bool ElcRTSPServer::setSource(const string& url, const string& iname)
{
LOG_INFO("ElcRTSPServer::setSource -url="<<url<<" -name="<<iname);
mUrl = url;
mName = iname;
Boolean reuseFirstSource = true;
if(!rtspServer)
{
LOG_ERROR("ElcRTSPServer::setSource failed! Rtspserver does not created propertly");
return false;
}
mediaSession = ServerMediaSession::createNew(*mEnv, mName.c_str(), mName.c_str(), "ELC");
if(!mediaSession)
return false;
// if multicast streaming
if(Config::isMulticast)
{
struct in_addr destinationAddress;
if(Config::streamingDestinationIp == "default" || Config::streamingDestinationIp =="")
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*mEnv);
else
destinationAddress.s_addr = our_inet_addr(Config::streamingDestinationIp.c_str());
Port rtpPort(Config::rtpPortNum);
Port rtcpPort(Config::rtcpPortNum);
rtpGroupsock = new Groupsock(*mEnv, destinationAddress, rtpPort, 255/*Config::ttl*/);
rtpGroupsock->multicastSendOnly(); // we're a SSM source
rtcpGroupsock = new Groupsock(*mEnv, destinationAddress, rtcpPort, 255/*Config::ttl*/);
rtcpGroupsock->multicastSendOnly(); // we're a SSM source
videoSink = MPEG4ESVideoRTPSink::createNew(*mEnv, rtpGroupsock, 96);
const unsigned estimatedSessionBandwidth = 500; // in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen+1];
gethostname((char*)CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
rtcp = RTCPInstance::createNew(*mEnv, rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
videoSink, NULL /* we're a server*/ ,
True /* we're a SSM source */);
mediaSession->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, rtcp));
rtspServer->addServerMediaSession(mediaSession);
LOG_INFO("ElcRTSPServer::setSource OK");
}
else // unicast streaming
{
mediaSubSession = Mpeg4LiveServerMediaSubSession::createNew(*mEnv, reuseFirstSource, mUrl);
mediaSession->addSubsession(mediaSubSession);
rtspServer->addServerMediaSession(mediaSession);
LOG_INFO("ElcRTSPServer::setSource OK");
}
return true;
}
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
// Define our destination (multicast) IP address:
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
// Note: This is a multicast address. If you wish instead to stream
// using unicast, then you should use the "testOnDemandRTSPServer"
// test program - not this test program - as a model.
// Create our RTSP server. (Receivers will need to use RTSP to access the stream.)
rtspServer = RTSPServer::createNew(*env, 8554);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
sms = ServerMediaSession::createNew(*env, "testStream", inputFileName,
"Session streamed by \"testMKVStreamer\"",
True /*SSM*/);
// Arrange to create a "MatroskaFile" object for the specified file.
// (Note that this object is not created immediately, but instead via a callback.)
MatroskaFile::createNew(*env, inputFileName, onMatroskaFileCreation, NULL, "jpn");
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
void CCreateRTPServer::createGroupsock()
{
m_prtcpGroupsockAudio = NULL;
m_prtpGroupsockAudio = NULL;
m_prtcpGroupsockVideo = NULL;
m_prtpGroupsockVideo = NULL;
struct in_addr destAddress;
//destAddress.s_addr = our_inet_addr(m_destAddressStr.c_str());
if(!m_rtpDestAddressStr.empty())
destAddress.s_addr = our_inet_addr(m_rtpDestAddressStr.c_str());
else
destAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
if(m_pAudioSourceQueue != NULL)
{
const unsigned short rtpPortNumAudio = m_rtpPortNum;
const unsigned short rtcpPortNumAudio = rtpPortNumAudio + 1;
const Port rtpPortAudio(rtpPortNumAudio);
const Port rtcpPortAudio(rtcpPortNumAudio);
m_prtpGroupsockAudio = new Groupsock(*env, destAddress, rtpPortAudio, m_ttl);
m_prtcpGroupsockAudio = new Groupsock(*env, destAddress, rtcpPortAudio, m_ttl);
if(m_prtpGroupsockAudio && m_prtcpGroupsockAudio)
{
m_rtpPortNum += 2;
}
}
if(m_pVideoSourceQueue != NULL)
{
const unsigned short rtpPortNumVideo = m_rtpPortNum;
const unsigned short rtcpPortNumVideo = rtpPortNumVideo + 1;
const Port rtpPortVideo(rtpPortNumVideo);
const Port rtcpPortVideo(rtcpPortNumVideo);
m_prtpGroupsockVideo = new Groupsock(*env, destAddress, rtpPortVideo, m_ttl);
m_prtcpGroupsockVideo = new Groupsock(*env, destAddress, rtcpPortVideo, m_ttl);
}
}
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
// Create 'groupsocks' for RTP and RTCP:
struct in_addr destinationAddress;
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
// Note: This is a multicast address. If you wish instead to stream
// using unicast, then you should use the "testOnDemandRTSPServer"
// test program - not this test program - as a model.
const unsigned short rtpPortNum = 18888;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 255;
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
rtpGroupsock.multicastSendOnly(); // we're a SSM source
Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
rtcpGroupsock.multicastSendOnly(); // we're a SSM source
// Create a 'H264 Video RTP' sink from the RTP 'groupsock':
OutPacketBuffer::maxSize = 100000;
videoSink = H264VideoRTPSink::createNew(*env, &rtpGroupsock, 96);
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 5000; // in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen+1];
gethostname((char*)CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
RTCPInstance* rtcp
= RTCPInstance::createNew(*env, &rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
videoSink, NULL /* we're a server */,
True /* we're a SSM source */);
// Note: This starts RTCP running automatically
RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, "testStream", inputFileName,
"Session streamed by \"testH264VideoStreamer\"",
True /*SSM*/);
sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, rtcp));
rtspServer->addServerMediaSession(sms);
char* url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
// Start the streaming:
*env << "Beginning streaming...\n";
play();
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
int main(int argc, char const** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
// Parse command-line options:
// (Unfortunately we can't use getopt() here; Windoze doesn't have it)
programName = argv[0];
while (argc > 2) {
char const* const opt = argv[1];
if (opt[0] != '-') break;
switch (opt[1]) {
case 'i': { // transmit video I-frames only
iFramesOnly = True;
break;
}
case 'a': { // transmit audio, but not video
mediaToStream &=~ VOB_VIDEO;
break;
}
case 'v': { // transmit video, but not audio
mediaToStream &=~ VOB_AUDIO;
break;
}
case 'p': { // specify port number for built-in RTSP server
int portArg;
if (sscanf(argv[2], "%d", &portArg) != 1) {
usage();
}
if (portArg <= 0 || portArg >= 65536) {
*env << "bad port number: " << portArg
<< " (must be in the range (0,65536))\n";
usage();
}
rtspServerPortNum = (unsigned short)portArg;
++argv; --argc;
break;
}
default: {
usage();
break;
}
}
++argv; --argc;
}
if (argc < 2) usage();
if (mediaToStream == 0) {
*env << "The -a and -v flags cannot both be used!\n";
usage();
}
if (iFramesOnly && (mediaToStream&VOB_VIDEO) == 0) {
*env << "Warning: Because we're not streaming video, the -i flag has no effect.\n";
}
inputFileNames = &argv[1];
curInputFileName = inputFileNames;
// Create 'groupsocks' for RTP and RTCP:
struct in_addr destinationAddress;
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
const unsigned short rtpPortNumAudio = 4444;
const unsigned short rtcpPortNumAudio = rtpPortNumAudio+1;
const unsigned short rtpPortNumVideo = 8888;
const unsigned short rtcpPortNumVideo = rtpPortNumVideo+1;
const unsigned char ttl = 255;
const Port rtpPortAudio(rtpPortNumAudio);
const Port rtcpPortAudio(rtcpPortNumAudio);
const Port rtpPortVideo(rtpPortNumVideo);
const Port rtcpPortVideo(rtcpPortNumVideo);
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen+1];
gethostname((char*)CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
