdr->F=n->forbidden_bit;
// nalu_hdr->NRI=n->nal_reference_idc>>5;
// nalu_hdr->TYPE=n->nal_unit_type;
// nalu_payload=&sendbuf[1];
// memcpy(nalu_payload,n->buf+1,n->len-1);
// ts_current=ts_current+timestamp_increase;
// if(n->nal_unit_type==1||n->nal_unit_type==5)
// {
// status=sess.SendPacket((void *)sendbuf,n->len,96,true,3600);
// }
// else
// {
// status=sess.SendPacket((void *)sendbuf,n->len,96,true,0);
// continue;
// }
// checkerror(status);
// }
// else if(n->len > MAX_RTP_PKT_LENGTH)
// {
// int k=0,l=0;
// k=n->len/MAX_RTP_PKT_LENGTH;
// l=n->len%MAX_RTP_PKT_LENGTH;
// int t=0;
// while(t<=k)
// {
// if(!t)//first pkt package
// {
// memset(sendbuf,0,1500);
// fu_ind=(FU_INDICATOR *)&sendbuf[0];
// fu_ind->F=n->forbidden_bit;
// fu_ind->NRI=n->nal_reference_idc>>5;
// fu_ind->TYPE=28;//FU-A
// fu_hdr=(FU_HEADER *)&sendbuf[1];
// fu_hdr->E=0;
// fu_hdr->R=0;
// fu_hdr->S=1;
// fu_hdr->TYPE=n->nal_unit_type;
// nalu_payload=&sendbuf[2];
// memcpy(nalu_payload,n->buf+1,MAX_RTP_PKT_LENGTH);
// status=sess.SendPacket((void *)sendbuf,MAX_RTP_PKT_LENGTH+2,96,false,0);
// checkerror(status);
// t++;
// }
// else if(t==k)//last package
// {
// memset(sendbuf,0,1500);
// fu_ind=(FU_INDICATOR *)&sendbuf[0];
// fu_ind->F=n->forbidden_bit;
// fu_ind->NRI=n->nal_reference_idc>>5;
// fu_ind->TYPE=28;//FU-A
// fu_hdr=(FU_HEADER *)&sendbuf[1];
// fu_hdr->R=0;
// fu_hdr->S=0;
// fu_hdr->E=1;
// fu_hdr->TYPE=n->nal_unit_type;
// nalu_payload=&sendbuf[2];
// memcpy(nalu_payload,n->buf+t*MAX_RTP_PKT_LENGTH+1,l-1);
// status=sess.SendPacket((void *)sendbuf,l+1,96,true,3600);
// checkerror(status);
// t++;
// }
// else if( (t<k) && (t!=0) ) //packageS between the first and the last
// {
// memset(sendbuf,0,1500);
// fu_ind=(FU_INDICATOR *)&sendbuf[0];
// fu_ind->F=n->forbidden_bit;
// fu_ind->NRI=n->nal_reference_idc>>5;
// fu_ind->TYPE=28;//FU-A
// fu_hdr=(FU_HEADER *)&sendbuf[1];
// fu_hdr->R=0;
// fu_hdr->S=0;
// fu_hdr->E=0;//E=1
// fu_hdr->TYPE=n->nal_unit_type;
// nalu_payload=&sendbuf[2];
// memcpy(nalu_payload,n->buf+t*MAX_RTP_PKT_LENGTH+1,MAX_RTP_PKT_LENGTH);
// status=sess.SendPacket((void *)sendbuf,MAX_RTP_PKT_LENGTH+2,96,false,0);
// checkerror(status);
// t++;
// }
// }
// }
// RTPTime::Wait(delay);//Wait(delay);
// RTPTime time=RTPTime::CurrentTime();
// time-=starttime;
// if(time>RTPTime(60.0))break;
// }
// // @4 Finish sending NALU package
// printf("Mission over\n");
// delay=RTPTime(10.0);
// sess.BYEDestroy(delay,"Time's up",9);
return 0;
}
int main(void)
{
int status,i;
RTPSession s;
RTPSessionParams sessparams;
RTPUDPv4TransmissionParams transparams;
char blaai[100];
transparams.SetPortbase(10000);
sessparams.SetOwnTimestampUnit(1.0/8000.0);
sessparams.SetUsePollThread(true);
sessparams.SetMaximumPacketSize(10000);
status = s.Create(sessparams,&transparams);
s.SetLocalName((const uint8_t *)"Jori Liesenborgs",16);
s.SetLocalEMail((const uint8_t *)"*****@*****.**",20);
s.SetLocalNote((const uint8_t *)"Blaai",5);
s.SetNameInterval(3);
s.SetEMailInterval(5);
s.SetNoteInterval(2);
status = s.AddDestination(RTPIPv4Address(ntohl(inet_addr("192.168.2.115")),5000));
//status = s.AddDestination(RTPIPv4Address(ntohl(inet_addr("127.0.0.1")),7000));
int snd = open("/dev/dsp",O_RDWR);
int val;
val = 0;
status = ioctl(snd,SNDCTL_DSP_STEREO,&val);
val = 8;
status = ioctl(snd,SNDCTL_DSP_SAMPLESIZE,&val);
val = 8000;
status = ioctl(snd,SNDCTL_DSP_SPEED,&val);
val = 7 | (128<<16);
ioctl(snd,SNDCTL_DSP_SETFRAGMENT,&val);
i = 0;
while (i++ < 40000)
{
if (i == 1000)
{
//std::cout <<"Disabling note" << std::endl;
s.SetNoteInterval(0);
}
uint8_t data[PACKSIZE];
RTPTime t1 = RTPTime::CurrentTime();
status = read(snd,data,PACKSIZE);
RTPTime t2 = RTPTime::CurrentTime();
t2 -= t1;
printf("%d.%06d\n",t2.GetSeconds(),t2.GetMicroSeconds());
status = s.SendPacket(data,PACKSIZE,1,false,PACKSIZE);
}
close(snd);
printf("Destroying...\n");
s.BYEDestroy(RTPTime(10,0),(const uint8_t *)"Leaving session",16);
return 0;
}
// This method does almost everything.
