RTPHandler::RTPHandler(char* di, char* dp)
{
portbase = RCV_PORT;
destip = inet_addr(di);
destport = (uint16_t) atoi(dp);
alive = true;
if (destip == INADDR_NONE)
{
cerr << "Bad IP address specified" << endl;
alive = false;
return;
}
destip = ntohl(destip);
RTPUDPv4TransmissionParams transparams;
RTPSessionParams sessparams;
sessparams.SetOwnTimestampUnit(1.0/CLK_RATE);
sessparams.SetAcceptOwnPackets(true);
transparams.SetPortbase(portbase);
int e = sess.Create(sessparams,&transparams);
if(error(e)) return;
cout << sessparams.GetMaximumPacketSize()<< endl;
RTPIPv4Address addr(destip,destport);
e = sess.AddDestination(addr);
if(error(e)) return;
}
int MyRTPSession::MyRTP_SetUp(MediaSession * media_session)
{
if(!media_session) {
fprintf(stderr, "%s: Invalid media session\n", __func__);
return RTP_ERROR;
}
if(0 == media_session->TimeRate) {
fprintf(stderr, "%s: Invalid MediaSession::TimeRate\n", __func__);
return RTP_ERROR;
}
if(0 == media_session->RTPPort) {
fprintf(stderr, "%s: Invalid MediaSession::RTPPort\n", __func__);
return RTP_ERROR;
}
int status;
// Now, we'll create a RTP session, set the destination
// and poll for incoming data.
RTPUDPv4TransmissionParams transparams;
RTPSessionParams sessparams;
// IMPORTANT: The local timestamp unit MUST be set, otherwise
// RTCP Sender Report info will be calculated wrong
// In this case, we'll be just use 8000 samples per second.
sessparams.SetOwnTimestampUnit(1.0/media_session->TimeRate);
sessparams.SetAcceptOwnPackets(true);
transparams.SetPortbase(media_session->RTPPort);
status = Create(sessparams,&transparams);
return IsError(status);
}
int CRTPRecv::InitRTP(int Port)
{
RTPUDPv4TransmissionParams TransParams;
RTPSessionParams SessionParams;
//设置本地通讯端口.
TransParams.SetPortbase(Port);
//设置时间戳,每秒钟发送的包数量.
SessionParams.SetOwnTimestampUnit(1.0/90000.0);
SessionParams.SetAcceptOwnPackets(true); //接收自己发送的数据包
//创建RTP对象.
int Result = Create(SessionParams, &TransParams);
if (Result < 0)
{
LOG_ERROR("RTP: "<<RTPGetErrorString(Result));
return -1;
}
//设置默认负载类型.
this->SetMaximumPacketSize(65535);
//this->SetDefaultMark(true);
//this->SetDefaultTimestampIncrement(TIMESTAMP);
//this->SetDefaultPayloadType(H264);
return 0;
}
// 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;
}
// Maybe this could be used here. Have a try .
void Sender::setRTPParams(){
int status;
RTPUDPv4TransmissionParams transparams;
RTPSessionParams sessparams;
sessparams.SetOwnTimestampUnit(1.0/25.0);
sessparams.SetAcceptOwnPackets(true);
sessparams.SetUsePredefinedSSRC(true);
// sessparams.SetPredefinedSSRC(SSRC);
transparams.SetPortbase(baseport);
status = this->Create(sessparams, &transparams);
checkerror(status);
RTPIPv4Address addr(dst_ip, destport);
status = this->AddDestination(addr);
checkerror(status);
}
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;
}
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;
}
bool MediaStream::start(QHostAddress ip, int port, int localPort, int codecPayload )
{
if ( isRunning() )
stop();
printf("%s %d | %d | %d\n",ip.toString().toAscii().data(), port, localPort, codecPayload);
if(d->processThread == NULL)
d->processThread = new MediaThread(this);
d->outBufferPos = 0;
d->outBufferTime = 0;
// int localPort = 3000;
printf("getFactory(%d)\n", codecPayload);
VoiceCodecFactory *factory = CodecsManager::instance()->codecFactory(codecPayload);
if ( !factory ) {
printf("VoiceCodecFactory not found!\n");
return true;
}
d->codecPayload = codecPayload;
d->decoder = factory->decoder();
d->encoder = factory->encoder();
printf("transparams\n");
// Now, we'll create a RTP session and set the destination
// d->transparams.mediaChannel = mediaChannel;
// d->transparams.incomingPackets = incomingPackets;
d->transparams.SetPortbase(localPort);
RTPSessionParams sessparams;
sessparams.SetReceiveMode(RTPTransmitter::AcceptAll);
sessparams.SetOwnTimestampUnit(1.0/8000.0); // 8KHz
sessparams.SetAcceptOwnPackets(true);
printf("session.Create()\n");
int status = d->session.Create( sessparams, &d->transparams, RTPTransmitter::SynapseProto );
if ( status<0 ) {
qDebug("can't create RTP session, %s", RTPGetErrorString(status).c_str() );
d->session.Destroy();
return false;
}
printf("session.AddDestination()\n");
RTPIPv4Address addr(ip.toIPv4Address(),port);
status = d->session.AddDestination(addr);
if ( status<0 ) {
qDebug("can't add rtp destination, %s", RTPGetErrorString(status).c_str() );
d->session.Destroy();
return false;
}
d->session.SetDefaultPayloadType(codecPayload);
d->session.SetDefaultMark(false);
d->session.SetDefaultTimestampIncrement(160);
//initialise audio
status = Pa_Initialize();
//////////////////// FOR TESTING
if( status != paNoError ) {
qDebug( "PortAudio error: %s", Pa_GetErrorText(status) );
// stop();
// return true;
}
if(status == paNoError) {
status = Pa_OpenDefaultStream(
&d->audioStream,/* passes back stream pointer */
1, /* 1 input channel */
1, /* mono output */
paInt16, /* 16 bit fixed point output */
8000, /* sample rate */
240, /* frames per buffer */
16, /* number of buffers, if zero then use default minimum */
audioCallback, /* specify our custom callback */
d ); /* pass our data through to callback */
status = Pa_StartStream( d->audioStream );
if( status != paNoError ) {
qDebug( "PortAudio error: %s", Pa_GetErrorText(status) );
// stop();
// return true;
}
}
// put something to dsp buffer
/*
char emptyData[160*8];
memset( emptyData, 1, sizeof(emptyData) );
d->dspBuffer->lock();
d->dspBuffer->put( emptyData, sizeof(emptyData) );
d->dspBuffer->unlock();
*/
//d->timer.start(1,false);
d->isRunning = true;
d->processThread->start();
// qDebug("mediastream started");
printf("mediastream started\n");
return true;
} // }}}
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;
}