if (mediaToStream&VOB_AUDIO) {
rtpGroupsockAudio
= new Groupsock(*env, destinationAddress, rtpPortAudio, ttl);
rtpGroupsockAudio->multicastSendOnly(); // because we're a SSM source
// Create an 'AC3 Audio RTP' sink from the RTP 'groupsock':
audioSink
= AC3AudioRTPSink::createNew(*env, rtpGroupsockAudio, 96, 0);
// set the RTP timestamp frequency 'for real' later
// Create (and start) a 'RTCP instance' for this RTP sink:
rtcpGroupsockAudio
= new Groupsock(*env, destinationAddress, rtcpPortAudio, ttl);
rtcpGroupsockAudio->multicastSendOnly(); // because we're a SSM source
const unsigned estimatedSessionBandwidthAudio
= 160; // in kbps; for RTCP b/w share
audioRTCP = RTCPInstance::createNew(*env, rtcpGroupsockAudio,
estimatedSessionBandwidthAudio, CNAME,
audioSink, NULL /* we're a server */,
True /* we're a SSM source */);
// Note: This starts RTCP running automatically
}
if (mediaToStream&VOB_VIDEO) {
rtpGroupsockVideo
= new Groupsock(*env, destinationAddress, rtpPortVideo, ttl);
rtpGroupsockVideo->multicastSendOnly(); // because we're a SSM source
// Create a 'MPEG Video RTP' sink from the RTP 'groupsock':
videoSink = MPEG1or2VideoRTPSink::createNew(*env, rtpGroupsockVideo);
// Create (and start) a 'RTCP instance' for this RTP sink:
rtcpGroupsockVideo
= new Groupsock(*env, destinationAddress, rtcpPortVideo, ttl);
rtcpGroupsockVideo->multicastSendOnly(); // because we're a SSM source
const unsigned estimatedSessionBandwidthVideo
= 4500; // in kbps; for RTCP b/w share
videoRTCP = RTCPInstance::createNew(*env, rtcpGroupsockVideo,
estimatedSessionBandwidthVideo, CNAME,
videoSink, NULL /* we're a server */,
True /* we're a SSM source */);
// Note: This starts RTCP running automatically
}
if (rtspServer == NULL) {
rtspServer = RTSPServer::createNew(*env, rtspServerPortNum);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
*env << "To change the RTSP server's port number, use the \"-p <port number>\" option.\n";
exit(1);
}
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, "vobStream", *curInputFileName,
"Session streamed by \"vobStreamer\"", True /*SSM*/);
if (audioSink != NULL) sms->addSubsession(PassiveServerMediaSubsession::createNew(*audioSink, audioRTCP));
if (videoSink != NULL) sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, videoRTCP));
rtspServer->addServerMediaSession(sms);
*env << "Created RTSP server.\n";
// Display our "rtsp://" URL, for clients to connect to:
char* url = rtspServer->rtspURL(sms);
*env << "Access this stream using the URL:\n\t" << url << "\n";
delete[] url;
}
// Finally, start the streaming:
*env << "Beginning streaming...\n";
play();
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
// -----------------------------------------
// entry point
// -----------------------------------------
int main(int argc, char** argv)
{
// default parameters
const char *dev_name = "/dev/video0";
int format = V4L2_PIX_FMT_H264;
int width = 640;
int height = 480;
int queueSize = 10;
int fps = 25;
unsigned short rtpPortNum = 20000;
unsigned short rtcpPortNum = rtpPortNum+1;
unsigned char ttl = 5;
struct in_addr destinationAddress;
unsigned short rtspPort = 8554;
unsigned short rtspOverHTTPPort = 0;
bool multicast = false;
int verbose = 0;
std::string outputFile;
bool useMmap = false;
// decode parameters
int c = 0;
while ((c = getopt (argc, argv, "hW:H:Q:P:F:v::O:T:mM")) != -1)
{
switch (c)
{
case 'O': outputFile = optarg; break;
case 'v': verbose = 1; if (optarg && *optarg=='v') verbose++; break;
case 'm': multicast = true; break;
case 'W': width = atoi(optarg); break;
case 'H': height = atoi(optarg); break;
case 'Q': queueSize = atoi(optarg); break;
case 'P': rtspPort = atoi(optarg); break;
case 'T': rtspOverHTTPPort = atoi(optarg); break;
case 'F': fps = atoi(optarg); break;
case 'M': useMmap = true; break;
case 'h':
{
std::cout << argv[0] << " [-v[v]][-m] [-P RTSP port][-P RTSP/HTTP port][-Q queueSize] [-M] [-W width] [-H height] [-F fps] [-O file] [device]" << std::endl;
std::cout << "\t -v : verbose " << std::endl;
std::cout << "\t -v v : very verbose " << std::endl;
std::cout << "\t -Q length: Number of frame queue (default "<< queueSize << ")" << std::endl;
std::cout << "\t -O file : Dump capture to a file" << std::endl;
std::cout << "\t RTSP options :" << std::endl;
std::cout << "\t -m : Enable multicast output" << std::endl;
std::cout << "\t -P port : RTSP port (default "<< rtspPort << ")" << std::endl;
std::cout << "\t -H port : RTSP over HTTP port (default "<< rtspOverHTTPPort << ")" << std::endl;
std::cout << "\t V4L2 options :" << std::endl;
std::cout << "\t -M : V4L2 capture using memory mapped buffers (default use read interface)" << std::endl;
std::cout << "\t -F fps : V4L2 capture framerate (default "<< fps << ")" << std::endl;
std::cout << "\t -W width : V4L2 capture width (default "<< width << ")" << std::endl;
std::cout << "\t -H height: V4L2 capture height (default "<< height << ")" << std::endl;
std::cout << "\t device : V4L2 capture device (default "<< dev_name << ")" << std::endl;
exit(0);
}
}
}
if (optind<argc)
{
dev_name = argv[optind];
}
// create live555 environment
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
// create RTSP server
RTSPServer* rtspServer = RTSPServer::createNew(*env, rtspPort);
if (rtspServer == NULL)
{
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
}
else
{
// set http tunneling
if (rtspOverHTTPPort)
{
rtspServer->setUpTunnelingOverHTTP(rtspOverHTTPPort);
}
// Init capture
*env << "Create V4L2 Source..." << dev_name << "\n";
V4L2DeviceParameters param(dev_name,format,width,height,fps,verbose);
V4L2Device* videoCapture = NULL;
if (useMmap)
{
videoCapture = V4L2MMAPDeviceSource::createNew(param);
}
else
{
videoCapture = V4L2READDeviceSource::createNew(param);
}
V4L2DeviceSource* videoES = V4L2DeviceSource::createNew(*env, param, videoCapture, outputFile, queueSize, verbose);
if (videoES == NULL)
{
*env << "Unable to create source for device " << dev_name << "\n";
}
else
{
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
OutPacketBuffer::maxSize = videoCapture->getBufferSize();
StreamReplicator* replicator = StreamReplicator::createNew(*env, videoES, false);
// Create Server Multicast Session
if (multicast)
{
addSession(rtspServer, "multicast", MulticastServerMediaSubsession::createNew(*env,destinationAddress, Port(rtpPortNum), Port(rtcpPortNum), ttl, 96, replicator,format));
}
// Create Server Unicast Session
addSession(rtspServer, "unicast", UnicastServerMediaSubsession::createNew(*env,replicator,format));
// main loop
signal(SIGINT,sighandler);
env->taskScheduler().doEventLoop(&quit);
*env << "Exiting..\n";
}
Medium::close(videoES);
delete videoCapture;
Medium::close(rtspServer);
}
env->reclaim();
delete scheduler;
return 0;
}
// -----------------------------------------
// entry point
// -----------------------------------------
int main(int argc, char** argv)
{
// default parameters
const char *dev_name = "/dev/video0";
int format = V4L2_PIX_FMT_H264;
int width = 640;
int height = 480;
int queueSize = 10;
int fps = 25;
unsigned short rtspPort = 8554;
unsigned short rtspOverHTTPPort = 0;
bool multicast = false;
int verbose = 0;
std::string outputFile;
bool useMmap = true;
std::string url = "unicast";
std::string murl = "multicast";
bool useThread = true;
std::string maddr;
bool repeatConfig = true;
int timeout = 65;
// decode parameters
int c = 0;
while ((c = getopt (argc, argv, "v::Q:O:" "I:P:T:m:u:M:ct:" "rsfF:W:H:" "h")) != -1)
{
switch (c)
{
case 'v': verbose = 1; if (optarg && *optarg=='v') verbose++; break;
case 'Q': queueSize = atoi(optarg); break;
case 'O': outputFile = optarg; break;
// RTSP/RTP
case 'I': ReceivingInterfaceAddr = inet_addr(optarg); break;
case 'P': rtspPort = atoi(optarg); break;
case 'T': rtspOverHTTPPort = atoi(optarg); break;
case 'u': url = optarg; break;
case 'm': multicast = true; murl = optarg; break;