void run(void) {
// redefined from Thread.
// Before using ccRTP you should learn something about other
// CommonC++ classes. We need InetHostAddress...
// Construct loopback address
InetHostAddress local_ip;
local_ip = "127.0.0.1";
// Is that correct?
if( ! local_ip ){
// this is equivalent to `! local_ip.isInetAddress()'
cerr << ": IP address is not correct!" << endl;
exit();
}
cout << local_ip.getHostname() <<
" is going to transmit audio to perself through " <<
local_ip << "..." << endl;
// ____Here comes the real RTP stuff____
// Construct the RTP socket.
socket = new RTPSession(local_ip,TRANSMITTER_BASE);
// Set up connection
socket->setSchedulingTimeout(10000);
if( !socket->addDestination(local_ip,RECEIVER_BASE) )
cerr << "I could not connect.";
socket->setPayloadFormat(StaticPayloadFormat(sptPCMU));
socket->startRunning();
cout << "The RTP queue service thread is ";
if( socket->isActive() )
cout << "active." << endl;
else
cerr << "not active." << endl;
cout << "Transmitting " << PACKET_SIZE
<< " octects long packets "
<< "every " << PERIOD << " milliseconds..." << endl;
unsigned char buffer[PACKET_SIZE];
int count=PACKET_SIZE;
// This will be useful for periodic execution
TimerPort::setTimer(PERIOD);
setCancel(cancelImmediate);
// This is the main loop, where packets are transmitted.
for( int i = 0 ; (!sendingfile || count > 0) ; i++ ){
count = ::read(audioinput,buffer,PACKET_SIZE);
if( count > 0 ){
// send an RTP packet, providing timestamp,
// payload type and payload.
socket->putData(PACKET_SIZE*i,buffer,
PACKET_SIZE);
}
cout << "." << flush;
// Let's wait for the next cycle
Thread::sleep(TimerPort::getTimer());
TimerPort::incTimer(PERIOD);
}
cout << endl << "I have got no more data to send. " <<endl;
}
int main(void)
{
int packetsPerSecond = 100;
MIPTime interval(1.0/(double)packetsPerSecond); // We'll use 10 millisecond intervals.
MIPAverageTimer timer(interval);
MIPOSCInput oscInput;
MIPOSCEncoder oscEnc;
MIPRTPOSCEncoder rtpEnc;
MIPRTPComponent rtpComp;
MIPRTPDecoder rtpDec;
MIPRTPOSCDecoder rtpOSCDec;
MIPOSCDecoder oscDec;
MIPOSCOutput oscOutput;
MyChain chain("OSC Sender");
RTPSession rtpSession;
bool returnValue;
// Convert Messages to MIPOSCMessages
returnValue = oscEnc.init();
checkError(returnValue, oscEnc);
// Initialize the RTP OSC encoder: this component will create
// RTP messages which can be sent to the RTP component.
returnValue = rtpEnc.init();
checkError(returnValue, rtpEnc);
// We'll initialize the RTPSession object which is needed by the
// RTP component.
RTPUDPv4TransmissionParams transmissionParams;
RTPSessionParams sessionParams;
int portBase = 60000;
int status;
transmissionParams.SetPortbase(portBase);
sessionParams.SetOwnTimestampUnit(1.0/((double)packetsPerSecond));
sessionParams.SetMaximumPacketSize(64000);
sessionParams.SetAcceptOwnPackets(true);
status = rtpSession.Create(sessionParams,&transmissionParams);
checkError(status);
// Instruct the RTP session to send data to ourselves.
status = rtpSession.AddDestination(RTPIPv4Address(ntohl(inet_addr("127.0.0.1")),portBase));
checkError(status);
// Tell the RTP component to use this RTPSession object.
returnValue = rtpComp.init(&rtpSession);
checkError(returnValue, rtpComp);
returnValue = rtpDec.init(false, 0, &rtpSession);
checkError(returnValue, rtpDec);
returnValue = rtpDec.setPacketDecoder(0, &rtpOSCDec);
checkError(returnValue, rtpDec);
returnValue = oscDec.init();
checkError(returnValue, oscDec);
// Next, we'll create the chain
returnValue = chain.setChainStart(&timer);
checkError(returnValue, chain);
returnValue = chain.addConnection(&timer, &oscInput);
checkError(returnValue, chain);
returnValue = chain.addConnection(&oscInput, &oscEnc);
checkError(returnValue, chain);
returnValue = chain.addConnection(&oscEnc, &rtpEnc);
checkError(returnValue, chain);
returnValue = chain.addConnection(&rtpEnc, &rtpComp);
checkError(returnValue, chain);
returnValue = chain.addConnection(&rtpComp, &rtpDec);
checkError(returnValue, chain);
returnValue = chain.addConnection(&rtpDec, &oscDec, true);
checkError(returnValue, chain);
returnValue = chain.addConnection(&oscDec, &oscOutput);
checkError(returnValue, chain);
// Start the chain
returnValue = chain.start();
checkError(returnValue, chain);
// We'll wait until enter is pressed
int counter = 0;
sleep(1);
for(int i=0; i<4; i++) {
lo_message m = lo_message_new();
lo_message_add_int32(m,counter++);
oscInput.push(m, "/testpfad");
sleep(1);
}
getc(stdin);
returnValue = chain.stop();
checkError(returnValue, chain);
rtpSession.Destroy();
return 0;
}
int main(int argc, char** argv)
{
#if 0
CRTPSender sender;
string destip_str = "127.0.0.1";
uint32_t dest_ip = inet_addr(destip_str.c_str());
SetRTPParams(sender,dest_ip,DEST_PORT,BASE_PORT);
sender.SetParamsForSendingH264();
#else
RTPSession session;
RTPSessionParams sessionparams;
sessionparams.SetOwnTimestampUnit(1.0/90000.0);
RTPUDPv4TransmissionParams transparams;
transparams.SetPortbase(8000);
int status = session.Create(sessionparams,&transparams);
if (status < 0)
{
std::cerr << RTPGetErrorString(status) << std::endl;
exit(-1);
}
uint8_t localip[]={127,0,0,1};
RTPIPv4Address addr(localip,9000);
status = session.AddDestination(addr);
if (status < 0)
{
std::cerr << RTPGetErrorString(status) << std::endl;
exit(-1);
}
session.SetDefaultPayloadType(96);
session.SetDefaultMark(false);
session.SetDefaultTimestampIncrement(90000.0 /25.0);
RTPTime delay(0.040);
RTPTime starttime = RTPTime::CurrentTime();
#endif
NALU_HEADER *nalu_hdr;
FU_INDICATOR *fu_ind;
FU_HEADER *fu_hdr;
char sendbuf[1500];
char* nalu_payload;
unsigned int timestamp_increse=0,ts_current=0;
#define ddd
OpenBitstreamFile("raw.264");//打开264文件，并将文件指针赋给bits,在此修改文件名实现打开别的264文件。
NALU_t *n;
n = AllocNALU(8000000);//为结构体nalu_t及其成员buf分配空间。返回值为指向nalu_t存储空间的指针
bool start=false;
while(!feof(bits))
{
int size=GetAnnexbNALU(n);//每执行一次，文件的指针指向本次找到的NALU的末尾，下一个位置即为下个NALU的起始码0x000001
if(size<4)
{
printf("get nul error!\n");
continue;
}
dump(n);//输出NALU长度和TYPE
if(!start)
{
if(n->nal_unit_type==5||n->nal_unit_type==6||
n->nal_unit_type==7||n->nal_unit_type==7)
{
printf("begin\n");
start=true;
}
}
//将编码数据写入文件t
//fwrite(pNals[i].p_payload, 1, pNals[i].i_payload, pFile);
//发送编码文件
#if 1
// 当一个NALU小于MAX_RTP_PKT_LENGTH字节的时候，采用一个单RTP包发送
if(n->len<=MAX_RTP_PKT_LENGTH)
{
//printf("ddd0\n");
//session.SetDefaultMark(false);
//设置NALU HEADER,并将这个HEADER填入sendbuf[12]
nalu_hdr =(NALU_HEADER*)&sendbuf[0]; //将sendbuf[12]的地址赋给nalu_hdr，之后对nalu_hdr的写入就将写入sendbuf中；
nalu_hdr->F=n->forbidden_bit;
nalu_hdr->NRI=n->nal_reference_idc>>5;//有效数据在n->nal_reference_idc的第6，7位，需要右移5位才能将其值赋给nalu_hdr->NRI。
nalu_hdr->TYPE=n->nal_unit_type;
nalu_payload=&sendbuf[1];//同理将sendbuf[13]赋给nalu_payload
memcpy(nalu_payload,n->buf+1,n->len-1);//去掉nalu头的nalu剩余内容写入sendbuf[13]开始的字符串。
ts_current=ts_current+timestamp_increse;
//status = session.SendPacket((void *)sendbuf,n->len);
if(n->nal_unit_type==1 || n->nal_unit_type==5)
{
status = session.SendPacket((void *)sendbuf,n->len,96,true,3600);
}
else
{
status = session.SendPacket((void *)sendbuf,n->len,96,true,0);\
//如果是6,7类型的包，不应该延时；之前有停顿，原因这在这
continue;
}
//发送RTP格式数据包并指定负载类型为96
if (status < 0)
{
std::cerr << RTPGetErrorString(status) << std::endl;
exit(-1);
}
}