case 'M': multicast = true; maddr = optarg; break;
case 'c': repeatConfig = false; break;
case 't': timeout = atoi(optarg); break;
// V4L2
case 'r': useMmap = false; break;
case 's': useThread = false; break;
case 'f': format = 0; break;
case 'F': fps = atoi(optarg); break;
case 'W': width = atoi(optarg); break;
case 'H': height = atoi(optarg); break;
case 'h':
default:
{
std::cout << argv[0] << " [-v[v]] [-Q queueSize] [-O file]" << std::endl;
std::cout << "\t [-I interface] [-P RTSP port] [-T RTSP/HTTP port] [-m multicast url] [-u unicast url] [-M multicast addr] [-c] [-t timeout]" << std::endl;
std::cout << "\t [-r] [-s] [-W width] [-H height] [-F fps] [device] [device]" << std::endl;
std::cout << "\t -v : verbose" << std::endl;
std::cout << "\t -vv : very verbose" << std::endl;
std::cout << "\t -Q length: Number of frame queue (default "<< queueSize << ")" << std::endl;
std::cout << "\t -O output: Copy captured frame to a file or a V4L2 device" << std::endl;
std::cout << "\t RTSP options :" << std::endl;
std::cout << "\t -I addr : RTSP interface (default autodetect)" << std::endl;
std::cout << "\t -P port : RTSP port (default "<< rtspPort << ")" << std::endl;
std::cout << "\t -T port : RTSP over HTTP port (default "<< rtspOverHTTPPort << ")" << std::endl;
std::cout << "\t -u url : unicast url (default " << url << ")" << std::endl;
std::cout << "\t -m url : multicast url (default " << murl << ")" << std::endl;
std::cout << "\t -M addr : multicast group:port (default is random_address:20000)" << std::endl;
std::cout << "\t -c : don't repeat config (default repeat config before IDR frame)" << std::endl;
std::cout << "\t -t secs : RTCP expiration timeout (default " << timeout << ")" << std::endl;
std::cout << "\t V4L2 options :" << std::endl;
std::cout << "\t -r : V4L2 capture using read interface (default use memory mapped buffers)" << std::endl;
std::cout << "\t -s : V4L2 capture using live555 mainloop (default use a reader thread)" << std::endl;
std::cout << "\t -f : V4L2 capture using current format (-W,-H,-F are ignore)" << std::endl;
std::cout << "\t -W width : V4L2 capture width (default "<< width << ")" << std::endl;
std::cout << "\t -H height: V4L2 capture height (default "<< height << ")" << std::endl;
std::cout << "\t -F fps : V4L2 capture framerate (default "<< fps << ")" << std::endl;
std::cout << "\t device : V4L2 capture device (default "<< dev_name << ")" << std::endl;
exit(0);
}
}
}
std::list<std::string> devList;
while (optind<argc)
{
devList.push_back(argv[optind]);
optind++;
}
if (devList.empty())
{
devList.push_back(dev_name);
}
// init logger
initLogger(verbose);
// create live555 environment
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
// split multicast info
std::istringstream is(maddr);
std::string ip;
getline(is, ip, ':');
struct in_addr destinationAddress;
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
if (!ip.empty())
{
destinationAddress.s_addr = inet_addr(ip.c_str());
}
std::string port;
getline(is, port, ':');
unsigned short rtpPortNum = 20000;
if (!port.empty())
{
rtpPortNum = atoi(port.c_str());
}
unsigned short rtcpPortNum = rtpPortNum+1;
unsigned char ttl = 5;
// create RTSP server
RTSPServer* rtspServer = createRTSPServer(*env, rtspPort, rtspOverHTTPPort, timeout);
if (rtspServer == NULL)
{
LOG(ERROR) << "Failed to create RTSP server: " << env->getResultMsg();
}
else
{
int nbSource = 0;
std::list<std::string>::iterator devIt;
for ( devIt=devList.begin() ; devIt!=devList.end() ; ++devIt)
{
std::string deviceName(*devIt);
// Init capture
LOG(NOTICE) << "Create V4L2 Source..." << deviceName;
V4L2DeviceParameters param(deviceName.c_str(),format,width,height,fps, verbose);
V4l2Capture* videoCapture = V4l2DeviceFactory::CreateVideoCapure(param, useMmap);
if (videoCapture)
{
nbSource++;
format = videoCapture->getFormat();
int outfd = -1;
V4l2Output* out = NULL;
if (!outputFile.empty())
{
V4L2DeviceParameters outparam(outputFile.c_str(), videoCapture->getFormat(), videoCapture->getWidth(), videoCapture->getHeight(), 0,verbose);
V4l2Output* out = V4l2DeviceFactory::CreateVideoOutput(outparam, useMmap);
if (out != NULL)
{
outfd = out->getFd();
}
}
LOG(NOTICE) << "Start V4L2 Capture..." << deviceName;
if (!videoCapture->captureStart())
{
LOG(NOTICE) << "Cannot start V4L2 Capture for:" << deviceName;
}
V4L2DeviceSource* videoES = NULL;
if (format == V4L2_PIX_FMT_H264)
{
videoES = H264_V4L2DeviceSource::createNew(*env, param, videoCapture, outfd, queueSize, useThread, repeatConfig);
}
else
{
videoES = V4L2DeviceSource::createNew(*env, param, videoCapture, outfd, queueSize, useThread);
}
if (videoES == NULL)
{
LOG(FATAL) << "Unable to create source for device " << deviceName;
delete videoCapture;
}
else
{
// extend buffer size if needed
if (videoCapture->getBufferSize() > OutPacketBuffer::maxSize)
{
OutPacketBuffer::maxSize = videoCapture->getBufferSize();
}
StreamReplicator* replicator = StreamReplicator::createNew(*env, videoES, false);
std::string baseUrl;
if (devList.size() > 1)
{
baseUrl = basename(deviceName.c_str());
baseUrl.append("/");
}
// Create Multicast Session
if (multicast)
{
LOG(NOTICE) << "RTP address " << inet_ntoa(destinationAddress) << ":" << rtpPortNum;
LOG(NOTICE) << "RTCP address " << inet_ntoa(destinationAddress) << ":" << rtcpPortNum;
addSession(rtspServer, baseUrl+murl, MulticastServerMediaSubsession::createNew(*env,destinationAddress, Port(rtpPortNum), Port(rtcpPortNum), ttl, replicator,format));
// increment ports for next sessions
rtpPortNum+=2;
rtcpPortNum+=2;
}
// Create Unicast Session
addSession(rtspServer, baseUrl+url, UnicastServerMediaSubsession::createNew(*env,replicator,format));
}
if (out)
{
delete out;
}
}
}
if (nbSource>0)
{
// main loop
signal(SIGINT,sighandler);
env->taskScheduler().doEventLoop(&quit);
LOG(NOTICE) << "Exiting....";
}
Medium::close(rtspServer);
}
env->reclaim();
delete scheduler;
return 0;
}
void play() {
// Open the file as a 'WAV' file:
WAVAudioFileSource* pcmSource
= WAVAudioFileSource::createNew(*env, inputFileName);
if (pcmSource == NULL) {
*env << "Unable to open file \"" << inputFileName
<< "\" as a WAV audio file source: "
<< env->getResultMsg() << "\n";
exit(1);
}
// Get attributes of the audio source:
unsigned char const bitsPerSample = pcmSource->bitsPerSample();
if (bitsPerSample != 8 && bitsPerSample != 16) {
*env << "The input file contains " << bitsPerSample
<< " bit-per-sample audio, which we don't handle\n";
exit(1);
}
sessionState.source = pcmSource;
unsigned const samplingFrequency = pcmSource->samplingFrequency();
unsigned char const numChannels = pcmSource->numChannels();
unsigned bitsPerSecond
= samplingFrequency*bitsPerSample*numChannels;
*env << "Audio source parameters:\n\t" << samplingFrequency << " Hz, ";
*env << bitsPerSample << " bits-per-sample, ";
*env << numChannels << " channels => ";
*env << bitsPerSecond << " bits-per-second\n";
// Add in any filter necessary to transform the data prior to streaming.
// (This is where any audio compression would get added.)
char const* mimeType;
unsigned char payloadFormatCode;
if (bitsPerSample == 16) {
#ifdef CONVERT_TO_ULAW
// Add a filter that converts from raw 16-bit PCM audio (in little-endian order)
// to 8-bit u-law audio:
sessionState.source
= uLawFromPCMAudioSource::createNew(*env, pcmSource, 1/*little-endian*/);
if (sessionState.source == NULL) {
*env << "Unable to create a u-law filter from the PCM audio source: "
<< env->getResultMsg() << "\n";
exit(1);
}
bitsPerSecond /= 2;
mimeType = "PCMU";
if (samplingFrequency == 8000 && numChannels == 1) {
payloadFormatCode = 0; // a static RTP payload type
} else {
payloadFormatCode = 96; // a dynamic RTP payload type
}
*env << "Converting to 8-bit u-law audio for streaming => "
<< bitsPerSecond << " bits-per-second\n";
#else
// The 16-bit samples in WAV files are in little-endian order.