else if(n->len>MAX_RTP_PKT_LENGTH)
/***********************************************************************************************************
**函数:SimpleH264SendPacket
**功能:
**输入参数:
**返回值:
***********************************************************************************************************/
int SimpleH264SendPacket(unsigned char *val,uint32_t length)
{
/*
*val:为264的原始数据，包含0x00 0x00 0x00 0x01信息，
*/
int status=0;
uint32_t TimestampIncrement;
uint32_t send_length,valid_len=length-4;
char NALU=val[4],*sendStartAddr=NULL;
#define MAX_STREAM_SLICE 1024
if(SimpleRtpIsActive()!=true)
{
printf("rtp Session is error,%s ,%d\n",__FUNCTION__, __LINE__);
return -1;
}
Rtp_Lock();
TimestampIncrement=sess.GetDefaultTimestampIncrement();
if(valid_len<=MAX_STREAM_SLICE)
{
status = sess.SendPacket((void *)&val[4],valid_len,PLOAD_TYPE,true,DefaultTimestampIncrement);
checkerror(status);
}
else
{
int k=0,l=0,len=valid_len,pos=0;
k=len/MAX_STREAM_SLICE;
l=len%MAX_STREAM_SLICE;
int t=0;
while(t<=k)
{
if(t==0)
{//发送第1包
sendStartAddr=(char *)(val+4);//发送数据的起始地址
sendStartAddr[pos-1]=(NALU&0x60)|28;//FU indicator
sendStartAddr[pos]=(NALU&0x1f)|0x80;//FU header
send_length=MAX_STREAM_SLICE+1;//要发送数据的长度
status = sess.SendPacket(sendStartAddr-1,send_length,PLOAD_TYPE,false,DefaultTimestampIncrement);
checkerror(status);
if(status<0)
{
goto end;
}
pos+=MAX_STREAM_SLICE;
}
else if(k==t&&l!=0)
{//最后一包
sendStartAddr[pos-2]=(NALU&0x60)|28;//FU indicator
sendStartAddr[pos-1]=(NALU&0x1f)|0x40;//FU header
send_length=l+2;//要发送数据的长度
status = sess.SendPacket(sendStartAddr+pos-2,send_length,PLOAD_TYPE,true,0);
checkerror(status);
if(status<0)
{
goto end;
}
pos+=l;
}
else if(t<k&&0!=t)
{//发送第2包-----倒数第2包
if(l==0&&t==k-1)
{
sendStartAddr[pos-2]=(NALU&0x60)|28;//FU indicator
sendStartAddr[pos-1]=(NALU&0x1f)|0x40;//FU header
}
else
{
sendStartAddr[pos-2]=(NALU&0x60)|28;//FU indicator
sendStartAddr[pos-1]=(NALU&0x1f);//FU header
}
send_length=MAX_STREAM_SLICE+2;//要发送数据的长度
status = sess.SendPacket(sendStartAddr+pos-2,send_length,PLOAD_TYPE,false,0);
checkerror(status);
if(status<0)
{
goto end;
}
pos+=MAX_STREAM_SLICE;
}
t++;
}
}
#ifndef RTP_SUPPORT_THREAD
checkerror(status);
#endif
end:
Rtp_UnLock();
return status;
}
int main(int argc, char ** argv)
{
uint16_t portbase,destport;
uint32_t destip;
std:string ipstr; //get the destiny ip from the std input stream
// NALU_t *n;
printf("Please Enter:./test framenum(500)\n");
if(argc!=2)
{
printf("Please Enter:./test framenum\n");
exit(EXIT_FAILURE);
}
//dev_name=argv[1];
frame_num=atoi(argv[1]); //convert a string to an integer
if(!frame_num)
{
printf("frame_num should be over 0 frame\n");
exit(EXIT_FAILURE);
}
// @0506 here ,there is no need to test YUYV | YUV420 | tmpH264 , for Module Test
//openFiles();
//*****************************************************************************
//****************** the capturing process *****************start*********
printf("capture video begin\n");
open_device();
init_device();
start_capturing();
//fp = fopen(filename, "wa+"); //yuv422
// ***********
//alloc_image();
// ***********
// ***********
//open_x264file("test420.yuv");
init_encoder();
init_picture(yuv_size_2);
// ***********
portbase = 33334;
destport = 9000;
//ipstr = "222.197.174.76";
ipstr = "202.115.11.128";
destip = inet_addr(ipstr.c_str());
//destip = ntohl(destip); ///************** very important **************
destip = htonl(destip);
//********************** get the rtp parameter from std input stream ****above ***
// @2 Setting basic parameter ----------------------------------
RTPUDPv4TransmissionParams transparams;
transparams.SetPortbase(portbase);
RTPSessionParams sessparams;
sessparams.SetOwnTimestampUnit(1.0/90000.0);
status=sess.Create(sessparams,&transparams);
checkerror(status);
RTPIPv4Address addr(destip,destport);
status = sess.AddDestination(addr);
checkerror(status);
sess.SetDefaultPayloadType(96);
sess.SetDefaultMark(false);
sess.SetDefaultTimestampIncrement(90000.0/10.0); // frame rate control 10 / 25
RTPTime delay(0.040);//RTPTime delay(0.040)
//RTPTime::Wait(delay);
RTPTime starttime=RTPTime::CurrentTime();
// // @3.0 Preparation for sending NALU package----------------------------------------------------
// char sendbuf[1500];
// char *nalu_payload;
// int size=0;
// unsigned int timestamp_increase=0,ts_current=0;
// OpenBitstreamFile("test.h264");
// n=AllocNALU(8000000);// alloc memory for the <struct NALU> n->maxSize = 8_000_000 = 8 MB
// int start=false;
// //****************************************************************
//****************** the capturing process ********************end ******
//****************** x264 encode ******************************* start ***
// open_yuvfile(filename);
// alloc_image();
// swscale_start(); // ****** yuv422 -> I420 *******
// swscale_close();
// open_x264file(dst_filename);
// init_encoder();
// init_picture(yuv_size);
// encode_frame(); // **** encode x264 format ***
// close_encoder();
//****************** x264 encode ******************************* end ***
//-----------------------------------------------------------
//@1223--01
struct timeval tpstart,tpend;
float timeuse;
gettimeofday(&tpstart,NULL);
///------ implement capture the pictures next line -----------------------------
tmpFP264 = fopen("softEncodeH264.h264","wb");
mainloop(frame_num); // **** capture frame *****
fclose(tmpFP264);
///-----------------------------------------------------------------------------
gettimeofday(&tpend,NULL);
timeuse=1000000 * (tpend.tv_sec-tpstart.tv_sec) + tpend.tv_usec - tpstart.tv_usec;
timeuse/=1000000;
printf("timeuse total Used Time( Second ):%f\n",timeuse);
//@1223--02
//-----------------------------------------------------------
stop_capturing();
uninit_device();
close_device();
close_encoder();
return 0;
}
// This method does almost everything.
void run(void){
// redefined from Thread.
// Before using ccRTP you should learn something about other
// CommonC++ classes. We need InetHostAddress...
// Construct loopback address
InetHostAddress local_ip;
local_ip = "127.0.0.1";
// Is that correct?
if( ! local_ip ){
// this is equivalent to `! local_ip.isInetAddress()'
cerr << ": IP address is not correct!" << endl;
exit();
}
cout << local_ip.getHostname() <<
" is going to listen to perself through " <<
local_ip << "..." << endl;
// ____Here comes the real RTP stuff____
// Construct the RTP socket
socket = new RTPSession(local_ip,RECEIVER_BASE,0);
// Set up receiver's connection
socket->setSchedulingTimeout(10000);
if( !socket->addDestination(local_ip,TRANSMITTER_BASE) )
cerr << "The receiver could not connect.";
// Let's check the queue (you should read the documentation
// so that you know what the queue is for).
socket->startRunning();
cout << "The RTP queue is ";
if( socket->isActive() )
cout << "active." << endl;
else
cerr << "not active." << endl;
cout << "Waiting for audio packets..." << endl;
// This will be useful for periodic execution.
TimerPort::setTimer(PERIOD);
setCancel(cancelImmediate);
// This is the main loop, where packets are sent and receipt.
socket->setPayloadFormat(StaticPayloadFormat(sptPCMU));
for( int i=0 ; true ; i++ ){
const AppDataUnit* adu;
do{
adu = socket->getData(socket->getFirstTimestamp());
if ( NULL == adu )
Thread::sleep(5);
else cout << ".";
}while ( (NULL == adu) || (adu->getSize() <= 0) );
// This is for buffering some packets at the
// receiver side, since playing smoothly
// without any reception buffer is almost
// impossible. Try commenting the two lines
// below, or stop transmission and continue
// later: you will probably hear noise or
// cracks.
if (i==0)
Thread::sleep(20);
::write(audiooutput,adu->getData(),adu->getSize());
cout << "." << flush;
// Let's wait for the next cycle
Thread::sleep(TimerPort::getTimer());
TimerPort::incTimer(PERIOD);
}
} // end of run
int main(void)
{
#ifdef NEED_PA_INIT
std::string errStr;
if (!MIPPAInputOutput::initializePortAudio(errStr))
{
std::cerr << "Can't initialize PortAudio: " << errStr << std::endl;
return -1;
}
#endif // NEED_PA_INIT
#ifdef WIN32
WSADATA dat;
WSAStartup(MAKEWORD(2,2),&dat);
#endif // WIN32
MIPTime interval(0.020); // We'll use 20 millisecond intervals.