// Add a filter that converts them to network (i.e., big-endian) order:
sessionState.source = EndianSwap16::createNew(*env, pcmSource);
if (sessionState.source == NULL) {
*env << "Unable to create a little->bit-endian order filter from the PCM audio source: "
<< env->getResultMsg() << "\n";
exit(1);
}
mimeType = "L16";
if (samplingFrequency == 44100 && numChannels == 2) {
payloadFormatCode = 10; // a static RTP payload type
} else if (samplingFrequency == 44100 && numChannels == 1) {
payloadFormatCode = 11; // a static RTP payload type
} else {
payloadFormatCode = 96; // a dynamic RTP payload type
}
*env << "Converting to network byte order for streaming\n";
#endif
} else { // bitsPerSample == 8
// Don't do any transformation; send the 8-bit PCM data 'as is':
mimeType = "L8";
payloadFormatCode = 96; // a dynamic RTP payload type
}
// Create 'groupsocks' for RTP and RTCP:
struct in_addr destinationAddress;
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
// Note: This is a multicast address. If you wish instead to stream
// using unicast, then you should use the "testOnDemandRTSPServer"
// test program - not this test program - as a model.
const unsigned short rtpPortNum = 2222;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 255;
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
sessionState.rtpGroupsock
= new Groupsock(*env, destinationAddress, rtpPort, ttl);
sessionState.rtpGroupsock->multicastSendOnly(); // we're a SSM source
sessionState.rtcpGroupsock
= new Groupsock(*env, destinationAddress, rtcpPort, ttl);
sessionState.rtcpGroupsock->multicastSendOnly(); // we're a SSM source
// Create an appropriate audio RTP sink (using "SimpleRTPSink")
// from the RTP 'groupsock':
sessionState.sink
= SimpleRTPSink::createNew(*env, sessionState.rtpGroupsock,
payloadFormatCode, samplingFrequency,
"audio", mimeType, numChannels);
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = bitsPerSecond/1000;
// in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen+1];
gethostname((char*)CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
sessionState.rtcpInstance
= RTCPInstance::createNew(*env, sessionState.rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
sessionState.sink, NULL /* we're a server */,
True /* we're a SSM source*/);
// Note: This starts RTCP running automatically
// Create and start a RTSP server to serve this stream:
sessionState.rtspServer = RTSPServer::createNew(*env, 8554);
if (sessionState.rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, "testStream", inputFileName,
"Session streamed by \"testWAVAudiotreamer\"", True/*SSM*/);
sms->addSubsession(PassiveServerMediaSubsession::createNew(*sessionState.sink, sessionState.rtcpInstance));
sessionState.rtspServer->addServerMediaSession(sms);
char* url = sessionState.rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
// Finally, start the streaming:
*env << "Beginning streaming...\n";
sessionState.sink->startPlaying(*sessionState.source, afterPlaying, NULL);
}
// -----------------------------------------
// entry point
// -----------------------------------------
int main(int argc, char** argv)
{
#if AUDIO_STREAM
StreamReplicator* audio_replicator = NULL;
#endif
// default parameters
const char *dev_name = "/dev/video1";
int format = V4L2_PIX_FMT_H264;
int width = 1280;
int height = 720;
int queueSize = 10;
int fps = 30;
int isp_fps = 30;
int bitrate = 1024; //(Kbps)
int mjpeg_qp = 120;
int m2m_en = 1;
int gop = fps;
unsigned short rtpPortNum = 20000;
unsigned short rtcpPortNum = rtpPortNum+1;
unsigned char ttl = 5;
struct in_addr destinationAddress;
unsigned short rtspPort = 554;
unsigned short rtspOverHTTPPort = 0;
bool multicast = false;
int verbose = 0;
std::string outputFile;
//bool useMmap = true;
std::string url = "unicast";
std::string murl = "multicast";
bool useThread = true;
in_addr_t maddr = INADDR_NONE;
bool audio_en = false;
// decode parameters
int c = 0;
//while ((c = getopt (argc, argv, "hW:H:Q:P:F:v::O:T:m:u:rsM:")) != -1)
#if AUDIO_STREAM
while ((c = getopt (argc, argv, "hb:W:H:g:Q:P:F:i:O:T:m:u:M:aj:")) != -1)
#else
while ((c = getopt (argc, argv, "hb:W:H:g:Q:P:F:i:O:T:m:u:M:j:")) != -1)
#endif
{
switch (c)
{
case 'O': outputFile = optarg; break;
//case 'v': verbose = 1; if (optarg && *optarg=='v') verbose++; break;
case 'm': multicast = true; if (optarg) murl = optarg; break;
case 'M': multicast = true; if (optarg) maddr = inet_addr(optarg); break;
case 'g': gop = atoi(optarg); break;
case 'b': bitrate = atoi(optarg); break;
case 'W': width = atoi(optarg); break;
case 'H': height = atoi(optarg); break;
case 'Q': queueSize = atoi(optarg); break;
case 'P': rtspPort = atoi(optarg); break;
case 'T': rtspOverHTTPPort = atoi(optarg); break;
case 'F': fps = atoi(optarg); break;
case 'i': isp_fps = atoi(optarg); break;
//case 'r': useMmap = false; break;
//case 's': useThread = false; break;
case 'u': url = optarg; break;
#if AUDIO_STREAM
case 'a': audio_en = true; break;
#endif
case 'j': format = V4L2_PIX_FMT_MJPEG; mjpeg_qp = atoi(optarg);break;
case 'h':
default:
{
std::cout << argv[0] << "Version:" << SNX_RTSP_SERVER_VERSION << std::endl;
std::cout << "Usage :" << std::endl;
std::cout << "\t " << argv[0] << " [-a] [-j mjpeg_qp] [-m] [-P RTSP port][-T RTSP/HTTP port][-Q queueSize] [-M groupaddress] [-b bitrate] [-W width] [-H height] [-F fps] [-i isp_fps] [device]" << std::endl;
std::cout << "\t -Q length: Number of frame queue (default "<< queueSize << ")" << std::endl;
std::cout << "\t RTSP options :" << std::endl;
std::cout << "\t -u url : unicast url (default " << url << ")" << std::endl;
std::cout << "\t -m url : multicast url (default " << murl << ")" << std::endl;
std::cout << "\t -M addr : multicast group (default is a random address)" << std::endl;
std::cout << "\t -P port : RTSP port (default "<< rtspPort << ")" << std::endl;
std::cout << "\t -T port : RTSP over HTTP port (default "<< rtspOverHTTPPort << ")" << std::endl;
std::cout << "\t V4L2 options :" << std::endl;
//std::cout << "\t -r : V4L2 capture using read interface (default use memory mapped buffers)" << std::endl;
//std::cout << "\t -s : V4L2 capture using live555 mainloop (default use a separated reading thread)" << std::endl;
std::cout << "\t -F fps : V4L2 capture framerate (default "<< fps << ")" << std::endl;
std::cout << "\t -i isp_fps : ISP capture framerate (default "<< isp_fps << ")" << std::endl;
std::cout << "\t -W width : V4L2 capture width (default "<< width << ")" << std::endl;
std::cout << "\t -H height : V4L2 capture height (default "<< height << ")" << std::endl;
std::cout << "\t V4L2 H264 options :" << std::endl;
std::cout << "\t -b bitrate : V4L2 capture bitrate kbps(default "<< bitrate << " kbps)" << std::endl;
std::cout << "\t -g gop : V4L2 capture gop (default "<< gop << " )" << std::endl;
std::cout << "\t device : V4L2 capture device (default "<< dev_name << ")" << std::endl;
std::cout << "\t V4L2 MJPEG options :" << std::endl;
std::cout << "\t -j mjpeg_qp : MJPEG streaming and qp (default is 60)" << std::endl;
#if AUDIO_STREAM
std::cout << "\t -a : enable A-law pcm streaming " << std::endl;
std::cout << "\t H264 example : "<< argv[0] << " -a -Q 5 -u media/stream1 -P 554" << std::endl;