MIPAverageTimer timer(interval);
MIPWAVInput sndFileInput;
MIPSamplingRateConverter sampConv, sampConv2;
MIPSampleEncoder sampEnc, sampEnc2, sampEnc3;
MIPULawEncoder uLawEnc;
MIPRTPULawEncoder rtpEnc;
MIPRTPComponent rtpComp;
MIPRTPDecoder rtpDec;
MIPRTPULawDecoder rtpULawDec;
MIPULawDecoder uLawDec;
MIPAudioMixer mixer;
MIPComponentAlias rtpCompAlias(&rtpComp);
ToggleOutputComponent sndToggleComponent(&sndFileInput);
#ifdef MIPCONFIG_SUPPORT_WINMM
MIPWinMMOutput sndCardOutput;
#else
#ifdef MIPCONFIG_SUPPORT_OSS
MIPOSSInputOutput sndCardOutput;
#else
MIPPAInputOutput sndCardOutput;
#endif
#endif
MyChain chain("Sound file player");
RTPSession rtpSession;
bool returnValue;
// We'll open the file 'soundfile.wav'.
returnValue = sndFileInput.open("soundfile.wav", interval);
checkError(returnValue, sndFileInput);
// We'll convert to a sampling rate of 8000Hz and mono sound.
int samplingRate = 8000;
int numChannels = 1;
returnValue = sampConv.init(samplingRate, numChannels);
checkError(returnValue, sampConv);
// Initialize the sample encoder: the RTP U-law audio encoder
// expects native endian signed 16 bit samples.
returnValue = sampEnc.init(MIPRAWAUDIOMESSAGE_TYPE_S16);
checkError(returnValue, sampEnc);
// Convert samples to U-law encoding
returnValue = uLawEnc.init();
checkError(returnValue, uLawEnc);
// Initialize the RTP audio encoder: this component will create
// RTP messages which can be sent to the RTP component.
returnValue = rtpEnc.init();
checkError(returnValue, rtpEnc);
// We'll initialize the RTPSession object which is needed by the
// RTP component.
RTPUDPv4TransmissionParams transmissionParams;
RTPSessionParams sessionParams;
int portBase = 60000;
int status;
transmissionParams.SetPortbase(portBase);
sessionParams.SetOwnTimestampUnit(1.0/((double)samplingRate));
sessionParams.SetMaximumPacketSize(64000);
sessionParams.SetAcceptOwnPackets(true);
status = rtpSession.Create(sessionParams,&transmissionParams);
checkError(status);
// Instruct the RTP session to send data to ourselves.
status = rtpSession.AddDestination(RTPIPv4Address(ntohl(inet_addr("127.0.0.1")),portBase));
checkError(status);
// Tell the RTP component to use this RTPSession object.
returnValue = rtpComp.init(&rtpSession, 160); // 20ms at 8000Hz = 160 samples per RTP packet
checkError(returnValue, rtpComp);
// Initialize the RTP audio decoder.
returnValue = rtpDec.init(true, 0, &rtpSession);
checkError(returnValue, rtpDec);
// Register the U-law decoder for payload type 0
returnValue = rtpDec.setPacketDecoder(0,&rtpULawDec);
checkError(returnValue, rtpDec);
// Convert U-law encoded samples to linear encoded samples
returnValue = uLawDec.init();
checkError(returnValue, uLawDec);
// Transform the received audio data to floating point format.
returnValue = sampEnc2.init(MIPRAWAUDIOMESSAGE_TYPE_FLOAT);
checkError(returnValue, sampEnc2);
// We'll make sure that received audio frames are converted to the right
// sampling rate.
returnValue = sampConv2.init(samplingRate, numChannels);
checkError(returnValue, sampConv2);
// Initialize the mixer.
returnValue = mixer.init(samplingRate, numChannels, interval);
checkError(returnValue, mixer);
// Initialize the soundcard output.
returnValue = sndCardOutput.open(samplingRate, numChannels, interval);
checkError(returnValue, sndCardOutput);
#ifdef MIPCONFIG_SUPPORT_WINMM
// The WinMM output component uses signed little endian 16 bit samples.
returnValue = sampEnc3.init(MIPRAWAUDIOMESSAGE_TYPE_S16LE);
#else
#ifdef MIPCONFIG_SUPPORT_OSS
// The OSS component can use several encoding types. We'll ask
// the component to which format samples should be converted.