#else
std::cout << "\t H264 example : "<< argv[0] << " -Q 5 -u media/stream1 -P 554" << std::endl;
#endif
std::cout << "\t MJPEG example : "<< argv[0] << " -W 640 -H 480 -j 120 -Q 5 -u media/stream1 -P 554" << std::endl;
exit(0);
}
}
}
if (optind<argc)
{
dev_name = argv[optind];
}
// create live555 environment
scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
// create RTSP server
rtspServer = RTSPServer::createNew(*env, rtspPort);
if (rtspServer == NULL)
{
//LOG(ERROR) << "Failed to create RTSP server: " << env->getResultMsg();
fprintf(stderr, "Failed to create RTSP server: %s \n", env->getResultMsg());
}
else
{
// set http tunneling
if (rtspOverHTTPPort)
{
rtspServer->setUpTunnelingOverHTTP(rtspOverHTTPPort);
}
// Init capture
//LOG(NOTICE) << "Create V4L2 Source..." << dev_name;
fprintf(stderr, "create Video source = %s \n", dev_name);
V4L2DeviceParameters param(dev_name,format,width,height,fps, isp_fps, verbose, bitrate, m2m_en, gop, mjpeg_qp, queueSize );
videoCapture = createVideoCapure(param);
#if AUDIO_STREAM
if (audio_en) {
audioCapture = createAudioCapure();
}
#endif
if (videoCapture)
{
int outputFd = -1;
//int outputFd = createOutput(outputFile, videoCapture->getFd());
//LOG(NOTICE) << "Start V4L2 Capture..." << dev_name;
fprintf(stderr, "Start V4L2 Capture... %s \n", dev_name);
//videoCapture->captureStart();
snx98600_video_start(videoCapture);
printf("\n\n------- V4L2 Infomation -------- \n");
printf("m2m_en: %d\n", videoCapture->m2m->m2m);
printf("codec_dev: %s\n", videoCapture->m2m->codec_dev);
printf("codec_fps: %d\n", videoCapture->m2m->codec_fps);
if(videoCapture->m2m->m2m)
printf("isp_fps: %d\n", videoCapture->m2m->isp_fps);
printf("width: %d\n", videoCapture->m2m->width);
printf("height: %d\n", videoCapture->m2m->height);
printf("scale: %d\n", videoCapture->m2m->scale);
printf("bit_rate: %d\n", videoCapture->m2m->bit_rate);
printf("dyn_fps_en: %d\n", videoCapture->m2m->dyn_fps_en);
if(videoCapture->m2m->dyn_fps_en) {
printf("framerate: %d\n", videoCapture->rate_ctl->framerate);
}
printf("GOP: %d\n", videoCapture->rate_ctl->gop);
printf("ds_font_num: %d\n", videoCapture->m2m->ds_font_num);
printf("\n----------------------------- \n\n");
#if AUDIO_STREAM
/*
Start Audio Device
*/
if (audio_en) {
int rc;
if (audioCapture) {
if ((rc = snx98600_record_audio_start(audioCapture))) {
fprintf(stderr, "failed to start audio source: %s\n", strerror(rc));
}
}
}
#endif
/* Determind which Class to use */
if (format == V4L2_PIX_FMT_H264)
videoES = H264_V4L2DeviceSource::createNew(*env, param, outputFd, useThread);
else {
videoES = V4L2DeviceSource::createNew(*env, param, outputFd, useThread);
}
/* check if create a Device source success */
if (videoES == NULL)
{
//LOG(FATAL) << "Unable to create source for device " << dev_name;
fprintf(stderr, "Unable to create source for device %s \n", dev_name);
}
else
{
videoCapture->devicesource = videoES;
// Setup the outpacket size;
if (m2m_en) {
//OutPacketBuffer::maxSize = (unsigned int)videoCapture->m2m->isp_buffers->length;
OutPacketBuffer::maxSize = bitrate << 8; //2X Bitrate as the max packet size
fprintf(stderr, "isp buffers: %u , outpack maxsize : %u\n", (unsigned int)videoCapture->m2m->isp_buffers->length, OutPacketBuffer::maxSize );
}else {
OutPacketBuffer::maxSize = width * height * 3 / 2;
}
#if AUDIO_STREAM
/*
create Alsa Device source Class
*/
if (audio_en && audioCapture) {
audioES = AlsaDeviceSource::createNew(*env, -1, queueSize, useThread);
if (audioES == NULL)
{
fprintf(stderr, "Unable to create audio devicesource \n");
}
else
{
audioCapture->devicesource = audioES;
}
}
#endif
replicator = StreamReplicator::createNew(*env, videoES, false);
#if AUDIO_STREAM
if (audio_en && audioCapture)
audio_replicator = StreamReplicator::createNew(*env, audioES, false);
#endif
// Create Server Multicast Session
if (multicast)
{
ServerMediaSubsession * multicast_video_subSession = NULL;
ServerMediaSubsession * multicast_audio_subSession = NULL;
if (maddr == INADDR_NONE) maddr = chooseRandomIPv4SSMAddress(*env);
destinationAddress.s_addr = maddr;
//LOG(NOTICE) << "Mutlicast address " << inet_ntoa(destinationAddress);
fprintf(stderr, "Mutlicast address %s \n", inet_ntoa(destinationAddress));
multicast_video_subSession = MulticastServerMediaSubsession::createNew(*env,destinationAddress, Port(rtpPortNum), Port(rtcpPortNum), ttl, replicator,format,param);
#if AUDIO_STREAM
if (audio_en && audioCapture)
multicast_audio_subSession = MulticastServerMediaSubsession::createNew(*env,destinationAddress, Port(rtpPortNum), Port(rtcpPortNum), ttl, audio_replicator,WA_PCMA,param);
#endif
addSession(rtspServer, murl.c_str(), multicast_video_subSession, multicast_audio_subSession);
}
ServerMediaSubsession * video_subSession = NULL;
ServerMediaSubsession * audio_subSession = NULL;
video_subSession = UnicastServerMediaSubsession::createNew(*env,replicator,format, param);
#if AUDIO_STREAM
if (audio_en && audioCapture)
audio_subSession = UnicastServerMediaSubsession::createNew(*env,audio_replicator,WA_PCMA, param);
#endif
// Create Server Unicast Session
addSession(rtspServer, url.c_str(), video_subSession, audio_subSession);
// main loop
signal(SIGINT,sighandler);
env->taskScheduler().doEventLoop(&quit);
fprintf(stderr, "Exiting.... \n");
#if AUDIO_STREAM
if (audioES)
{
Medium::close(audioES);
}
#endif
Medium::close(videoES);
}
#if AUDIO_STREAM
if (audio_en && audioCapture)
closeAudioCapure(audioCapture);
#endif
if (videoCapture)
closeVideoCapure(videoCapture);
//delete videoCapture;
if (outputFd != -1)
{
close(outputFd);
}
}
Medium::close(rtspServer);
}
env->reclaim();
delete scheduler;
return 0;
}
static void RtspServerStart(RTSP_PARAM_INFO *cRtspParamInfo)
{
enum {ENUM_UNICAST = 0, ENUM_MULTICAST};
scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
Boolean reuseFirstSource = true;
OutPacketBuffer::maxSize = 500000;
const unsigned short statusPort = 10086;
Port RTSPStatusPort(statusPort);
Port RTSPServerPort(cRtspParamInfo->iRTSPServerPort);
RTPSink *videoSinkMajor = NULL;
RTPSink *videoSinkMinor = NULL;
char streamName[RTSP_STRING_LEN] = {0};
char secondStreamName[RTSP_STRING_LEN] = {0};
strncpy(streamName, cRtspParamInfo->rgStreamName[RTSP_STREAM_MAJOR], RTSP_STRING_LEN - 1);
strncpy(secondStreamName, cRtspParamInfo->rgStreamName[RTSP_STREAM_MINOR], RTSP_STRING_LEN - 1);
// Create 'groupsocks' for RTP and RTCP:
struct in_addr destinationAddressMajor;
struct in_addr destinationAddressMinor;
UserAuthenticationDatabase *authDB = NULL;
//#ifdef ACCESS_CONTROL
if(cRtspParamInfo->iAuthenticateEnable){
authDB = new UserAuthenticationDatabase;
// authDB->addUserRecord(cRtspParamInfo->rgUserName, cRtspParamInfo->rgPassword);
authDB->addUserRecord("admin", "admin");
fprintf(stdout, "%s %d Authentication Enable!\n", __FILE__, __LINE__);
}
//#endif
destinationAddressMajor.s_addr = chooseRandomIPv4SSMAddress(*env);
destinationAddressMinor.s_addr = chooseRandomIPv4SSMAddress(*env);
// Note: This is a multicast address. If you wish instead to stream
// using unicast, then you should use the "testOnDemandRTSPServer"
// test program - not this test program - as a model.
const unsigned short rtpPortNumMajor = 18888;
const unsigned short rtcpPortNumMajor = rtpPortNumMajor + 1;
const unsigned short rtpPortNumMinor = rtcpPortNumMajor + 1;
const unsigned short rtcpPortNumMinor = rtpPortNumMinor + 1;
const unsigned char ttl = 255;
const Port rtpPortMajor(rtpPortNumMajor);
const Port rtcpPortMajor(rtcpPortNumMajor);
const Port rtpPortMinor(rtpPortNumMinor);
const Port rtcpPortMinor(rtcpPortNumMinor);
Groupsock rtpGroupsockMajor(*env, destinationAddressMajor, rtpPortMajor, ttl);
rtpGroupsockMajor.multicastSendOnly(); // we're a SSM source
Groupsock rtcpGroupsockMajor(*env, destinationAddressMajor, rtcpPortMajor, ttl);
rtcpGroupsockMajor.multicastSendOnly(); // we're a SSM source
Groupsock rtpGroupsockMinor(*env, destinationAddressMinor, rtpPortMinor, ttl);
rtpGroupsockMinor.multicastSendOnly();
Groupsock rtcpGroupsockMinor(*env, destinationAddressMinor, rtcpPortMinor, ttl);
rtcpGroupsockMinor.multicastSendOnly();
// Create a 'H264 Video RTP' sink from the RTP 'groupsock':
OutPacketBuffer::maxSize = 1000000;
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 500; // in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen+1];
gethostname((char*)CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
RTSPServer* rtspServer = RTSPServer::createNew(*env, RTSPServerPort, authDB);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: "
<< env->getResultMsg() << "\n";
exit(1);
}
//first stream
if(cRtspParamInfo->rgCommunicationMode[RTSP_STREAM_MAJOR] == ENUM_UNICAST){
ServerMediaSession* smsMajor
= ServerMediaSession::createNew(*env, streamName,
streamName, "Session streamed by \"testH264VideoStreamer\"");
smsMajor->addSubsession(H264VideoFileServerMediaSubsession::createNew(*env,
firstInputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(smsMajor);
}else if(cRtspParamInfo->rgCommunicationMode[RTSP_STREAM_MAJOR] == ENUM_MULTICAST){
videoSinkMajor = H264VideoRTPSink::createNew(*env, &rtpGroupsockMajor, 96);
RTCPInstance* rtcpMajor = RTCPInstance::createNew(*env, &rtcpGroupsockMajor,
estimatedSessionBandwidth, CNAME,
videoSinkMajor, NULL /* we're a server */,
True /* we're a SSM source */);
ServerMediaSession* smsMajor = ServerMediaSession::createNew(*env, streamName, firstInputFileName,
"Session streamed by \"swH264VideoStreamer\"",
True /*SSM*/);
smsMajor->addSubsession(PassiveServerMediaSubsession::createNew(*videoSinkMajor, rtcpMajor));
rtspServer->addServerMediaSession(smsMajor);
play(videoSinkMajor, firstInputFileName);
}
if((cRtspParamInfo->iRTSPStreamNum > 1) &&
(cRtspParamInfo->rgCommunicationMode[RTSP_STREAM_MINOR] == ENUM_UNICAST)){
ServerMediaSession* smsMinor
= ServerMediaSession::createNew(*env, secondStreamName,
secondStreamName,
"Session streamed by \"testH264VideoStreamer\"");
smsMinor->addSubsession(H264VideoFileServerMediaSubsession::createNew(*env,
secondInputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(smsMinor);
}else if((cRtspParamInfo->iRTSPStreamNum > 1) &&
(cRtspParamInfo->rgCommunicationMode[RTSP_STREAM_MINOR] == ENUM_MULTICAST)){
videoSinkMinor = H264VideoRTPSink::createNew(*env, &rtpGroupsockMinor, 96);
RTCPInstance* rtcpMinor = RTCPInstance::createNew(*env, &rtcpGroupsockMinor,
estimatedSessionBandwidth, CNAME,
videoSinkMinor, NULL /* we're a server */,
True /* we're a SSM source */);
ServerMediaSession *smsMinor = ServerMediaSession::createNew(*env, secondStreamName, secondInputFileName,
"Session streamed by \"swH264VideoStreamer\"",
True /*SSM*/);
smsMinor->addSubsession(PassiveServerMediaSubsession::createNew(*videoSinkMinor, rtcpMinor));
rtspServer->addServerMediaSession(smsMinor);
play(videoSinkMinor, secondInputFileName);
}
rtspServer->setUpConnectionStatus(RTSPStatusPort);
rtspServer->setStreamName(0, streamName, strlen(streamName)); //0 for major
rtspServer->setStreamName(1, secondStreamName, strlen(secondStreamName));
rtspServer->setAutoControlBitrate(cRtspParamInfo->iAutoControlBitrateEnable);
if(rtspServer->setUpTunnelingOverHTTP(80) ||
rtspServer->setUpTunnelingOverHTTP(8000) ||
rtspServer->setUpTunnelingOverHTTP(8080)) {
*env << "\n(We use port " << rtspServer->httpServerPortNum()
<< " for optional RTSP-over-HTTP tunneling.)\n";
} else {
*env << "\n(RTSP-over-HTTP tunneling is not available.)\n";
}
env->taskScheduler().doEventLoop(); // does not return
}
int main(int argc, char** argv) {
init_signals();
setpriority(PRIO_PROCESS, 0, 0);
int IsSilence = 0;
int svcEnable = 0;
int cnt=0;
int activePortCnt=0;
if( GetSampleRate() == 16000 )
{
audioOutputBitrate = 128000;
audioSamplingFrequency = 16000;
}else{
audioOutputBitrate = 64000;
audioSamplingFrequency = 8000;
}
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
int msg_type, video_type;
APPROInput* MjpegInputDevice = NULL;
APPROInput* H264InputDevice = NULL;
APPROInput* Mpeg4InputDevice = NULL;
static pid_t child[4] = {
-1,-1,-1,-1
};
StreamingMode streamingMode = STREAMING_UNICAST;
netAddressBits multicastAddress = 0;//our_inet_addr("224.1.4.6");
portNumBits videoRTPPortNum = 0;
portNumBits audioRTPPortNum = 0;
IsSilence = 0;
svcEnable = 0;
audioType = AUDIO_G711;
streamingMode = STREAMING_UNICAST;
for( cnt = 1; cnt < argc ;cnt++ )
{
if( strcmp( argv[cnt],"-m" )== 0 )
{
streamingMode = STREAMING_MULTICAST_SSM;
}
if( strcmp( argv[cnt],"-s" )== 0 )
{
IsSilence = 1;
}
if( strcmp( argv[cnt],"-a" )== 0 )
{
audioType = AUDIO_AAC;
}
if( strcmp( argv[cnt],"-v" )== 0 )
{
svcEnable = 1;
}
}
#if 0
printf("###########IsSilence = %d ################\n",IsSilence);
printf("###########streamingMode = %d ################\n",streamingMode);
printf("###########audioType = %d ################\n",audioType);
printf("###########svcEnable = %d ################\n",svcEnable);
#endif
child[0] = fork();
if( child[0] != 0 )
{
child[1] = fork();
}
if( child[0] != 0 && child[1] != 0 )
{
child[2] = fork();
}
if( child[0] != 0 && child[1] != 0 && child[2] != 0 )
{
child[3] = fork();
}
if(svcEnable) {
if( child[0] != 0 && child[1] != 0 && child[2] != 0 && child[3] != 0)
{
child[4] = fork();
}
if( child[0] != 0 && child[1] != 0 && child[2] != 0 && child[3] != 0 && child[4] != 0)
{
child[5] = fork();
}
if( child[0] != 0 && child[1] != 0 && child[2] != 0 && child[3] != 0 && child[4] != 0 && child[5] != 0)
{
child[6] = fork();
}
if( child[0] != 0 && child[1] != 0 && child[2] != 0 && child[3] != 0 && child[4] != 0 && child[5] != 0 && child[6] != 0)
{
child[7] = fork();
}
}
if( child[0] == 0 )
{
/* parent, success */