returnValue = sampEnc3.init(sndCardOutput.getRawAudioSubtype());
#else
// The PortAudio output component uses signed 16 bit samples
returnValue = sampEnc3.init(MIPRAWAUDIOMESSAGE_TYPE_S16);
#endif
#endif
checkError(returnValue, sampEnc3);
// Next, we'll create the chain
returnValue = chain.setChainStart(&timer);
checkError(returnValue, chain);
returnValue = chain.addConnection(&timer, &sndToggleComponent);
checkError(returnValue, chain);
returnValue = chain.addConnection(&sndToggleComponent, &sampConv);
checkError(returnValue, chain);
returnValue = chain.addConnection(&sampConv, &sampEnc);
checkError(returnValue, chain);
returnValue = chain.addConnection(&sampEnc, &uLawEnc);
checkError(returnValue, chain);
returnValue = chain.addConnection(&uLawEnc, &rtpEnc);
checkError(returnValue, chain);
returnValue = chain.addConnection(&rtpEnc, &rtpComp);
checkError(returnValue, chain);
returnValue = chain.addConnection(&timer, &rtpCompAlias);
checkError(returnValue, chain);
returnValue = chain.addConnection(&rtpCompAlias, &rtpDec);
checkError(returnValue, chain);
// This is where the feedback chain is specified: we want
// feedback from the mixer to reach the RTP audio decoder,
// so we'll specify that over the links in between, feedback
// should be transferred.
returnValue = chain.addConnection(&rtpDec, &uLawDec, true);
checkError(returnValue, chain);
returnValue = chain.addConnection(&uLawDec, &sampEnc2, true);
checkError(returnValue, chain);
returnValue = chain.addConnection(&sampEnc2, &sampConv2, true);
checkError(returnValue, chain);
returnValue = chain.addConnection(&sampConv2, &mixer, true);
checkError(returnValue, chain);
returnValue = chain.addConnection(&mixer, &sampEnc3);
checkError(returnValue, chain);
returnValue = chain.addConnection(&sampEnc3, &sndCardOutput);
checkError(returnValue, chain);
// Start the chain
returnValue = chain.start();
checkError(returnValue, chain);
// We'll wait until enter is pressed
int num = 10;
for (int i = 0 ; i < num ; i++)
{
std::cout << "iteration " << (i+1) << "/" << num << std::endl;
std::cout << "Press enter for silence" << std::endl;
getc(stdin);
sndToggleComponent.lock();
sndToggleComponent.setEnabled(false);
sndToggleComponent.unlock();
std::cout << "Press enter for sound" << std::endl;
getc(stdin);
sndToggleComponent.lock();
sndToggleComponent.setEnabled(true);
sndToggleComponent.unlock();
}
returnValue = chain.stop();
checkError(returnValue, chain);
rtpSession.Destroy();
// We'll let the destructors of the other components take care
// of their de-initialization.
sndCardOutput.close(); // In case we're using PortAudio
#ifdef NEED_PA_INIT
MIPPAInputOutput::terminatePortAudio();
#endif // NEED_PA_INIT
#ifdef WIN32
WSACleanup();
#endif
return 0;
}
int main(void)
{
RTPSession sess;
int portbase;
unsigned long destip;
int destport;
char ipstr[256];
int status;
char dummybuffer[1024];
struct timeval rttprev = {0,0},rtt,tv;
int sock1,sock2;
bool done;
fd_set fdset;
/*
First, we'll ask for the necessary information
*/
printf("Enter the local portbase\n");
scanf("%d",&portbase);
printf("\n");
printf("Enter the destination IP address\n");
scanf("%s",ipstr);
destip = inet_addr(ipstr);
if (destip == INADDR_NONE)
{
printf("Bad IP address specified\n");
return -1;
}
// The inet_addr function returns a value in network byte order, but
// we need the IP address in host byte order, so we use a call to
// ntohl
destip = ntohl(destip);
printf("Enter the destination port\n");
scanf("%d",&destport);
/*
Now, we'll create a RTP session, set the destination, send some
packets and poll for incoming data.
*/
status = sess.Create(portbase);
checkerror(status);
/* Get the sockets, so we can use them in a 'select' call */
sess.GetRTPSocket(&sock1);
sess.GetRTCPSocket(&sock2);
status = sess.AddDestination(destip,destport);
checkerror(status);
printf("Press return to quit...\n");
done = false;
while (!done)
{
/* Just send something */
status = sess.SendPacket("1234567890",10,0,false,10);
checkerror(status);
/* Wait for incoming data, or wait just one second */
tv.tv_sec = 1;
tv.tv_usec = 0;
FD_ZERO(&fdset);
FD_SET(sock1,&fdset);
FD_SET(sock2,&fdset);
FD_SET(0,&fdset); // check for keypress
select(FD_SETSIZE,&fdset,NULL,NULL,&tv);
if (FD_ISSET(0,&fdset))
done = true;
/* poll for incoming data */
status = sess.PollData();
/* check incoming packets */
if (sess.GotoFirstSourceWithData())
{
do
{
RTPSourceData *srcdat;
srcdat = sess.GetCurrentSourceInfo();
srcdat->FlushPackets(); // we don't need the actual data
rtt = srcdat->INF_GetRoundTripTime();
if (rtt.tv_sec != 0 || rtt.tv_usec != 0)
{
if ((rtt.tv_sec != rttprev.tv_sec) || (rtt.tv_usec != rttprev.tv_usec))
{
double t;
t = (double)rtt.tv_sec;
t += ((double)rtt.tv_usec)/1000000.0;
t *= 1000.0; // we want milliseconds;
printf("rtt: %f ms\n",(float)t);
rttprev = rtt;
}
}
} while (sess.GotoNextSourceWithData());
}
}
return 0;
}
// Thread function for RTP session.