msg_type = LIVE_MSG_TYPE4;
video_type = VIDEO_TYPE_H264_CIF;
rtspServerPortNum = 8556;
H264VideoBitrate = 12000000;
videoRTPPortNum = 6012;
audioRTPPortNum = 6014;
}
if( child[1] == 0 )
{
/* parent, success */
msg_type = LIVE_MSG_TYPE3;
video_type = VIDEO_TYPE_MJPEG;
rtspServerPortNum = 8555;
MjpegVideoBitrate = 12000000;
videoRTPPortNum = 6008;
audioRTPPortNum = 6010;
}
if( child[2] == 0 )
{
/* parent, success */
msg_type = LIVE_MSG_TYPE;
video_type = VIDEO_TYPE_MPEG4;
rtspServerPortNum = 8553;
Mpeg4VideoBitrate = 12000000;
videoRTPPortNum = 6000;
audioRTPPortNum = 6002;
}
if( child[3] == 0 )
{
/* parent, success */
msg_type = LIVE_MSG_TYPE2;
video_type = VIDEO_TYPE_MPEG4_CIF;
rtspServerPortNum = 8554;
Mpeg4VideoBitrate = 12000000;
videoRTPPortNum = 6004;
audioRTPPortNum = 6006;
}
if(svcEnable) {
if( child[4] == 0 )
{
/* parent, success */
msg_type = LIVE_MSG_TYPE5;
video_type = VIDEO_TYPE_H264_SVC_30FPS;
rtspServerPortNum = 8601;
H264VideoBitrate = 12000000;
videoRTPPortNum = 6016;
audioRTPPortNum = 6018;
}
if( child[5] == 0 )
{
/* parent, success */
msg_type = LIVE_MSG_TYPE6;
video_type = VIDEO_TYPE_H264_SVC_15FPS;
rtspServerPortNum = 8602;
H264VideoBitrate = 12000000;
videoRTPPortNum = 6020;
audioRTPPortNum = 6022;
}
if( child[6] == 0 )
{
/* parent, success */
msg_type = LIVE_MSG_TYPE7;
video_type = VIDEO_TYPE_H264_SVC_7FPS;
rtspServerPortNum = 8603;
H264VideoBitrate = 12000000;
videoRTPPortNum = 6024;
audioRTPPortNum = 6026;
}
if( child[7] == 0 )
{
/* parent, success */
msg_type = LIVE_MSG_TYPE8;
video_type = VIDEO_TYPE_H264_SVC_3FPS;
rtspServerPortNum = 8604;
H264VideoBitrate = 12000000;
videoRTPPortNum = 6028;
audioRTPPortNum = 6030;
}
if( child[0] != 0 && child[1] != 0 && child[2] != 0 && child[3] != 0 && child[4] != 0 && child[5] != 0 && child[6] != 0 && child[7] != 0)
{
/* parent, success */
msg_type = LIVE_MSG_TYPE9;
video_type = VIDEO_TYPE_H264;
rtspServerPortNum = 8557;
H264VideoBitrate = 12000000;
videoRTPPortNum = 6032;
audioRTPPortNum = 6034;
}
}
else {
if( child[0] != 0 && child[1] != 0 && child[2] != 0 && child[3] != 0)
{
/* parent, success */
msg_type = LIVE_MSG_TYPE5;
video_type = VIDEO_TYPE_H264;
rtspServerPortNum = 8557;
H264VideoBitrate = 12000000;
videoRTPPortNum = 6032;
audioRTPPortNum = 6034;
}
}
videoType = video_type;
// Objects used for multicast streaming:
static Groupsock* rtpGroupsockAudio = NULL;
static Groupsock* rtcpGroupsockAudio = NULL;
static Groupsock* rtpGroupsockVideo = NULL;
static Groupsock* rtcpGroupsockVideo = NULL;
static FramedSource* sourceAudio = NULL;
static RTPSink* sinkAudio = NULL;
static RTCPInstance* rtcpAudio = NULL;
static FramedSource* sourceVideo = NULL;
static RTPSink* sinkVideo = NULL;
static RTCPInstance* rtcpVideo = NULL;
share_memory_init(msg_type);
//init_signals();
*env << "Initializing...\n";
// Initialize the WIS input device:
if( video_type == VIDEO_TYPE_MJPEG)
{
MjpegInputDevice = APPROInput::createNew(*env, VIDEO_TYPE_MJPEG);
if (MjpegInputDevice == NULL) {
err(*env) << "Failed to create MJPEG input device\n";
exit(1);
}
}
if( video_type == VIDEO_TYPE_H264 || video_type == VIDEO_TYPE_H264_CIF || video_type == VIDEO_TYPE_H264_SVC_30FPS ||
video_type == VIDEO_TYPE_H264_SVC_15FPS || video_type == VIDEO_TYPE_H264_SVC_7FPS || video_type == VIDEO_TYPE_H264_SVC_3FPS)
{
H264InputDevice = APPROInput::createNew(*env, video_type);
if (H264InputDevice == NULL) {
err(*env) << "Failed to create MJPEG input device\n";
exit(1);
}
}
if( video_type == VIDEO_TYPE_MPEG4 || video_type == VIDEO_TYPE_MPEG4_CIF )
{
Mpeg4InputDevice = APPROInput::createNew(*env, video_type);
if (Mpeg4InputDevice == NULL) {
err(*env) << "Failed to create MPEG4 input device\n";
exit(1);
}
}
// Create the RTSP server:
RTSPServer* rtspServer = NULL;
// Normal case: Streaming from a built-in RTSP server:
rtspServer = RTSPServer::createNew(*env, rtspServerPortNum, NULL);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
*env << "...done initializing\n";
if( streamingMode == STREAMING_UNICAST )
{
if( video_type == VIDEO_TYPE_MJPEG)
{
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, MjpegStreamName, MjpegStreamName, streamDescription,streamingMode == STREAMING_MULTICAST_SSM);
sms->addSubsession(WISJPEGVideoServerMediaSubsession
::createNew(sms->envir(), *MjpegInputDevice, MjpegVideoBitrate));
if( IsSilence == 0)
{
sms->addSubsession(WISPCMAudioServerMediaSubsession::createNew(sms->envir(), *MjpegInputDevice));
}
rtspServer->addServerMediaSession(sms);
char *url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL:\n\t" << url << "\n";
delete[] url;
}
if( video_type == VIDEO_TYPE_H264 || video_type == VIDEO_TYPE_H264_CIF || video_type == VIDEO_TYPE_H264_SVC_30FPS ||
video_type == VIDEO_TYPE_H264_SVC_15FPS || video_type == VIDEO_TYPE_H264_SVC_7FPS || video_type ==VIDEO_TYPE_H264_SVC_3FPS)
{
ServerMediaSession* sms;
sms
= ServerMediaSession::createNew(*env, H264StreamName, H264StreamName, streamDescription,streamingMode == STREAMING_MULTICAST_SSM);
sms->addSubsession(WISH264VideoServerMediaSubsession
::createNew(sms->envir(), *H264InputDevice, H264VideoBitrate));
if( IsSilence == 0)
{
sms->addSubsession(WISPCMAudioServerMediaSubsession::createNew(sms->envir(), *H264InputDevice));
}
rtspServer->addServerMediaSession(sms);
char *url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL:\n\t" << url << "\n";
delete[] url;
}
// Create a record describing the media to be streamed:
if( video_type == VIDEO_TYPE_MPEG4 || video_type == VIDEO_TYPE_MPEG4_CIF )
{
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, Mpeg4StreamName, Mpeg4StreamName, streamDescription,streamingMode == STREAMING_MULTICAST_SSM);
sms->addSubsession(WISMPEG4VideoServerMediaSubsession
::createNew(sms->envir(), *Mpeg4InputDevice, Mpeg4VideoBitrate));
if( IsSilence == 0)
{
sms->addSubsession(WISPCMAudioServerMediaSubsession::createNew(sms->envir(), *Mpeg4InputDevice));
}
rtspServer->addServerMediaSession(sms);
char *url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL:\n\t" << url << "\n";
delete[] url;
}
}else{
if (streamingMode == STREAMING_MULTICAST_SSM)
{
if (multicastAddress == 0)
multicastAddress = chooseRandomIPv4SSMAddress(*env);
} else if (multicastAddress != 0) {
streamingMode = STREAMING_MULTICAST_ASM;
}
struct in_addr dest; dest.s_addr = multicastAddress;
const unsigned char ttl = 255;
// For RTCP:
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen + 1];
gethostname((char *) CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