static void* rtpSessionThread(void *arg)
{
VideoStream* video = reinterpret_cast<VideoStream*>(arg);
u_int8_t bigBuffer[MAX_FRAME_SIZE];
unsigned int lastPacketTimestamp = 0;
unsigned int currentIndex = 0;
double last_time = 0;
RTPSession session = createRtpSession(video->getMulticastIP(),
video->getPort());
while (1) {
session.Poll();
// Distribute data from the session to connected clients if we've
// got anything
session.BeginDataAccess();
if (session.GotoFirstSourceWithData()) {
do {
RTPPacket *packet = NULL;
while ((packet = session.GetNextPacket()) != NULL) {
if ((packet->GetPayloadLength() > sizeof(bigBuffer)) ||
(packet->GetPayloadLength() == 0)) {
// Free the packet, we're not going to use it.
session.DeletePacket(packet);
continue; // Exit this level of the loop and drop it
}
// Check timestamps for new data. A new timestamp means
// this is from a different time.
if (packet->GetTimestamp() != lastPacketTimestamp) {
video->decode((uint8_t*)&bigBuffer[0], currentIndex);
currentIndex = 0;
memset(&bigBuffer[0], 0, sizeof(bigBuffer));
} // End new timestamp optimization.
// Copy data into buffer
if (currentIndex + packet->GetPayloadLength() > sizeof(bigBuffer)) {
throw std::runtime_error("Frame buffer overflow");
}
memcpy(&bigBuffer[currentIndex], packet->GetPayloadData(),
packet->GetPayloadLength());
currentIndex += packet->GetPayloadLength();
// Mark our last timestamp
lastPacketTimestamp = packet->GetTimestamp();
// Free the packet.
session.DeletePacket(packet);
}
} while (session.GotoNextSourceWithData());
}
session.EndDataAccess();
RTPTime delay(0, 100); // 100usec
// Update More Data
bool moreData;
session.WaitForIncomingData(delay, &moreData);
}
// Leave the session while sending BYE.
RTPTime timeout(0.75f); // Wait briefly.
const char* reason = "Session Destroyed.";
unsigned int reasonlen = strlen(reason);
if (session.IsActive())
session.BYEDestroy(timeout, reason, reasonlen);
}
/****************************************
* RecAudio
* Obtiene los packetes y los muestra
*****************************************/
int AudioStream::RecAudio()
{
BYTE lost=0;
int recBytes=0;
struct timeval before;
BYTE data[512];
WORD playBuffer[512];
DWORD size=512;
AudioCodec* codec=NULL;
AudioCodec::Type type;
DWORD timeStamp=0;
DWORD frameTime=0;
DWORD lastTime=0;
Log(">RecAudio\n");
//Inicializamos el tiempo
gettimeofday(&before,NULL);
//Empezamos a reproducir
audioOutput->StartPlaying();
//Mientras tengamos que capturar
while(receivingAudio)
{
//POnemos el tam�o
size=512;
lost=0;
//Obtenemos el paquete
if (!rtp.GetAudioPacket((BYTE *)data,&size,&lost,&type,&timeStamp))
{
//Log(">Process audio error\n");
switch (errno)
{
case 11:
case 4:
case 0:
break;
default:
Log("Error obteniendo el packete [%d]\n",errno);
}
continue;
}
// add for rtsp watcher by liuhong start
if (!muted) {
Rtps::iterator iter;
for(iter = rtps.begin(); iter != rtps.end(); iter++)
{
RTPSession *rtp = iter->second;
int ret = rtp->ForwardPacket((BYTE *)data, size, 0, timeStamp);
//Log(">Send audio to watcher %d %ld\n", ret, timeStamp);
}
}
//Comprobamos el tipo
if ((codec==NULL) || (type!=codec->type))
{
//Si habia uno nos lo cargamos
if (codec!=NULL)
delete codec;
//Creamos uno dependiendo del tipo
if ((codec = CreateAudioCodec(type))==NULL)
{
Log("Error creando nuevo codec de audio [%d]\n",type);
continue;
}
// add for rtsp watcher by liuhong
decodeCodec = type;
}
//Lo decodificamos
int len = codec->Decode(data,size,playBuffer,512);
//Obtenemos el tiempo del frame
frameTime = timeStamp - lastTime;
//Actualizamos el ultimo envio
lastTime = timeStamp;
//Check muted
if (!muted)
//Y lo reproducimos
audioOutput->PlayBuffer((WORD *)playBuffer,len,frameTime);
//Aumentamos el numero de bytes recividos
recBytes+=size;
}
//Terminamos de reproducir
audioOutput->StopPlaying();
//Borramos el codec
if (codec!=NULL)
delete codec;
//Salimos
Log("<RecAudio\n");
pthread_exit(0);
}