ServerMediaSession* sms=NULL;
if( video_type == VIDEO_TYPE_MJPEG)
{
sms = ServerMediaSession::createNew(*env, MjpegStreamName, MjpegStreamName, streamDescription,streamingMode == STREAMING_MULTICAST_SSM);
sourceAudio = MjpegInputDevice->audioSource();
sourceVideo = WISJPEGStreamSource::createNew(MjpegInputDevice->videoSource());
// Create 'groupsocks' for RTP and RTCP:
const Port rtpPortVideo(videoRTPPortNum);
const Port rtcpPortVideo(videoRTPPortNum+1);
rtpGroupsockVideo = new Groupsock(*env, dest, rtpPortVideo, ttl);
rtcpGroupsockVideo = new Groupsock(*env, dest, rtcpPortVideo, ttl);
if (streamingMode == STREAMING_MULTICAST_SSM) {
rtpGroupsockVideo->multicastSendOnly();
rtcpGroupsockVideo->multicastSendOnly();
}
setVideoRTPSinkBufferSize();
sinkVideo = JPEGVideoRTPSink::createNew(*env, rtpGroupsockVideo);
}
if( video_type == VIDEO_TYPE_H264 || video_type == VIDEO_TYPE_H264_CIF ||
video_type == VIDEO_TYPE_H264_SVC_30FPS || video_type == VIDEO_TYPE_H264_SVC_15FPS ||
video_type == VIDEO_TYPE_H264_SVC_7FPS || video_type == VIDEO_TYPE_H264_SVC_3FPS)
{
sms = ServerMediaSession::createNew(*env, H264StreamName, H264StreamName, streamDescription,streamingMode == STREAMING_MULTICAST_SSM);
sourceAudio = H264InputDevice->audioSource();
sourceVideo = H264VideoStreamFramer::createNew(*env, H264InputDevice->videoSource());
// Create 'groupsocks' for RTP and RTCP:
const Port rtpPortVideo(videoRTPPortNum);
const Port rtcpPortVideo(videoRTPPortNum+1);
rtpGroupsockVideo = new Groupsock(*env, dest, rtpPortVideo, ttl);
rtcpGroupsockVideo = new Groupsock(*env, dest, rtcpPortVideo, ttl);
if (streamingMode == STREAMING_MULTICAST_SSM) {
rtpGroupsockVideo->multicastSendOnly();
rtcpGroupsockVideo->multicastSendOnly();
}
setVideoRTPSinkBufferSize();
{
char BuffStr[200];
extern int GetSprop(void *pBuff, char vType);
GetSprop(BuffStr,video_type);
sinkVideo = H264VideoRTPSink::createNew(*env, rtpGroupsockVideo,96, 0x64001F,BuffStr);
}
}
// Create a record describing the media to be streamed:
if( video_type == VIDEO_TYPE_MPEG4 || video_type == VIDEO_TYPE_MPEG4_CIF )
{
sms = ServerMediaSession::createNew(*env, Mpeg4StreamName, Mpeg4StreamName, streamDescription,streamingMode == STREAMING_MULTICAST_SSM);
sourceAudio = Mpeg4InputDevice->audioSource();
sourceVideo = MPEG4VideoStreamDiscreteFramer::createNew(*env, Mpeg4InputDevice->videoSource());
// Create 'groupsocks' for RTP and RTCP:
const Port rtpPortVideo(videoRTPPortNum);
const Port rtcpPortVideo(videoRTPPortNum+1);
rtpGroupsockVideo = new Groupsock(*env, dest, rtpPortVideo, ttl);
rtcpGroupsockVideo = new Groupsock(*env, dest, rtcpPortVideo, ttl);
if (streamingMode == STREAMING_MULTICAST_SSM) {
rtpGroupsockVideo->multicastSendOnly();
rtcpGroupsockVideo->multicastSendOnly();
}
setVideoRTPSinkBufferSize();
sinkVideo = MPEG4ESVideoRTPSink::createNew(*env, rtpGroupsockVideo,97);
}
/* VIDEO Channel initial */
if(1)
{
// Create (and start) a 'RTCP instance' for this RTP sink:
unsigned totalSessionBandwidthVideo = (Mpeg4VideoBitrate+500)/1000; // in kbps; for RTCP b/w share
rtcpVideo = RTCPInstance::createNew(*env, rtcpGroupsockVideo,
totalSessionBandwidthVideo, CNAME,
sinkVideo, NULL /* we're a server */ ,
streamingMode == STREAMING_MULTICAST_SSM);
// Note: This starts RTCP running automatically
sms->addSubsession(PassiveServerMediaSubsession::createNew(*sinkVideo, rtcpVideo));
// Start streaming:
sinkVideo->startPlaying(*sourceVideo, NULL, NULL);
}
/* AUDIO Channel initial */
if( IsSilence == 0)
{
// there's a separate RTP stream for audio
// Create 'groupsocks' for RTP and RTCP:
const Port rtpPortAudio(audioRTPPortNum);
const Port rtcpPortAudio(audioRTPPortNum+1);
rtpGroupsockAudio = new Groupsock(*env, dest, rtpPortAudio, ttl);
rtcpGroupsockAudio = new Groupsock(*env, dest, rtcpPortAudio, ttl);
if (streamingMode == STREAMING_MULTICAST_SSM)
{
rtpGroupsockAudio->multicastSendOnly();
rtcpGroupsockAudio->multicastSendOnly();
}
if( audioSamplingFrequency == 16000 )
{
if( audioType == AUDIO_G711)
{
sinkAudio = SimpleRTPSink::createNew(*env, rtpGroupsockAudio, 96, audioSamplingFrequency, "audio", "PCMU", 1);
}
else
{
char const* encoderConfigStr = "1408";// (2<<3)|(8>>1) = 0x14 ; ((8<<7)&0xFF)|(1<<3)=0x08 ;
sinkAudio = MPEG4GenericRTPSink::createNew(*env, rtpGroupsockAudio,
96,
audioSamplingFrequency,
"audio", "AAC-hbr",
encoderConfigStr, audioNumChannels);
}
}
else{
if(audioType == AUDIO_G711)
{
sinkAudio = SimpleRTPSink::createNew(*env, rtpGroupsockAudio, 0, audioSamplingFrequency, "audio", "PCMU", 1);
}
else{
char const* encoderConfigStr = "1588";// (2<<3)|(11>>1) = 0x15 ; ((11<<7)&0xFF)|(1<<3)=0x88 ;
sinkAudio = MPEG4GenericRTPSink::createNew(*env, rtpGroupsockAudio,
96,
audioSamplingFrequency,
"audio", "AAC-hbr",
encoderConfigStr, audioNumChannels);
}
}
// Create (and start) a 'RTCP instance' for this RTP sink:
unsigned totalSessionBandwidthAudio = (audioOutputBitrate+500)/1000; // in kbps; for RTCP b/w share
rtcpAudio = RTCPInstance::createNew(*env, rtcpGroupsockAudio,
totalSessionBandwidthAudio, CNAME,
sinkAudio, NULL /* we're a server */,
streamingMode == STREAMING_MULTICAST_SSM);
// Note: This starts RTCP running automatically
sms->addSubsession(PassiveServerMediaSubsession::createNew(*sinkAudio, rtcpAudio));
// Start streaming:
sinkAudio->startPlaying(*sourceAudio, NULL, NULL);
}
rtspServer->addServerMediaSession(sms);
{
struct in_addr dest; dest.s_addr = multicastAddress;
char *url = rtspServer->rtspURL(sms);
//char *url2 = inet_ntoa(dest);
*env << "Mulicast Play this stream using the URL:\n\t" << url << "\n";
//*env << "2 Mulicast addr:\n\t" << url2 << "\n";
delete[] url;
}
}
// Begin the LIVE555 event loop:
env->taskScheduler().doEventLoop(&watchVariable); // does not return
if( streamingMode!= STREAMING_UNICAST )
{
Medium::close(rtcpAudio);
Medium::close(sinkAudio);
Medium::close(sourceAudio);
delete rtpGroupsockAudio;
delete rtcpGroupsockAudio;
Medium::close(rtcpVideo);
Medium::close(sinkVideo);
Medium::close(sourceVideo);
delete rtpGroupsockVideo;
delete rtcpGroupsockVideo;
}
Medium::close(rtspServer); // will also reclaim "sms" and its "ServerMediaSubsession"s
if( MjpegInputDevice != NULL )
{
Medium::close(MjpegInputDevice);
}
if( H264InputDevice != NULL )
{
Medium::close(H264InputDevice);
}
if( Mpeg4InputDevice != NULL )
{
Medium::close(Mpeg4InputDevice);
}
env->reclaim();
delete scheduler;
ApproInterfaceExit();
return 0; // only to prevent compiler warning
}
